758 files changed, 150508 insertions, 0 deletions

diff --git a/dom/media/webrtc/CubebDeviceEnumerator.cpp b/dom/media/webrtc/CubebDeviceEnumerator.cpp
new file mode 100644
index 0000000000..b5e705ea7d
--- /dev/null
+++ b/dom/media/webrtc/CubebDeviceEnumerator.cpp
@@ -0,0 +1,334 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "CubebDeviceEnumerator.h"
+
+#include "mozilla/ClearOnShutdown.h"
+#include "mozilla/SchedulerGroup.h"
+#include "mozilla/StaticMutex.h"
+#include "mozilla/StaticPtr.h"
+#include "mozilla/media/MediaUtils.h"
+#include "nsThreadUtils.h"
+#ifdef XP_WIN
+#  include "mozilla/mscom/EnsureMTA.h"
+#endif
+
+namespace mozilla {
+
+using namespace CubebUtils;
+using AudioDeviceSet = CubebDeviceEnumerator::AudioDeviceSet;
+
+/* static */
+static StaticRefPtr<CubebDeviceEnumerator> sInstance;
+static StaticMutex sInstanceMutex MOZ_UNANNOTATED;
+
+/* static */
+CubebDeviceEnumerator* CubebDeviceEnumerator::GetInstance() {
+  StaticMutexAutoLock lock(sInstanceMutex);
+  if (!sInstance) {
+    sInstance = new CubebDeviceEnumerator();
+    static bool clearOnShutdownSetup = []() -> bool {
+      auto setClearOnShutdown = []() -> void {
+        ClearOnShutdown(&sInstance, ShutdownPhase::XPCOMShutdownThreads);
+      };
+      if (NS_IsMainThread()) {
+        setClearOnShutdown();
+      } else {
+        SchedulerGroup::Dispatch(
+            TaskCategory::Other,
+            NS_NewRunnableFunction("CubebDeviceEnumerator::::GetInstance()",
+                                   std::move(setClearOnShutdown)));
+      }
+      return true;
+    }();
+    Unused << clearOnShutdownSetup;
+  }
+  return sInstance.get();
+}
+
+CubebDeviceEnumerator::CubebDeviceEnumerator()
+    : mMutex("CubebDeviceListMutex"),
+      mManualInputInvalidation(false),
+      mManualOutputInvalidation(false) {
+#ifdef XP_WIN
+  // Ensure the MTA thread exists and gets instantiated before the
+  // CubebDeviceEnumerator so that this instance will always gets destructed
+  // before the MTA thread gets shutdown.
+  mozilla::mscom::EnsureMTA([&]() -> void {
+#endif
+    int rv = cubeb_register_device_collection_changed(
+        GetCubebContext(), CUBEB_DEVICE_TYPE_OUTPUT,
+        &OutputAudioDeviceListChanged_s, this);
+    if (rv != CUBEB_OK) {
+      NS_WARNING(
+          "Could not register the audio output"
+          " device collection changed callback.");
+      mManualOutputInvalidation = true;
+    }
+    rv = cubeb_register_device_collection_changed(
+        GetCubebContext(), CUBEB_DEVICE_TYPE_INPUT,
+        &InputAudioDeviceListChanged_s, this);
+    if (rv != CUBEB_OK) {
+      NS_WARNING(
+          "Could not register the audio input"
+          " device collection changed callback.");
+      mManualInputInvalidation = true;
+    }
+#ifdef XP_WIN
+  });
+#endif
+}
+
+/* static */
+void CubebDeviceEnumerator::Shutdown() {
+  StaticMutexAutoLock lock(sInstanceMutex);
+  if (sInstance) {
+    sInstance = nullptr;
+  }
+}
+
+CubebDeviceEnumerator::~CubebDeviceEnumerator() {
+#ifdef XP_WIN
+  mozilla::mscom::EnsureMTA([&]() -> void {
+#endif
+    int rv = cubeb_register_device_collection_changed(
+        GetCubebContext(), CUBEB_DEVICE_TYPE_OUTPUT, nullptr, this);
+    if (rv != CUBEB_OK) {
+      NS_WARNING(
+          "Could not unregister the audio output"
+          " device collection changed callback.");
+    }
+    rv = cubeb_register_device_collection_changed(
+        GetCubebContext(), CUBEB_DEVICE_TYPE_INPUT, nullptr, this);
+    if (rv != CUBEB_OK) {
+      NS_WARNING(
+          "Could not unregister the audio input"
+          " device collection changed callback.");
+    }
+#ifdef XP_WIN
+  });
+#endif
+}
+
+RefPtr<const AudioDeviceSet>
+CubebDeviceEnumerator::EnumerateAudioInputDevices() {
+  return EnumerateAudioDevices(Side::INPUT);
+}
+
+RefPtr<const AudioDeviceSet>
+CubebDeviceEnumerator::EnumerateAudioOutputDevices() {
+  return EnumerateAudioDevices(Side::OUTPUT);
+}
+
+#ifndef ANDROID
+static uint16_t ConvertCubebType(cubeb_device_type aType) {
+  uint16_t map[] = {
+      nsIAudioDeviceInfo::TYPE_UNKNOWN,  // CUBEB_DEVICE_TYPE_UNKNOWN
+      nsIAudioDeviceInfo::TYPE_INPUT,    // CUBEB_DEVICE_TYPE_INPUT,
+      nsIAudioDeviceInfo::TYPE_OUTPUT    // CUBEB_DEVICE_TYPE_OUTPUT
+  };
+  return map[aType];
+}
+
+static uint16_t ConvertCubebState(cubeb_device_state aState) {
+  uint16_t map[] = {
+      nsIAudioDeviceInfo::STATE_DISABLED,   // CUBEB_DEVICE_STATE_DISABLED
+      nsIAudioDeviceInfo::STATE_UNPLUGGED,  // CUBEB_DEVICE_STATE_UNPLUGGED
+      nsIAudioDeviceInfo::STATE_ENABLED     // CUBEB_DEVICE_STATE_ENABLED
+  };
+  return map[aState];
+}
+
+static uint16_t ConvertCubebPreferred(cubeb_device_pref aPreferred) {
+  if (aPreferred == CUBEB_DEVICE_PREF_NONE) {
+    return nsIAudioDeviceInfo::PREF_NONE;
+  }
+  if (aPreferred == CUBEB_DEVICE_PREF_ALL) {
+    return nsIAudioDeviceInfo::PREF_ALL;
+  }
+
+  uint16_t preferred = 0;
+  if (aPreferred & CUBEB_DEVICE_PREF_MULTIMEDIA) {
+    preferred |= nsIAudioDeviceInfo::PREF_MULTIMEDIA;
+  }
+  if (aPreferred & CUBEB_DEVICE_PREF_VOICE) {
+    preferred |= nsIAudioDeviceInfo::PREF_VOICE;
+  }
+  if (aPreferred & CUBEB_DEVICE_PREF_NOTIFICATION) {
+    preferred |= nsIAudioDeviceInfo::PREF_NOTIFICATION;
+  }
+  return preferred;
+}
+
+static uint16_t ConvertCubebFormat(cubeb_device_fmt aFormat) {
+  uint16_t format = 0;
+  if (aFormat & CUBEB_DEVICE_FMT_S16LE) {
+    format |= nsIAudioDeviceInfo::FMT_S16LE;
+  }
+  if (aFormat & CUBEB_DEVICE_FMT_S16BE) {
+    format |= nsIAudioDeviceInfo::FMT_S16BE;
+  }
+  if (aFormat & CUBEB_DEVICE_FMT_F32LE) {
+    format |= nsIAudioDeviceInfo::FMT_F32LE;
+  }
+  if (aFormat & CUBEB_DEVICE_FMT_F32BE) {
+    format |= nsIAudioDeviceInfo::FMT_F32BE;
+  }
+  return format;
+}
+
+static RefPtr<AudioDeviceSet> GetDeviceCollection(Side aSide) {
+  RefPtr set = new AudioDeviceSet();
+  cubeb* context = GetCubebContext();
+  if (context) {
+    cubeb_device_collection collection = {nullptr, 0};
+#  ifdef XP_WIN
+    mozilla::mscom::EnsureMTA([&]() -> void {
+#  endif
+      if (cubeb_enumerate_devices(context,
+                                  aSide == Input ? CUBEB_DEVICE_TYPE_INPUT
+                                                 : CUBEB_DEVICE_TYPE_OUTPUT,
+                                  &collection) == CUBEB_OK) {
+        for (unsigned int i = 0; i < collection.count; ++i) {
+          auto device = collection.device[i];
+          if (device.max_channels == 0) {
+            continue;
+          }
+          RefPtr<AudioDeviceInfo> info = new AudioDeviceInfo(
+              device.devid, NS_ConvertUTF8toUTF16(device.friendly_name),
+              NS_ConvertUTF8toUTF16(device.group_id),
+              NS_ConvertUTF8toUTF16(device.vendor_name),
+              ConvertCubebType(device.type), ConvertCubebState(device.state),
+              ConvertCubebPreferred(device.preferred),
+              ConvertCubebFormat(device.format),
+              ConvertCubebFormat(device.default_format), device.max_channels,
+              device.default_rate, device.max_rate, device.min_rate,
+              device.latency_hi, device.latency_lo);
+          set->AppendElement(std::move(info));
+        }
+      }
+      cubeb_device_collection_destroy(context, &collection);
+#  ifdef XP_WIN
+    });
+#  endif
+  }
+  return set;
+}
+#endif  // non ANDROID
+
+RefPtr<const AudioDeviceSet> CubebDeviceEnumerator::EnumerateAudioDevices(
+    CubebDeviceEnumerator::Side aSide) {
+  MOZ_ASSERT(aSide == Side::INPUT || aSide == Side::OUTPUT);
+
+  RefPtr<const AudioDeviceSet>* devicesCache;
+  bool manualInvalidation = true;
+
+  if (aSide == Side::INPUT) {
+    devicesCache = &mInputDevices;
+    manualInvalidation = mManualInputInvalidation;
+  } else {
+    MOZ_ASSERT(aSide == Side::OUTPUT);
+    devicesCache = &mOutputDevices;
+    manualInvalidation = mManualOutputInvalidation;
+  }
+
+  cubeb* context = GetCubebContext();
+  if (!context) {
+    return new AudioDeviceSet();
+  }
+  if (!manualInvalidation) {
+    MutexAutoLock lock(mMutex);
+    if (*devicesCache) {
+      return *devicesCache;
+    }
+  }
+
+#ifdef ANDROID
+  cubeb_device_type type = CUBEB_DEVICE_TYPE_UNKNOWN;
+  uint32_t channels = 0;
+  nsAutoString name;
+  if (aSide == Side::INPUT) {
+    type = CUBEB_DEVICE_TYPE_INPUT;
+    channels = 1;
+    name = u"Default audio input device"_ns;
+  } else {
+    MOZ_ASSERT(aSide == Side::OUTPUT);
+    type = CUBEB_DEVICE_TYPE_OUTPUT;
+    channels = 2;
+    name = u"Default audio output device"_ns;
+  }
+  RefPtr devices = new AudioDeviceSet();
+  // Bug 1473346: enumerating devices is not supported on Android in cubeb,
+  // simply state that there is a single sink, that it is the default, and has
+  // a single channel. All the other values are made up and are not to be used.
+  // Bug 1660391: we can't use fluent here yet to get localized strings, so
+  // those are hard-coded en_US strings for now.
+  RefPtr<AudioDeviceInfo> info = new AudioDeviceInfo(
+      nullptr, name, u""_ns, u""_ns, type, CUBEB_DEVICE_STATE_ENABLED,
+      CUBEB_DEVICE_PREF_ALL, CUBEB_DEVICE_FMT_ALL, CUBEB_DEVICE_FMT_S16NE,
+      channels, 44100, 44100, 44100, 441, 128);
+  devices->AppendElement(std::move(info));
+#else
+  RefPtr devices = GetDeviceCollection(
+      (aSide == Side::INPUT) ? CubebUtils::Input : CubebUtils::Output);
+#endif
+  {
+    MutexAutoLock lock(mMutex);
+    *devicesCache = devices;
+  }
+  return devices;
+}
+
+already_AddRefed<AudioDeviceInfo> CubebDeviceEnumerator::DeviceInfoFromName(
+    const nsString& aName, Side aSide) {
+  RefPtr devices = EnumerateAudioDevices(aSide);
+  for (const RefPtr<AudioDeviceInfo>& device : *devices) {
+    if (device->Name().Equals(aName)) {
+      RefPtr<AudioDeviceInfo> other = device;
+      return other.forget();
+    }
+  }
+
+  return nullptr;
+}
+
+RefPtr<AudioDeviceInfo> CubebDeviceEnumerator::DefaultDevice(Side aSide) {
+  RefPtr devices = EnumerateAudioDevices(aSide);
+  for (const RefPtr<AudioDeviceInfo>& device : *devices) {
+    if (device->Preferred()) {
+      RefPtr<AudioDeviceInfo> other = device;
+      return other.forget();
+    }
+  }
+
+  return nullptr;
+}
+
+void CubebDeviceEnumerator::InputAudioDeviceListChanged_s(cubeb* aContext,
+                                                          void* aUser) {
+  CubebDeviceEnumerator* self = reinterpret_cast<CubebDeviceEnumerator*>(aUser);
+  self->AudioDeviceListChanged(CubebDeviceEnumerator::Side::INPUT);
+}
+
+void CubebDeviceEnumerator::OutputAudioDeviceListChanged_s(cubeb* aContext,
+                                                           void* aUser) {
+  CubebDeviceEnumerator* self = reinterpret_cast<CubebDeviceEnumerator*>(aUser);
+  self->AudioDeviceListChanged(CubebDeviceEnumerator::Side::OUTPUT);
+}
+
+void CubebDeviceEnumerator::AudioDeviceListChanged(Side aSide) {
+  MutexAutoLock lock(mMutex);
+  if (aSide == Side::INPUT) {
+    mInputDevices = nullptr;
+    mOnInputDeviceListChange.Notify();
+  } else {
+    MOZ_ASSERT(aSide == Side::OUTPUT);
+    mOutputDevices = nullptr;
+    mOnOutputDeviceListChange.Notify();
+  }
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/CubebDeviceEnumerator.h b/dom/media/webrtc/CubebDeviceEnumerator.h
new file mode 100644
index 0000000000..6b6499f728
--- /dev/null
+++ b/dom/media/webrtc/CubebDeviceEnumerator.h
@@ -0,0 +1,87 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef CUBEBDEVICEENUMERATOR_H_
+#define CUBEBDEVICEENUMERATOR_H_
+
+#include "AudioDeviceInfo.h"
+#include "cubeb/cubeb.h"
+#include "MediaEventSource.h"
+#include "mozilla/Mutex.h"
+#include "nsTArray.h"
+
+namespace mozilla {
+
+namespace media {
+template <typename T>
+class Refcountable;
+}
+
+// This class implements a cache for accessing the audio device list.
+// It can be accessed on any thread.
+class CubebDeviceEnumerator final {
+ public:
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(CubebDeviceEnumerator)
+
+  static CubebDeviceEnumerator* GetInstance();
+  static void Shutdown();
+  using AudioDeviceSet = media::Refcountable<nsTArray<RefPtr<AudioDeviceInfo>>>;
+  // This method returns a list of all the input audio devices
+  // (sources) available on this machine.
+  // This method is safe to call from all threads.
+  RefPtr<const AudioDeviceSet> EnumerateAudioInputDevices();
+  // Similar for the audio audio devices (sinks). Also thread safe.
+  RefPtr<const AudioDeviceSet> EnumerateAudioOutputDevices();
+  // From a device name, return the info for this device, if it's a valid name,
+  // or nullptr otherwise.
+  // This method is safe to call from any thread.
+  enum class Side {
+    INPUT,
+    OUTPUT,
+  };
+  already_AddRefed<AudioDeviceInfo> DeviceInfoFromName(const nsString& aName,
+                                                       Side aSide);
+  // Event source to listen for changes to the audio input device list on.
+  MediaEventSource<void>& OnAudioInputDeviceListChange() {
+    return mOnInputDeviceListChange;
+  }
+
+  // Event source to listen for changes to the audio output device list on.
+  MediaEventSource<void>& OnAudioOutputDeviceListChange() {
+    return mOnOutputDeviceListChange;
+  }
+
+  // Return the default device for a particular side.
+  RefPtr<AudioDeviceInfo> DefaultDevice(Side aSide);
+
+ private:
+  CubebDeviceEnumerator();
+  ~CubebDeviceEnumerator();
+  // Static functions called by cubeb when the audio device list changes
+  // (i.e. when a new device is made available, or non-available). This
+  // simply calls `AudioDeviceListChanged` below.
+  static void InputAudioDeviceListChanged_s(cubeb* aContext, void* aUser);
+  static void OutputAudioDeviceListChanged_s(cubeb* aContext, void* aUser);
+  // Invalidates the cached audio input device list, can be called on any
+  // thread.
+  void AudioDeviceListChanged(Side aSide);
+  RefPtr<const AudioDeviceSet> EnumerateAudioDevices(Side aSide);
+  // Synchronize access to mInputDevices and mOutputDevices;
+  Mutex mMutex MOZ_UNANNOTATED;
+  RefPtr<const AudioDeviceSet> mInputDevices;
+  RefPtr<const AudioDeviceSet> mOutputDevices;
+  // If mManual*Invalidation is true, then it is necessary to query the device
+  // list each time instead of relying on automatic invalidation of the cache by
+  // cubeb itself. Set in the constructor and then can be access on any thread.
+  bool mManualInputInvalidation;
+  bool mManualOutputInvalidation;
+  MediaEventProducer<void> mOnInputDeviceListChange;
+  MediaEventProducer<void> mOnOutputDeviceListChange;
+};
+
+typedef CubebDeviceEnumerator Enumerator;
+typedef CubebDeviceEnumerator::Side EnumeratorSide;
+}  // namespace mozilla
+
+#endif  // CUBEBDEVICEENUMERATOR_H_
diff --git a/dom/media/webrtc/MediaEngine.h b/dom/media/webrtc/MediaEngine.h
new file mode 100644
index 0000000000..d3e1ece452
--- /dev/null
+++ b/dom/media/webrtc/MediaEngine.h
@@ -0,0 +1,66 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef MEDIAENGINE_H_
+#define MEDIAENGINE_H_
+
+#include "DOMMediaStream.h"
+#include "MediaEventSource.h"
+#include "MediaTrackGraph.h"
+#include "MediaTrackConstraints.h"
+#include "mozilla/dom/MediaStreamTrackBinding.h"
+#include "mozilla/dom/VideoStreamTrack.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/ThreadSafeWeakPtr.h"
+
+namespace mozilla {
+
+namespace dom {
+class Blob;
+}  // namespace dom
+
+class AllocationHandle;
+class MediaDevice;
+class MediaEngineSource;
+
+enum MediaSinkEnum {
+  Speaker,
+  Other,
+};
+
+enum { kVideoTrack = 1, kAudioTrack = 2, kTrackCount };
+
+class MediaEngine {
+ public:
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(MediaEngine)
+  NS_DECL_OWNINGEVENTTARGET
+
+  void AssertIsOnOwningThread() const { NS_ASSERT_OWNINGTHREAD(MediaEngine); }
+
+  /**
+   * Populate an array of sources of the requested type in the nsTArray.
+   * Also include devices that are currently unavailable.
+   */
+  virtual void EnumerateDevices(dom::MediaSourceEnum, MediaSinkEnum,
+                                nsTArray<RefPtr<MediaDevice>>*) = 0;
+
+  virtual void Shutdown() = 0;
+
+  virtual RefPtr<MediaEngineSource> CreateSource(
+      const MediaDevice* aDevice) = 0;
+
+  virtual MediaEventSource<void>& DeviceListChangeEvent() = 0;
+  /**
+   * Return true if devices returned from EnumerateDevices are emulated media
+   * devices.
+   */
+  virtual bool IsFake() const = 0;
+
+ protected:
+  virtual ~MediaEngine() = default;
+};
+
+}  // namespace mozilla
+
+#endif /* MEDIAENGINE_H_ */
diff --git a/dom/media/webrtc/MediaEngineFake.cpp b/dom/media/webrtc/MediaEngineFake.cpp
new file mode 100644
index 0000000000..bba6c18694
--- /dev/null
+++ b/dom/media/webrtc/MediaEngineFake.cpp
@@ -0,0 +1,653 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "MediaEngineFake.h"
+
+#include "AudioSegment.h"
+#include "DOMMediaStream.h"
+#include "ImageContainer.h"
+#include "ImageTypes.h"
+#include "MediaEnginePrefs.h"
+#include "MediaEngineSource.h"
+#include "MediaTrackGraph.h"
+#include "MediaTrackListener.h"
+#include "MediaTrackConstraints.h"
+#include "mozilla/dom/File.h"
+#include "mozilla/MediaManager.h"
+#include "mozilla/Mutex.h"
+#include "mozilla/SyncRunnable.h"
+#include "mozilla/UniquePtr.h"
+#include "nsComponentManagerUtils.h"
+#include "nsCOMPtr.h"
+#include "nsContentUtils.h"
+#include "nsITimer.h"
+#include "SineWaveGenerator.h"
+#include "Tracing.h"
+#include "VideoSegment.h"
+#include "VideoUtils.h"
+
+#ifdef MOZ_WIDGET_ANDROID
+#  include "nsISupportsUtils.h"
+#endif
+
+#ifdef MOZ_WEBRTC
+#  include "YuvStamper.h"
+#endif
+
+#define VIDEO_WIDTH_MIN 160
+#define VIDEO_WIDTH_MAX 4096
+#define VIDEO_HEIGHT_MIN 90
+#define VIDEO_HEIGHT_MAX 2160
+#define DEFAULT_AUDIO_TIMER_MS 10
+namespace mozilla {
+
+using namespace mozilla::gfx;
+using dom::MediaSourceEnum;
+using dom::MediaTrackConstraints;
+using dom::MediaTrackSettings;
+using dom::VideoFacingModeEnum;
+
+static nsString FakeVideoName() {
+  // For the purpose of testing we allow to change the name of the fake device
+  // by pref.
+  nsAutoString cameraNameFromPref;
+  nsresult rv;
+  auto getPref = [&]() {
+    rv = Preferences::GetString("media.getusermedia.fake-camera-name",
+                                cameraNameFromPref);
+  };
+  if (NS_IsMainThread()) {
+    getPref();
+  } else {
+    // Here it is preferred a "hard" block, instead of "soft" block provided
+    // by sync dispatch, which allows the waiting thread to spin its event
+    // loop. The latter would allow multiple enumeration requests being
+    // processed out-of-order.
+    RefPtr runnable = NS_NewRunnableFunction(__func__, getPref);
+    SyncRunnable::DispatchToThread(GetMainThreadSerialEventTarget(), runnable);
+  }
+
+  if (NS_SUCCEEDED(rv)) {
+    return std::move(cameraNameFromPref);
+  }
+  return u"Default Video Device"_ns;
+}
+
+/**
+ * Fake video source.
+ */
+class MediaEngineFakeVideoSource : public MediaEngineSource {
+ public:
+  MediaEngineFakeVideoSource();
+
+  static nsString GetGroupId();
+
+  nsresult Allocate(const dom::MediaTrackConstraints& aConstraints,
+                    const MediaEnginePrefs& aPrefs, uint64_t aWindowID,
+                    const char** aOutBadConstraint) override;
+  void SetTrack(const RefPtr<MediaTrack>& aTrack,
+                const PrincipalHandle& aPrincipal) override;
+  nsresult Start() override;
+  nsresult Reconfigure(const dom::MediaTrackConstraints& aConstraints,
+                       const MediaEnginePrefs& aPrefs,
+                       const char** aOutBadConstraint) override;
+  nsresult Stop() override;
+  nsresult Deallocate() override;
+
+  uint32_t GetBestFitnessDistance(
+      const nsTArray<const NormalizedConstraintSet*>& aConstraintSets)
+      const override;
+  void GetSettings(dom::MediaTrackSettings& aOutSettings) const override;
+
+  bool IsFake() const override { return true; }
+
+ protected:
+  ~MediaEngineFakeVideoSource() = default;
+
+  /**
+   * Called by mTimer when it's time to generate a new frame.
+   */
+  void GenerateFrame();
+
+  nsCOMPtr<nsITimer> mTimer;
+
+  RefPtr<layers::ImageContainer> mImageContainer;
+
+  // Current state of this source.
+  MediaEngineSourceState mState = kReleased;
+  RefPtr<layers::Image> mImage;
+  RefPtr<SourceMediaTrack> mTrack;
+  PrincipalHandle mPrincipalHandle = PRINCIPAL_HANDLE_NONE;
+
+  MediaEnginePrefs mOpts;
+  int mCb = 16;
+  int mCr = 16;
+
+  // Main thread only.
+  const RefPtr<media::Refcountable<dom::MediaTrackSettings>> mSettings;
+};
+
+MediaEngineFakeVideoSource::MediaEngineFakeVideoSource()
+    : mTimer(nullptr),
+      mSettings(MakeAndAddRef<media::Refcountable<MediaTrackSettings>>()) {
+  mSettings->mWidth.Construct(
+      int32_t(MediaEnginePrefs::DEFAULT_43_VIDEO_WIDTH));
+  mSettings->mHeight.Construct(
+      int32_t(MediaEnginePrefs::DEFAULT_43_VIDEO_HEIGHT));
+  mSettings->mFrameRate.Construct(double(MediaEnginePrefs::DEFAULT_VIDEO_FPS));
+  mSettings->mFacingMode.Construct(
+      NS_ConvertASCIItoUTF16(dom::VideoFacingModeEnumValues::strings
+                                 [uint8_t(VideoFacingModeEnum::Environment)]
+                                     .value));
+}
+
+nsString MediaEngineFakeVideoSource::GetGroupId() {
+  return u"Fake Video Group"_ns;
+}
+
+uint32_t MediaEngineFakeVideoSource::GetBestFitnessDistance(
+    const nsTArray<const NormalizedConstraintSet*>& aConstraintSets) const {
+  AssertIsOnOwningThread();
+
+  uint64_t distance = 0;
+
+#ifdef MOZ_WEBRTC
+  // distance is read from first entry only
+  if (aConstraintSets.Length() >= 1) {
+    const auto* cs = aConstraintSets.ElementAt(0);
+    Maybe<nsString> facingMode = Nothing();
+    distance +=
+        MediaConstraintsHelper::FitnessDistance(facingMode, cs->mFacingMode);
+
+    if (cs->mWidth.mMax < VIDEO_WIDTH_MIN ||
+        cs->mWidth.mMin > VIDEO_WIDTH_MAX) {
+      distance += UINT32_MAX;
+    }
+
+    if (cs->mHeight.mMax < VIDEO_HEIGHT_MIN ||
+        cs->mHeight.mMin > VIDEO_HEIGHT_MAX) {
+      distance += UINT32_MAX;
+    }
+  }
+#endif
+
+  return uint32_t(std::min(distance, uint64_t(UINT32_MAX)));
+}
+
+void MediaEngineFakeVideoSource::GetSettings(
+    MediaTrackSettings& aOutSettings) const {
+  MOZ_ASSERT(NS_IsMainThread());
+  aOutSettings = *mSettings;
+}
+
+nsresult MediaEngineFakeVideoSource::Allocate(
+    const MediaTrackConstraints& aConstraints, const MediaEnginePrefs& aPrefs,
+    uint64_t aWindowID, const char** aOutBadConstraint) {
+  AssertIsOnOwningThread();
+
+  MOZ_ASSERT(mState == kReleased);
+
+  FlattenedConstraints c(aConstraints);
+
+  // emulator debug is very, very slow; reduce load on it with smaller/slower
+  // fake video
+  mOpts = aPrefs;
+  mOpts.mWidth =
+      c.mWidth.Get(aPrefs.mWidth ? aPrefs.mWidth :
+#ifdef DEBUG
+                                 MediaEnginePrefs::DEFAULT_43_VIDEO_WIDTH / 2
+#else
+                                 MediaEnginePrefs::DEFAULT_43_VIDEO_WIDTH
+#endif
+      );
+  mOpts.mHeight =
+      c.mHeight.Get(aPrefs.mHeight ? aPrefs.mHeight :
+#ifdef DEBUG
+                                   MediaEnginePrefs::DEFAULT_43_VIDEO_HEIGHT / 2
+#else
+                                   MediaEnginePrefs::DEFAULT_43_VIDEO_HEIGHT
+#endif
+      );
+  mOpts.mWidth =
+      std::max(VIDEO_WIDTH_MIN, std::min(mOpts.mWidth, VIDEO_WIDTH_MAX)) & ~1;
+  mOpts.mHeight =
+      std::max(VIDEO_HEIGHT_MIN, std::min(mOpts.mHeight, VIDEO_HEIGHT_MAX)) &
+      ~1;
+
+  NS_DispatchToMainThread(NS_NewRunnableFunction(
+      __func__, [settings = mSettings, frameRate = mOpts.mFPS,
+                 width = mOpts.mWidth, height = mOpts.mHeight]() {
+        settings->mFrameRate.Value() = frameRate;
+        settings->mWidth.Value() = width;
+        settings->mHeight.Value() = height;
+      }));
+
+  mState = kAllocated;
+  return NS_OK;
+}
+
+nsresult MediaEngineFakeVideoSource::Deallocate() {
+  AssertIsOnOwningThread();
+
+  MOZ_ASSERT(!mImage);
+  MOZ_ASSERT(mState == kStopped || mState == kAllocated);
+
+  if (mTrack) {
+    mTrack->End();
+    mTrack = nullptr;
+    mPrincipalHandle = PRINCIPAL_HANDLE_NONE;
+  }
+  mState = kReleased;
+  mImageContainer = nullptr;
+
+  return NS_OK;
+}
+
+static bool AllocateSolidColorFrame(layers::PlanarYCbCrData& aData, int aWidth,
+                                    int aHeight, int aY, int aCb, int aCr) {
+  MOZ_ASSERT(!(aWidth & 1));
+  MOZ_ASSERT(!(aHeight & 1));
+  // Allocate a single frame with a solid color
+  int yLen = aWidth * aHeight;
+  int cbLen = yLen >> 2;
+  int crLen = cbLen;
+  uint8_t* frame = (uint8_t*)malloc(yLen + cbLen + crLen);
+  if (!frame) {
+    return false;
+  }
+  memset(frame, aY, yLen);
+  memset(frame + yLen, aCb, cbLen);
+  memset(frame + yLen + cbLen, aCr, crLen);
+
+  aData.mYChannel = frame;
+  aData.mYStride = aWidth;
+  aData.mCbCrStride = aWidth >> 1;
+  aData.mCbChannel = frame + yLen;
+  aData.mCrChannel = aData.mCbChannel + cbLen;
+  aData.mPictureRect = IntRect(0, 0, aWidth, aHeight);
+  aData.mStereoMode = StereoMode::MONO;
+  aData.mYUVColorSpace = gfx::YUVColorSpace::BT601;
+  aData.mChromaSubsampling = gfx::ChromaSubsampling::HALF_WIDTH_AND_HEIGHT;
+  return true;
+}
+
+static void ReleaseFrame(layers::PlanarYCbCrData& aData) {
+  free(aData.mYChannel);
+}
+
+void MediaEngineFakeVideoSource::SetTrack(const RefPtr<MediaTrack>& aTrack,
+                                          const PrincipalHandle& aPrincipal) {
+  AssertIsOnOwningThread();
+
+  MOZ_ASSERT(mState == kAllocated);
+  MOZ_ASSERT(!mTrack);
+  MOZ_ASSERT(aTrack->AsSourceTrack());
+
+  mTrack = aTrack->AsSourceTrack();
+  mPrincipalHandle = aPrincipal;
+}
+
+nsresult MediaEngineFakeVideoSource::Start() {
+  AssertIsOnOwningThread();
+
+  MOZ_ASSERT(mState == kAllocated || mState == kStopped);
+  MOZ_ASSERT(mTrack, "SetTrack() must happen before Start()");
+
+  mTimer = NS_NewTimer(GetCurrentSerialEventTarget());
+  if (!mTimer) {
+    return NS_ERROR_FAILURE;
+  }
+
+  if (!mImageContainer) {
+    mImageContainer = MakeAndAddRef<layers::ImageContainer>(
+        layers::ImageContainer::ASYNCHRONOUS);
+  }
+
+  // Start timer for subsequent frames
+  uint32_t interval;
+#if defined(MOZ_WIDGET_ANDROID) && defined(DEBUG)
+  // emulator debug is very, very slow and has problems dealing with realtime
+  // audio inputs
+  interval = 10 * (1000 / mOpts.mFPS);
+#else
+  interval = 1000 / mOpts.mFPS;
+#endif
+  mTimer->InitWithNamedFuncCallback(
+      [](nsITimer* aTimer, void* aClosure) {
+        RefPtr<MediaEngineFakeVideoSource> source =
+            static_cast<MediaEngineFakeVideoSource*>(aClosure);
+        source->GenerateFrame();
+      },
+      this, interval, nsITimer::TYPE_REPEATING_SLACK,
+      "MediaEngineFakeVideoSource::GenerateFrame");
+
+  mState = kStarted;
+  return NS_OK;
+}
+
+nsresult MediaEngineFakeVideoSource::Stop() {
+  AssertIsOnOwningThread();
+
+  if (mState == kStopped || mState == kAllocated) {
+    return NS_OK;
+  }
+
+  MOZ_ASSERT(mState == kStarted);
+  MOZ_ASSERT(mTimer);
+  MOZ_ASSERT(mTrack);
+
+  mTimer->Cancel();
+  mTimer = nullptr;
+
+  mState = kStopped;
+
+  return NS_OK;
+}
+
+nsresult MediaEngineFakeVideoSource::Reconfigure(
+    const MediaTrackConstraints& aConstraints, const MediaEnginePrefs& aPrefs,
+    const char** aOutBadConstraint) {
+  return NS_OK;
+}
+
+void MediaEngineFakeVideoSource::GenerateFrame() {
+  AssertIsOnOwningThread();
+
+  // Update the target color
+  if (mCr <= 16) {
+    if (mCb < 240) {
+      mCb++;
+    } else {
+      mCr++;
+    }
+  } else if (mCb >= 240) {
+    if (mCr < 240) {
+      mCr++;
+    } else {
+      mCb--;
+    }
+  } else if (mCr >= 240) {
+    if (mCb > 16) {
+      mCb--;
+    } else {
+      mCr--;
+    }
+  } else {
+    mCr--;
+  }
+
+  // Allocate a single solid color image
+  RefPtr<layers::PlanarYCbCrImage> ycbcr_image =
+      mImageContainer->CreatePlanarYCbCrImage();
+  layers::PlanarYCbCrData data;
+  if (NS_WARN_IF(!AllocateSolidColorFrame(data, mOpts.mWidth, mOpts.mHeight,
+                                          0x80, mCb, mCr))) {
+    return;
+  }
+
+#ifdef MOZ_WEBRTC
+  uint64_t timestamp = PR_Now();
+  YuvStamper::Encode(mOpts.mWidth, mOpts.mHeight, mOpts.mWidth, data.mYChannel,
+                     reinterpret_cast<unsigned char*>(&timestamp),
+                     sizeof(timestamp), 0, 0);
+#endif
+
+  bool setData = ycbcr_image->CopyData(data);
+  MOZ_ASSERT(setData);
+
+  // SetData copies data, so we can free the frame
+  ReleaseFrame(data);
+
+  if (!setData) {
+    return;
+  }
+
+  VideoSegment segment;
+  segment.AppendFrame(ycbcr_image.forget(),
+                      gfx::IntSize(mOpts.mWidth, mOpts.mHeight),
+                      mPrincipalHandle);
+  mTrack->AppendData(&segment);
+}
+
+// This class is created on the media thread, as part of Start(), then entirely
+// self-sustained until destruction, just forwarding calls to Pull().
+class AudioSourcePullListener : public MediaTrackListener {
+ public:
+  AudioSourcePullListener(RefPtr<SourceMediaTrack> aTrack,
+                          const PrincipalHandle& aPrincipalHandle,
+                          uint32_t aFrequency)
+      : mTrack(std::move(aTrack)),
+        mPrincipalHandle(aPrincipalHandle),
+        mSineGenerator(MakeUnique<SineWaveGenerator<int16_t>>(
+            mTrack->mSampleRate, aFrequency)) {
+    MOZ_COUNT_CTOR(AudioSourcePullListener);
+  }
+
+  MOZ_COUNTED_DTOR(AudioSourcePullListener)
+
+  void NotifyPull(MediaTrackGraph* aGraph, TrackTime aEndOfAppendedData,
+                  TrackTime aDesiredTime) override;
+
+  const RefPtr<SourceMediaTrack> mTrack;
+  const PrincipalHandle mPrincipalHandle;
+  const UniquePtr<SineWaveGenerator<int16_t>> mSineGenerator;
+};
+
+/**
+ * Fake audio source.
+ */
+class MediaEngineFakeAudioSource : public MediaEngineSource {
+ public:
+  MediaEngineFakeAudioSource() = default;
+
+  static nsString GetUUID();
+  static nsString GetGroupId();
+
+  nsresult Allocate(const dom::MediaTrackConstraints& aConstraints,
+                    const MediaEnginePrefs& aPrefs, uint64_t aWindowID,
+                    const char** aOutBadConstraint) override;
+  void SetTrack(const RefPtr<MediaTrack>& aTrack,
+                const PrincipalHandle& aPrincipal) override;
+  nsresult Start() override;
+  nsresult Reconfigure(const dom::MediaTrackConstraints& aConstraints,
+                       const MediaEnginePrefs& aPrefs,
+                       const char** aOutBadConstraint) override;
+  nsresult Stop() override;
+  nsresult Deallocate() override;
+
+  bool IsFake() const override { return true; }
+
+  void GetSettings(dom::MediaTrackSettings& aOutSettings) const override;
+
+ protected:
+  ~MediaEngineFakeAudioSource() = default;
+
+  // Current state of this source.
+  MediaEngineSourceState mState = kReleased;
+  RefPtr<SourceMediaTrack> mTrack;
+  PrincipalHandle mPrincipalHandle = PRINCIPAL_HANDLE_NONE;
+  uint32_t mFrequency = 1000;
+  RefPtr<AudioSourcePullListener> mPullListener;
+};
+
+nsString MediaEngineFakeAudioSource::GetUUID() {
+  return u"B7CBD7C1-53EF-42F9-8353-73F61C70C092"_ns;
+}
+
+nsString MediaEngineFakeAudioSource::GetGroupId() {
+  return u"Fake Audio Group"_ns;
+}
+
+void MediaEngineFakeAudioSource::GetSettings(
+    MediaTrackSettings& aOutSettings) const {
+  MOZ_ASSERT(NS_IsMainThread());
+  aOutSettings.mAutoGainControl.Construct(false);
+  aOutSettings.mEchoCancellation.Construct(false);
+  aOutSettings.mNoiseSuppression.Construct(false);
+  aOutSettings.mChannelCount.Construct(1);
+}
+
+nsresult MediaEngineFakeAudioSource::Allocate(
+    const MediaTrackConstraints& aConstraints, const MediaEnginePrefs& aPrefs,
+    uint64_t aWindowID, const char** aOutBadConstraint) {
+  AssertIsOnOwningThread();
+
+  MOZ_ASSERT(mState == kReleased);
+
+  mFrequency = aPrefs.mFreq ? aPrefs.mFreq : 1000;
+
+  mState = kAllocated;
+  return NS_OK;
+}
+
+nsresult MediaEngineFakeAudioSource::Deallocate() {
+  AssertIsOnOwningThread();
+
+  MOZ_ASSERT(mState == kStopped || mState == kAllocated);
+
+  if (mTrack) {
+    mTrack->End();
+    mTrack = nullptr;
+    mPrincipalHandle = PRINCIPAL_HANDLE_NONE;
+  }
+  mState = kReleased;
+  return NS_OK;
+}
+
+void MediaEngineFakeAudioSource::SetTrack(const RefPtr<MediaTrack>& aTrack,
+                                          const PrincipalHandle& aPrincipal) {
+  AssertIsOnOwningThread();
+
+  MOZ_ASSERT(mState == kAllocated);
+  MOZ_ASSERT(!mTrack);
+  MOZ_ASSERT(aTrack->AsSourceTrack());
+
+  mTrack = aTrack->AsSourceTrack();
+  mPrincipalHandle = aPrincipal;
+}
+
+nsresult MediaEngineFakeAudioSource::Start() {
+  AssertIsOnOwningThread();
+
+  if (mState == kStarted) {
+    return NS_OK;
+  }
+
+  MOZ_ASSERT(mState == kAllocated || mState == kStopped);
+  MOZ_ASSERT(mTrack, "SetTrack() must happen before Start()");
+
+  if (!mPullListener) {
+    mPullListener = MakeAndAddRef<AudioSourcePullListener>(
+        mTrack, mPrincipalHandle, mFrequency);
+  }
+
+  mState = kStarted;
+
+  NS_DispatchToMainThread(NS_NewRunnableFunction(
+      __func__, [track = mTrack, listener = mPullListener]() {
+        if (track->IsDestroyed()) {
+          return;
+        }
+        track->AddListener(listener);
+        track->SetPullingEnabled(true);
+      }));
+
+  return NS_OK;
+}
+
+nsresult MediaEngineFakeAudioSource::Stop() {
+  AssertIsOnOwningThread();
+
+  if (mState == kStopped || mState == kAllocated) {
+    return NS_OK;
+  }
+  MOZ_ASSERT(mState == kStarted);
+  mState = kStopped;
+
+  NS_DispatchToMainThread(NS_NewRunnableFunction(
+      __func__, [track = mTrack, listener = std::move(mPullListener)]() {
+        if (track->IsDestroyed()) {
+          return;
+        }
+        track->RemoveListener(listener);
+        track->SetPullingEnabled(false);
+      }));
+  return NS_OK;
+}
+
+nsresult MediaEngineFakeAudioSource::Reconfigure(
+    const MediaTrackConstraints& aConstraints, const MediaEnginePrefs& aPrefs,
+    const char** aOutBadConstraint) {
+  return NS_OK;
+}
+
+void AudioSourcePullListener::NotifyPull(MediaTrackGraph* aGraph,
+                                         TrackTime aEndOfAppendedData,
+                                         TrackTime aDesiredTime) {
+  TRACE_COMMENT("SourceMediaTrack::NotifyPull", "SourceMediaTrack %p",
+                mTrack.get());
+  AudioSegment segment;
+  TrackTicks delta = aDesiredTime - aEndOfAppendedData;
+  CheckedInt<size_t> bufferSize(sizeof(int16_t));
+  bufferSize *= delta;
+  RefPtr<SharedBuffer> buffer = SharedBuffer::Create(bufferSize);
+  int16_t* dest = static_cast<int16_t*>(buffer->Data());
+  mSineGenerator->generate(dest, delta);
+  AutoTArray<const int16_t*, 1> channels;
+  channels.AppendElement(dest);
+  segment.AppendFrames(buffer.forget(), channels, delta, mPrincipalHandle);
+  mTrack->AppendData(&segment);
+}
+
+MediaEngineFake::MediaEngineFake() = default;
+MediaEngineFake::~MediaEngineFake() = default;
+
+void MediaEngineFake::EnumerateDevices(
+    MediaSourceEnum aMediaSource, MediaSinkEnum aMediaSink,
+    nsTArray<RefPtr<MediaDevice>>* aDevices) {
+  AssertIsOnOwningThread();
+  using IsScary = MediaDevice::IsScary;
+  using OsPromptable = MediaDevice::OsPromptable;
+
+  if (aMediaSink == MediaSinkEnum::Speaker) {
+    NS_WARNING("No default implementation for MediaSinkEnum::Speaker");
+  }
+
+  switch (aMediaSource) {
+    case MediaSourceEnum::Camera: {
+      nsString name = FakeVideoName();
+      aDevices->EmplaceBack(
+          new MediaDevice(this, aMediaSource, name, /*aRawId=*/name,
+                          MediaEngineFakeVideoSource::GetGroupId(), IsScary::No,
+                          OsPromptable::No));
+      return;
+    }
+    case MediaSourceEnum::Microphone:
+      aDevices->EmplaceBack(
+          new MediaDevice(this, aMediaSource, u"Default Audio Device"_ns,
+                          MediaEngineFakeAudioSource::GetUUID(),
+                          MediaEngineFakeAudioSource::GetGroupId(), IsScary::No,
+                          OsPromptable::No));
+      return;
+    default:
+      MOZ_ASSERT_UNREACHABLE("Unsupported source type");
+      return;
+  }
+}
+
+RefPtr<MediaEngineSource> MediaEngineFake::CreateSource(
+    const MediaDevice* aMediaDevice) {
+  MOZ_ASSERT(aMediaDevice->mEngine == this);
+  switch (aMediaDevice->mMediaSource) {
+    case MediaSourceEnum::Camera:
+      return new MediaEngineFakeVideoSource();
+    case MediaSourceEnum::Microphone:
+      return new MediaEngineFakeAudioSource();
+    default:
+      MOZ_ASSERT_UNREACHABLE("Unsupported source type");
+      return nullptr;
+  }
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/MediaEngineFake.h b/dom/media/webrtc/MediaEngineFake.h
new file mode 100644
index 0000000000..611529a067
--- /dev/null
+++ b/dom/media/webrtc/MediaEngineFake.h
@@ -0,0 +1,40 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef MEDIAENGINEFAKE_H_
+#define MEDIAENGINEFAKE_H_
+
+#include "nsTArrayForwardDeclare.h"
+#include "MediaEngine.h"
+
+namespace mozilla {
+
+template <typename...>
+class MediaEventProducer;
+
+/**
+ * The fake implementation of the MediaEngine interface.
+ */
+class MediaEngineFake : public MediaEngine {
+ public:
+  MediaEngineFake();
+
+  void EnumerateDevices(dom::MediaSourceEnum, MediaSinkEnum,
+                        nsTArray<RefPtr<MediaDevice>>*) override;
+  void Shutdown() override {}
+  RefPtr<MediaEngineSource> CreateSource(const MediaDevice* aDevice) override;
+
+  MediaEventSource<void>& DeviceListChangeEvent() override {
+    return mDeviceListChangeEvent;
+  }
+  bool IsFake() const override { return true; }
+
+ private:
+  ~MediaEngineFake();
+  MediaEventProducer<void> mDeviceListChangeEvent;
+};
+
+}  // namespace mozilla
+
+#endif /* NSMEDIAENGINEFAKE_H_ */
diff --git a/dom/media/webrtc/MediaEnginePrefs.h b/dom/media/webrtc/MediaEnginePrefs.h
new file mode 100644
index 0000000000..6c9775a238
--- /dev/null
+++ b/dom/media/webrtc/MediaEnginePrefs.h
@@ -0,0 +1,101 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef MediaEnginePrefs_h
+#define MediaEnginePrefs_h
+
+#include <stdint.h>
+#include <string.h>
+
+namespace mozilla {
+
+/**
+ * Video source and friends.
+ */
+class MediaEnginePrefs {
+ public:
+  static const int DEFAULT_VIDEO_FPS = 30;
+  static const int DEFAULT_43_VIDEO_WIDTH = 640;
+  static const int DEFAULT_43_VIDEO_HEIGHT = 480;
+  static const int DEFAULT_169_VIDEO_WIDTH = 1280;
+  static const int DEFAULT_169_VIDEO_HEIGHT = 720;
+
+  MediaEnginePrefs()
+      : mWidth(0),
+        mHeight(0),
+        mFPS(0),
+        mFreq(0),
+        mAecOn(false),
+        mUseAecMobile(false),
+        mAgcOn(false),
+        mHPFOn(false),
+        mNoiseOn(false),
+        mTransientOn(false),
+        mResidualEchoOn(false),
+        mAgc2Forced(false),
+        mAgc(0),
+        mNoise(0),
+        mChannels(0) {}
+
+  int32_t mWidth;
+  int32_t mHeight;
+  int32_t mFPS;
+  int32_t mFreq;  // for test tones (fake:true)
+  bool mAecOn;
+  bool mUseAecMobile;
+  bool mAgcOn;
+  bool mHPFOn;
+  bool mNoiseOn;
+  bool mTransientOn;
+  bool mResidualEchoOn;
+  bool mAgc2Forced;
+  int32_t mAgc;
+  int32_t mNoise;
+  int32_t mChannels;
+
+  bool operator==(const MediaEnginePrefs& aRhs) {
+    return memcmp(this, &aRhs, sizeof(MediaEnginePrefs)) == 0;
+  };
+
+  // mWidth and/or mHeight may be zero (=adaptive default), so use functions.
+
+  int32_t GetWidth(bool aHD = false) const {
+    return mWidth ? mWidth
+                  : (mHeight ? (mHeight * GetDefWidth(aHD)) / GetDefHeight(aHD)
+                             : GetDefWidth(aHD));
+  }
+
+  int32_t GetHeight(bool aHD = false) const {
+    return mHeight ? mHeight
+                   : (mWidth ? (mWidth * GetDefHeight(aHD)) / GetDefWidth(aHD)
+                             : GetDefHeight(aHD));
+  }
+
+ private:
+  static int32_t GetDefWidth(bool aHD = false) {
+    // It'd be nice if we could use the ternary operator here, but we can't
+    // because of bug 1002729.
+    if (aHD) {
+      return DEFAULT_169_VIDEO_WIDTH;
+    }
+
+    return DEFAULT_43_VIDEO_WIDTH;
+  }
+
+  static int32_t GetDefHeight(bool aHD = false) {
+    // It'd be nice if we could use the ternary operator here, but we can't
+    // because of bug 1002729.
+    if (aHD) {
+      return DEFAULT_169_VIDEO_HEIGHT;
+    }
+
+    return DEFAULT_43_VIDEO_HEIGHT;
+  }
+};
+
+}  // namespace mozilla
+
+#endif  // MediaEnginePrefs_h
diff --git a/dom/media/webrtc/MediaEngineRemoteVideoSource.cpp b/dom/media/webrtc/MediaEngineRemoteVideoSource.cpp
new file mode 100644
index 0000000000..ce9f4ad9a8
--- /dev/null
+++ b/dom/media/webrtc/MediaEngineRemoteVideoSource.cpp
@@ -0,0 +1,907 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "MediaEngineRemoteVideoSource.h"
+
+#include "CamerasChild.h"
+#include "MediaManager.h"
+#include "MediaTrackConstraints.h"
+#include "mozilla/dom/MediaTrackSettingsBinding.h"
+#include "mozilla/ErrorNames.h"
+#include "mozilla/gfx/Point.h"
+#include "mozilla/RefPtr.h"
+#include "PerformanceRecorder.h"
+#include "Tracing.h"
+#include "VideoFrameUtils.h"
+#include "VideoUtils.h"
+#include "ImageContainer.h"
+#include "common_video/include/video_frame_buffer.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+
+namespace mozilla {
+
+extern LazyLogModule gMediaManagerLog;
+#define LOG(...) MOZ_LOG(gMediaManagerLog, LogLevel::Debug, (__VA_ARGS__))
+#define LOG_FRAME(...) \
+  MOZ_LOG(gMediaManagerLog, LogLevel::Verbose, (__VA_ARGS__))
+
+using dom::ConstrainLongRange;
+using dom::MediaSourceEnum;
+using dom::MediaTrackConstraints;
+using dom::MediaTrackConstraintSet;
+using dom::MediaTrackSettings;
+using dom::VideoFacingModeEnum;
+
+/* static */
+camera::CaptureEngine MediaEngineRemoteVideoSource::CaptureEngine(
+    MediaSourceEnum aMediaSource) {
+  switch (aMediaSource) {
+    case MediaSourceEnum::Browser:
+      return camera::BrowserEngine;
+    case MediaSourceEnum::Camera:
+      return camera::CameraEngine;
+    case MediaSourceEnum::Screen:
+      return camera::ScreenEngine;
+    case MediaSourceEnum::Window:
+      return camera::WinEngine;
+    default:
+      MOZ_CRASH();
+  }
+}
+
+static Maybe<VideoFacingModeEnum> GetFacingMode(const nsString& aDeviceName) {
+  // Set facing mode based on device name.
+#if defined(ANDROID)
+  // Names are generated. Example: "Camera 0, Facing back, Orientation 90"
+  //
+  // See media/webrtc/trunk/webrtc/modules/video_capture/android/java/src/org/
+  // webrtc/videoengine/VideoCaptureDeviceInfoAndroid.java
+
+  if (aDeviceName.Find(u"Facing back"_ns) != kNotFound) {
+    return Some(VideoFacingModeEnum::Environment);
+  }
+  if (aDeviceName.Find(u"Facing front"_ns) != kNotFound) {
+    return Some(VideoFacingModeEnum::User);
+  }
+#endif  // ANDROID
+#ifdef XP_MACOSX
+  // Kludge to test user-facing cameras on OSX.
+  if (aDeviceName.Find(u"Face"_ns) != -1) {
+    return Some(VideoFacingModeEnum::User);
+  }
+#endif
+#ifdef XP_WIN
+  // The cameras' name of Surface book are "Microsoft Camera Front" and
+  // "Microsoft Camera Rear" respectively.
+
+  if (aDeviceName.Find(u"Front"_ns) != kNotFound) {
+    return Some(VideoFacingModeEnum::User);
+  }
+  if (aDeviceName.Find(u"Rear"_ns) != kNotFound) {
+    return Some(VideoFacingModeEnum::Environment);
+  }
+#endif  // WINDOWS
+
+  return Nothing();
+}
+
+MediaEngineRemoteVideoSource::MediaEngineRemoteVideoSource(
+    const MediaDevice* aMediaDevice)
+    : mCapEngine(CaptureEngine(aMediaDevice->mMediaSource)),
+      mTrackingId(CaptureEngineToTrackingSourceStr(mCapEngine), 0),
+      mMutex("MediaEngineRemoteVideoSource::mMutex"),
+      mRescalingBufferPool(/* zero_initialize */ false,
+                           /* max_number_of_buffers */ 1),
+      mSettingsUpdatedByFrame(MakeAndAddRef<media::Refcountable<AtomicBool>>()),
+      mSettings(MakeAndAddRef<media::Refcountable<MediaTrackSettings>>()),
+      mFirstFramePromise(mFirstFramePromiseHolder.Ensure(__func__)),
+      mMediaDevice(aMediaDevice),
+      mDeviceUUID(NS_ConvertUTF16toUTF8(aMediaDevice->mRawID)) {
+  LOG("%s", __PRETTY_FUNCTION__);
+  mSettings->mWidth.Construct(0);
+  mSettings->mHeight.Construct(0);
+  mSettings->mFrameRate.Construct(0);
+  if (mCapEngine == camera::CameraEngine) {
+    // Only cameras can have a facing mode.
+    Maybe<VideoFacingModeEnum> facingMode =
+        GetFacingMode(mMediaDevice->mRawName);
+    if (facingMode.isSome()) {
+      NS_ConvertASCIItoUTF16 facingString(
+          dom::VideoFacingModeEnumValues::GetString(*facingMode));
+      mSettings->mFacingMode.Construct(facingString);
+      mFacingMode.emplace(facingString);
+    }
+  }
+}
+
+MediaEngineRemoteVideoSource::~MediaEngineRemoteVideoSource() {
+  mFirstFramePromiseHolder.RejectIfExists(NS_ERROR_ABORT, __func__);
+}
+
+nsresult MediaEngineRemoteVideoSource::Allocate(
+    const MediaTrackConstraints& aConstraints, const MediaEnginePrefs& aPrefs,
+    uint64_t aWindowID, const char** aOutBadConstraint) {
+  LOG("%s", __PRETTY_FUNCTION__);
+  AssertIsOnOwningThread();
+
+  MOZ_ASSERT(mState == kReleased);
+
+  NormalizedConstraints constraints(aConstraints);
+  webrtc::CaptureCapability newCapability;
+  LOG("ChooseCapability(kFitness) for mCapability (Allocate) ++");
+  if (!ChooseCapability(constraints, aPrefs, newCapability, kFitness)) {
+    *aOutBadConstraint =
+        MediaConstraintsHelper::FindBadConstraint(constraints, mMediaDevice);
+    return NS_ERROR_FAILURE;
+  }
+  LOG("ChooseCapability(kFitness) for mCapability (Allocate) --");
+
+  mCaptureId =
+      camera::GetChildAndCall(&camera::CamerasChild::AllocateCapture,
+                              mCapEngine, mDeviceUUID.get(), aWindowID);
+  if (mCaptureId < 0) {
+    return NS_ERROR_FAILURE;
+  }
+
+  {
+    MutexAutoLock lock(mMutex);
+    mState = kAllocated;
+    mCapability = newCapability;
+    mTrackingId =
+        TrackingId(CaptureEngineToTrackingSourceStr(mCapEngine), mCaptureId);
+  }
+
+  LOG("Video device %d allocated", mCaptureId);
+  return NS_OK;
+}
+
+nsresult MediaEngineRemoteVideoSource::Deallocate() {
+  LOG("%s", __PRETTY_FUNCTION__);
+  AssertIsOnOwningThread();
+
+  MOZ_ASSERT(mState == kStopped || mState == kAllocated);
+
+  if (mTrack) {
+    mTrack->End();
+  }
+
+  {
+    MutexAutoLock lock(mMutex);
+
+    mTrack = nullptr;
+    mPrincipal = PRINCIPAL_HANDLE_NONE;
+    mState = kReleased;
+  }
+
+  // Stop() has stopped capture synchronously on the media thread before we get
+  // here, so there are no longer any callbacks on an IPC thread accessing
+  // mImageContainer or mRescalingBufferPool.
+  mImageContainer = nullptr;
+  mRescalingBufferPool.Release();
+
+  LOG("Video device %d deallocated", mCaptureId);
+
+  if (camera::GetChildAndCall(&camera::CamerasChild::ReleaseCapture, mCapEngine,
+                              mCaptureId)) {
+    // Failure can occur when the parent process is shutting down.
+    return NS_ERROR_FAILURE;
+  }
+  return NS_OK;
+}
+
+void MediaEngineRemoteVideoSource::SetTrack(const RefPtr<MediaTrack>& aTrack,
+                                            const PrincipalHandle& aPrincipal) {
+  LOG("%s", __PRETTY_FUNCTION__);
+  AssertIsOnOwningThread();
+
+  MOZ_ASSERT(mState == kAllocated);
+  MOZ_ASSERT(!mTrack);
+  MOZ_ASSERT(aTrack);
+  MOZ_ASSERT(aTrack->AsSourceTrack());
+
+  if (!mImageContainer) {
+    mImageContainer = MakeAndAddRef<layers::ImageContainer>(
+        layers::ImageContainer::ASYNCHRONOUS);
+  }
+
+  {
+    MutexAutoLock lock(mMutex);
+    mTrack = aTrack->AsSourceTrack();
+    mPrincipal = aPrincipal;
+  }
+}
+
+nsresult MediaEngineRemoteVideoSource::Start() {
+  LOG("%s", __PRETTY_FUNCTION__);
+  AssertIsOnOwningThread();
+
+  MOZ_ASSERT(mState == kAllocated || mState == kStopped);
+  MOZ_ASSERT(mTrack);
+
+  {
+    MutexAutoLock lock(mMutex);
+    mState = kStarted;
+  }
+
+  mSettingsUpdatedByFrame->mValue = false;
+
+  if (camera::GetChildAndCall(&camera::CamerasChild::StartCapture, mCapEngine,
+                              mCaptureId, mCapability, this)) {
+    LOG("StartCapture failed");
+    MutexAutoLock lock(mMutex);
+    mState = kStopped;
+    return NS_ERROR_FAILURE;
+  }
+
+  NS_DispatchToMainThread(NS_NewRunnableFunction(
+      "MediaEngineRemoteVideoSource::SetLastCapability",
+      [settings = mSettings, updated = mSettingsUpdatedByFrame,
+       capEngine = mCapEngine, cap = mCapability]() mutable {
+        switch (capEngine) {
+          case camera::ScreenEngine:
+          case camera::WinEngine:
+            // Undo the hack where ideal and max constraints are crammed
+            // together in mCapability for consumption by low-level code. We
+            // don't actually know the real resolution yet, so report min(ideal,
+            // max) for now.
+            // TODO: This can be removed in bug 1453269.
+            cap.width = std::min(cap.width >> 16, cap.width & 0xffff);
+            cap.height = std::min(cap.height >> 16, cap.height & 0xffff);
+            break;
+          default:
+            break;
+        }
+
+        if (!updated->mValue) {
+          settings->mWidth.Value() = cap.width;
+          settings->mHeight.Value() = cap.height;
+        }
+        settings->mFrameRate.Value() = cap.maxFPS;
+      }));
+
+  return NS_OK;
+}
+
+nsresult MediaEngineRemoteVideoSource::FocusOnSelectedSource() {
+  LOG("%s", __PRETTY_FUNCTION__);
+  AssertIsOnOwningThread();
+
+  int result;
+  result = camera::GetChildAndCall(&camera::CamerasChild::FocusOnSelectedSource,
+                                   mCapEngine, mCaptureId);
+  return result == 0 ? NS_OK : NS_ERROR_FAILURE;
+}
+
+nsresult MediaEngineRemoteVideoSource::Stop() {
+  LOG("%s", __PRETTY_FUNCTION__);
+  AssertIsOnOwningThread();
+
+  if (mState == kStopped || mState == kAllocated) {
+    return NS_OK;
+  }
+
+  MOZ_ASSERT(mState == kStarted);
+
+  if (camera::GetChildAndCall(&camera::CamerasChild::StopCapture, mCapEngine,
+                              mCaptureId)) {
+    // Failure can occur when the parent process is shutting down.
+    return NS_ERROR_FAILURE;
+  }
+
+  {
+    MutexAutoLock lock(mMutex);
+    mState = kStopped;
+  }
+
+  return NS_OK;
+}
+
+nsresult MediaEngineRemoteVideoSource::Reconfigure(
+    const MediaTrackConstraints& aConstraints, const MediaEnginePrefs& aPrefs,
+    const char** aOutBadConstraint) {
+  LOG("%s", __PRETTY_FUNCTION__);
+  AssertIsOnOwningThread();
+
+  NormalizedConstraints constraints(aConstraints);
+  webrtc::CaptureCapability newCapability;
+  LOG("ChooseCapability(kFitness) for mTargetCapability (Reconfigure) ++");
+  if (!ChooseCapability(constraints, aPrefs, newCapability, kFitness)) {
+    *aOutBadConstraint =
+        MediaConstraintsHelper::FindBadConstraint(constraints, mMediaDevice);
+    return NS_ERROR_INVALID_ARG;
+  }
+  LOG("ChooseCapability(kFitness) for mTargetCapability (Reconfigure) --");
+
+  if (mCapability == newCapability) {
+    return NS_OK;
+  }
+
+  bool started = mState == kStarted;
+  if (started) {
+    nsresult rv = Stop();
+    if (NS_WARN_IF(NS_FAILED(rv))) {
+      nsAutoCString name;
+      GetErrorName(rv, name);
+      LOG("Video source %p for video device %d Reconfigure() failed "
+          "unexpectedly in Stop(). rv=%s",
+          this, mCaptureId, name.Data());
+      return NS_ERROR_UNEXPECTED;
+    }
+  }
+
+  {
+    MutexAutoLock lock(mMutex);
+    // Start() applies mCapability on the device.
+    mCapability = newCapability;
+  }
+
+  if (started) {
+    nsresult rv = Start();
+    if (NS_WARN_IF(NS_FAILED(rv))) {
+      nsAutoCString name;
+      GetErrorName(rv, name);
+      LOG("Video source %p for video device %d Reconfigure() failed "
+          "unexpectedly in Start(). rv=%s",
+          this, mCaptureId, name.Data());
+      return NS_ERROR_UNEXPECTED;
+    }
+  }
+
+  return NS_OK;
+}
+
+size_t MediaEngineRemoteVideoSource::NumCapabilities() const {
+  AssertIsOnOwningThread();
+
+  if (!mCapabilities.IsEmpty()) {
+    return mCapabilities.Length();
+  }
+
+  int num = camera::GetChildAndCall(&camera::CamerasChild::NumberOfCapabilities,
+                                    mCapEngine, mDeviceUUID.get());
+  if (num > 0) {
+    mCapabilities.SetLength(num);
+  } else {
+    // The default for devices that don't return discrete capabilities: treat
+    // them as supporting all capabilities orthogonally. E.g. screensharing.
+    // CaptureCapability defaults key values to 0, which means accept any value.
+    mCapabilities.AppendElement(MakeUnique<webrtc::CaptureCapability>());
+    mCapabilitiesAreHardcoded = true;
+  }
+
+  return mCapabilities.Length();
+}
+
+webrtc::CaptureCapability& MediaEngineRemoteVideoSource::GetCapability(
+    size_t aIndex) const {
+  AssertIsOnOwningThread();
+  MOZ_RELEASE_ASSERT(aIndex < mCapabilities.Length());
+  if (!mCapabilities[aIndex]) {
+    mCapabilities[aIndex] = MakeUnique<webrtc::CaptureCapability>();
+    camera::GetChildAndCall(&camera::CamerasChild::GetCaptureCapability,
+                            mCapEngine, mDeviceUUID.get(), aIndex,
+                            mCapabilities[aIndex].get());
+  }
+  return *mCapabilities[aIndex];
+}
+
+const TrackingId& MediaEngineRemoteVideoSource::GetTrackingId() const {
+  AssertIsOnOwningThread();
+  MOZ_ASSERT(mState != kReleased);
+  return mTrackingId;
+}
+
+int MediaEngineRemoteVideoSource::DeliverFrame(
+    uint8_t* aBuffer, const camera::VideoFrameProperties& aProps) {
+  // Cameras IPC thread - take great care with accessing members!
+
+  Maybe<int32_t> req_max_width;
+  Maybe<int32_t> req_max_height;
+  Maybe<int32_t> req_ideal_width;
+  Maybe<int32_t> req_ideal_height;
+  {
+    MutexAutoLock lock(mMutex);
+    MOZ_ASSERT(mState == kStarted);
+    // TODO: These can be removed in bug 1453269.
+    const int32_t max_width = mCapability.width & 0xffff;
+    const int32_t max_height = mCapability.height & 0xffff;
+    const int32_t ideal_width = (mCapability.width >> 16) & 0xffff;
+    const int32_t ideal_height = (mCapability.height >> 16) & 0xffff;
+
+    req_max_width = max_width ? Some(max_width) : Nothing();
+    req_max_height = max_height ? Some(max_height) : Nothing();
+    req_ideal_width = ideal_width ? Some(ideal_width) : Nothing();
+    req_ideal_height = ideal_height ? Some(ideal_height) : Nothing();
+    if (!mFrameDeliveringTrackingId) {
+      mFrameDeliveringTrackingId = Some(mTrackingId);
+    }
+  }
+
+  // This is only used in the case of screen sharing, see bug 1453269.
+
+  if (aProps.rotation() == 90 || aProps.rotation() == 270) {
+    // This frame is rotated, so what was negotiated as width is now height,
+    // and vice versa.
+    std::swap(req_max_width, req_max_height);
+    std::swap(req_ideal_width, req_ideal_height);
+  }
+
+  int32_t dst_max_width =
+      std::min(aProps.width(), req_max_width.valueOr(aProps.width()));
+  int32_t dst_max_height =
+      std::min(aProps.height(), req_max_height.valueOr(aProps.height()));
+  // This logic works for both camera and screen sharing case.
+  // for camera case, req_ideal_width and req_ideal_height are absent.
+  int32_t dst_width = req_ideal_width.valueOr(aProps.width());
+  int32_t dst_height = req_ideal_height.valueOr(aProps.height());
+
+  if (!req_ideal_width && req_ideal_height) {
+    dst_width = *req_ideal_height * aProps.width() / aProps.height();
+  } else if (!req_ideal_height && req_ideal_width) {
+    dst_height = *req_ideal_width * aProps.height() / aProps.width();
+  }
+  dst_width = std::min(dst_width, dst_max_width);
+  dst_height = std::min(dst_height, dst_max_height);
+
+  // Apply scaling for screen sharing, see bug 1453269.
+  switch (mCapEngine) {
+    case camera::ScreenEngine:
+    case camera::WinEngine: {
+      // scale to average of portrait and landscape
+      float scale_width = (float)dst_width / (float)aProps.width();
+      float scale_height = (float)dst_height / (float)aProps.height();
+      float scale = (scale_width + scale_height) / 2;
+      // If both req_ideal_width & req_ideal_height are absent, scale is 1, but
+      // if one is present and the other not, scale precisely to the one present
+      if (!req_ideal_width) {
+        scale = scale_height;
+      } else if (!req_ideal_height) {
+        scale = scale_width;
+      }
+      dst_width = int32_t(scale * (float)aProps.width());
+      dst_height = int32_t(scale * (float)aProps.height());
+
+      // if scaled rectangle exceeds max rectangle, scale to minimum of portrait
+      // and landscape
+      if (dst_width > dst_max_width || dst_height > dst_max_height) {
+        scale_width = (float)dst_max_width / (float)dst_width;
+        scale_height = (float)dst_max_height / (float)dst_height;
+        scale = std::min(scale_width, scale_height);
+        dst_width = int32_t(scale * dst_width);
+        dst_height = int32_t(scale * dst_height);
+      }
+      break;
+    }
+    default: {
+      break;
+    }
+  }
+
+  // Ensure width and height are at least two. Smaller frames can lead to
+  // problems with scaling and video encoding.
+  dst_width = std::max(2, dst_width);
+  dst_height = std::max(2, dst_height);
+
+  std::function<void()> callback_unused = []() {};
+  rtc::scoped_refptr<webrtc::I420BufferInterface> buffer =
+      webrtc::WrapI420Buffer(
+          aProps.width(), aProps.height(), aBuffer, aProps.yStride(),
+          aBuffer + aProps.yAllocatedSize(), aProps.uStride(),
+          aBuffer + aProps.yAllocatedSize() + aProps.uAllocatedSize(),
+          aProps.vStride(), callback_unused);
+
+  if ((dst_width != aProps.width() || dst_height != aProps.height()) &&
+      dst_width <= aProps.width() && dst_height <= aProps.height()) {
+    PerformanceRecorder<CopyVideoStage> rec("MERVS::CropAndScale"_ns,
+                                            *mFrameDeliveringTrackingId,
+                                            dst_width, dst_height);
+    // Destination resolution is smaller than source buffer. We'll rescale.
+    rtc::scoped_refptr<webrtc::I420Buffer> scaledBuffer =
+        mRescalingBufferPool.CreateI420Buffer(dst_width, dst_height);
+    if (!scaledBuffer) {
+      MOZ_ASSERT_UNREACHABLE(
+          "We might fail to allocate a buffer, but with this "
+          "being a recycling pool that shouldn't happen");
+      return 0;
+    }
+    scaledBuffer->CropAndScaleFrom(*buffer);
+    buffer = scaledBuffer;
+    rec.Record();
+  }
+
+  layers::PlanarYCbCrData data;
+  data.mYChannel = const_cast<uint8_t*>(buffer->DataY());
+  data.mYStride = buffer->StrideY();
+  MOZ_ASSERT(buffer->StrideU() == buffer->StrideV());
+  data.mCbCrStride = buffer->StrideU();
+  data.mCbChannel = const_cast<uint8_t*>(buffer->DataU());
+  data.mCrChannel = const_cast<uint8_t*>(buffer->DataV());
+  data.mPictureRect = gfx::IntRect(0, 0, buffer->width(), buffer->height());
+  data.mYUVColorSpace = gfx::YUVColorSpace::BT601;
+  data.mChromaSubsampling = gfx::ChromaSubsampling::HALF_WIDTH_AND_HEIGHT;
+
+  RefPtr<layers::PlanarYCbCrImage> image;
+  {
+    PerformanceRecorder<CopyVideoStage> rec(
+        "MERVS::Copy"_ns, *mFrameDeliveringTrackingId, dst_width, dst_height);
+    image = mImageContainer->CreatePlanarYCbCrImage();
+    if (!image->CopyData(data)) {
+      MOZ_ASSERT_UNREACHABLE(
+          "We might fail to allocate a buffer, but with this "
+          "being a recycling container that shouldn't happen");
+      return 0;
+    }
+    rec.Record();
+  }
+
+#ifdef DEBUG
+  static uint32_t frame_num = 0;
+  LOG_FRAME(
+      "frame %d (%dx%d)->(%dx%d); rotation %d, timeStamp %u, ntpTimeMs %" PRIu64
+      ", renderTimeMs %" PRIu64,
+      frame_num++, aProps.width(), aProps.height(), dst_width, dst_height,
+      aProps.rotation(), aProps.timeStamp(), aProps.ntpTimeMs(),
+      aProps.renderTimeMs());
+#endif
+
+  if (mImageSize.width != dst_width || mImageSize.height != dst_height) {
+    NS_DispatchToMainThread(NS_NewRunnableFunction(
+        "MediaEngineRemoteVideoSource::FrameSizeChange",
+        [settings = mSettings, updated = mSettingsUpdatedByFrame,
+         holder = std::move(mFirstFramePromiseHolder), dst_width,
+         dst_height]() mutable {
+          settings->mWidth.Value() = dst_width;
+          settings->mHeight.Value() = dst_height;
+          updated->mValue = true;
+          // Since mImageSize was initialized to (0,0), we end up here on the
+          // arrival of the first frame. We resolve the promise representing
+          // arrival of first frame, after correct settings values have been
+          // made available (Resolve() is idempotent if already resolved).
+          holder.ResolveIfExists(true, __func__);
+        }));
+  }
+
+  {
+    MutexAutoLock lock(mMutex);
+    MOZ_ASSERT(mState == kStarted);
+    VideoSegment segment;
+    mImageSize = image->GetSize();
+    segment.AppendFrame(image.forget(), mImageSize, mPrincipal);
+    mTrack->AppendData(&segment);
+  }
+
+  return 0;
+}
+
+uint32_t MediaEngineRemoteVideoSource::GetDistance(
+    const webrtc::CaptureCapability& aCandidate,
+    const NormalizedConstraintSet& aConstraints,
+    const DistanceCalculation aCalculate) const {
+  if (aCalculate == kFeasibility) {
+    return GetFeasibilityDistance(aCandidate, aConstraints);
+  }
+  return GetFitnessDistance(aCandidate, aConstraints);
+}
+
+uint32_t MediaEngineRemoteVideoSource::GetFitnessDistance(
+    const webrtc::CaptureCapability& aCandidate,
+    const NormalizedConstraintSet& aConstraints) const {
+  AssertIsOnOwningThread();
+
+  // Treat width|height|frameRate == 0 on capability as "can do any".
+  // This allows for orthogonal capabilities that are not in discrete steps.
+
+  typedef MediaConstraintsHelper H;
+  uint64_t distance =
+      uint64_t(H::FitnessDistance(mFacingMode, aConstraints.mFacingMode)) +
+      uint64_t(aCandidate.width ? H::FitnessDistance(int32_t(aCandidate.width),
+                                                     aConstraints.mWidth)
+                                : 0) +
+      uint64_t(aCandidate.height
+                   ? H::FitnessDistance(int32_t(aCandidate.height),
+                                        aConstraints.mHeight)
+                   : 0) +
+      uint64_t(aCandidate.maxFPS ? H::FitnessDistance(double(aCandidate.maxFPS),
+                                                      aConstraints.mFrameRate)
+                                 : 0);
+  return uint32_t(std::min(distance, uint64_t(UINT32_MAX)));
+}
+
+uint32_t MediaEngineRemoteVideoSource::GetFeasibilityDistance(
+    const webrtc::CaptureCapability& aCandidate,
+    const NormalizedConstraintSet& aConstraints) const {
+  AssertIsOnOwningThread();
+
+  // Treat width|height|frameRate == 0 on capability as "can do any".
+  // This allows for orthogonal capabilities that are not in discrete steps.
+
+  typedef MediaConstraintsHelper H;
+  uint64_t distance =
+      uint64_t(H::FitnessDistance(mFacingMode, aConstraints.mFacingMode)) +
+      uint64_t(aCandidate.width
+                   ? H::FeasibilityDistance(int32_t(aCandidate.width),
+                                            aConstraints.mWidth)
+                   : 0) +
+      uint64_t(aCandidate.height
+                   ? H::FeasibilityDistance(int32_t(aCandidate.height),
+                                            aConstraints.mHeight)
+                   : 0) +
+      uint64_t(aCandidate.maxFPS
+                   ? H::FeasibilityDistance(double(aCandidate.maxFPS),
+                                            aConstraints.mFrameRate)
+                   : 0);
+  return uint32_t(std::min(distance, uint64_t(UINT32_MAX)));
+}
+
+// Find best capability by removing inferiors. May leave >1 of equal distance
+
+/* static */
+void MediaEngineRemoteVideoSource::TrimLessFitCandidates(
+    nsTArray<CapabilityCandidate>& aSet) {
+  uint32_t best = UINT32_MAX;
+  for (auto& candidate : aSet) {
+    if (best > candidate.mDistance) {
+      best = candidate.mDistance;
+    }
+  }
+  aSet.RemoveElementsBy(
+      [best](const auto& set) { return set.mDistance > best; });
+  MOZ_ASSERT(aSet.Length());
+}
+
+uint32_t MediaEngineRemoteVideoSource::GetBestFitnessDistance(
+    const nsTArray<const NormalizedConstraintSet*>& aConstraintSets) const {
+  AssertIsOnOwningThread();
+
+  size_t num = NumCapabilities();
+  nsTArray<CapabilityCandidate> candidateSet;
+  for (size_t i = 0; i < num; i++) {
+    candidateSet.AppendElement(CapabilityCandidate(GetCapability(i)));
+  }
+
+  bool first = true;
+  for (const NormalizedConstraintSet* ns : aConstraintSets) {
+    for (size_t i = 0; i < candidateSet.Length();) {
+      auto& candidate = candidateSet[i];
+      uint32_t distance = GetFitnessDistance(candidate.mCapability, *ns);
+      if (distance == UINT32_MAX) {
+        candidateSet.RemoveElementAt(i);
+      } else {
+        ++i;
+        if (first) {
+          candidate.mDistance = distance;
+        }
+      }
+    }
+    first = false;
+  }
+  if (!candidateSet.Length()) {
+    return UINT32_MAX;
+  }
+  TrimLessFitCandidates(candidateSet);
+  return candidateSet[0].mDistance;
+}
+
+static const char* ConvertVideoTypeToCStr(webrtc::VideoType aType) {
+  switch (aType) {
+    case webrtc::VideoType::kI420:
+      return "I420";
+    case webrtc::VideoType::kIYUV:
+    case webrtc::VideoType::kYV12:
+      return "YV12";
+    case webrtc::VideoType::kRGB24:
+      return "24BG";
+    case webrtc::VideoType::kABGR:
+      return "ABGR";
+    case webrtc::VideoType::kARGB:
+      return "ARGB";
+    case webrtc::VideoType::kARGB4444:
+      return "R444";
+    case webrtc::VideoType::kRGB565:
+      return "RGBP";
+    case webrtc::VideoType::kARGB1555:
+      return "RGBO";
+    case webrtc::VideoType::kYUY2:
+      return "YUY2";
+    case webrtc::VideoType::kUYVY:
+      return "UYVY";
+    case webrtc::VideoType::kMJPEG:
+      return "MJPG";
+    case webrtc::VideoType::kNV21:
+      return "NV21";
+    case webrtc::VideoType::kNV12:
+      return "NV12";
+    case webrtc::VideoType::kBGRA:
+      return "BGRA";
+    case webrtc::VideoType::kUnknown:
+    default:
+      return "unknown";
+  }
+}
+
+static void LogCapability(const char* aHeader,
+                          const webrtc::CaptureCapability& aCapability,
+                          uint32_t aDistance) {
+  LOG("%s: %4u x %4u x %2u maxFps, %s. Distance = %" PRIu32, aHeader,
+      aCapability.width, aCapability.height, aCapability.maxFPS,
+      ConvertVideoTypeToCStr(aCapability.videoType), aDistance);
+}
+
+bool MediaEngineRemoteVideoSource::ChooseCapability(
+    const NormalizedConstraints& aConstraints, const MediaEnginePrefs& aPrefs,
+    webrtc::CaptureCapability& aCapability,
+    const DistanceCalculation aCalculate) {
+  LOG("%s", __PRETTY_FUNCTION__);
+  AssertIsOnOwningThread();
+
+  if (MOZ_LOG_TEST(gMediaManagerLog, LogLevel::Debug)) {
+    LOG("ChooseCapability: prefs: %dx%d @%dfps", aPrefs.GetWidth(),
+        aPrefs.GetHeight(), aPrefs.mFPS);
+    MediaConstraintsHelper::LogConstraints(aConstraints);
+    if (!aConstraints.mAdvanced.empty()) {
+      LOG("Advanced array[%zu]:", aConstraints.mAdvanced.size());
+      for (auto& advanced : aConstraints.mAdvanced) {
+        MediaConstraintsHelper::LogConstraints(advanced);
+      }
+    }
+  }
+
+  switch (mCapEngine) {
+    case camera::ScreenEngine:
+    case camera::WinEngine: {
+      FlattenedConstraints c(aConstraints);
+      // The actual resolution to constrain around is not easy to find ahead of
+      // time (and may in fact change over time), so as a hack, we push ideal
+      // and max constraints down to desktop_capture_impl.cc and finish the
+      // algorithm there.
+      // TODO: This can be removed in bug 1453269.
+      aCapability.width =
+          (std::min(0xffff, c.mWidth.mIdeal.valueOr(0)) & 0xffff) << 16 |
+          (std::min(0xffff, c.mWidth.mMax) & 0xffff);
+      aCapability.height =
+          (std::min(0xffff, c.mHeight.mIdeal.valueOr(0)) & 0xffff) << 16 |
+          (std::min(0xffff, c.mHeight.mMax) & 0xffff);
+      aCapability.maxFPS =
+          c.mFrameRate.Clamp(c.mFrameRate.mIdeal.valueOr(aPrefs.mFPS));
+      return true;
+    }
+    case camera::BrowserEngine: {
+      FlattenedConstraints c(aConstraints);
+      aCapability.maxFPS =
+          c.mFrameRate.Clamp(c.mFrameRate.mIdeal.valueOr(aPrefs.mFPS));
+      return true;
+    }
+    default:
+      break;
+  }
+
+  nsTArray<CapabilityCandidate> candidateSet;
+  size_t num = NumCapabilities();
+  for (size_t i = 0; i < num; i++) {
+    candidateSet.AppendElement(CapabilityCandidate(GetCapability(i)));
+  }
+
+  if (mCapabilitiesAreHardcoded && mCapEngine == camera::CameraEngine) {
+    // We have a hardcoded capability, which means this camera didn't report
+    // discrete capabilities. It might still allow a ranged capability, so we
+    // add a couple of default candidates based on prefs and constraints.
+    // The chosen candidate will be propagated to StartCapture() which will fail
+    // for an invalid candidate.
+    MOZ_DIAGNOSTIC_ASSERT(mCapabilities.Length() == 1);
+    MOZ_DIAGNOSTIC_ASSERT(candidateSet.Length() == 1);
+    candidateSet.Clear();
+
+    FlattenedConstraints c(aConstraints);
+    // Reuse the code across both the low-definition (`false`) pref and
+    // the high-definition (`true`) pref.
+    // If there are constraints we try to satisfy them but we default to prefs.
+    // Note that since constraints are from content and can literally be
+    // anything we put (rather generous) caps on them.
+    for (bool isHd : {false, true}) {
+      webrtc::CaptureCapability cap;
+      int32_t prefWidth = aPrefs.GetWidth(isHd);
+      int32_t prefHeight = aPrefs.GetHeight(isHd);
+
+      cap.width = c.mWidth.Get(prefWidth);
+      cap.width = std::max(0, std::min(cap.width, 7680));
+
+      cap.height = c.mHeight.Get(prefHeight);
+      cap.height = std::max(0, std::min(cap.height, 4320));
+
+      cap.maxFPS = c.mFrameRate.Get(aPrefs.mFPS);
+      cap.maxFPS = std::max(0, std::min(cap.maxFPS, 480));
+
+      if (cap.width != prefWidth) {
+        // Width was affected by constraints.
+        // We'll adjust the height too so the aspect ratio is retained.
+        cap.height = cap.width * prefHeight / prefWidth;
+      } else if (cap.height != prefHeight) {
+        // Height was affected by constraints but not width.
+        // We'll adjust the width too so the aspect ratio is retained.
+        cap.width = cap.height * prefWidth / prefHeight;
+      }
+
+      if (candidateSet.Contains(cap, CapabilityComparator())) {
+        continue;
+      }
+      LogCapability("Hardcoded capability", cap, 0);
+      candidateSet.AppendElement(cap);
+    }
+  }
+
+  // First, filter capabilities by required constraints (min, max, exact).
+
+  for (size_t i = 0; i < candidateSet.Length();) {
+    auto& candidate = candidateSet[i];
+    candidate.mDistance =
+        GetDistance(candidate.mCapability, aConstraints, aCalculate);
+    LogCapability("Capability", candidate.mCapability, candidate.mDistance);
+    if (candidate.mDistance == UINT32_MAX) {
+      candidateSet.RemoveElementAt(i);
+    } else {
+      ++i;
+    }
+  }
+
+  if (candidateSet.IsEmpty()) {
+    LOG("failed to find capability match from %zu choices",
+        candidateSet.Length());
+    return false;
+  }
+
+  // Filter further with all advanced constraints (that don't overconstrain).
+
+  for (const auto& cs : aConstraints.mAdvanced) {
+    nsTArray<CapabilityCandidate> rejects;
+    for (size_t i = 0; i < candidateSet.Length();) {
+      if (GetDistance(candidateSet[i].mCapability, cs, aCalculate) ==
+          UINT32_MAX) {
+        rejects.AppendElement(candidateSet[i]);
+        candidateSet.RemoveElementAt(i);
+      } else {
+        ++i;
+      }
+    }
+    if (!candidateSet.Length()) {
+      candidateSet.AppendElements(std::move(rejects));
+    }
+  }
+  MOZ_ASSERT(
+      candidateSet.Length(),
+      "advanced constraints filtering step can't reduce candidates to zero");
+
+  // Remaining algorithm is up to the UA.
+
+  TrimLessFitCandidates(candidateSet);
+
+  // Any remaining multiples all have the same distance. A common case of this
+  // occurs when no ideal is specified. Lean toward defaults.
+  uint32_t sameDistance = candidateSet[0].mDistance;
+  {
+    MediaTrackConstraintSet prefs;
+    prefs.mWidth.Construct().SetAsLong() = aPrefs.GetWidth();
+    prefs.mHeight.Construct().SetAsLong() = aPrefs.GetHeight();
+    prefs.mFrameRate.Construct().SetAsDouble() = aPrefs.mFPS;
+    NormalizedConstraintSet normPrefs(prefs, false);
+
+    for (auto& candidate : candidateSet) {
+      candidate.mDistance =
+          GetDistance(candidate.mCapability, normPrefs, aCalculate);
+    }
+    TrimLessFitCandidates(candidateSet);
+  }
+
+  aCapability = candidateSet[0].mCapability;
+
+  LogCapability("Chosen capability", aCapability, sameDistance);
+  return true;
+}
+
+void MediaEngineRemoteVideoSource::GetSettings(
+    MediaTrackSettings& aOutSettings) const {
+  aOutSettings = *mSettings;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/MediaEngineRemoteVideoSource.h b/dom/media/webrtc/MediaEngineRemoteVideoSource.h
new file mode 100644
index 0000000000..7e4f852701
--- /dev/null
+++ b/dom/media/webrtc/MediaEngineRemoteVideoSource.h
@@ -0,0 +1,242 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set sw=2 ts=8 et ft=cpp : */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef MEDIAENGINE_REMOTE_VIDEO_SOURCE_H_
+#define MEDIAENGINE_REMOTE_VIDEO_SOURCE_H_
+
+#include "prcvar.h"
+#include "prthread.h"
+
+#include "mozilla/Mutex.h"
+#include "nsCOMPtr.h"
+#include "nsThreadUtils.h"
+#include "DOMMediaStream.h"
+#include "nsDirectoryServiceDefs.h"
+#include "nsComponentManagerUtils.h"
+
+// Avoid warnings about redefinition of WARN_UNUSED_RESULT
+#include "ipc/IPCMessageUtils.h"
+#include "VideoUtils.h"
+#include "MediaEngineSource.h"
+#include "VideoSegment.h"
+#include "AudioSegment.h"
+#include "MediaTrackGraph.h"
+
+#include "MediaEngineWrapper.h"
+#include "mozilla/dom/MediaStreamTrackBinding.h"
+
+// Camera Access via IPC
+#include "CamerasChild.h"
+
+#include "NullTransport.h"
+
+// WebRTC includes
+#include "common_video/include/video_frame_buffer_pool.h"
+#include "modules/video_capture/video_capture_defines.h"
+
+namespace webrtc {
+using CaptureCapability = VideoCaptureCapability;
+}
+
+namespace mozilla {
+
+// Fitness distance is defined in
+// https://w3c.github.io/mediacapture-main/getusermedia.html#dfn-selectsettings
+
+// The main difference of feasibility and fitness distance is that if the
+// constraint is required ('max', or 'exact'), and the settings dictionary's
+// value for the constraint does not satisfy the constraint, the fitness
+// distance is positive infinity. Given a continuous space of settings
+// dictionaries comprising all discrete combinations of dimension and frame-rate
+// related properties, the feasibility distance is still in keeping with the
+// constraints algorithm.
+enum DistanceCalculation { kFitness, kFeasibility };
+
+/**
+ * The WebRTC implementation of the MediaEngine interface.
+ */
+class MediaEngineRemoteVideoSource : public MediaEngineSource,
+                                     public camera::FrameRelay {
+  ~MediaEngineRemoteVideoSource();
+
+  struct CapabilityCandidate {
+    explicit CapabilityCandidate(webrtc::CaptureCapability aCapability,
+                                 uint32_t aDistance = 0)
+        : mCapability(aCapability), mDistance(aDistance) {}
+
+    const webrtc::CaptureCapability mCapability;
+    uint32_t mDistance;
+  };
+
+  class CapabilityComparator {
+   public:
+    bool Equals(const CapabilityCandidate& aCandidate,
+                const webrtc::CaptureCapability& aCapability) const {
+      return aCandidate.mCapability == aCapability;
+    }
+  };
+
+  bool ChooseCapability(const NormalizedConstraints& aConstraints,
+                        const MediaEnginePrefs& aPrefs,
+                        webrtc::CaptureCapability& aCapability,
+                        const DistanceCalculation aCalculate);
+
+  uint32_t GetDistance(const webrtc::CaptureCapability& aCandidate,
+                       const NormalizedConstraintSet& aConstraints,
+                       const DistanceCalculation aCalculate) const;
+
+  uint32_t GetFitnessDistance(
+      const webrtc::CaptureCapability& aCandidate,
+      const NormalizedConstraintSet& aConstraints) const;
+
+  uint32_t GetFeasibilityDistance(
+      const webrtc::CaptureCapability& aCandidate,
+      const NormalizedConstraintSet& aConstraints) const;
+
+  static void TrimLessFitCandidates(nsTArray<CapabilityCandidate>& aSet);
+
+ public:
+  explicit MediaEngineRemoteVideoSource(const MediaDevice* aMediaDevice);
+
+  // ExternalRenderer
+  int DeliverFrame(uint8_t* aBuffer,
+                   const camera::VideoFrameProperties& aProps) override;
+
+  // MediaEngineSource
+  nsresult Allocate(const dom::MediaTrackConstraints& aConstraints,
+                    const MediaEnginePrefs& aPrefs, uint64_t aWindowID,
+                    const char** aOutBadConstraint) override;
+  nsresult Deallocate() override;
+  void SetTrack(const RefPtr<MediaTrack>& aTrack,
+                const PrincipalHandle& aPrincipal) override;
+  nsresult Start() override;
+  nsresult Reconfigure(const dom::MediaTrackConstraints& aConstraints,
+                       const MediaEnginePrefs& aPrefs,
+                       const char** aOutBadConstraint) override;
+  nsresult FocusOnSelectedSource() override;
+  nsresult Stop() override;
+
+  uint32_t GetBestFitnessDistance(
+      const nsTArray<const NormalizedConstraintSet*>& aConstraintSets)
+      const override;
+  void GetSettings(dom::MediaTrackSettings& aOutSettings) const override;
+
+  RefPtr<GenericNonExclusivePromise> GetFirstFramePromise() const override {
+    return mFirstFramePromise;
+  }
+
+  const TrackingId& GetTrackingId() const override;
+
+  static camera::CaptureEngine CaptureEngine(dom::MediaSourceEnum aMediaSource);
+
+ private:
+  /**
+   * Returns the number of capabilities for the underlying device.
+   *
+   * Guaranteed to return at least one capability.
+   */
+  size_t NumCapabilities() const;
+
+  /**
+   * Returns the capability with index `aIndex` for our assigned device.
+   *
+   * It is an error to call this with `aIndex >= NumCapabilities()`.
+   *
+   * The lifetime of the returned capability is the same as for this source.
+   */
+  webrtc::CaptureCapability& GetCapability(size_t aIndex) const;
+
+  int mCaptureId = -1;
+  const camera::CaptureEngine mCapEngine;  // source of media (cam, screen etc)
+
+  // A tracking id used to uniquely identify the source of video frames.
+  // Set under mMutex on the owning thread. Accessed under one of the two.
+  TrackingId mTrackingId;
+
+  // Mirror of mTrackingId on the frame-delivering thread (Cameras IPC).
+  Maybe<TrackingId> mFrameDeliveringTrackingId;
+
+  // mMutex protects certain members on 3 threads:
+  // MediaManager, Cameras IPC and MediaTrackGraph.
+  Mutex mMutex MOZ_UNANNOTATED;
+
+  // Current state of this source.
+  // Set under mMutex on the owning thread. Accessed under one of the two.
+  MediaEngineSourceState mState = kReleased;
+
+  // The source track that we feed video data to.
+  // Set under mMutex on the owning thread. Accessed under one of the two.
+  RefPtr<SourceMediaTrack> mTrack;
+
+  // The PrincipalHandle that gets attached to the frames we feed to mTrack.
+  // Set under mMutex on the owning thread. Accessed under one of the two.
+  PrincipalHandle mPrincipal = PRINCIPAL_HANDLE_NONE;
+
+  // Set in Start() and Deallocate() on the owning thread.
+  // Accessed in DeliverFrame() on the camera IPC thread, guaranteed to happen
+  // after Start() and before the end of Stop().
+  RefPtr<layers::ImageContainer> mImageContainer;
+
+  // A buffer pool used to manage the temporary buffer used when rescaling
+  // incoming images. Cameras IPC thread only.
+  webrtc::VideoFrameBufferPool mRescalingBufferPool;
+
+  // The intrinsic size of the latest captured image, so we can feed black
+  // images of the same size while stopped.
+  // Set under mMutex on the Cameras IPC thread. Accessed under one of the two.
+  gfx::IntSize mImageSize = gfx::IntSize(0, 0);
+
+  struct AtomicBool {
+    Atomic<bool> mValue;
+  };
+
+  // True when resolution settings have been updated from a real frame's
+  // resolution. Threadsafe.
+  // TODO: This can be removed in bug 1453269.
+  const RefPtr<media::Refcountable<AtomicBool>> mSettingsUpdatedByFrame;
+
+  // The current settings of this source.
+  // Note that these may be different from the settings of the underlying device
+  // since we scale frames to avoid fingerprinting.
+  // Members are main thread only.
+  const RefPtr<media::Refcountable<dom::MediaTrackSettings>> mSettings;
+  MozPromiseHolder<GenericNonExclusivePromise> mFirstFramePromiseHolder;
+  RefPtr<GenericNonExclusivePromise> mFirstFramePromise;
+
+  // The capability currently chosen by constraints of the user of this source.
+  // Set under mMutex on the owning thread. Accessed under one of the two.
+  webrtc::CaptureCapability mCapability;
+
+  /**
+   * Capabilities that we choose between when applying constraints.
+   *
+   * This allows for memoization of capabilities as they're requested from the
+   * parent process.
+   *
+   * This is mutable so that the const methods NumCapabilities() and
+   * GetCapability() can reset it. Owning thread only.
+   */
+  mutable nsTArray<UniquePtr<webrtc::CaptureCapability>> mCapabilities;
+
+  /**
+   * True if mCapabilities only contains hardcoded capabilities. This can happen
+   * if the underlying device is not reporting any capabilities. These can be
+   * affected by constraints, so they're evaluated in ChooseCapability() rather
+   * than GetCapability().
+   *
+   * This is mutable so that the const methods NumCapabilities() and
+   * GetCapability() can reset it. Owning thread only.
+   */
+  mutable bool mCapabilitiesAreHardcoded = false;
+
+  const RefPtr<const MediaDevice> mMediaDevice;
+  const nsCString mDeviceUUID;
+  Maybe<nsString> mFacingMode;
+};
+
+}  // namespace mozilla
+
+#endif /* MEDIAENGINE_REMOTE_VIDEO_SOURCE_H_ */
diff --git a/dom/media/webrtc/MediaEngineSource.cpp b/dom/media/webrtc/MediaEngineSource.cpp
new file mode 100644
index 0000000000..d0e5e23ae7
--- /dev/null
+++ b/dom/media/webrtc/MediaEngineSource.cpp
@@ -0,0 +1,69 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "MediaEngineSource.h"
+
+#include "mozilla/dom/MediaTrackSettingsBinding.h"
+
+namespace mozilla {
+
+using dom::MediaSourceEnum;
+using dom::MediaTrackSettings;
+
+// These need a definition somewhere because template
+// code is allowed to take their address, and they aren't
+// guaranteed to have one without this.
+const unsigned int MediaEngineSource::kMaxDeviceNameLength;
+const unsigned int MediaEngineSource::kMaxUniqueIdLength;
+
+/* static */
+bool MediaEngineSource::IsVideo(MediaSourceEnum aSource) {
+  switch (aSource) {
+    case MediaSourceEnum::Camera:
+    case MediaSourceEnum::Screen:
+    case MediaSourceEnum::Window:
+    case MediaSourceEnum::Browser:
+      return true;
+    case MediaSourceEnum::Microphone:
+    case MediaSourceEnum::AudioCapture:
+    case MediaSourceEnum::Other:
+      return false;
+    default:
+      MOZ_ASSERT_UNREACHABLE("Unknown type");
+      return false;
+  }
+}
+
+/* static */
+bool MediaEngineSource::IsAudio(MediaSourceEnum aSource) {
+  switch (aSource) {
+    case MediaSourceEnum::Microphone:
+    case MediaSourceEnum::AudioCapture:
+      return true;
+    case MediaSourceEnum::Camera:
+    case MediaSourceEnum::Screen:
+    case MediaSourceEnum::Window:
+    case MediaSourceEnum::Browser:
+    case MediaSourceEnum::Other:
+      return false;
+    default:
+      MOZ_ASSERT_UNREACHABLE("Unknown type");
+      return false;
+  }
+}
+
+bool MediaEngineSource::IsFake() const { return false; }
+
+nsresult MediaEngineSource::FocusOnSelectedSource() {
+  return NS_ERROR_NOT_AVAILABLE;
+}
+
+nsresult MediaEngineSource::TakePhoto(MediaEnginePhotoCallback* aCallback) {
+  return NS_ERROR_NOT_IMPLEMENTED;
+}
+
+MediaEngineSource::~MediaEngineSource() = default;
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/MediaEngineSource.h b/dom/media/webrtc/MediaEngineSource.h
new file mode 100644
index 0000000000..d32796b094
--- /dev/null
+++ b/dom/media/webrtc/MediaEngineSource.h
@@ -0,0 +1,255 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef MediaEngineSource_h
+#define MediaEngineSource_h
+
+#include "MediaSegment.h"
+#include "MediaTrackConstraints.h"
+#include "mozilla/dom/MediaStreamTrackBinding.h"
+#include "mozilla/media/MediaUtils.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/ThreadSafeWeakPtr.h"
+#include "nsStringFwd.h"
+#include "PerformanceRecorder.h"
+
+namespace mozilla {
+
+namespace dom {
+class Blob;
+struct MediaTrackSettings;
+}  // namespace dom
+
+namespace ipc {
+class PrincipalInfo;
+}  // namespace ipc
+
+class MediaEnginePhotoCallback;
+class MediaEnginePrefs;
+class MediaTrack;
+
+/**
+ * Callback interface for TakePhoto(). Either PhotoComplete() or PhotoError()
+ * should be called.
+ */
+class MediaEnginePhotoCallback {
+ public:
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(MediaEnginePhotoCallback)
+
+  // aBlob is the image captured by MediaEngineSource. It is
+  // called on main thread.
+  virtual nsresult PhotoComplete(already_AddRefed<dom::Blob> aBlob) = 0;
+
+  // It is called on main thread. aRv is the error code.
+  virtual nsresult PhotoError(nsresult aRv) = 0;
+
+ protected:
+  virtual ~MediaEnginePhotoCallback() = default;
+};
+
+/**
+ * Lifecycle state of MediaEngineSource.
+ */
+enum MediaEngineSourceState {
+  kAllocated,  // Allocated, not yet started.
+  kStarted,    // Previously allocated or stopped, then started.
+  kStopped,    // Previously started, then stopped.
+  kReleased    // Not allocated.
+};
+
+/**
+ * The pure interface of a MediaEngineSource.
+ *
+ * Most sources are helped by the defaults implemented in MediaEngineSource.
+ */
+class MediaEngineSourceInterface {
+ public:
+  /**
+   * Return true if this is a fake source. I.e., if it is generating media
+   * itself rather than being an interface to underlying hardware.
+   */
+  virtual bool IsFake() const = 0;
+
+  /**
+   * Override w/a promise if source has frames, in order to potentially allow
+   * deferring success of source acquisition until first frame has arrived.
+   */
+  virtual RefPtr<GenericNonExclusivePromise> GetFirstFramePromise() const {
+    return nullptr;
+  }
+
+  /**
+   * Get an id uniquely identifying the source of video frames that this
+   * MediaEngineSource represents. This can be used in profiler markers to
+   * separate markers from different sources into different lanes.
+   */
+  virtual const TrackingId& GetTrackingId() const = 0;
+
+  /**
+   * Called by MediaEngine to allocate an instance of this source.
+   */
+  virtual nsresult Allocate(const dom::MediaTrackConstraints& aConstraints,
+                            const MediaEnginePrefs& aPrefs, uint64_t aWindowID,
+                            const char** aOutBadConstraint) = 0;
+
+  /**
+   * Called by MediaEngine when a MediaTrack has been provided for the source to
+   * feed data to.
+   *
+   * This must be called before Start.
+   */
+  virtual void SetTrack(const RefPtr<MediaTrack>& aTrack,
+                        const PrincipalHandle& aPrincipal) = 0;
+
+  /**
+   * Called by MediaEngine to start feeding data to the track.
+   *
+   * NB: Audio sources handle the enabling of pulling themselves.
+   */
+  virtual nsresult Start() = 0;
+
+  /**
+   * This brings focus to the selected source, e.g. to bring a captured window
+   * to the front.
+   *
+   * We return one of the following:
+   * NS_OK                    - Success.
+   * NS_ERROR_NOT_AVAILABLE   - For backends where focusing does not make sense.
+   * NS_ERROR_NOT_IMPLEMENTED - For backends where focusing makes sense, but
+   *                            is not yet implemented.
+   * NS_ERROR_FAILURE         - Failures reported from underlying code.
+   */
+  virtual nsresult FocusOnSelectedSource() = 0;
+
+  /**
+   * Applies new constraints to the capability selection for the underlying
+   * device.
+   *
+   * Should the constraints lead to choosing a new capability while the device
+   * is actively being captured, the device will restart using the new
+   * capability.
+   *
+   * We return one of the following:
+   * NS_OK                - Successful reconfigure.
+   * NS_ERROR_INVALID_ARG - Couldn't find a capability fitting aConstraints.
+   *                        See aBadConstraint for details.
+   * NS_ERROR_UNEXPECTED  - Reconfiguring the underlying device failed
+   *                        unexpectedly. This leaves the device in a stopped
+   *                        state.
+   */
+  virtual nsresult Reconfigure(const dom::MediaTrackConstraints& aConstraints,
+                               const MediaEnginePrefs& aPrefs,
+                               const char** aOutBadConstraint) = 0;
+
+  /**
+   * Called by MediaEngine to stop feeding data to the track.
+   *
+   * Double-stopping is allowed and will return NS_OK. This is necessary
+   * sometimes during shutdown.
+   *
+   * NB: Audio sources handle the disabling of pulling themselves.
+   */
+  virtual nsresult Stop() = 0;
+
+  /**
+   * Called by MediaEngine to deallocate an underlying device.
+   */
+  virtual nsresult Deallocate() = 0;
+
+  /**
+   * If implementation of MediaEngineSource supports TakePhoto(), the picture
+   * should be returned via aCallback object. Otherwise, it returns
+   * NS_ERROR_NOT_IMPLEMENTED.
+   */
+  virtual nsresult TakePhoto(MediaEnginePhotoCallback* aCallback) = 0;
+
+  /**
+   * GetBestFitnessDistance returns the best distance the capture device can
+   * offer as a whole, given an accumulated number of ConstraintSets. Ideal
+   * values are considered in the first ConstraintSet only. Plain values are
+   * treated as Ideal in the first ConstraintSet. Plain values are treated as
+   * Exact in subsequent ConstraintSets. Infinity = UINT32_MAX e.g. device
+   * cannot satisfy accumulated ConstraintSets. A finite result may be used to
+   * calculate this device's ranking as a choice.
+   */
+  virtual uint32_t GetBestFitnessDistance(
+      const nsTArray<const NormalizedConstraintSet*>& aConstraintSets)
+      const = 0;
+
+  /**
+   * Returns the current settings of the underlying device.
+   *
+   * Note that this might not be the settings of the underlying hardware.
+   * In case of a camera where we intervene and scale frames to avoid
+   * leaking information from other documents than the current one,
+   * GetSettings() will return the scaled resolution. I.e., the
+   * device settings as seen by js.
+   */
+  virtual void GetSettings(dom::MediaTrackSettings& aOutSettings) const = 0;
+};
+
+/**
+ * Abstract base class for MediaEngineSources.
+ *
+ * Implements defaults for some common MediaEngineSourceInterface methods below.
+ * Also implements RefPtr support and an owning-thread model for thread safety
+ * checks in subclasses.
+ */
+class MediaEngineSource : public MediaEngineSourceInterface {
+ public:
+  // code inside webrtc.org assumes these sizes; don't use anything smaller
+  // without verifying it's ok
+  static const unsigned int kMaxDeviceNameLength = 128;
+  static const unsigned int kMaxUniqueIdLength = 256;
+
+  /**
+   * Returns true if the given source type is for video, false otherwise.
+   * Only call with real types.
+   */
+  static bool IsVideo(dom::MediaSourceEnum aSource);
+
+  /**
+   * Returns true if the given source type is for audio, false otherwise.
+   * Only call with real types.
+   */
+  static bool IsAudio(dom::MediaSourceEnum aSource);
+
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(MediaEngineSource)
+  NS_DECL_OWNINGEVENTTARGET
+
+  void AssertIsOnOwningThread() const {
+    NS_ASSERT_OWNINGTHREAD(MediaEngineSource);
+  }
+
+  const TrackingId& GetTrackingId() const override {
+    static auto notImplementedId = TrackingId();
+    return notImplementedId;
+  }
+
+  // Not fake by default.
+  bool IsFake() const override;
+
+  // Returns NS_ERROR_NOT_AVAILABLE by default.
+  nsresult FocusOnSelectedSource() override;
+
+  // TakePhoto returns NS_ERROR_NOT_IMPLEMENTED by default,
+  // to tell the caller to fallback to other methods.
+  nsresult TakePhoto(MediaEnginePhotoCallback* aCallback) override;
+
+  // Returns a default distance of 0 for devices that don't have capabilities.
+  uint32_t GetBestFitnessDistance(
+      const nsTArray<const NormalizedConstraintSet*>& aConstraintSets)
+      const override {
+    return 0;
+  }
+
+ protected:
+  virtual ~MediaEngineSource();
+};
+
+}  // namespace mozilla
+
+#endif /* MediaEngineSource_h */
diff --git a/dom/media/webrtc/MediaEngineWebRTC.cpp b/dom/media/webrtc/MediaEngineWebRTC.cpp
new file mode 100644
index 0000000000..cf638497a9
--- /dev/null
+++ b/dom/media/webrtc/MediaEngineWebRTC.cpp
@@ -0,0 +1,299 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set sw=2 ts=8 et ft=cpp : */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "MediaEngineWebRTC.h"
+
+#include "CamerasChild.h"
+#include "MediaEngineRemoteVideoSource.h"
+#include "MediaEngineWebRTCAudio.h"
+#include "MediaManager.h"
+#include "mozilla/Logging.h"
+
+// Pipewire detection support
+#if defined(WEBRTC_USE_PIPEWIRE)
+#  include "mozilla/StaticPrefs_media.h"
+#  include "modules/desktop_capture/desktop_capturer.h"
+#endif
+
+#define FAKE_ONDEVICECHANGE_EVENT_PERIOD_IN_MS 500
+
+static mozilla::LazyLogModule sGetUserMediaLog("GetUserMedia");
+#undef LOG
+#define LOG(args) MOZ_LOG(sGetUserMediaLog, mozilla::LogLevel::Debug, args)
+
+namespace mozilla {
+
+using AudioDeviceSet = CubebDeviceEnumerator::AudioDeviceSet;
+using camera::CamerasChild;
+using camera::GetChildAndCall;
+using dom::MediaSourceEnum;
+
+CubebDeviceEnumerator* GetEnumerator() {
+  return CubebDeviceEnumerator::GetInstance();
+}
+
+MediaEngineWebRTC::MediaEngineWebRTC() {
+  AssertIsOnOwningThread();
+
+  GetChildAndCall(
+      &CamerasChild::ConnectDeviceListChangeListener<MediaEngineWebRTC>,
+      &mCameraListChangeListener, AbstractThread::MainThread(), this,
+      &MediaEngineWebRTC::DeviceListChanged);
+  mMicrophoneListChangeListener =
+      GetEnumerator()->OnAudioInputDeviceListChange().Connect(
+          AbstractThread::MainThread(), this,
+          &MediaEngineWebRTC::DeviceListChanged);
+  mSpeakerListChangeListener =
+      GetEnumerator()->OnAudioOutputDeviceListChange().Connect(
+          AbstractThread::MainThread(), this,
+          &MediaEngineWebRTC::DeviceListChanged);
+}
+
+void MediaEngineWebRTC::EnumerateVideoDevices(
+    MediaSourceEnum aMediaSource, nsTArray<RefPtr<MediaDevice>>* aDevices) {
+  AssertIsOnOwningThread();
+  // flag sources with cross-origin exploit potential
+  bool scaryKind = (aMediaSource == MediaSourceEnum::Screen ||
+                    aMediaSource == MediaSourceEnum::Browser);
+#if defined(WEBRTC_USE_PIPEWIRE)
+  bool canRequestOsLevelPrompt =
+      mozilla::StaticPrefs::media_webrtc_capture_allow_pipewire() &&
+      webrtc::DesktopCapturer::IsRunningUnderWayland() &&
+      (aMediaSource == MediaSourceEnum::Application ||
+       aMediaSource == MediaSourceEnum::Screen ||
+       aMediaSource == MediaSourceEnum::Window);
+#else
+  bool canRequestOsLevelPrompt = false;
+#endif
+  /*
+   * We still enumerate every time, in case a new device was plugged in since
+   * the last call. TODO: Verify that WebRTC actually does deal with hotplugging
+   * new devices (with or without new engine creation) and accordingly adjust.
+   * Enumeration is not neccessary if GIPS reports the same set of devices
+   * for a given instance of the engine.
+   */
+  int num;
+#if defined(_ARM64_) && defined(XP_WIN)
+  // There are problems with using DirectShow on versions of Windows before
+  // 19H1 on arm64. This disables the camera on older versions of Windows.
+  if (aMediaSource == MediaSourceEnum::Camera) {
+    typedef ULONG (*RtlGetVersionFn)(LPOSVERSIONINFOEXW);
+    RtlGetVersionFn RtlGetVersion;
+    RtlGetVersion = (RtlGetVersionFn)GetProcAddress(GetModuleHandleA("ntdll"),
+                                                    "RtlGetVersion");
+    if (RtlGetVersion) {
+      OSVERSIONINFOEXW info;
+      info.dwOSVersionInfoSize = sizeof(info);
+      RtlGetVersion(&info);
+      // 19H1 is 18346
+      if (info.dwBuildNumber < 18346) {
+        return;
+      }
+    }
+  }
+#endif
+  camera::CaptureEngine capEngine =
+      MediaEngineRemoteVideoSource::CaptureEngine(aMediaSource);
+  num = GetChildAndCall(&CamerasChild::NumberOfCaptureDevices, capEngine);
+
+  for (int i = 0; i < num; i++) {
+    char deviceName[MediaEngineSource::kMaxDeviceNameLength];
+    char uniqueId[MediaEngineSource::kMaxUniqueIdLength];
+    bool scarySource = false;
+
+    // paranoia
+    deviceName[0] = '\0';
+    uniqueId[0] = '\0';
+    int error;
+
+    error = GetChildAndCall(&CamerasChild::GetCaptureDevice, capEngine, i,
+                            deviceName, sizeof(deviceName), uniqueId,
+                            sizeof(uniqueId), &scarySource);
+    if (error) {
+      LOG(("camera:GetCaptureDevice: Failed %d", error));
+      continue;
+    }
+#ifdef DEBUG
+    LOG(("  Capture Device Index %d, Name %s", i, deviceName));
+
+    webrtc::CaptureCapability cap;
+    int numCaps = GetChildAndCall(&CamerasChild::NumberOfCapabilities,
+                                  capEngine, uniqueId);
+    LOG(("Number of Capabilities %d", numCaps));
+    for (int j = 0; j < numCaps; j++) {
+      if (GetChildAndCall(&CamerasChild::GetCaptureCapability, capEngine,
+                          uniqueId, j, &cap) != 0) {
+        break;
+      }
+      LOG(("type=%d width=%d height=%d maxFPS=%d",
+           static_cast<int>(cap.videoType), cap.width, cap.height, cap.maxFPS));
+    }
+#endif
+
+    NS_ConvertUTF8toUTF16 name(deviceName);
+    NS_ConvertUTF8toUTF16 uuid(uniqueId);
+    // The remote video backend doesn't implement group id. We return the
+    // device name and higher layers will correlate this with the name of
+    // audio devices.
+    aDevices->EmplaceBack(new MediaDevice(
+        this, aMediaSource, name, uuid, uuid,
+        MediaDevice::IsScary(scaryKind || scarySource),
+        canRequestOsLevelPrompt ? MediaDevice::OsPromptable::Yes
+                                : MediaDevice::OsPromptable::No));
+  }
+}
+
+void MediaEngineWebRTC::EnumerateMicrophoneDevices(
+    nsTArray<RefPtr<MediaDevice>>* aDevices) {
+  AssertIsOnOwningThread();
+
+  RefPtr<const AudioDeviceSet> devices =
+      GetEnumerator()->EnumerateAudioInputDevices();
+
+  DebugOnly<bool> foundPreferredDevice = false;
+
+  for (const auto& deviceInfo : *devices) {
+#ifndef ANDROID
+    MOZ_ASSERT(deviceInfo->DeviceID());
+#endif
+    LOG(("Cubeb device: type 0x%x, state 0x%x, name %s, id %p",
+         deviceInfo->Type(), deviceInfo->State(),
+         NS_ConvertUTF16toUTF8(deviceInfo->Name()).get(),
+         deviceInfo->DeviceID()));
+
+    if (deviceInfo->State() == CUBEB_DEVICE_STATE_ENABLED) {
+      MOZ_ASSERT(deviceInfo->Type() == CUBEB_DEVICE_TYPE_INPUT);
+      // Lie and provide the name as UUID
+      RefPtr device = new MediaDevice(this, deviceInfo, deviceInfo->Name());
+      if (deviceInfo->Preferred()) {
+#ifdef DEBUG
+        if (!foundPreferredDevice) {
+          foundPreferredDevice = true;
+        } else {
+          // This is possible on windows, there is a default communication
+          // device, and a default device:
+          // See https://bugzilla.mozilla.org/show_bug.cgi?id=1542739
+#  ifndef XP_WIN
+          MOZ_ASSERT(!foundPreferredDevice,
+                     "Found more than one preferred audio input device"
+                     "while enumerating");
+#  endif
+        }
+#endif
+        aDevices->InsertElementAt(0, std::move(device));
+      } else {
+        aDevices->AppendElement(std::move(device));
+      }
+    }
+  }
+}
+
+void MediaEngineWebRTC::EnumerateSpeakerDevices(
+    nsTArray<RefPtr<MediaDevice>>* aDevices) {
+  AssertIsOnOwningThread();
+
+  RefPtr<const AudioDeviceSet> devices =
+      GetEnumerator()->EnumerateAudioOutputDevices();
+
+#ifndef XP_WIN
+  DebugOnly<bool> preferredDeviceFound = false;
+#endif
+  for (const auto& deviceInfo : *devices) {
+    LOG(("Cubeb device: type 0x%x, state 0x%x, name %s, id %p",
+         deviceInfo->Type(), deviceInfo->State(),
+         NS_ConvertUTF16toUTF8(deviceInfo->Name()).get(),
+         deviceInfo->DeviceID()));
+    if (deviceInfo->State() == CUBEB_DEVICE_STATE_ENABLED) {
+      MOZ_ASSERT(deviceInfo->Type() == CUBEB_DEVICE_TYPE_OUTPUT);
+      nsString uuid(deviceInfo->Name());
+      // If, for example, input and output are in the same device, uuid
+      // would be the same for both which ends up to create the same
+      // deviceIDs (in JS).
+      uuid.Append(u"_Speaker"_ns);
+      RefPtr device = new MediaDevice(this, deviceInfo, uuid);
+      if (deviceInfo->Preferred()) {
+        // In windows is possible to have more than one preferred device
+#if defined(DEBUG) && !defined(XP_WIN)
+        MOZ_ASSERT(!preferredDeviceFound, "More than one preferred device");
+        preferredDeviceFound = true;
+#endif
+        aDevices->InsertElementAt(0, std::move(device));
+      } else {
+        aDevices->AppendElement(std::move(device));
+      }
+    }
+  }
+}
+
+void MediaEngineWebRTC::EnumerateDevices(
+    MediaSourceEnum aMediaSource, MediaSinkEnum aMediaSink,
+    nsTArray<RefPtr<MediaDevice>>* aDevices) {
+  AssertIsOnOwningThread();
+  MOZ_ASSERT(aMediaSource != MediaSourceEnum::Other ||
+             aMediaSink != MediaSinkEnum::Other);
+  if (MediaEngineSource::IsVideo(aMediaSource)) {
+    switch (aMediaSource) {
+      case MediaSourceEnum::Window:
+        // Since the mediaSource constraint is deprecated, treat the Window
+        // value as a request for getDisplayMedia-equivalent sharing: Combine
+        // window and fullscreen into a single list of choices. The other values
+        // are still useful for testing.
+        EnumerateVideoDevices(MediaSourceEnum::Window, aDevices);
+        EnumerateVideoDevices(MediaSourceEnum::Browser, aDevices);
+        EnumerateVideoDevices(MediaSourceEnum::Screen, aDevices);
+        break;
+      case MediaSourceEnum::Screen:
+      case MediaSourceEnum::Browser:
+      case MediaSourceEnum::Camera:
+        EnumerateVideoDevices(aMediaSource, aDevices);
+        break;
+      default:
+        MOZ_CRASH("No valid video source");
+        break;
+    }
+  } else if (aMediaSource == MediaSourceEnum::AudioCapture) {
+    aDevices->EmplaceBack(new MediaDevice(
+        this, aMediaSource, u"AudioCapture"_ns,
+        MediaEngineWebRTCAudioCaptureSource::GetUUID(),
+        MediaEngineWebRTCAudioCaptureSource::GetGroupId(),
+        MediaDevice::IsScary::No, MediaDevice::OsPromptable::No));
+  } else if (aMediaSource == MediaSourceEnum::Microphone) {
+    EnumerateMicrophoneDevices(aDevices);
+  }
+
+  if (aMediaSink == MediaSinkEnum::Speaker) {
+    EnumerateSpeakerDevices(aDevices);
+  }
+}
+
+RefPtr<MediaEngineSource> MediaEngineWebRTC::CreateSource(
+    const MediaDevice* aMediaDevice) {
+  MOZ_ASSERT(aMediaDevice->mEngine == this);
+  if (MediaEngineSource::IsVideo(aMediaDevice->mMediaSource)) {
+    return new MediaEngineRemoteVideoSource(aMediaDevice);
+  }
+  switch (aMediaDevice->mMediaSource) {
+    case MediaSourceEnum::AudioCapture:
+      return new MediaEngineWebRTCAudioCaptureSource(aMediaDevice);
+    case MediaSourceEnum::Microphone:
+      return new MediaEngineWebRTCMicrophoneSource(aMediaDevice);
+    default:
+      MOZ_CRASH("Unsupported source type");
+      return nullptr;
+  }
+}
+
+void MediaEngineWebRTC::Shutdown() {
+  AssertIsOnOwningThread();
+  mCameraListChangeListener.DisconnectIfExists();
+  mMicrophoneListChangeListener.DisconnectIfExists();
+  mSpeakerListChangeListener.DisconnectIfExists();
+
+  LOG(("%s", __FUNCTION__));
+  mozilla::camera::Shutdown();
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/MediaEngineWebRTC.h b/dom/media/webrtc/MediaEngineWebRTC.h
new file mode 100644
index 0000000000..918558dfab
--- /dev/null
+++ b/dom/media/webrtc/MediaEngineWebRTC.h
@@ -0,0 +1,53 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef MEDIAENGINEWEBRTC_H_
+#define MEDIAENGINEWEBRTC_H_
+
+#include "MediaEngine.h"
+#include "MediaEventSource.h"
+#include "MediaEngineSource.h"
+#include "nsTArray.h"
+#include "CubebDeviceEnumerator.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/dom/MediaStreamTrackBinding.h"
+
+namespace mozilla {
+
+class MediaEngineWebRTC : public MediaEngine {
+ public:
+  MediaEngineWebRTC();
+
+  // Clients should ensure to clean-up sources video/audio sources
+  // before invoking Shutdown on this class.
+  void Shutdown() override;
+
+  void EnumerateDevices(dom::MediaSourceEnum, MediaSinkEnum,
+                        nsTArray<RefPtr<MediaDevice>>*) override;
+  RefPtr<MediaEngineSource> CreateSource(const MediaDevice* aDevice) override;
+
+  MediaEventSource<void>& DeviceListChangeEvent() override {
+    return mDeviceListChangeEvent;
+  }
+  bool IsFake() const override { return false; }
+
+ private:
+  ~MediaEngineWebRTC() = default;
+  void EnumerateVideoDevices(dom::MediaSourceEnum,
+                             nsTArray<RefPtr<MediaDevice>>*);
+  void EnumerateMicrophoneDevices(nsTArray<RefPtr<MediaDevice>>*);
+  void EnumerateSpeakerDevices(nsTArray<RefPtr<MediaDevice>>*);
+
+  void DeviceListChanged() { mDeviceListChangeEvent.Notify(); }
+
+  MediaEventListener mCameraListChangeListener;
+  MediaEventListener mMicrophoneListChangeListener;
+  MediaEventListener mSpeakerListChangeListener;
+  MediaEventProducer<void> mDeviceListChangeEvent;
+};
+
+}  // namespace mozilla
+
+#endif /* NSMEDIAENGINEWEBRTC_H_ */
diff --git a/dom/media/webrtc/MediaEngineWebRTCAudio.cpp b/dom/media/webrtc/MediaEngineWebRTCAudio.cpp
new file mode 100644
index 0000000000..64ed88c625
--- /dev/null
+++ b/dom/media/webrtc/MediaEngineWebRTCAudio.cpp
@@ -0,0 +1,1329 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "MediaEngineWebRTCAudio.h"
+
+#include <stdio.h>
+#include <algorithm>
+
+#include "AudioConverter.h"
+#include "MediaManager.h"
+#include "MediaTrackGraphImpl.h"
+#include "MediaTrackConstraints.h"
+#include "mozilla/Assertions.h"
+#include "mozilla/ErrorNames.h"
+#include "nsIDUtils.h"
+#include "transport/runnable_utils.h"
+#include "Tracing.h"
+#include "mozilla/Sprintf.h"
+#include "mozilla/Logging.h"
+
+#include "common_audio/include/audio_util.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+using namespace webrtc;
+
+// These are restrictions from the webrtc.org code
+#define MAX_CHANNELS 2
+#define MONO 1
+#define MAX_SAMPLING_FREQ 48000  // Hz - multiple of 100
+
+namespace mozilla {
+
+using dom::MediaSourceEnum;
+
+extern LazyLogModule gMediaManagerLog;
+#define LOG(...) MOZ_LOG(gMediaManagerLog, LogLevel::Debug, (__VA_ARGS__))
+#define LOG_FRAME(...) \
+  MOZ_LOG(gMediaManagerLog, LogLevel::Verbose, (__VA_ARGS__))
+#define LOG_ERROR(...) MOZ_LOG(gMediaManagerLog, LogLevel::Error, (__VA_ARGS__))
+
+/**
+ * WebRTC Microphone MediaEngineSource.
+ */
+
+MediaEngineWebRTCMicrophoneSource::MediaEngineWebRTCMicrophoneSource(
+    const MediaDevice* aMediaDevice)
+    : mPrincipal(PRINCIPAL_HANDLE_NONE),
+      mDeviceInfo(aMediaDevice->mAudioDeviceInfo),
+      mDeviceMaxChannelCount(mDeviceInfo->MaxChannels()),
+      mSettings(new nsMainThreadPtrHolder<
+                media::Refcountable<dom::MediaTrackSettings>>(
+          "MediaEngineWebRTCMicrophoneSource::mSettings",
+          new media::Refcountable<dom::MediaTrackSettings>(),
+          // Non-strict means it won't assert main thread for us.
+          // It would be great if it did but we're already on the media thread.
+          /* aStrict = */ false)) {
+  MOZ_ASSERT(aMediaDevice->mMediaSource == MediaSourceEnum::Microphone);
+#ifndef ANDROID
+  MOZ_ASSERT(mDeviceInfo->DeviceID());
+#endif
+
+  // We'll init lazily as needed
+  mSettings->mEchoCancellation.Construct(0);
+  mSettings->mAutoGainControl.Construct(0);
+  mSettings->mNoiseSuppression.Construct(0);
+  mSettings->mChannelCount.Construct(0);
+
+  mState = kReleased;
+}
+
+nsresult MediaEngineWebRTCMicrophoneSource::EvaluateSettings(
+    const NormalizedConstraints& aConstraintsUpdate,
+    const MediaEnginePrefs& aInPrefs, MediaEnginePrefs* aOutPrefs,
+    const char** aOutBadConstraint) {
+  AssertIsOnOwningThread();
+
+  FlattenedConstraints c(aConstraintsUpdate);
+  MediaEnginePrefs prefs = aInPrefs;
+
+  prefs.mAecOn = c.mEchoCancellation.Get(aInPrefs.mAecOn);
+  prefs.mAgcOn = c.mAutoGainControl.Get(aInPrefs.mAgcOn && prefs.mAecOn);
+  prefs.mNoiseOn = c.mNoiseSuppression.Get(aInPrefs.mNoiseOn && prefs.mAecOn);
+
+  // Determine an actual channel count to use for this source. Three factors at
+  // play here: the device capabilities, the constraints passed in by content,
+  // and a pref that can force things (for testing)
+  int32_t maxChannels = static_cast<int32_t>(mDeviceInfo->MaxChannels());
+
+  // First, check channelCount violation wrt constraints. This fails in case of
+  // error.
+  if (c.mChannelCount.mMin > maxChannels) {
+    *aOutBadConstraint = "channelCount";
+    return NS_ERROR_FAILURE;
+  }
+  // A pref can force the channel count to use. If the pref has a value of zero
+  // or lower, it has no effect.
+  if (aInPrefs.mChannels <= 0) {
+    prefs.mChannels = maxChannels;
+  }
+
+  // Get the number of channels asked for by content, and clamp it between the
+  // pref and the maximum number of channels that the device supports.
+  prefs.mChannels = c.mChannelCount.Get(std::min(prefs.mChannels, maxChannels));
+  prefs.mChannels = std::max(1, std::min(prefs.mChannels, maxChannels));
+
+  LOG("Audio config: agc: %d, noise: %d, channels: %d",
+      prefs.mAgcOn ? prefs.mAgc : -1, prefs.mNoiseOn ? prefs.mNoise : -1,
+      prefs.mChannels);
+
+  *aOutPrefs = prefs;
+
+  return NS_OK;
+}
+
+nsresult MediaEngineWebRTCMicrophoneSource::Reconfigure(
+    const dom::MediaTrackConstraints& aConstraints,
+    const MediaEnginePrefs& aPrefs, const char** aOutBadConstraint) {
+  AssertIsOnOwningThread();
+  MOZ_ASSERT(mTrack);
+
+  LOG("Mic source %p Reconfigure ", this);
+
+  NormalizedConstraints constraints(aConstraints);
+  MediaEnginePrefs outputPrefs;
+  nsresult rv =
+      EvaluateSettings(constraints, aPrefs, &outputPrefs, aOutBadConstraint);
+  if (NS_FAILED(rv)) {
+    if (aOutBadConstraint) {
+      return NS_ERROR_INVALID_ARG;
+    }
+
+    nsAutoCString name;
+    GetErrorName(rv, name);
+    LOG("Mic source %p Reconfigure() failed unexpectedly. rv=%s", this,
+        name.Data());
+    Stop();
+    return NS_ERROR_UNEXPECTED;
+  }
+
+  ApplySettings(outputPrefs);
+
+  mCurrentPrefs = outputPrefs;
+
+  return NS_OK;
+}
+
+void MediaEngineWebRTCMicrophoneSource::ApplySettings(
+    const MediaEnginePrefs& aPrefs) {
+  AssertIsOnOwningThread();
+
+  TRACE("ApplySettings");
+  MOZ_ASSERT(
+      mTrack,
+      "ApplySetting is to be called only after SetTrack has been called");
+
+  mAudioProcessingConfig.pipeline.multi_channel_render = true;
+  mAudioProcessingConfig.pipeline.multi_channel_capture = true;
+
+  mAudioProcessingConfig.echo_canceller.enabled = aPrefs.mAecOn;
+  mAudioProcessingConfig.echo_canceller.mobile_mode = aPrefs.mUseAecMobile;
+
+  if ((mAudioProcessingConfig.gain_controller1.enabled =
+           aPrefs.mAgcOn && !aPrefs.mAgc2Forced)) {
+    auto mode = static_cast<AudioProcessing::Config::GainController1::Mode>(
+        aPrefs.mAgc);
+    if (mode != AudioProcessing::Config::GainController1::kAdaptiveAnalog &&
+        mode != AudioProcessing::Config::GainController1::kAdaptiveDigital &&
+        mode != AudioProcessing::Config::GainController1::kFixedDigital) {
+      LOG_ERROR("AudioInputProcessing %p Attempt to set invalid AGC mode %d",
+                mInputProcessing.get(), static_cast<int>(mode));
+      mode = AudioProcessing::Config::GainController1::kAdaptiveDigital;
+    }
+#if defined(WEBRTC_IOS) || defined(ATA) || defined(WEBRTC_ANDROID)
+    if (mode == AudioProcessing::Config::GainController1::kAdaptiveAnalog) {
+      LOG_ERROR(
+          "AudioInputProcessing %p Invalid AGC mode kAdaptiveAnalog on "
+          "mobile",
+          mInputProcessing.get());
+      MOZ_ASSERT_UNREACHABLE(
+          "Bad pref set in all.js or in about:config"
+          " for the auto gain, on mobile.");
+      mode = AudioProcessing::Config::GainController1::kFixedDigital;
+    }
+#endif
+    mAudioProcessingConfig.gain_controller1.mode = mode;
+  }
+  mAudioProcessingConfig.gain_controller2.enabled =
+      mAudioProcessingConfig.gain_controller2.adaptive_digital.enabled =
+          aPrefs.mAgcOn && aPrefs.mAgc2Forced;
+
+  if ((mAudioProcessingConfig.noise_suppression.enabled = aPrefs.mNoiseOn)) {
+    auto level = static_cast<AudioProcessing::Config::NoiseSuppression::Level>(
+        aPrefs.mNoise);
+    if (level != AudioProcessing::Config::NoiseSuppression::kLow &&
+        level != AudioProcessing::Config::NoiseSuppression::kModerate &&
+        level != AudioProcessing::Config::NoiseSuppression::kHigh &&
+        level != AudioProcessing::Config::NoiseSuppression::kVeryHigh) {
+      LOG_ERROR(
+          "AudioInputProcessing %p Attempt to set invalid noise suppression "
+          "level %d",
+          mInputProcessing.get(), static_cast<int>(level));
+
+      level = AudioProcessing::Config::NoiseSuppression::kModerate;
+    }
+    mAudioProcessingConfig.noise_suppression.level = level;
+  }
+
+  mAudioProcessingConfig.transient_suppression.enabled = aPrefs.mTransientOn;
+
+  mAudioProcessingConfig.high_pass_filter.enabled = aPrefs.mHPFOn;
+
+  // See https://bugs.chromium.org/p/webrtc/issues/detail?id=11539 for more
+  // info.  Our pref defaults to false, and if this is truly as unhelpful
+  // as the upstream bug claim, we could delete the pref that drive this:
+  // media.getusermedia.residual_echo_enabled.  See Bug 1779498.
+  // mAudioProcessingConfig.residual_echo_detector.enabled =
+  //    aPrefs.mResidualEchoOn;
+
+  RefPtr<MediaEngineWebRTCMicrophoneSource> that = this;
+  CubebUtils::AudioDeviceID deviceID = mDeviceInfo->DeviceID();
+  NS_DispatchToMainThread(NS_NewRunnableFunction(
+      __func__, [this, that, deviceID, track = mTrack, prefs = aPrefs,
+                 audioProcessingConfig = mAudioProcessingConfig] {
+        mSettings->mEchoCancellation.Value() = prefs.mAecOn;
+        mSettings->mAutoGainControl.Value() = prefs.mAgcOn;
+        mSettings->mNoiseSuppression.Value() = prefs.mNoiseOn;
+        mSettings->mChannelCount.Value() = prefs.mChannels;
+
+        class Message : public ControlMessage {
+          CubebUtils::AudioDeviceID mDeviceID;
+          const RefPtr<AudioInputProcessing> mInputProcessing;
+          const AudioProcessing::Config mAudioProcessingConfig;
+          const bool mPassThrough;
+          const uint32_t mRequestedInputChannelCount;
+
+         public:
+          Message(MediaTrack* aTrack, CubebUtils::AudioDeviceID aDeviceID,
+                  AudioInputProcessing* aInputProcessing,
+                  const AudioProcessing::Config& aAudioProcessingConfig,
+                  bool aPassThrough, uint32_t aRequestedInputChannelCount)
+              : ControlMessage(aTrack),
+                mDeviceID(aDeviceID),
+                mInputProcessing(aInputProcessing),
+                mAudioProcessingConfig(aAudioProcessingConfig),
+                mPassThrough(aPassThrough),
+                mRequestedInputChannelCount(aRequestedInputChannelCount) {}
+
+          void Run() override {
+            mInputProcessing->ApplyConfig(mTrack->GraphImpl(),
+                                          mAudioProcessingConfig);
+            {
+              TRACE("SetRequestedInputChannelCount");
+              mInputProcessing->SetRequestedInputChannelCount(
+                  mTrack->GraphImpl(), mDeviceID, mRequestedInputChannelCount);
+            }
+            {
+              TRACE("SetPassThrough")
+              mInputProcessing->SetPassThrough(mTrack->GraphImpl(),
+                                               mPassThrough);
+            }
+          }
+        };
+
+        // The high-pass filter is not taken into account when activating the
+        // pass through, since it's not controllable from content.
+        bool passThrough = !(prefs.mAecOn || prefs.mAgcOn || prefs.mNoiseOn);
+
+        if (track->IsDestroyed()) {
+          return;
+        }
+        track->GraphImpl()->AppendMessage(MakeUnique<Message>(
+            track, deviceID, mInputProcessing, audioProcessingConfig,
+            passThrough, prefs.mChannels));
+      }));
+}
+
+nsresult MediaEngineWebRTCMicrophoneSource::Allocate(
+    const dom::MediaTrackConstraints& aConstraints,
+    const MediaEnginePrefs& aPrefs, uint64_t aWindowID,
+    const char** aOutBadConstraint) {
+  AssertIsOnOwningThread();
+
+  mState = kAllocated;
+
+  NormalizedConstraints normalized(aConstraints);
+  MediaEnginePrefs outputPrefs;
+  nsresult rv =
+      EvaluateSettings(normalized, aPrefs, &outputPrefs, aOutBadConstraint);
+  if (NS_FAILED(rv)) {
+    return rv;
+  }
+
+  NS_DispatchToMainThread(NS_NewRunnableFunction(
+      __func__, [settings = mSettings, prefs = outputPrefs] {
+        settings->mEchoCancellation.Value() = prefs.mAecOn;
+        settings->mAutoGainControl.Value() = prefs.mAgcOn;
+        settings->mNoiseSuppression.Value() = prefs.mNoiseOn;
+        settings->mChannelCount.Value() = prefs.mChannels;
+      }));
+
+  mCurrentPrefs = outputPrefs;
+
+  return rv;
+}
+
+nsresult MediaEngineWebRTCMicrophoneSource::Deallocate() {
+  AssertIsOnOwningThread();
+
+  MOZ_ASSERT(mState == kStopped || mState == kAllocated);
+
+  class EndTrackMessage : public ControlMessage {
+    const RefPtr<AudioInputProcessing> mInputProcessing;
+
+   public:
+    explicit EndTrackMessage(AudioInputProcessing* aAudioInputProcessing)
+        : ControlMessage(nullptr), mInputProcessing(aAudioInputProcessing) {}
+
+    void Run() override {
+      TRACE("mInputProcessing::End");
+      mInputProcessing->End();
+    }
+  };
+
+  if (mTrack) {
+    NS_DispatchToMainThread(NS_NewRunnableFunction(
+        __func__,
+        [track = std::move(mTrack), inputProcessing = mInputProcessing] {
+          if (track->IsDestroyed()) {
+            // This track has already been destroyed on main thread by its
+            // DOMMediaStream. No cleanup left to do.
+            return;
+          }
+          track->GraphImpl()->AppendMessage(
+              MakeUnique<EndTrackMessage>(inputProcessing));
+        }));
+  }
+
+  // Reset all state. This is not strictly necessary, this instance will get
+  // destroyed soon.
+  mTrack = nullptr;
+  mPrincipal = PRINCIPAL_HANDLE_NONE;
+
+  // If empty, no callbacks to deliver data should be occuring
+  MOZ_ASSERT(mState != kReleased, "Source not allocated");
+  MOZ_ASSERT(mState != kStarted, "Source not stopped");
+
+  mState = kReleased;
+  LOG("Mic source %p Audio device %s deallocated", this,
+      NS_ConvertUTF16toUTF8(mDeviceInfo->Name()).get());
+  return NS_OK;
+}
+
+void MediaEngineWebRTCMicrophoneSource::SetTrack(
+    const RefPtr<MediaTrack>& aTrack, const PrincipalHandle& aPrincipal) {
+  AssertIsOnOwningThread();
+  MOZ_ASSERT(aTrack);
+  MOZ_ASSERT(aTrack->AsAudioProcessingTrack());
+
+  MOZ_ASSERT(!mTrack);
+  MOZ_ASSERT(mPrincipal == PRINCIPAL_HANDLE_NONE);
+  mTrack = aTrack->AsAudioProcessingTrack();
+  mPrincipal = aPrincipal;
+
+  mInputProcessing =
+      MakeAndAddRef<AudioInputProcessing>(mDeviceMaxChannelCount);
+
+  NS_DispatchToMainThread(NS_NewRunnableFunction(
+      __func__, [track = mTrack, processing = mInputProcessing]() mutable {
+        track->SetInputProcessing(std::move(processing));
+        track->Resume();  // Suspended by MediaManager
+      }));
+
+  LOG("Mic source %p Track %p registered for microphone capture", this,
+      aTrack.get());
+}
+
+class StartStopMessage : public ControlMessage {
+ public:
+  enum StartStop { Start, Stop };
+
+  StartStopMessage(MediaTrack* aTrack, AudioInputProcessing* aInputProcessing,
+                   StartStop aAction)
+      : ControlMessage(aTrack),
+        mInputProcessing(aInputProcessing),
+        mAction(aAction) {}
+
+  void Run() override {
+    if (mAction == StartStopMessage::Start) {
+      TRACE("InputProcessing::Start")
+      mInputProcessing->Start(mTrack->GraphImpl());
+    } else if (mAction == StartStopMessage::Stop) {
+      TRACE("InputProcessing::Stop")
+      mInputProcessing->Stop(mTrack->GraphImpl());
+    } else {
+      MOZ_CRASH("Invalid enum value");
+    }
+  }
+
+ protected:
+  const RefPtr<AudioInputProcessing> mInputProcessing;
+  const StartStop mAction;
+};
+
+nsresult MediaEngineWebRTCMicrophoneSource::Start() {
+  AssertIsOnOwningThread();
+
+  // This spans setting both the enabled state and mState.
+  if (mState == kStarted) {
+    return NS_OK;
+  }
+
+  MOZ_ASSERT(mState == kAllocated || mState == kStopped);
+
+  ApplySettings(mCurrentPrefs);
+
+  CubebUtils::AudioDeviceID deviceID = mDeviceInfo->DeviceID();
+  NS_DispatchToMainThread(NS_NewRunnableFunction(
+      __func__, [inputProcessing = mInputProcessing, deviceID, track = mTrack,
+                 principal = mPrincipal] {
+        if (track->IsDestroyed()) {
+          return;
+        }
+
+        track->GraphImpl()->AppendMessage(MakeUnique<StartStopMessage>(
+            track, inputProcessing, StartStopMessage::Start));
+        track->ConnectDeviceInput(deviceID, inputProcessing.get(), principal);
+      }));
+
+  MOZ_ASSERT(mState != kReleased);
+  mState = kStarted;
+
+  return NS_OK;
+}
+
+nsresult MediaEngineWebRTCMicrophoneSource::Stop() {
+  AssertIsOnOwningThread();
+
+  LOG("Mic source %p Stop()", this);
+  MOZ_ASSERT(mTrack, "SetTrack must have been called before ::Stop");
+
+  if (mState == kStopped) {
+    // Already stopped - this is allowed
+    return NS_OK;
+  }
+
+  NS_DispatchToMainThread(NS_NewRunnableFunction(
+      __func__, [inputProcessing = mInputProcessing, deviceInfo = mDeviceInfo,
+                 track = mTrack] {
+        if (track->IsDestroyed()) {
+          return;
+        }
+
+        MOZ_ASSERT(track->DeviceId().value() == deviceInfo->DeviceID());
+        track->DisconnectDeviceInput();
+        track->GraphImpl()->AppendMessage(MakeUnique<StartStopMessage>(
+            track, inputProcessing, StartStopMessage::Stop));
+      }));
+
+  MOZ_ASSERT(mState == kStarted, "Should be started when stopping");
+  mState = kStopped;
+
+  return NS_OK;
+}
+
+void MediaEngineWebRTCMicrophoneSource::GetSettings(
+    dom::MediaTrackSettings& aOutSettings) const {
+  MOZ_ASSERT(NS_IsMainThread());
+  aOutSettings = *mSettings;
+}
+
+AudioInputProcessing::AudioInputProcessing(uint32_t aMaxChannelCount)
+    : mAudioProcessing(AudioProcessingBuilder().Create().release()),
+      mRequestedInputChannelCount(aMaxChannelCount),
+      mSkipProcessing(false),
+      mInputDownmixBuffer(MAX_SAMPLING_FREQ * MAX_CHANNELS / 100),
+      mEnabled(false),
+      mEnded(false),
+      mPacketCount(0) {}
+
+void AudioInputProcessing::Disconnect(MediaTrackGraphImpl* aGraph) {
+  // This method is just for asserts.
+  MOZ_ASSERT(aGraph->OnGraphThread());
+}
+
+bool AudioInputProcessing::PassThrough(MediaTrackGraphImpl* aGraph) const {
+  MOZ_ASSERT(aGraph->OnGraphThread());
+  return mSkipProcessing;
+}
+
+void AudioInputProcessing::SetPassThrough(MediaTrackGraphImpl* aGraph,
+                                          bool aPassThrough) {
+  MOZ_ASSERT(aGraph->OnGraphThread());
+
+  if (aPassThrough == mSkipProcessing) {
+    return;
+  }
+  mSkipProcessing = aPassThrough;
+
+  if (!mEnabled) {
+    MOZ_ASSERT(!mPacketizerInput);
+    return;
+  }
+
+  if (aPassThrough) {
+    // Turn on pass-through
+    ResetAudioProcessing(aGraph);
+  } else {
+    // Turn off pass-through
+    MOZ_ASSERT(!mPacketizerInput);
+    EnsureAudioProcessing(aGraph, mRequestedInputChannelCount);
+  }
+}
+
+uint32_t AudioInputProcessing::GetRequestedInputChannelCount() {
+  return mRequestedInputChannelCount;
+}
+
+void AudioInputProcessing::SetRequestedInputChannelCount(
+    MediaTrackGraphImpl* aGraph, CubebUtils::AudioDeviceID aDeviceId,
+    uint32_t aRequestedInputChannelCount) {
+  mRequestedInputChannelCount = aRequestedInputChannelCount;
+
+  aGraph->ReevaluateInputDevice(aDeviceId);
+}
+
+void AudioInputProcessing::Start(MediaTrackGraphImpl* aGraph) {
+  MOZ_ASSERT(aGraph->OnGraphThread());
+
+  if (mEnabled) {
+    return;
+  }
+  mEnabled = true;
+
+  if (mSkipProcessing) {
+    return;
+  }
+
+  MOZ_ASSERT(!mPacketizerInput);
+  EnsureAudioProcessing(aGraph, mRequestedInputChannelCount);
+}
+
+void AudioInputProcessing::Stop(MediaTrackGraphImpl* aGraph) {
+  MOZ_ASSERT(aGraph->OnGraphThread());
+
+  if (!mEnabled) {
+    return;
+  }
+
+  mEnabled = false;
+
+  if (mSkipProcessing) {
+    return;
+  }
+
+  // Packetizer is active and we were just stopped. Stop the packetizer and
+  // processing.
+  ResetAudioProcessing(aGraph);
+}
+
+// The following is how how Process() works in pass-through and non-pass-through
+// mode. In both mode, Process() outputs the same amount of the frames as its
+// input data.
+//
+// I. In non-pass-through mode:
+//
+// We will use webrtc::AudioProcessing to process the input audio data in this
+// mode. The data input in webrtc::AudioProcessing needs to be a 10ms chunk,
+// while the input data passed to Process() is not necessary to have times of
+// 10ms-chunk length. To divide the input data into 10ms chunks,
+// mPacketizerInput is introduced.
+//
+// We will add one 10ms-chunk silence into the internal buffer before Process()
+// works. Those extra frames is called pre-buffering. It aims to avoid glitches
+// we may have when producing data in mPacketizerInput. Without pre-buffering,
+// when the input data length is not 10ms-times, we could end up having no
+// enough output needs since mPacketizerInput would keep some input data, which
+// is the remainder of the 10ms-chunk length. To force processing those data
+// left in mPacketizerInput, we would need to add some extra frames to make
+// mPacketizerInput produce a 10ms-chunk. For example, if the sample rate is
+// 44100 Hz, then the packet-size is 441 frames. When we only have 384 input
+// frames, we would need to put additional 57 frames to mPacketizerInput to
+// produce a packet. However, those extra 57 frames result in a glitch sound.
+//
+// By adding one 10ms-chunk silence in advance to the internal buffer, we won't
+// need to add extra frames between the input data no matter what data length it
+// is. The only drawback is the input data won't be processed and send to output
+// immediately. Process() will consume pre-buffering data for its output first.
+// The below describes how it works:
+//
+//
+//                          Process()
+//               +-----------------------------+
+//   input D(N)  |   +--------+   +--------+   |  output D(N)
+// --------------|-->|  P(N)  |-->|  S(N)  |---|-------------->
+//               |   +--------+   +--------+   |
+//               |   packetizer    mSegment    |
+//               +-----------------------------+
+//               <------ internal buffer ------>
+//
+//
+//   D(N): number of frames from the input and the output needs in the N round
+//      Z: number of frames of a 10ms chunk(packet) in mPacketizerInput, Z >= 1
+//         (if Z = 1, packetizer has no effect)
+//   P(N): number of frames left in mPacketizerInput after the N round. Once the
+//         frames in packetizer >= Z, packetizer will produce a packet to
+//         mSegment, so P(N) = (P(N-1) + D(N)) % Z, 0 <= P(N) <= Z-1
+//   S(N): number of frames left in mSegment after the N round. The input D(N)
+//         frames will be passed to mPacketizerInput first, and then
+//         mPacketizerInput may append some packets to mSegment, so
+//         S(N) = S(N-1) + Z * floor((P(N-1) + D(N)) / Z) - D(N)
+//
+// At the first, we set P(0) = 0, S(0) = X, where X >= Z-1. X is the
+// pre-buffering put in the internal buffer. With this settings, P(K) + S(K) = X
+// always holds.
+//
+// Intuitively, this seems true: We put X frames in the internal buffer at
+// first. If the data won't be blocked in packetizer, after the Process(), the
+// internal buffer should still hold X frames since the number of frames coming
+// from input is the same as the output needs. The key of having enough data for
+// output needs, while the input data is piled up in packetizer, is by putting
+// at least Z-1 frames as pre-buffering, since the maximum number of frames
+// stuck in the packetizer before it can emit a packet is packet-size - 1.
+// Otherwise, we don't have enough data for output if the new input data plus
+// the data left in packetizer produces a smaller-than-10ms chunk, which will be
+// left in packetizer. Thus we must have some pre-buffering frames in the
+// mSegment to make up the length of the left chunk we need for output. This can
+// also be told by by induction:
+//   (1) This holds when K = 0
+//   (2) Assume this holds when K = N: so P(N) + S(N) = X
+//       => P(N) + S(N) = X >= Z-1 => S(N) >= Z-1-P(N)
+//   (3) When K = N+1, we have D(N+1) input frames comes
+//     a. if P(N) + D(N+1) < Z, then packetizer has no enough data for one
+//        packet. No data produced by packertizer, so the mSegment now has
+//        S(N) >= Z-1-P(N) frames. Output needs D(N+1) < Z-P(N) frames. So it
+//        needs at most Z-P(N)-1 frames, and mSegment has enough frames for
+//        output, Then, P(N+1) = P(N) + D(N+1) and S(N+1) = S(N) - D(N+1)
+//        => P(N+1) + S(N+1) = P(N) + S(N) = X
+//     b. if P(N) + D(N+1) = Z, then packetizer will produce one packet for
+//        mSegment, so mSegment now has S(N) + Z frames. Output needs D(N+1)
+//        = Z-P(N) frames. S(N) has at least Z-1-P(N)+Z >= Z-P(N) frames, since
+//        Z >= 1. So mSegment has enough frames for output. Then, P(N+1) = 0 and
+//        S(N+1) = S(N) + Z - D(N+1) = S(N) + P(N)
+//        => P(N+1) + S(N+1) = P(N) + S(N) = X
+//     c. if P(N) + D(N+1) > Z, and let P(N) + D(N+1) = q * Z + r, where q >= 1
+//        and 0 <= r <= Z-1, then packetizer will produce can produce q packets
+//        for mSegment. Output needs D(N+1) = q * Z - P(N) + r frames and
+//        mSegment has S(N) + q * z >= q * z - P(N) + Z-1 >= q*z -P(N) + r,
+//        since r <= Z-1. So mSegment has enough frames for output. Then,
+//        P(N+1) = r and S(N+1) = S(N) + q * Z - D(N+1)
+//         => P(N+1) + S(N+1) = S(N) + (q * Z + r - D(N+1)) =  S(N) + P(N) = X
+//   => P(K) + S(K) = X always holds
+//
+// Since P(K) + S(K) = X and P(K) is in [0, Z-1], the S(K) is in [X-Z+1, X]
+// range. In our implementation, X is set to Z so S(K) is in [1, Z].
+// By the above workflow, we always have enough data for output and no extra
+// frames put into packetizer. It means we don't have any glitch!
+//
+// II. In pass-through mode:
+//
+//                Process()
+//               +--------+
+//   input D(N)  |        |  output D(N)
+// -------------->-------->--------------->
+//               |        |
+//               +--------+
+//
+// The D(N) frames of data are just forwarded from input to output without any
+// processing
+void AudioInputProcessing::Process(MediaTrackGraphImpl* aGraph, GraphTime aFrom,
+                                   GraphTime aTo, AudioSegment* aInput,
+                                   AudioSegment* aOutput) {
+  MOZ_ASSERT(aGraph->OnGraphThread());
+  MOZ_ASSERT(aFrom <= aTo);
+  MOZ_ASSERT(!mEnded);
+
+  TrackTime need = aTo - aFrom;
+  if (need == 0) {
+    return;
+  }
+
+  if (!mEnabled) {
+    LOG_FRAME("(Graph %p, Driver %p) AudioInputProcessing %p Filling %" PRId64
+              " frames of silence to output (disabled)",
+              aGraph, aGraph->CurrentDriver(), this, need);
+    aOutput->AppendNullData(need);
+    return;
+  }
+
+  MOZ_ASSERT(aInput->GetDuration() == need,
+             "Wrong data length from input port source");
+
+  if (PassThrough(aGraph)) {
+    LOG_FRAME(
+        "(Graph %p, Driver %p) AudioInputProcessing %p Forwarding %" PRId64
+        " frames of input data to output directly (PassThrough)",
+        aGraph, aGraph->CurrentDriver(), this, aInput->GetDuration());
+    aOutput->AppendSegment(aInput);
+    return;
+  }
+
+  // SetPassThrough(false) must be called before reaching here.
+  MOZ_ASSERT(mPacketizerInput);
+  // If mRequestedInputChannelCount is updated, create a new packetizer. No
+  // need to change the pre-buffering since the rate is always the same. The
+  // frames left in the packetizer would be replaced by null data and then
+  // transferred to mSegment.
+  EnsureAudioProcessing(aGraph, mRequestedInputChannelCount);
+
+  // Preconditions of the audio-processing logic.
+  MOZ_ASSERT(static_cast<uint32_t>(mSegment.GetDuration()) +
+                 mPacketizerInput->FramesAvailable() ==
+             mPacketizerInput->mPacketSize);
+  // We pre-buffer mPacketSize frames, but the maximum number of frames stuck in
+  // the packetizer before it can emit a packet is mPacketSize-1. Thus that
+  // remaining 1 frame will always be present in mSegment.
+  MOZ_ASSERT(mSegment.GetDuration() >= 1);
+  MOZ_ASSERT(mSegment.GetDuration() <= mPacketizerInput->mPacketSize);
+
+  PacketizeAndProcess(aGraph, *aInput);
+  LOG_FRAME("(Graph %p, Driver %p) AudioInputProcessing %p Buffer has %" PRId64
+            " frames of data now, after packetizing and processing",
+            aGraph, aGraph->CurrentDriver(), this, mSegment.GetDuration());
+
+  // By setting pre-buffering to the number of frames of one packet, and
+  // because the maximum number of frames stuck in the packetizer before
+  // it can emit a packet is the mPacketSize-1, we always have at least
+  // one more frame than output needs.
+  MOZ_ASSERT(mSegment.GetDuration() > need);
+  aOutput->AppendSlice(mSegment, 0, need);
+  mSegment.RemoveLeading(need);
+  LOG_FRAME("(Graph %p, Driver %p) AudioInputProcessing %p moving %" PRId64
+            " frames of data to output, leaving %" PRId64 " frames in buffer",
+            aGraph, aGraph->CurrentDriver(), this, need,
+            mSegment.GetDuration());
+
+  // Postconditions of the audio-processing logic.
+  MOZ_ASSERT(static_cast<uint32_t>(mSegment.GetDuration()) +
+                 mPacketizerInput->FramesAvailable() ==
+             mPacketizerInput->mPacketSize);
+  MOZ_ASSERT(mSegment.GetDuration() >= 1);
+  MOZ_ASSERT(mSegment.GetDuration() <= mPacketizerInput->mPacketSize);
+}
+
+void AudioInputProcessing::ProcessOutputData(MediaTrackGraphImpl* aGraph,
+                                             AudioDataValue* aBuffer,
+                                             size_t aFrames, TrackRate aRate,
+                                             uint32_t aChannels) {
+  MOZ_ASSERT(aGraph->OnGraphThread());
+
+  if (!mEnabled || PassThrough(aGraph)) {
+    return;
+  }
+
+  if (!mPacketizerOutput ||
+      mPacketizerOutput->mPacketSize != GetPacketSize(aRate) ||
+      mPacketizerOutput->mChannels != aChannels) {
+    // It's ok to drop the audio still in the packetizer here: if this changes,
+    // we changed devices or something.
+    mPacketizerOutput = Nothing();
+    mPacketizerOutput.emplace(GetPacketSize(aRate), aChannels);
+  }
+
+  mPacketizerOutput->Input(aBuffer, aFrames);
+
+  while (mPacketizerOutput->PacketsAvailable()) {
+    uint32_t samplesPerPacket =
+        mPacketizerOutput->mPacketSize * mPacketizerOutput->mChannels;
+    if (mOutputBuffer.Length() < samplesPerPacket) {
+      mOutputBuffer.SetLength(samplesPerPacket);
+    }
+    if (mDeinterleavedBuffer.Length() < samplesPerPacket) {
+      mDeinterleavedBuffer.SetLength(samplesPerPacket);
+    }
+    float* packet = mOutputBuffer.Data();
+    mPacketizerOutput->Output(packet);
+
+    AutoTArray<float*, MAX_CHANNELS> deinterleavedPacketDataChannelPointers;
+    float* interleavedFarend = nullptr;
+    uint32_t channelCountFarend = 0;
+    uint32_t framesPerPacketFarend = 0;
+
+    // Downmix from aChannels to MAX_CHANNELS if needed. We always have
+    // floats here, the packetized performed the conversion.
+    if (aChannels > MAX_CHANNELS) {
+      AudioConverter converter(
+          AudioConfig(aChannels, 0, AudioConfig::FORMAT_FLT),
+          AudioConfig(MAX_CHANNELS, 0, AudioConfig::FORMAT_FLT));
+      framesPerPacketFarend = mPacketizerOutput->mPacketSize;
+      framesPerPacketFarend =
+          converter.Process(mInputDownmixBuffer, packet, framesPerPacketFarend);
+      interleavedFarend = mInputDownmixBuffer.Data();
+      channelCountFarend = MAX_CHANNELS;
+      deinterleavedPacketDataChannelPointers.SetLength(MAX_CHANNELS);
+    } else {
+      interleavedFarend = packet;
+      channelCountFarend = aChannels;
+      framesPerPacketFarend = mPacketizerOutput->mPacketSize;
+      deinterleavedPacketDataChannelPointers.SetLength(aChannels);
+    }
+
+    MOZ_ASSERT(interleavedFarend &&
+               (channelCountFarend == 1 || channelCountFarend == 2) &&
+               framesPerPacketFarend);
+
+    if (mInputBuffer.Length() < framesPerPacketFarend * channelCountFarend) {
+      mInputBuffer.SetLength(framesPerPacketFarend * channelCountFarend);
+    }
+
+    size_t offset = 0;
+    for (size_t i = 0; i < deinterleavedPacketDataChannelPointers.Length();
+         ++i) {
+      deinterleavedPacketDataChannelPointers[i] = mInputBuffer.Data() + offset;
+      offset += framesPerPacketFarend;
+    }
+
+    // Deinterleave, prepare a channel pointers array, with enough storage for
+    // the frames.
+    DeinterleaveAndConvertBuffer(
+        interleavedFarend, framesPerPacketFarend, channelCountFarend,
+        deinterleavedPacketDataChannelPointers.Elements());
+
+    // Having the same config for input and output means we potentially save
+    // some CPU.
+    StreamConfig inputConfig(aRate, channelCountFarend);
+    StreamConfig outputConfig = inputConfig;
+
+    // Passing the same pointers here saves a copy inside this function.
+    DebugOnly<int> err = mAudioProcessing->ProcessReverseStream(
+        deinterleavedPacketDataChannelPointers.Elements(), inputConfig,
+        outputConfig, deinterleavedPacketDataChannelPointers.Elements());
+
+    MOZ_ASSERT(!err, "Could not process the reverse stream.");
+  }
+}
+
+// Only called if we're not in passthrough mode
+void AudioInputProcessing::PacketizeAndProcess(MediaTrackGraphImpl* aGraph,
+                                               const AudioSegment& aSegment) {
+  MOZ_ASSERT(!PassThrough(aGraph),
+             "This should be bypassed when in PassThrough mode.");
+  MOZ_ASSERT(mEnabled);
+  MOZ_ASSERT(mPacketizerInput);
+  MOZ_ASSERT(mPacketizerInput->mPacketSize ==
+             GetPacketSize(aGraph->GraphRate()));
+
+  // Calculate number of the pending frames in mChunksInPacketizer.
+  auto pendingFrames = [&]() {
+    TrackTime frames = 0;
+    for (const auto& p : mChunksInPacketizer) {
+      frames += p.first;
+    }
+    return frames;
+  };
+
+  // Precondition of the Principal-labelling logic below.
+  MOZ_ASSERT(mPacketizerInput->FramesAvailable() ==
+             static_cast<uint32_t>(pendingFrames()));
+
+  // The WriteToInterleavedBuffer will do upmix or downmix if the channel-count
+  // in aSegment's chunks is different from mPacketizerInput->mChannels
+  // WriteToInterleavedBuffer could be avoided once Bug 1729041 is done.
+  size_t sampleCount = aSegment.WriteToInterleavedBuffer(
+      mInterleavedBuffer, mPacketizerInput->mChannels);
+  size_t frameCount =
+      sampleCount / static_cast<size_t>(mPacketizerInput->mChannels);
+
+  // Packetize our input data into 10ms chunks, deinterleave into planar channel
+  // buffers, process, and append to the right MediaStreamTrack.
+  mPacketizerInput->Input(mInterleavedBuffer.Elements(),
+                          static_cast<uint32_t>(frameCount));
+
+  // Update mChunksInPacketizer and make sure the precondition for the
+  // Principal-labelling logic still holds.
+  for (AudioSegment::ConstChunkIterator iter(aSegment); !iter.IsEnded();
+       iter.Next()) {
+    MOZ_ASSERT(iter->mDuration > 0);
+    mChunksInPacketizer.emplace_back(
+        std::make_pair(iter->mDuration, iter->mPrincipalHandle));
+  }
+  MOZ_ASSERT(mPacketizerInput->FramesAvailable() ==
+             static_cast<uint32_t>(pendingFrames()));
+
+  LOG_FRAME(
+      "(Graph %p, Driver %p) AudioInputProcessing %p Packetizing %zu frames. "
+      "Packetizer has %u frames (enough for %u packets) now",
+      aGraph, aGraph->CurrentDriver(), this, frameCount,
+      mPacketizerInput->FramesAvailable(),
+      mPacketizerInput->PacketsAvailable());
+
+  size_t offset = 0;
+
+  while (mPacketizerInput->PacketsAvailable()) {
+    mPacketCount++;
+    uint32_t samplesPerPacket =
+        mPacketizerInput->mPacketSize * mPacketizerInput->mChannels;
+    if (mInputBuffer.Length() < samplesPerPacket) {
+      mInputBuffer.SetLength(samplesPerPacket);
+    }
+    if (mDeinterleavedBuffer.Length() < samplesPerPacket) {
+      mDeinterleavedBuffer.SetLength(samplesPerPacket);
+    }
+    float* packet = mInputBuffer.Data();
+    mPacketizerInput->Output(packet);
+
+    // Downmix from mPacketizerInput->mChannels to mono if needed. We always
+    // have floats here, the packetizer performed the conversion.
+    AutoTArray<float*, 8> deinterleavedPacketizedInputDataChannelPointers;
+    uint32_t channelCountInput = 0;
+    if (mPacketizerInput->mChannels > MAX_CHANNELS) {
+      channelCountInput = MONO;
+      deinterleavedPacketizedInputDataChannelPointers.SetLength(
+          channelCountInput);
+      deinterleavedPacketizedInputDataChannelPointers[0] =
+          mDeinterleavedBuffer.Data();
+      // Downmix to mono (and effectively have a planar buffer) by summing all
+      // channels in the first channel, and scaling by the number of channels to
+      // avoid clipping.
+      float gain = 1.f / mPacketizerInput->mChannels;
+      size_t readIndex = 0;
+      for (size_t i = 0; i < mPacketizerInput->mPacketSize; i++) {
+        mDeinterleavedBuffer.Data()[i] = 0.;
+        for (size_t j = 0; j < mPacketizerInput->mChannels; j++) {
+          mDeinterleavedBuffer.Data()[i] += gain * packet[readIndex++];
+        }
+      }
+    } else {
+      channelCountInput = mPacketizerInput->mChannels;
+      // Deinterleave the input data
+      // Prepare an array pointing to deinterleaved channels.
+      deinterleavedPacketizedInputDataChannelPointers.SetLength(
+          channelCountInput);
+      offset = 0;
+      for (size_t i = 0;
+           i < deinterleavedPacketizedInputDataChannelPointers.Length(); ++i) {
+        deinterleavedPacketizedInputDataChannelPointers[i] =
+            mDeinterleavedBuffer.Data() + offset;
+        offset += mPacketizerInput->mPacketSize;
+      }
+      // Deinterleave to mInputBuffer, pointed to by inputBufferChannelPointers.
+      Deinterleave(packet, mPacketizerInput->mPacketSize, channelCountInput,
+                   deinterleavedPacketizedInputDataChannelPointers.Elements());
+    }
+
+    StreamConfig inputConfig(aGraph->GraphRate(), channelCountInput);
+    StreamConfig outputConfig = inputConfig;
+
+    // Bug 1404965: Get the right delay here, it saves some work down the line.
+    mAudioProcessing->set_stream_delay_ms(0);
+
+    // Bug 1414837: find a way to not allocate here.
+    CheckedInt<size_t> bufferSize(sizeof(float));
+    bufferSize *= mPacketizerInput->mPacketSize;
+    bufferSize *= channelCountInput;
+    RefPtr<SharedBuffer> buffer = SharedBuffer::Create(bufferSize);
+
+    // Prepare channel pointers to the SharedBuffer created above.
+    AutoTArray<float*, 8> processedOutputChannelPointers;
+    AutoTArray<const float*, 8> processedOutputChannelPointersConst;
+    processedOutputChannelPointers.SetLength(channelCountInput);
+    processedOutputChannelPointersConst.SetLength(channelCountInput);
+
+    offset = 0;
+    for (size_t i = 0; i < processedOutputChannelPointers.Length(); ++i) {
+      processedOutputChannelPointers[i] =
+          static_cast<float*>(buffer->Data()) + offset;
+      processedOutputChannelPointersConst[i] =
+          static_cast<float*>(buffer->Data()) + offset;
+      offset += mPacketizerInput->mPacketSize;
+    }
+
+    mAudioProcessing->ProcessStream(
+        deinterleavedPacketizedInputDataChannelPointers.Elements(), inputConfig,
+        outputConfig, processedOutputChannelPointers.Elements());
+
+    // If logging is enabled, dump the audio processing stats twice a second
+    if (MOZ_LOG_TEST(gMediaManagerLog, LogLevel::Debug) &&
+        !(mPacketCount % 50)) {
+      AudioProcessingStats stats = mAudioProcessing->GetStatistics();
+      char msg[1024];
+      size_t offset = 0;
+#define AddIfValue(format, member)                                       \
+  if (stats.member.has_value()) {                                        \
+    offset += SprintfBuf(msg + offset, sizeof(msg) - offset,             \
+                         #member ":" format ", ", stats.member.value()); \
+  }
+      AddIfValue("%d", voice_detected);
+      AddIfValue("%lf", echo_return_loss);
+      AddIfValue("%lf", echo_return_loss_enhancement);
+      AddIfValue("%lf", divergent_filter_fraction);
+      AddIfValue("%d", delay_median_ms);
+      AddIfValue("%d", delay_standard_deviation_ms);
+      AddIfValue("%lf", residual_echo_likelihood);
+      AddIfValue("%lf", residual_echo_likelihood_recent_max);
+      AddIfValue("%d", delay_ms);
+#undef AddIfValue
+      LOG("AudioProcessing statistics: %s", msg);
+    }
+
+    if (mEnded) {
+      continue;
+    }
+
+    // We already have planar audio data of the right format. Insert into the
+    // MTG.
+    MOZ_ASSERT(processedOutputChannelPointers.Length() == channelCountInput);
+
+    // Insert the processed data chunk by chunk to mSegment with the paired
+    // PrincipalHandle value. The chunks are tracked in mChunksInPacketizer.
+
+    auto getAudioChunk = [&](TrackTime aStart, TrackTime aEnd,
+                             const PrincipalHandle& aPrincipalHandle) {
+      if (aStart == aEnd) {
+        return AudioChunk();
+      }
+      RefPtr<SharedBuffer> other = buffer;
+      AudioChunk c =
+          AudioChunk(other.forget(), processedOutputChannelPointersConst,
+                     static_cast<TrackTime>(mPacketizerInput->mPacketSize),
+                     aPrincipalHandle);
+      c.SliceTo(aStart, aEnd);
+      return c;
+    };
+
+    // The number of frames of data that needs to be labelled with Principal
+    // values.
+    TrackTime len = static_cast<TrackTime>(mPacketizerInput->mPacketSize);
+    // The start offset of the unlabelled chunk.
+    TrackTime start = 0;
+    // By mChunksInPacketizer's information, we can keep labelling the
+    // unlabelled frames chunk by chunk.
+    while (!mChunksInPacketizer.empty()) {
+      auto& [frames, principal] = mChunksInPacketizer.front();
+      const TrackTime end = start + frames;
+      if (end > len) {
+        // If the left unlabelled frames are part of this chunk, then we need to
+        // adjust the number of frames in the chunk.
+        if (len > start) {
+          mSegment.AppendAndConsumeChunk(getAudioChunk(start, len, principal));
+          frames -= len - start;
+        }
+        break;
+      }
+      // Otherwise, the number of unlabelled frames is larger than or equal to
+      // this chunk. We can label the whole chunk directly.
+      mSegment.AppendAndConsumeChunk(getAudioChunk(start, end, principal));
+      start = end;
+      mChunksInPacketizer.pop_front();
+    }
+
+    LOG_FRAME(
+        "(Graph %p, Driver %p) AudioInputProcessing %p Appending %u frames of "
+        "packetized audio, leaving %u frames in packetizer (%" PRId64
+        " frames in mChunksInPacketizer)",
+        aGraph, aGraph->CurrentDriver(), this, mPacketizerInput->mPacketSize,
+        mPacketizerInput->FramesAvailable(), pendingFrames());
+
+    // Postcondition of the Principal-labelling logic.
+    MOZ_ASSERT(mPacketizerInput->FramesAvailable() ==
+               static_cast<uint32_t>(pendingFrames()));
+  }
+}
+
+void AudioInputProcessing::DeviceChanged(MediaTrackGraphImpl* aGraph) {
+  MOZ_ASSERT(aGraph->OnGraphThread());
+
+  // Reset some processing
+  mAudioProcessing->Initialize();
+  LOG_FRAME(
+      "(Graph %p, Driver %p) AudioInputProcessing %p Reinitializing audio "
+      "processing",
+      aGraph, aGraph->CurrentDriver(), this);
+}
+
+void AudioInputProcessing::ApplyConfig(MediaTrackGraphImpl* aGraph,
+                                       const AudioProcessing::Config& aConfig) {
+  MOZ_ASSERT(aGraph->OnGraphThread());
+  mAudioProcessing->ApplyConfig(aConfig);
+}
+
+void AudioInputProcessing::End() {
+  mEnded = true;
+  mSegment.Clear();
+}
+
+TrackTime AudioInputProcessing::NumBufferedFrames(
+    MediaTrackGraphImpl* aGraph) const {
+  MOZ_ASSERT(aGraph->OnGraphThread());
+  return mSegment.GetDuration();
+}
+
+void AudioInputProcessing::EnsureAudioProcessing(MediaTrackGraphImpl* aGraph,
+                                                 uint32_t aChannels) {
+  MOZ_ASSERT(aGraph->OnGraphThread());
+  MOZ_ASSERT(aChannels > 0);
+  MOZ_ASSERT(mEnabled);
+  MOZ_ASSERT(!mSkipProcessing);
+
+  if (mPacketizerInput && mPacketizerInput->mChannels == aChannels) {
+    return;
+  }
+
+  // If mPacketizerInput exists but with different channel-count, there is no
+  // need to change pre-buffering since the packet size is the same as the old
+  // one, since the rate is a constant.
+  MOZ_ASSERT_IF(mPacketizerInput, mPacketizerInput->mPacketSize ==
+                                      GetPacketSize(aGraph->GraphRate()));
+  bool needPreBuffering = !mPacketizerInput;
+  if (mPacketizerInput) {
+    const TrackTime numBufferedFrames =
+        static_cast<TrackTime>(mPacketizerInput->FramesAvailable());
+    mSegment.AppendNullData(numBufferedFrames);
+    mPacketizerInput = Nothing();
+    mChunksInPacketizer.clear();
+  }
+
+  mPacketizerInput.emplace(GetPacketSize(aGraph->GraphRate()), aChannels);
+
+  if (needPreBuffering) {
+    LOG_FRAME(
+        "(Graph %p, Driver %p) AudioInputProcessing %p: Adding %u frames of "
+        "silence as pre-buffering",
+        aGraph, aGraph->CurrentDriver(), this, mPacketizerInput->mPacketSize);
+
+    AudioSegment buffering;
+    buffering.AppendNullData(
+        static_cast<TrackTime>(mPacketizerInput->mPacketSize));
+    PacketizeAndProcess(aGraph, buffering);
+  }
+}
+
+void AudioInputProcessing::ResetAudioProcessing(MediaTrackGraphImpl* aGraph) {
+  MOZ_ASSERT(aGraph->OnGraphThread());
+  MOZ_ASSERT(mSkipProcessing || !mEnabled);
+  MOZ_ASSERT(mPacketizerInput);
+
+  LOG_FRAME(
+      "(Graph %p, Driver %p) AudioInputProcessing %p Resetting audio "
+      "processing",
+      aGraph, aGraph->CurrentDriver(), this);
+
+  // Reset AudioProcessing so that if we resume processing in the future it
+  // doesn't depend on old state.
+  mAudioProcessing->Initialize();
+
+  MOZ_ASSERT(static_cast<uint32_t>(mSegment.GetDuration()) +
+                 mPacketizerInput->FramesAvailable() ==
+             mPacketizerInput->mPacketSize);
+
+  // It's ok to clear all the internal buffer here since we won't use mSegment
+  // in pass-through mode or when audio processing is disabled.
+  LOG_FRAME(
+      "(Graph %p, Driver %p) AudioInputProcessing %p Emptying out %" PRId64
+      " frames of data",
+      aGraph, aGraph->CurrentDriver(), this, mSegment.GetDuration());
+  mSegment.Clear();
+
+  mPacketizerInput = Nothing();
+  mChunksInPacketizer.clear();
+}
+
+void AudioProcessingTrack::Destroy() {
+  MOZ_ASSERT(NS_IsMainThread());
+  DisconnectDeviceInput();
+
+  MediaTrack::Destroy();
+}
+
+void AudioProcessingTrack::SetInputProcessing(
+    RefPtr<AudioInputProcessing> aInputProcessing) {
+  class Message : public ControlMessage {
+    const RefPtr<AudioProcessingTrack> mTrack;
+    const RefPtr<AudioInputProcessing> mProcessing;
+
+   public:
+    Message(RefPtr<AudioProcessingTrack> aTrack,
+            RefPtr<AudioInputProcessing> aProcessing)
+        : ControlMessage(aTrack),
+          mTrack(std::move(aTrack)),
+          mProcessing(std::move(aProcessing)) {}
+    void Run() override {
+      TRACE("AudioProcessingTrack::SetInputProcessingImpl");
+      mTrack->SetInputProcessingImpl(mProcessing);
+    }
+  };
+
+  if (IsDestroyed()) {
+    return;
+  }
+  GraphImpl()->AppendMessage(
+      MakeUnique<Message>(std::move(this), std::move(aInputProcessing)));
+}
+
+AudioProcessingTrack* AudioProcessingTrack::Create(MediaTrackGraph* aGraph) {
+  MOZ_ASSERT(NS_IsMainThread());
+  AudioProcessingTrack* track = new AudioProcessingTrack(aGraph->GraphRate());
+  aGraph->AddTrack(track);
+  return track;
+}
+
+void AudioProcessingTrack::DestroyImpl() {
+  ProcessedMediaTrack::DestroyImpl();
+  if (mInputProcessing) {
+    mInputProcessing->End();
+  }
+}
+
+void AudioProcessingTrack::ProcessInput(GraphTime aFrom, GraphTime aTo,
+                                        uint32_t aFlags) {
+  TRACE_COMMENT("AudioProcessingTrack::ProcessInput", "AudioProcessingTrack %p",
+                this);
+  MOZ_ASSERT(mInputProcessing);
+
+  LOG_FRAME(
+      "(Graph %p, Driver %p) AudioProcessingTrack %p ProcessInput from %" PRId64
+      " to %" PRId64 ", needs %" PRId64 " frames",
+      mGraph, mGraph->CurrentDriver(), this, aFrom, aTo, aTo - aFrom);
+
+  if (aFrom >= aTo) {
+    return;
+  }
+
+  if (!mInputProcessing->IsEnded()) {
+    MOZ_ASSERT(TrackTimeToGraphTime(GetEnd()) == aFrom);
+    if (mInputs.IsEmpty()) {
+      GetData<AudioSegment>()->AppendNullData(aTo - aFrom);
+      LOG_FRAME("(Graph %p, Driver %p) AudioProcessingTrack %p Filling %" PRId64
+                " frames of null data (no input source)",
+                mGraph, mGraph->CurrentDriver(), this, aTo - aFrom);
+    } else {
+      MOZ_ASSERT(mInputs.Length() == 1);
+      AudioSegment data;
+      DeviceInputConsumerTrack::GetInputSourceData(data, mInputs[0], aFrom,
+                                                   aTo);
+      mInputProcessing->Process(GraphImpl(), aFrom, aTo, &data,
+                                GetData<AudioSegment>());
+    }
+    MOZ_ASSERT(TrackTimeToGraphTime(GetEnd()) == aTo);
+
+    ApplyTrackDisabling(mSegment.get());
+  } else if (aFlags & ALLOW_END) {
+    mEnded = true;
+  }
+}
+
+void AudioProcessingTrack::NotifyOutputData(MediaTrackGraphImpl* aGraph,
+                                            AudioDataValue* aBuffer,
+                                            size_t aFrames, TrackRate aRate,
+                                            uint32_t aChannels) {
+  MOZ_ASSERT(mGraph == aGraph, "Cannot feed audio output to another graph");
+  MOZ_ASSERT(mGraph->OnGraphThread());
+  if (mInputProcessing) {
+    mInputProcessing->ProcessOutputData(aGraph, aBuffer, aFrames, aRate,
+                                        aChannels);
+  }
+}
+
+void AudioProcessingTrack::SetInputProcessingImpl(
+    RefPtr<AudioInputProcessing> aInputProcessing) {
+  MOZ_ASSERT(GraphImpl()->OnGraphThread());
+  mInputProcessing = std::move(aInputProcessing);
+}
+
+MediaEngineWebRTCAudioCaptureSource::MediaEngineWebRTCAudioCaptureSource(
+    const MediaDevice* aMediaDevice) {
+  MOZ_ASSERT(aMediaDevice->mMediaSource == MediaSourceEnum::AudioCapture);
+}
+
+/* static */
+nsString MediaEngineWebRTCAudioCaptureSource::GetUUID() {
+  nsID uuid{};
+  char uuidBuffer[NSID_LENGTH];
+  nsCString asciiString;
+  ErrorResult rv;
+
+  rv = nsID::GenerateUUIDInPlace(uuid);
+  if (rv.Failed()) {
+    return u""_ns;
+  }
+
+  uuid.ToProvidedString(uuidBuffer);
+  asciiString.AssignASCII(uuidBuffer);
+
+  // Remove {} and the null terminator
+  return NS_ConvertASCIItoUTF16(Substring(asciiString, 1, NSID_LENGTH - 3));
+}
+
+/* static */
+nsString MediaEngineWebRTCAudioCaptureSource::GetGroupId() {
+  return u"AudioCaptureGroup"_ns;
+}
+
+void MediaEngineWebRTCAudioCaptureSource::SetTrack(
+    const RefPtr<MediaTrack>& aTrack, const PrincipalHandle& aPrincipalHandle) {
+  AssertIsOnOwningThread();
+  // Nothing to do here. aTrack is a placeholder dummy and not exposed.
+}
+
+nsresult MediaEngineWebRTCAudioCaptureSource::Start() {
+  AssertIsOnOwningThread();
+  return NS_OK;
+}
+
+nsresult MediaEngineWebRTCAudioCaptureSource::Stop() {
+  AssertIsOnOwningThread();
+  return NS_OK;
+}
+
+nsresult MediaEngineWebRTCAudioCaptureSource::Reconfigure(
+    const dom::MediaTrackConstraints& aConstraints,
+    const MediaEnginePrefs& aPrefs, const char** aOutBadConstraint) {
+  return NS_OK;
+}
+
+void MediaEngineWebRTCAudioCaptureSource::GetSettings(
+    dom::MediaTrackSettings& aOutSettings) const {
+  aOutSettings.mAutoGainControl.Construct(false);
+  aOutSettings.mEchoCancellation.Construct(false);
+  aOutSettings.mNoiseSuppression.Construct(false);
+  aOutSettings.mChannelCount.Construct(1);
+}
+
+}  // namespace mozilla
+
+// Don't allow our macros to leak into other cpps in our unified build unit.
+#undef MAX_CHANNELS
+#undef MONO
+#undef MAX_SAMPLING_FREQ
diff --git a/dom/media/webrtc/MediaEngineWebRTCAudio.h b/dom/media/webrtc/MediaEngineWebRTCAudio.h
new file mode 100644
index 0000000000..d9a602962d
--- /dev/null
+++ b/dom/media/webrtc/MediaEngineWebRTCAudio.h
@@ -0,0 +1,295 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef MediaEngineWebRTCAudio_h
+#define MediaEngineWebRTCAudio_h
+
+#include "AudioPacketizer.h"
+#include "AudioSegment.h"
+#include "AudioDeviceInfo.h"
+#include "DeviceInputTrack.h"
+#include "MediaEngineWebRTC.h"
+#include "MediaEnginePrefs.h"
+#include "MediaTrackListener.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+namespace mozilla {
+
+class AudioInputProcessing;
+class AudioProcessingTrack;
+
+// This class is created and used exclusively on the Media Manager thread, with
+// exactly two exceptions:
+// - Pull is always called on the MTG thread. It only ever uses
+//   mInputProcessing. mInputProcessing is set, then a message is sent first to
+//   the main thread and then the MTG thread so that it can be used as part of
+//   the graph processing. On destruction, similarly, a message is sent to the
+//   graph so that it stops using it, and then it is deleted.
+// - mSettings is created on the MediaManager thread is always ever accessed on
+//   the Main Thread. It is const.
+class MediaEngineWebRTCMicrophoneSource : public MediaEngineSource {
+ public:
+  explicit MediaEngineWebRTCMicrophoneSource(const MediaDevice* aMediaDevice);
+
+  nsresult Allocate(const dom::MediaTrackConstraints& aConstraints,
+                    const MediaEnginePrefs& aPrefs, uint64_t aWindowID,
+                    const char** aOutBadConstraint) override;
+  nsresult Deallocate() override;
+  void SetTrack(const RefPtr<MediaTrack>& aTrack,
+                const PrincipalHandle& aPrincipal) override;
+  nsresult Start() override;
+  nsresult Stop() override;
+  nsresult Reconfigure(const dom::MediaTrackConstraints& aConstraints,
+                       const MediaEnginePrefs& aPrefs,
+                       const char** aOutBadConstraint) override;
+
+  /**
+   * Assigns the current settings of the capture to aOutSettings.
+   * Main thread only.
+   */
+  void GetSettings(dom::MediaTrackSettings& aOutSettings) const override;
+
+  nsresult TakePhoto(MediaEnginePhotoCallback* aCallback) override {
+    return NS_ERROR_NOT_IMPLEMENTED;
+  }
+
+ protected:
+  ~MediaEngineWebRTCMicrophoneSource() = default;
+
+ private:
+  /**
+   * From a set of constraints and about:config preferences, output the correct
+   * set of preferences that can be sent to AudioInputProcessing.
+   *
+   * This can fail if the number of channels requested is zero, negative, or
+   * more than the device supports.
+   */
+  nsresult EvaluateSettings(const NormalizedConstraints& aConstraintsUpdate,
+                            const MediaEnginePrefs& aInPrefs,
+                            MediaEnginePrefs* aOutPrefs,
+                            const char** aOutBadConstraint);
+  /**
+   * From settings output by EvaluateSettings, send those settings to the
+   * AudioInputProcessing instance and the main thread (for use in GetSettings).
+   */
+  void ApplySettings(const MediaEnginePrefs& aPrefs);
+
+  PrincipalHandle mPrincipal = PRINCIPAL_HANDLE_NONE;
+
+  const RefPtr<AudioDeviceInfo> mDeviceInfo;
+
+  // The maximum number of channels that this device supports.
+  const uint32_t mDeviceMaxChannelCount;
+  // The current settings for the underlying device.
+  // Constructed on the MediaManager thread, and then only ever accessed on the
+  // main thread.
+  const nsMainThreadPtrHandle<media::Refcountable<dom::MediaTrackSettings>>
+      mSettings;
+
+  // Current state of the resource for this source.
+  MediaEngineSourceState mState;
+
+  // The current preferences that will be forwarded to mAudioProcessingConfig
+  // below.
+  MediaEnginePrefs mCurrentPrefs;
+
+  // The AudioProcessingTrack used to inteface with the MediaTrackGraph. Set in
+  // SetTrack as part of the initialization, and nulled in ::Deallocate.
+  RefPtr<AudioProcessingTrack> mTrack;
+
+  // See note at the top of this class.
+  RefPtr<AudioInputProcessing> mInputProcessing;
+
+  // Copy of the config currently applied to AudioProcessing through
+  // mInputProcessing.
+  webrtc::AudioProcessing::Config mAudioProcessingConfig;
+};
+
+// This class is created on the MediaManager thread, and then exclusively used
+// on the MTG thread.
+// All communication is done via message passing using MTG ControlMessages
+class AudioInputProcessing : public AudioDataListener {
+ public:
+  explicit AudioInputProcessing(uint32_t aMaxChannelCount);
+  void Process(MediaTrackGraphImpl* aGraph, GraphTime aFrom, GraphTime aTo,
+               AudioSegment* aInput, AudioSegment* aOutput);
+
+  void ProcessOutputData(MediaTrackGraphImpl* aGraph, AudioDataValue* aBuffer,
+                         size_t aFrames, TrackRate aRate, uint32_t aChannels);
+  bool IsVoiceInput(MediaTrackGraphImpl* aGraph) const override {
+    // If we're passing data directly without AEC or any other process, this
+    // means that all voice-processing has been disabled intentionaly. In this
+    // case, consider that the device is not used for voice input.
+    return !PassThrough(aGraph);
+  }
+
+  void Start(MediaTrackGraphImpl* aGraph);
+  void Stop(MediaTrackGraphImpl* aGraph);
+
+  void DeviceChanged(MediaTrackGraphImpl* aGraph) override;
+
+  uint32_t RequestedInputChannelCount(MediaTrackGraphImpl*) override {
+    return GetRequestedInputChannelCount();
+  }
+
+  void Disconnect(MediaTrackGraphImpl* aGraph) override;
+
+  void PacketizeAndProcess(MediaTrackGraphImpl* aGraph,
+                           const AudioSegment& aSegment);
+
+  void SetPassThrough(MediaTrackGraphImpl* aGraph, bool aPassThrough);
+  uint32_t GetRequestedInputChannelCount();
+  void SetRequestedInputChannelCount(MediaTrackGraphImpl* aGraph,
+                                     CubebUtils::AudioDeviceID aDeviceId,
+                                     uint32_t aRequestedInputChannelCount);
+  // This is true when all processing is disabled, we can skip
+  // packetization, resampling and other processing passes.
+  bool PassThrough(MediaTrackGraphImpl* aGraphImpl) const;
+
+  // This allow changing the APM options, enabling or disabling processing
+  // steps. The config gets applied the next time we're about to process input
+  // data.
+  void ApplyConfig(MediaTrackGraphImpl* aGraph,
+                   const webrtc::AudioProcessing::Config& aConfig);
+
+  void End();
+
+  TrackTime NumBufferedFrames(MediaTrackGraphImpl* aGraph) const;
+
+  // The packet size contains samples in 10ms. The unit of aRate is hz.
+  constexpr static uint32_t GetPacketSize(TrackRate aRate) {
+    return static_cast<uint32_t>(aRate) / 100u;
+  }
+
+  bool IsEnded() const { return mEnded; }
+
+ private:
+  ~AudioInputProcessing() = default;
+  void EnsureAudioProcessing(MediaTrackGraphImpl* aGraph, uint32_t aChannels);
+  void ResetAudioProcessing(MediaTrackGraphImpl* aGraph);
+  PrincipalHandle GetCheckedPrincipal(const AudioSegment& aSegment);
+  // This implements the processing algoritm to apply to the input (e.g. a
+  // microphone). If all algorithms are disabled, this class in not used. This
+  // class only accepts audio chunks of 10ms. It has two inputs and one output:
+  // it is fed the speaker data and the microphone data. It outputs processed
+  // input data.
+  const UniquePtr<webrtc::AudioProcessing> mAudioProcessing;
+  // Packetizer to be able to feed 10ms packets to the input side of
+  // mAudioProcessing. Not used if the processing is bypassed.
+  Maybe<AudioPacketizer<AudioDataValue, float>> mPacketizerInput;
+  // Packetizer to be able to feed 10ms packets to the output side of
+  // mAudioProcessing. Not used if the processing is bypassed.
+  Maybe<AudioPacketizer<AudioDataValue, float>> mPacketizerOutput;
+  // The number of channels asked for by content, after clamping to the range of
+  // legal channel count for this particular device.
+  uint32_t mRequestedInputChannelCount;
+  // mSkipProcessing is true if none of the processing passes are enabled,
+  // because of prefs or constraints. This allows simply copying the audio into
+  // the MTG, skipping resampling and the whole webrtc.org code.
+  bool mSkipProcessing;
+  // Stores the mixed audio output for the reverse-stream of the AEC (the
+  // speaker data).
+  AlignedFloatBuffer mOutputBuffer;
+  // Stores the input audio, to be processed by the APM.
+  AlignedFloatBuffer mInputBuffer;
+  // Stores the deinterleaved microphone audio
+  AlignedFloatBuffer mDeinterleavedBuffer;
+  // Stores the mixed down input audio
+  AlignedFloatBuffer mInputDownmixBuffer;
+  // Stores data waiting to be pulled.
+  AudioSegment mSegment;
+  // Whether or not this MediaEngine is enabled. If it's not enabled, it
+  // operates in "pull" mode, and we append silence only, releasing the audio
+  // input track.
+  bool mEnabled;
+  // Whether or not we've ended and removed the AudioProcessingTrack.
+  bool mEnded;
+  // When processing is enabled, the number of packets received by this
+  // instance, to implement periodic logging.
+  uint64_t mPacketCount;
+  // A storage holding the interleaved audio data converted the AudioSegment.
+  // This will be used as an input parameter for PacketizeAndProcess. This
+  // should be removed once bug 1729041 is done.
+  AutoTArray<AudioDataValue,
+             SilentChannel::AUDIO_PROCESSING_FRAMES * GUESS_AUDIO_CHANNELS>
+      mInterleavedBuffer;
+  // Tracks the pending frames with paired principals piled up in packetizer.
+  std::deque<std::pair<TrackTime, PrincipalHandle>> mChunksInPacketizer;
+};
+
+// MediaTrack subclass tailored for MediaEngineWebRTCMicrophoneSource.
+class AudioProcessingTrack : public DeviceInputConsumerTrack {
+  // Only accessed on the graph thread.
+  RefPtr<AudioInputProcessing> mInputProcessing;
+
+  explicit AudioProcessingTrack(TrackRate aSampleRate)
+      : DeviceInputConsumerTrack(aSampleRate) {}
+
+  ~AudioProcessingTrack() = default;
+
+ public:
+  // Main Thread API
+  void Destroy() override;
+  void SetInputProcessing(RefPtr<AudioInputProcessing> aInputProcessing);
+  static AudioProcessingTrack* Create(MediaTrackGraph* aGraph);
+
+  // Graph Thread API
+  void DestroyImpl() override;
+  void ProcessInput(GraphTime aFrom, GraphTime aTo, uint32_t aFlags) override;
+  uint32_t NumberOfChannels() const override {
+    MOZ_DIAGNOSTIC_ASSERT(
+        mInputProcessing,
+        "Must set mInputProcessing before exposing to content");
+    return mInputProcessing->GetRequestedInputChannelCount();
+  }
+  // Pass the graph's mixed audio output to mInputProcessing for processing as
+  // the reverse stream.
+  void NotifyOutputData(MediaTrackGraphImpl* aGraph, AudioDataValue* aBuffer,
+                        size_t aFrames, TrackRate aRate, uint32_t aChannels);
+
+  // Any thread
+  AudioProcessingTrack* AsAudioProcessingTrack() override { return this; }
+
+ private:
+  // Graph thread API
+  void SetInputProcessingImpl(RefPtr<AudioInputProcessing> aInputProcessing);
+};
+
+class MediaEngineWebRTCAudioCaptureSource : public MediaEngineSource {
+ public:
+  explicit MediaEngineWebRTCAudioCaptureSource(const MediaDevice* aMediaDevice);
+  static nsString GetUUID();
+  static nsString GetGroupId();
+  nsresult Allocate(const dom::MediaTrackConstraints& aConstraints,
+                    const MediaEnginePrefs& aPrefs, uint64_t aWindowID,
+                    const char** aOutBadConstraint) override {
+    // Nothing to do here, everything is managed in MediaManager.cpp
+    return NS_OK;
+  }
+  nsresult Deallocate() override {
+    // Nothing to do here, everything is managed in MediaManager.cpp
+    return NS_OK;
+  }
+  void SetTrack(const RefPtr<MediaTrack>& aTrack,
+                const PrincipalHandle& aPrincipal) override;
+  nsresult Start() override;
+  nsresult Stop() override;
+  nsresult Reconfigure(const dom::MediaTrackConstraints& aConstraints,
+                       const MediaEnginePrefs& aPrefs,
+                       const char** aOutBadConstraint) override;
+
+  nsresult TakePhoto(MediaEnginePhotoCallback* aCallback) override {
+    return NS_ERROR_NOT_IMPLEMENTED;
+  }
+
+  void GetSettings(dom::MediaTrackSettings& aOutSettings) const override;
+
+ protected:
+  virtual ~MediaEngineWebRTCAudioCaptureSource() = default;
+};
+
+}  // end namespace mozilla
+
+#endif  // MediaEngineWebRTCAudio_h
diff --git a/dom/media/webrtc/MediaTrackConstraints.cpp b/dom/media/webrtc/MediaTrackConstraints.cpp
new file mode 100644
index 0000000000..101cd0240a
--- /dev/null
+++ b/dom/media/webrtc/MediaTrackConstraints.cpp
@@ -0,0 +1,560 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "MediaTrackConstraints.h"
+
+#include <limits>
+#include <algorithm>
+#include <iterator>
+
+#include "MediaEngineSource.h"
+#include "mozilla/dom/MediaStreamTrackBinding.h"
+#include "mozilla/MediaManager.h"
+
+#ifdef MOZ_WEBRTC
+namespace mozilla {
+extern LazyLogModule gMediaManagerLog;
+}
+#else
+static mozilla::LazyLogModule gMediaManagerLog("MediaManager");
+#endif
+#define LOG(...) MOZ_LOG(gMediaManagerLog, LogLevel::Debug, (__VA_ARGS__))
+
+namespace mozilla {
+
+using dom::CallerType;
+using dom::ConstrainBooleanParameters;
+
+template <class ValueType>
+template <class ConstrainRange>
+void NormalizedConstraintSet::Range<ValueType>::SetFrom(
+    const ConstrainRange& aOther) {
+  if (aOther.mIdeal.WasPassed()) {
+    mIdeal.emplace(aOther.mIdeal.Value());
+  }
+  if (aOther.mExact.WasPassed()) {
+    mMin = aOther.mExact.Value();
+    mMax = aOther.mExact.Value();
+  } else {
+    if (aOther.mMin.WasPassed()) {
+      mMin = aOther.mMin.Value();
+    }
+    if (aOther.mMax.WasPassed()) {
+      mMax = aOther.mMax.Value();
+    }
+  }
+}
+
+// The Range code works surprisingly well for bool, except when averaging
+// ideals.
+template <>
+bool NormalizedConstraintSet::Range<bool>::Merge(const Range& aOther) {
+  if (!Intersects(aOther)) {
+    return false;
+  }
+  Intersect(aOther);
+
+  // To avoid "unsafe use of type 'bool'", we keep counter in mMergeDenominator
+  uint32_t counter = mMergeDenominator >> 16;
+  uint32_t denominator = mMergeDenominator & 0xffff;
+
+  if (aOther.mIdeal.isSome()) {
+    if (mIdeal.isNothing()) {
+      mIdeal.emplace(aOther.Get(false));
+      counter = aOther.Get(false);
+      denominator = 1;
+    } else {
+      if (!denominator) {
+        counter = Get(false);
+        denominator = 1;
+      }
+      counter += aOther.Get(false);
+      denominator++;
+    }
+  }
+  mMergeDenominator = ((counter & 0xffff) << 16) + (denominator & 0xffff);
+  return true;
+}
+
+template <>
+void NormalizedConstraintSet::Range<bool>::FinalizeMerge() {
+  if (mMergeDenominator) {
+    uint32_t counter = mMergeDenominator >> 16;
+    uint32_t denominator = mMergeDenominator & 0xffff;
+
+    *mIdeal = !!(counter / denominator);
+    mMergeDenominator = 0;
+  }
+}
+
+NormalizedConstraintSet::LongRange::LongRange(
+    LongPtrType aMemberPtr, const char* aName,
+    const dom::Optional<dom::OwningLongOrConstrainLongRange>& aOther,
+    bool advanced, nsTArray<MemberPtrType>* aList)
+    : Range<int32_t>((MemberPtrType)aMemberPtr, aName, 1 + INT32_MIN,
+                     INT32_MAX,  // +1 avoids Windows compiler bug
+                     aList) {
+  if (!aOther.WasPassed()) {
+    return;
+  }
+  auto& other = aOther.Value();
+  if (other.IsLong()) {
+    if (advanced) {
+      mMin = mMax = other.GetAsLong();
+    } else {
+      mIdeal.emplace(other.GetAsLong());
+    }
+  } else {
+    SetFrom(other.GetAsConstrainLongRange());
+  }
+}
+
+NormalizedConstraintSet::LongLongRange::LongLongRange(
+    LongLongPtrType aMemberPtr, const char* aName, const long long& aOther,
+    nsTArray<MemberPtrType>* aList)
+    : Range<int64_t>((MemberPtrType)aMemberPtr, aName, 1 + INT64_MIN,
+                     INT64_MAX,  // +1 avoids Windows compiler bug
+                     aList) {
+  mIdeal.emplace(aOther);
+}
+
+NormalizedConstraintSet::DoubleRange::DoubleRange(
+    DoublePtrType aMemberPtr, const char* aName,
+    const dom::Optional<dom::OwningDoubleOrConstrainDoubleRange>& aOther,
+    bool advanced, nsTArray<MemberPtrType>* aList)
+    : Range<double>((MemberPtrType)aMemberPtr, aName,
+                    -std::numeric_limits<double>::infinity(),
+                    std::numeric_limits<double>::infinity(), aList) {
+  if (!aOther.WasPassed()) {
+    return;
+  }
+  auto& other = aOther.Value();
+  if (other.IsDouble()) {
+    if (advanced) {
+      mMin = mMax = other.GetAsDouble();
+    } else {
+      mIdeal.emplace(other.GetAsDouble());
+    }
+  } else {
+    SetFrom(other.GetAsConstrainDoubleRange());
+  }
+}
+
+NormalizedConstraintSet::BooleanRange::BooleanRange(
+    BooleanPtrType aMemberPtr, const char* aName,
+    const dom::Optional<dom::OwningBooleanOrConstrainBooleanParameters>& aOther,
+    bool advanced, nsTArray<MemberPtrType>* aList)
+    : Range<bool>((MemberPtrType)aMemberPtr, aName, false, true, aList) {
+  if (!aOther.WasPassed()) {
+    return;
+  }
+  auto& other = aOther.Value();
+  if (other.IsBoolean()) {
+    if (advanced) {
+      mMin = mMax = other.GetAsBoolean();
+    } else {
+      mIdeal.emplace(other.GetAsBoolean());
+    }
+  } else {
+    auto& r = other.GetAsConstrainBooleanParameters();
+    if (r.mIdeal.WasPassed()) {
+      mIdeal.emplace(r.mIdeal.Value());
+    }
+    if (r.mExact.WasPassed()) {
+      mMin = r.mExact.Value();
+      mMax = r.mExact.Value();
+    }
+  }
+}
+
+NormalizedConstraintSet::StringRange::StringRange(
+    StringPtrType aMemberPtr, const char* aName,
+    const dom::Optional<
+        dom::OwningStringOrStringSequenceOrConstrainDOMStringParameters>&
+        aOther,
+    bool advanced, nsTArray<MemberPtrType>* aList)
+    : BaseRange((MemberPtrType)aMemberPtr, aName, aList) {
+  if (!aOther.WasPassed()) {
+    return;
+  }
+  auto& other = aOther.Value();
+  if (other.IsString()) {
+    if (advanced) {
+      mExact.insert(other.GetAsString());
+    } else {
+      mIdeal.insert(other.GetAsString());
+    }
+  } else if (other.IsStringSequence()) {
+    if (advanced) {
+      mExact.clear();
+      for (auto& str : other.GetAsStringSequence()) {
+        mExact.insert(str);
+      }
+    } else {
+      mIdeal.clear();
+      for (auto& str : other.GetAsStringSequence()) {
+        mIdeal.insert(str);
+      }
+    }
+  } else {
+    SetFrom(other.GetAsConstrainDOMStringParameters());
+  }
+}
+
+void NormalizedConstraintSet::StringRange::SetFrom(
+    const dom::ConstrainDOMStringParameters& aOther) {
+  if (aOther.mIdeal.WasPassed()) {
+    mIdeal.clear();
+    if (aOther.mIdeal.Value().IsString()) {
+      mIdeal.insert(aOther.mIdeal.Value().GetAsString());
+    } else {
+      for (auto& str : aOther.mIdeal.Value().GetAsStringSequence()) {
+        mIdeal.insert(str);
+      }
+    }
+  }
+  if (aOther.mExact.WasPassed()) {
+    mExact.clear();
+    if (aOther.mExact.Value().IsString()) {
+      mExact.insert(aOther.mExact.Value().GetAsString());
+    } else {
+      for (auto& str : aOther.mExact.Value().GetAsStringSequence()) {
+        mExact.insert(str);
+      }
+    }
+  }
+}
+
+auto NormalizedConstraintSet::StringRange::Clamp(const ValueType& n) const
+    -> ValueType {
+  if (mExact.empty()) {
+    return n;
+  }
+  ValueType result;
+  for (auto& entry : n) {
+    if (mExact.find(entry) != mExact.end()) {
+      result.insert(entry);
+    }
+  }
+  return result;
+}
+
+bool NormalizedConstraintSet::StringRange::Intersects(
+    const StringRange& aOther) const {
+  if (mExact.empty() || aOther.mExact.empty()) {
+    return true;
+  }
+
+  ValueType intersection;
+  set_intersection(mExact.begin(), mExact.end(), aOther.mExact.begin(),
+                   aOther.mExact.end(),
+                   std::inserter(intersection, intersection.begin()));
+  return !intersection.empty();
+}
+
+void NormalizedConstraintSet::StringRange::Intersect(
+    const StringRange& aOther) {
+  if (aOther.mExact.empty()) {
+    return;
+  }
+
+  ValueType intersection;
+  set_intersection(mExact.begin(), mExact.end(), aOther.mExact.begin(),
+                   aOther.mExact.end(),
+                   std::inserter(intersection, intersection.begin()));
+  mExact = intersection;
+}
+
+bool NormalizedConstraintSet::StringRange::Merge(const StringRange& aOther) {
+  if (!Intersects(aOther)) {
+    return false;
+  }
+  Intersect(aOther);
+
+  ValueType unioned;
+  set_union(mIdeal.begin(), mIdeal.end(), aOther.mIdeal.begin(),
+            aOther.mIdeal.end(), std::inserter(unioned, unioned.begin()));
+  mIdeal = unioned;
+  return true;
+}
+
+NormalizedConstraints::NormalizedConstraints(
+    const dom::MediaTrackConstraints& aOther, nsTArray<MemberPtrType>* aList)
+    : NormalizedConstraintSet(aOther, false, aList), mBadConstraint(nullptr) {
+  if (aOther.mAdvanced.WasPassed()) {
+    for (auto& entry : aOther.mAdvanced.Value()) {
+      mAdvanced.push_back(NormalizedConstraintSet(entry, true));
+    }
+  }
+}
+
+FlattenedConstraints::FlattenedConstraints(const NormalizedConstraints& aOther)
+    : NormalizedConstraintSet(aOther) {
+  for (auto& set : aOther.mAdvanced) {
+    // Must only apply compatible i.e. inherently non-overconstraining sets
+    // This rule is pretty much why this code is centralized here.
+    if (mWidth.Intersects(set.mWidth) && mHeight.Intersects(set.mHeight) &&
+        mFrameRate.Intersects(set.mFrameRate)) {
+      mWidth.Intersect(set.mWidth);
+      mHeight.Intersect(set.mHeight);
+      mFrameRate.Intersect(set.mFrameRate);
+    }
+    if (mEchoCancellation.Intersects(set.mEchoCancellation)) {
+      mEchoCancellation.Intersect(set.mEchoCancellation);
+    }
+    if (mNoiseSuppression.Intersects(set.mNoiseSuppression)) {
+      mNoiseSuppression.Intersect(set.mNoiseSuppression);
+    }
+    if (mAutoGainControl.Intersects(set.mAutoGainControl)) {
+      mAutoGainControl.Intersect(set.mAutoGainControl);
+    }
+    if (mChannelCount.Intersects(set.mChannelCount)) {
+      mChannelCount.Intersect(set.mChannelCount);
+    }
+  }
+}
+
+// MediaEngine helper
+//
+// The full algorithm for all devices.
+//
+// Fitness distance returned as integer math * 1000. Infinity = UINT32_MAX
+
+// First, all devices have a minimum distance based on their deviceId.
+// If you have no other constraints, use this one. Reused by all device types.
+
+/* static */
+bool MediaConstraintsHelper::SomeSettingsFit(
+    const NormalizedConstraints& aConstraints,
+    const nsTArray<RefPtr<LocalMediaDevice>>& aDevices) {
+  nsTArray<const NormalizedConstraintSet*> sets;
+  sets.AppendElement(&aConstraints);
+
+  MOZ_ASSERT(!aDevices.IsEmpty());
+  for (auto& device : aDevices) {
+    auto distance = device->GetBestFitnessDistance(sets, CallerType::NonSystem);
+    if (distance != UINT32_MAX) {
+      return true;
+    }
+  }
+  return false;
+}
+
+// Fitness distance returned as integer math * 1000. Infinity = UINT32_MAX
+
+/* static */
+uint32_t MediaConstraintsHelper::FitnessDistance(
+    const Maybe<nsString>& aN,
+    const NormalizedConstraintSet::StringRange& aParams) {
+  if (!aParams.mExact.empty() &&
+      (aN.isNothing() || aParams.mExact.find(*aN) == aParams.mExact.end())) {
+    return UINT32_MAX;
+  }
+  if (!aParams.mIdeal.empty() &&
+      (aN.isNothing() || aParams.mIdeal.find(*aN) == aParams.mIdeal.end())) {
+    return 1000;
+  }
+  return 0;
+}
+
+/* static */ const char* MediaConstraintsHelper::SelectSettings(
+    const NormalizedConstraints& aConstraints,
+    nsTArray<RefPtr<LocalMediaDevice>>& aDevices, CallerType aCallerType) {
+  auto& c = aConstraints;
+  LogConstraints(c);
+
+  // First apply top-level constraints.
+
+  // Stack constraintSets that pass, starting with the required one, because the
+  // whole stack must be re-satisfied each time a capability-set is ruled out
+  // (this avoids storing state or pushing algorithm into the lower-level code).
+  nsTArray<RefPtr<LocalMediaDevice>> unsatisfactory;
+  nsTArray<const NormalizedConstraintSet*> aggregateConstraints;
+  aggregateConstraints.AppendElement(&c);
+
+  std::multimap<uint32_t, RefPtr<LocalMediaDevice>> ordered;
+
+  for (uint32_t i = 0; i < aDevices.Length();) {
+    uint32_t distance =
+        aDevices[i]->GetBestFitnessDistance(aggregateConstraints, aCallerType);
+    if (distance == UINT32_MAX) {
+      unsatisfactory.AppendElement(std::move(aDevices[i]));
+      aDevices.RemoveElementAt(i);
+    } else {
+      ordered.insert(std::make_pair(distance, aDevices[i]));
+      ++i;
+    }
+  }
+  if (aDevices.IsEmpty()) {
+    return FindBadConstraint(c, unsatisfactory);
+  }
+
+  // Order devices by shortest distance
+  for (auto& ordinal : ordered) {
+    aDevices.RemoveElement(ordinal.second);
+    aDevices.AppendElement(ordinal.second);
+  }
+
+  // Then apply advanced constraints.
+
+  for (const auto& advanced : c.mAdvanced) {
+    aggregateConstraints.AppendElement(&advanced);
+    nsTArray<RefPtr<LocalMediaDevice>> rejects;
+    for (uint32_t j = 0; j < aDevices.Length();) {
+      uint32_t distance = aDevices[j]->GetBestFitnessDistance(
+          aggregateConstraints, aCallerType);
+      if (distance == UINT32_MAX) {
+        rejects.AppendElement(std::move(aDevices[j]));
+        aDevices.RemoveElementAt(j);
+      } else {
+        ++j;
+      }
+    }
+    if (aDevices.IsEmpty()) {
+      aDevices.AppendElements(std::move(rejects));
+      aggregateConstraints.RemoveLastElement();
+    }
+  }
+  return nullptr;
+}
+
+/* static */ const char* MediaConstraintsHelper::FindBadConstraint(
+    const NormalizedConstraints& aConstraints,
+    const nsTArray<RefPtr<LocalMediaDevice>>& aDevices) {
+  // The spec says to report a constraint that satisfies NONE
+  // of the sources. Unfortunately, this is a bit laborious to find out, and
+  // requires updating as new constraints are added!
+  auto& c = aConstraints;
+  dom::MediaTrackConstraints empty;
+
+  if (aDevices.IsEmpty() ||
+      !SomeSettingsFit(NormalizedConstraints(empty), aDevices)) {
+    return "";
+  }
+  {
+    NormalizedConstraints fresh(empty);
+    fresh.mDeviceId = c.mDeviceId;
+    if (!SomeSettingsFit(fresh, aDevices)) {
+      return "deviceId";
+    }
+  }
+  {
+    NormalizedConstraints fresh(empty);
+    fresh.mGroupId = c.mGroupId;
+    if (!SomeSettingsFit(fresh, aDevices)) {
+      return "groupId";
+    }
+  }
+  {
+    NormalizedConstraints fresh(empty);
+    fresh.mWidth = c.mWidth;
+    if (!SomeSettingsFit(fresh, aDevices)) {
+      return "width";
+    }
+  }
+  {
+    NormalizedConstraints fresh(empty);
+    fresh.mHeight = c.mHeight;
+    if (!SomeSettingsFit(fresh, aDevices)) {
+      return "height";
+    }
+  }
+  {
+    NormalizedConstraints fresh(empty);
+    fresh.mFrameRate = c.mFrameRate;
+    if (!SomeSettingsFit(fresh, aDevices)) {
+      return "frameRate";
+    }
+  }
+  {
+    NormalizedConstraints fresh(empty);
+    fresh.mFacingMode = c.mFacingMode;
+    if (!SomeSettingsFit(fresh, aDevices)) {
+      return "facingMode";
+    }
+  }
+  return "";
+}
+
+/* static */
+const char* MediaConstraintsHelper::FindBadConstraint(
+    const NormalizedConstraints& aConstraints,
+    const MediaDevice* aMediaDevice) {
+  NormalizedConstraints c(aConstraints);
+  NormalizedConstraints empty((dom::MediaTrackConstraints()));
+  c.mDeviceId = empty.mDeviceId;
+  c.mGroupId = empty.mGroupId;
+  AutoTArray<RefPtr<LocalMediaDevice>, 1> devices;
+  devices.EmplaceBack(
+      new LocalMediaDevice(aMediaDevice, u""_ns, u""_ns, u""_ns));
+  return FindBadConstraint(c, devices);
+}
+
+static void LogConstraintStringRange(
+    const NormalizedConstraintSet::StringRange& aRange) {
+  if (aRange.mExact.size() <= 1 && aRange.mIdeal.size() <= 1) {
+    LOG("  %s: { exact: [%s], ideal: [%s] }", aRange.mName,
+        (aRange.mExact.empty()
+             ? ""
+             : NS_ConvertUTF16toUTF8(*aRange.mExact.begin()).get()),
+        (aRange.mIdeal.empty()
+             ? ""
+             : NS_ConvertUTF16toUTF8(*aRange.mIdeal.begin()).get()));
+  } else {
+    LOG("  %s: { exact: [", aRange.mName);
+    for (auto& entry : aRange.mExact) {
+      LOG("      %s,", NS_ConvertUTF16toUTF8(entry).get());
+    }
+    LOG("    ], ideal: [");
+    for (auto& entry : aRange.mIdeal) {
+      LOG("      %s,", NS_ConvertUTF16toUTF8(entry).get());
+    }
+    LOG("    ]}");
+  }
+}
+
+template <typename T>
+static void LogConstraintRange(
+    const NormalizedConstraintSet::Range<T>& aRange) {
+  if (aRange.mIdeal.isSome()) {
+    LOG("  %s: { min: %d, max: %d, ideal: %d }", aRange.mName, aRange.mMin,
+        aRange.mMax, aRange.mIdeal.valueOr(0));
+  } else {
+    LOG("  %s: { min: %d, max: %d }", aRange.mName, aRange.mMin, aRange.mMax);
+  }
+}
+
+template <>
+void LogConstraintRange(const NormalizedConstraintSet::Range<double>& aRange) {
+  if (aRange.mIdeal.isSome()) {
+    LOG("  %s: { min: %f, max: %f, ideal: %f }", aRange.mName, aRange.mMin,
+        aRange.mMax, aRange.mIdeal.valueOr(0));
+  } else {
+    LOG("  %s: { min: %f, max: %f }", aRange.mName, aRange.mMin, aRange.mMax);
+  }
+}
+
+/* static */
+void MediaConstraintsHelper::LogConstraints(
+    const NormalizedConstraintSet& aConstraints) {
+  auto& c = aConstraints;
+  LOG("Constraints: {");
+  LOG("%s", [&]() {
+    LogConstraintRange(c.mWidth);
+    LogConstraintRange(c.mHeight);
+    LogConstraintRange(c.mFrameRate);
+    LogConstraintStringRange(c.mMediaSource);
+    LogConstraintStringRange(c.mFacingMode);
+    LogConstraintStringRange(c.mDeviceId);
+    LogConstraintStringRange(c.mGroupId);
+    LogConstraintRange(c.mEchoCancellation);
+    LogConstraintRange(c.mAutoGainControl);
+    LogConstraintRange(c.mNoiseSuppression);
+    LogConstraintRange(c.mChannelCount);
+    return "}";
+  }());
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/MediaTrackConstraints.h b/dom/media/webrtc/MediaTrackConstraints.h
new file mode 100644
index 0000000000..61e0ed85ea
--- /dev/null
+++ b/dom/media/webrtc/MediaTrackConstraints.h
@@ -0,0 +1,371 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// This file should not be included by other includes, as it contains code
+
+#ifndef MEDIATRACKCONSTRAINTS_H_
+#define MEDIATRACKCONSTRAINTS_H_
+
+#include <map>
+#include <set>
+#include <vector>
+
+#include "mozilla/Attributes.h"
+#include "mozilla/dom/MediaStreamTrackBinding.h"
+#include "mozilla/dom/MediaTrackSupportedConstraintsBinding.h"
+
+namespace mozilla {
+
+class LocalMediaDevice;
+class MediaDevice;
+
+template <class EnumValuesStrings, class Enum>
+static Enum StringToEnum(const EnumValuesStrings& aStrings,
+                         const nsAString& aValue, Enum aDefaultValue) {
+  for (size_t i = 0; aStrings[i].value; i++) {
+    if (aValue.EqualsASCII(aStrings[i].value)) {
+      return Enum(i);
+    }
+  }
+  return aDefaultValue;
+}
+
+// Helper classes for orthogonal constraints without interdependencies.
+// Instead of constraining values, constrain the constraints themselves.
+class NormalizedConstraintSet {
+ protected:
+  class BaseRange {
+   protected:
+    typedef BaseRange NormalizedConstraintSet::*MemberPtrType;
+
+    BaseRange(MemberPtrType aMemberPtr, const char* aName,
+              nsTArray<MemberPtrType>* aList)
+        : mName(aName) {
+      if (aList) {
+        aList->AppendElement(aMemberPtr);
+      }
+    }
+    virtual ~BaseRange() = default;
+
+   public:
+    virtual bool Merge(const BaseRange& aOther) = 0;
+    virtual void FinalizeMerge() = 0;
+
+    const char* mName;
+  };
+
+  typedef BaseRange NormalizedConstraintSet::*MemberPtrType;
+
+ public:
+  template <class ValueType>
+  class Range : public BaseRange {
+   public:
+    ValueType mMin, mMax;
+    Maybe<ValueType> mIdeal;
+
+    Range(MemberPtrType aMemberPtr, const char* aName, ValueType aMin,
+          ValueType aMax, nsTArray<MemberPtrType>* aList)
+        : BaseRange(aMemberPtr, aName, aList),
+          mMin(aMin),
+          mMax(aMax),
+          mMergeDenominator(0) {}
+    virtual ~Range() = default;
+
+    template <class ConstrainRange>
+    void SetFrom(const ConstrainRange& aOther);
+    ValueType Clamp(ValueType n) const {
+      return std::max(mMin, std::min(n, mMax));
+    }
+    ValueType Get(ValueType defaultValue) const {
+      return Clamp(mIdeal.valueOr(defaultValue));
+    }
+    bool Intersects(const Range& aOther) const {
+      return mMax >= aOther.mMin && mMin <= aOther.mMax;
+    }
+    void Intersect(const Range& aOther) {
+      mMin = std::max(mMin, aOther.mMin);
+      if (Intersects(aOther)) {
+        mMax = std::min(mMax, aOther.mMax);
+      } else {
+        // If there is no intersection, we will down-scale or drop frame
+        mMax = std::max(mMax, aOther.mMax);
+      }
+    }
+    bool Merge(const Range& aOther) {
+      if (strcmp(mName, "width") != 0 && strcmp(mName, "height") != 0 &&
+          strcmp(mName, "frameRate") != 0 && !Intersects(aOther)) {
+        return false;
+      }
+      Intersect(aOther);
+
+      if (aOther.mIdeal.isSome()) {
+        // Ideal values, as stored, may be outside their min max range, so use
+        // clamped values in averaging, to avoid extreme outliers skewing
+        // results.
+        if (mIdeal.isNothing()) {
+          mIdeal.emplace(aOther.Get(0));
+          mMergeDenominator = 1;
+        } else {
+          if (!mMergeDenominator) {
+            *mIdeal = Get(0);
+            mMergeDenominator = 1;
+          }
+          *mIdeal += aOther.Get(0);
+          mMergeDenominator++;
+        }
+      }
+      return true;
+    }
+    void FinalizeMerge() override {
+      if (mMergeDenominator) {
+        *mIdeal /= mMergeDenominator;
+        mMergeDenominator = 0;
+      }
+    }
+    void TakeHighestIdeal(const Range& aOther) {
+      if (aOther.mIdeal.isSome()) {
+        if (mIdeal.isNothing()) {
+          mIdeal.emplace(aOther.Get(0));
+        } else {
+          *mIdeal = std::max(Get(0), aOther.Get(0));
+        }
+      }
+    }
+
+   private:
+    bool Merge(const BaseRange& aOther) override {
+      return Merge(static_cast<const Range&>(aOther));
+    }
+
+    uint32_t mMergeDenominator;
+  };
+
+  struct LongRange : public Range<int32_t> {
+    typedef LongRange NormalizedConstraintSet::*LongPtrType;
+
+    LongRange(LongPtrType aMemberPtr, const char* aName,
+              const dom::Optional<dom::OwningLongOrConstrainLongRange>& aOther,
+              bool advanced, nsTArray<MemberPtrType>* aList);
+  };
+
+  struct LongLongRange : public Range<int64_t> {
+    typedef LongLongRange NormalizedConstraintSet::*LongLongPtrType;
+
+    LongLongRange(LongLongPtrType aMemberPtr, const char* aName,
+                  const long long& aOther, nsTArray<MemberPtrType>* aList);
+  };
+
+  struct DoubleRange : public Range<double> {
+    typedef DoubleRange NormalizedConstraintSet::*DoublePtrType;
+
+    DoubleRange(
+        DoublePtrType aMemberPtr, const char* aName,
+        const dom::Optional<dom::OwningDoubleOrConstrainDoubleRange>& aOther,
+        bool advanced, nsTArray<MemberPtrType>* aList);
+  };
+
+  struct BooleanRange : public Range<bool> {
+    typedef BooleanRange NormalizedConstraintSet::*BooleanPtrType;
+
+    BooleanRange(
+        BooleanPtrType aMemberPtr, const char* aName,
+        const dom::Optional<dom::OwningBooleanOrConstrainBooleanParameters>&
+            aOther,
+        bool advanced, nsTArray<MemberPtrType>* aList);
+
+    BooleanRange(BooleanPtrType aMemberPtr, const char* aName,
+                 const bool& aOther, nsTArray<MemberPtrType>* aList)
+        : Range<bool>((MemberPtrType)aMemberPtr, aName, false, true, aList) {
+      mIdeal.emplace(aOther);
+    }
+  };
+
+  struct StringRange : public BaseRange {
+    typedef std::set<nsString> ValueType;
+    ValueType mExact, mIdeal;
+
+    typedef StringRange NormalizedConstraintSet::*StringPtrType;
+
+    StringRange(
+        StringPtrType aMemberPtr, const char* aName,
+        const dom::Optional<
+            dom::OwningStringOrStringSequenceOrConstrainDOMStringParameters>&
+            aOther,
+        bool advanced, nsTArray<MemberPtrType>* aList);
+
+    StringRange(StringPtrType aMemberPtr, const char* aName,
+                const dom::Optional<nsString>& aOther,
+                nsTArray<MemberPtrType>* aList)
+        : BaseRange((MemberPtrType)aMemberPtr, aName, aList) {
+      if (aOther.WasPassed()) {
+        mIdeal.insert(aOther.Value());
+      }
+    }
+
+    ~StringRange() = default;
+
+    void SetFrom(const dom::ConstrainDOMStringParameters& aOther);
+    ValueType Clamp(const ValueType& n) const;
+    ValueType Get(const ValueType& defaultValue) const {
+      return Clamp(mIdeal.empty() ? defaultValue : mIdeal);
+    }
+    bool Intersects(const StringRange& aOther) const;
+    void Intersect(const StringRange& aOther);
+    bool Merge(const StringRange& aOther);
+    void FinalizeMerge() override {}
+
+   private:
+    bool Merge(const BaseRange& aOther) override {
+      return Merge(static_cast<const StringRange&>(aOther));
+    }
+  };
+
+  // All new constraints should be added here whether they use flattening or not
+  LongRange mWidth, mHeight;
+  DoubleRange mFrameRate;
+  StringRange mFacingMode;
+  StringRange mMediaSource;
+  LongLongRange mBrowserWindow;
+  StringRange mDeviceId;
+  StringRange mGroupId;
+  LongRange mViewportOffsetX, mViewportOffsetY, mViewportWidth, mViewportHeight;
+  BooleanRange mEchoCancellation, mNoiseSuppression, mAutoGainControl;
+  LongRange mChannelCount;
+
+ private:
+  typedef NormalizedConstraintSet T;
+
+ public:
+  NormalizedConstraintSet(const dom::MediaTrackConstraintSet& aOther,
+                          bool advanced,
+                          nsTArray<MemberPtrType>* aList = nullptr)
+      : mWidth(&T::mWidth, "width", aOther.mWidth, advanced, aList),
+        mHeight(&T::mHeight, "height", aOther.mHeight, advanced, aList),
+        mFrameRate(&T::mFrameRate, "frameRate", aOther.mFrameRate, advanced,
+                   aList),
+        mFacingMode(&T::mFacingMode, "facingMode", aOther.mFacingMode, advanced,
+                    aList),
+        mMediaSource(&T::mMediaSource, "mediaSource", aOther.mMediaSource,
+                     aList),
+        mBrowserWindow(&T::mBrowserWindow, "browserWindow",
+                       aOther.mBrowserWindow.WasPassed()
+                           ? aOther.mBrowserWindow.Value()
+                           : 0,
+                       aList),
+        mDeviceId(&T::mDeviceId, "deviceId", aOther.mDeviceId, advanced, aList),
+        mGroupId(&T::mGroupId, "groupId", aOther.mGroupId, advanced, aList),
+        mViewportOffsetX(&T::mViewportOffsetX, "viewportOffsetX",
+                         aOther.mViewportOffsetX, advanced, aList),
+        mViewportOffsetY(&T::mViewportOffsetY, "viewportOffsetY",
+                         aOther.mViewportOffsetY, advanced, aList),
+        mViewportWidth(&T::mViewportWidth, "viewportWidth",
+                       aOther.mViewportWidth, advanced, aList),
+        mViewportHeight(&T::mViewportHeight, "viewportHeight",
+                        aOther.mViewportHeight, advanced, aList),
+        mEchoCancellation(&T::mEchoCancellation, "echoCancellation",
+                          aOther.mEchoCancellation, advanced, aList),
+        mNoiseSuppression(&T::mNoiseSuppression, "noiseSuppression",
+                          aOther.mNoiseSuppression, advanced, aList),
+        mAutoGainControl(&T::mAutoGainControl, "autoGainControl",
+                         aOther.mAutoGainControl, advanced, aList),
+        mChannelCount(&T::mChannelCount, "channelCount", aOther.mChannelCount,
+                      advanced, aList) {}
+};
+
+template <>
+bool NormalizedConstraintSet::Range<bool>::Merge(const Range& aOther);
+template <>
+void NormalizedConstraintSet::Range<bool>::FinalizeMerge();
+
+// Used instead of MediaTrackConstraints in lower-level code.
+struct NormalizedConstraints : public NormalizedConstraintSet {
+  explicit NormalizedConstraints(const dom::MediaTrackConstraints& aOther,
+                                 nsTArray<MemberPtrType>* aList = nullptr);
+
+  std::vector<NormalizedConstraintSet> mAdvanced;
+  const char* mBadConstraint;
+};
+
+// Flattened version is used in low-level code with orthogonal constraints only.
+struct FlattenedConstraints : public NormalizedConstraintSet {
+  explicit FlattenedConstraints(const NormalizedConstraints& aOther);
+
+  explicit FlattenedConstraints(const dom::MediaTrackConstraints& aOther)
+      : FlattenedConstraints(NormalizedConstraints(aOther)) {}
+};
+
+// A helper class for MediaEngineSources
+class MediaConstraintsHelper {
+ public:
+  template <class ValueType, class NormalizedRange>
+  static uint32_t FitnessDistance(ValueType aN, const NormalizedRange& aRange) {
+    if (aRange.mMin > aN || aRange.mMax < aN) {
+      return UINT32_MAX;
+    }
+    if (aN == aRange.mIdeal.valueOr(aN)) {
+      return 0;
+    }
+    return uint32_t(
+        ValueType((std::abs(aN - aRange.mIdeal.value()) * 1000) /
+                  std::max(std::abs(aN), std::abs(aRange.mIdeal.value()))));
+  }
+
+  template <class ValueType, class NormalizedRange>
+  static uint32_t FeasibilityDistance(ValueType aN,
+                                      const NormalizedRange& aRange) {
+    if (aRange.mMin > aN) {
+      return UINT32_MAX;
+    }
+    // We prefer larger resolution because now we support downscaling
+    if (aN == aRange.mIdeal.valueOr(aN)) {
+      return 0;
+    }
+
+    if (aN > aRange.mIdeal.value()) {
+      return uint32_t(
+          ValueType((std::abs(aN - aRange.mIdeal.value()) * 1000) /
+                    std::max(std::abs(aN), std::abs(aRange.mIdeal.value()))));
+    }
+
+    return 10000 +
+           uint32_t(ValueType(
+               (std::abs(aN - aRange.mIdeal.value()) * 1000) /
+               std::max(std::abs(aN), std::abs(aRange.mIdeal.value()))));
+  }
+
+  static uint32_t FitnessDistance(
+      const Maybe<nsString>& aN,
+      const NormalizedConstraintSet::StringRange& aParams);
+
+ protected:
+  static bool SomeSettingsFit(
+      const NormalizedConstraints& aConstraints,
+      const nsTArray<RefPtr<LocalMediaDevice>>& aDevices);
+
+ public:
+  static uint32_t GetMinimumFitnessDistance(
+      const NormalizedConstraintSet& aConstraints, const nsString& aDeviceId,
+      const nsString& aGroupId);
+
+  // Apply constrains to a supplied list of devices (removes items from the
+  // list)
+  static const char* SelectSettings(
+      const NormalizedConstraints& aConstraints,
+      nsTArray<RefPtr<LocalMediaDevice>>& aDevices,
+      dom::CallerType aCallerType);
+
+  static const char* FindBadConstraint(
+      const NormalizedConstraints& aConstraints,
+      const nsTArray<RefPtr<LocalMediaDevice>>& aDevices);
+
+  static const char* FindBadConstraint(
+      const NormalizedConstraints& aConstraints,
+      const MediaDevice* aMediaDevice);
+
+  static void LogConstraints(const NormalizedConstraintSet& aConstraints);
+};
+
+}  // namespace mozilla
+
+#endif /* MEDIATRACKCONSTRAINTS_H_ */
diff --git a/dom/media/webrtc/MediaTransportChild.h b/dom/media/webrtc/MediaTransportChild.h
new file mode 100644
index 0000000000..b84dce70b2
--- /dev/null
+++ b/dom/media/webrtc/MediaTransportChild.h
@@ -0,0 +1,38 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _MTRANSPORTCHILD_H__
+#define _MTRANSPORTCHILD_H__
+
+#include "mozilla/dom/PMediaTransportChild.h"
+
+namespace mozilla {
+class MediaTransportHandlerIPC;
+
+class MediaTransportChild : public dom::PMediaTransportChild {
+ public:
+#ifdef MOZ_WEBRTC
+  explicit MediaTransportChild(MediaTransportHandlerIPC* aUser);
+  virtual ~MediaTransportChild();
+  mozilla::ipc::IPCResult RecvOnCandidate(const string& transportId,
+                                          const CandidateInfo& candidateInfo);
+  mozilla::ipc::IPCResult RecvOnAlpnNegotiated(const string& alpn);
+  mozilla::ipc::IPCResult RecvOnGatheringStateChange(const int& state);
+  mozilla::ipc::IPCResult RecvOnConnectionStateChange(const int& state);
+  mozilla::ipc::IPCResult RecvOnPacketReceived(const string& transportId,
+                                               const MediaPacket& packet);
+  mozilla::ipc::IPCResult RecvOnEncryptedSending(const string& transportId,
+                                                 const MediaPacket& packet);
+  mozilla::ipc::IPCResult RecvOnStateChange(const string& transportId,
+                                            const int& state);
+  mozilla::ipc::IPCResult RecvOnRtcpStateChange(const string& transportId,
+                                                const int& state);
+
+ private:
+  RefPtr<MediaTransportHandlerIPC> mUser;
+#endif  // MOZ_WEBRTC
+};
+
+}  // namespace mozilla
+#endif
diff --git a/dom/media/webrtc/MediaTransportParent.h b/dom/media/webrtc/MediaTransportParent.h
new file mode 100644
index 0000000000..8c698d0f54
--- /dev/null
+++ b/dom/media/webrtc/MediaTransportParent.h
@@ -0,0 +1,69 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _MTRANSPORTHANDLER_PARENT_H__
+#define _MTRANSPORTHANDLER_PARENT_H__
+
+#include "mozilla/dom/PMediaTransportParent.h"
+#include <memory>
+
+namespace mozilla {
+
+class MediaTransportParent : public dom::PMediaTransportParent {
+ public:
+#ifdef MOZ_WEBRTC
+  MediaTransportParent();
+  virtual ~MediaTransportParent();
+
+  mozilla::ipc::IPCResult RecvGetIceLog(const nsCString& pattern,
+                                        GetIceLogResolver&& aResolve);
+  mozilla::ipc::IPCResult RecvClearIceLog();
+  mozilla::ipc::IPCResult RecvEnterPrivateMode();
+  mozilla::ipc::IPCResult RecvExitPrivateMode();
+  mozilla::ipc::IPCResult RecvCreateIceCtx(const string& name);
+  mozilla::ipc::IPCResult RecvSetIceConfig(
+      nsTArray<RTCIceServer>&& iceServers,
+      const RTCIceTransportPolicy& icePolicy);
+  mozilla::ipc::IPCResult RecvSetProxyConfig(
+      const net::WebrtcProxyConfig& aProxyConfig);
+  mozilla::ipc::IPCResult RecvEnsureProvisionalTransport(
+      const string& transportId, const string& localUfrag,
+      const string& localPwd, const int& componentCount);
+  mozilla::ipc::IPCResult RecvSetTargetForDefaultLocalAddressLookup(
+      const string& targetIp, uint16_t targetPort);
+  mozilla::ipc::IPCResult RecvStartIceGathering(
+      const bool& defaultRouteOnly, const bool& obfuscateAddresses,
+      const net::NrIceStunAddrArray& stunAddrs);
+  mozilla::ipc::IPCResult RecvActivateTransport(
+      const string& transportId, const string& localUfrag,
+      const string& localPwd, const int& componentCount,
+      const string& remoteUfrag, const string& remotePwd,
+      nsTArray<uint8_t>&& keyDer, nsTArray<uint8_t>&& certDer,
+      const int& authType, const bool& dtlsClient,
+      const DtlsDigestList& digests, const bool& privacyRequested);
+  mozilla::ipc::IPCResult RecvRemoveTransportsExcept(
+      const StringVector& transportIds);
+  mozilla::ipc::IPCResult RecvStartIceChecks(const bool& isControlling,
+                                             const StringVector& iceOptions);
+  mozilla::ipc::IPCResult RecvSendPacket(const string& transportId,
+                                         MediaPacket&& packet);
+  mozilla::ipc::IPCResult RecvAddIceCandidate(const string& transportId,
+                                              const string& candidate,
+                                              const string& ufrag,
+                                              const string& obfuscatedAddress);
+  mozilla::ipc::IPCResult RecvUpdateNetworkState(const bool& online);
+  mozilla::ipc::IPCResult RecvGetIceStats(const string& transportId,
+                                          const double& now,
+                                          GetIceStatsResolver&& aResolve);
+
+  void ActorDestroy(ActorDestroyReason aWhy);
+
+ private:
+  // Hide the sigslot/MediaTransportHandler stuff from IPC.
+  class Impl;
+  std::unique_ptr<Impl> mImpl;
+#endif  // MOZ_WEBRTC
+};
+}  // namespace mozilla
+#endif  //_MTRANSPORTHANDLER_PARENT_H__
diff --git a/dom/media/webrtc/PMediaTransport.ipdl b/dom/media/webrtc/PMediaTransport.ipdl
new file mode 100644
index 0000000000..b754d34ad2
--- /dev/null
+++ b/dom/media/webrtc/PMediaTransport.ipdl
@@ -0,0 +1,105 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+include protocol PBackground;
+
+#ifdef MOZ_WEBRTC
+include WebrtcProxyConfig;
+
+// ParamTraits stuff for generated code and other classes we don't want to change
+include "mozilla/net/NrIceStunAddrMessageUtils.h";
+include "mozilla/media/webrtc/WebrtcIPCTraits.h";
+using mozilla::StringVector from "mozilla/media/webrtc/WebrtcIPCTraits.h";
+using mozilla::CandidateInfo from "mozilla/media/webrtc/WebrtcIPCTraits.h";
+using mozilla::DtlsDigestList from "mozilla/media/webrtc/WebrtcIPCTraits.h";
+using std::string from "string";
+using struct mozilla::dom::RTCStatsCollection from "mozilla/dom/RTCStatsReportBinding.h";
+using WebrtcGlobalLog from "mozilla/media/webrtc/WebrtcGlobal.h";
+using mozilla::dom::RTCIceServer from "mozilla/dom/RTCConfigurationBinding.h";
+using mozilla::dom::RTCIceTransportPolicy from "mozilla/dom/RTCConfigurationBinding.h";
+
+// ParamTraits stuff for our own classes
+using mozilla::MediaPacket from "transport/mediapacket.h";
+include "mozilla/net/NrIceStunAddrMessageUtils.h";
+using mozilla::net::NrIceStunAddrArray from "mozilla/net/PStunAddrsParams.h";
+#endif // MOZ_WEBRTC
+
+namespace mozilla {
+namespace dom {
+
+[ManualDealloc]
+async protocol PMediaTransport {
+  manager PBackground;
+
+parent:
+  async __delete__();
+
+#ifdef MOZ_WEBRTC
+  async GetIceLog(nsCString pattern) returns (WebrtcGlobalLog loglines);
+  async ClearIceLog();
+  async EnterPrivateMode();
+  async ExitPrivateMode();
+
+  async CreateIceCtx(string name);
+
+  async SetIceConfig(RTCIceServer[] iceServers,
+                     RTCIceTransportPolicy icePolicy);
+
+  async SetProxyConfig(WebrtcProxyConfig proxyConfig);
+
+  async EnsureProvisionalTransport(string transportId,
+                                   string localUfrag,
+                                   string localPwd,
+                                   int componentCount);
+
+  async SetTargetForDefaultLocalAddressLookup(string targetIp,
+                                              uint16_t targetPort);
+
+  async StartIceGathering(bool defaultRouteOnly,
+                          bool obfuscateHostAddresses,
+                          NrIceStunAddrArray stunAddrs);
+
+  async ActivateTransport(string transportId,
+                          string localUfrag,
+                          string localPwd,
+                          int componentCount,
+                          string remoteUfrag,
+                          string remotePwd,
+                          uint8_t[] keyDer,
+                          uint8_t[] certDer,
+                          int authType,
+                          bool dtlsClient,
+                          DtlsDigestList digests,
+                          bool privacyRequested);
+
+  async RemoveTransportsExcept(StringVector transportIds);
+
+  async StartIceChecks(bool isControlling,
+                       StringVector iceOptions);
+
+  async SendPacket(string transportId, MediaPacket packet);
+
+  async AddIceCandidate(string transportId,
+                        string candidate,
+                        string ufrag,
+                        string obfuscatedAddr);
+
+  async UpdateNetworkState(bool online);
+
+  async GetIceStats(string transportId, double now) returns (UniquePtr<RTCStatsCollection> stats);
+
+child:
+  async OnCandidate(string transportId, CandidateInfo candidateInfo);
+  async OnAlpnNegotiated(string alpn);
+  async OnGatheringStateChange(int state);
+  async OnConnectionStateChange(int state);
+  async OnPacketReceived(string transportId, MediaPacket packet);
+  async OnEncryptedSending(string transportId, MediaPacket packet);
+  async OnStateChange(string transportId, int state);
+  async OnRtcpStateChange(string transportId, int state);
+
+#endif // MOZ_WEBRTC
+};
+} // namespace dom
+} // namespace mozilla
diff --git a/dom/media/webrtc/PWebrtcGlobal.ipdl b/dom/media/webrtc/PWebrtcGlobal.ipdl
new file mode 100644
index 0000000000..e274b23636
--- /dev/null
+++ b/dom/media/webrtc/PWebrtcGlobal.ipdl
@@ -0,0 +1,41 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+include protocol PContent;
+
+#ifdef MOZ_WEBRTC
+include "mozilla/media/webrtc/WebrtcGlobal.h";
+
+using struct mozilla::dom::RTCStatsReportInternal from "mozilla/dom/RTCStatsReportBinding.h";
+using WebrtcGlobalLog from "mozilla/media/webrtc/WebrtcGlobal.h";
+#endif
+
+namespace mozilla {
+namespace dom {
+
+[ManualDealloc, ChildImpl=virtual, ParentImpl=virtual]
+async protocol PWebrtcGlobal {
+  manager PContent;
+
+parent: // child -> parent messages
+  async __delete__();
+#ifdef MOZ_WEBRTC
+  async PeerConnectionCreated(nsString aPcId, bool aIsLongTermStatsDisabled);
+  async PeerConnectionDestroyed(nsString aPcId);
+  async PeerConnectionFinalStats(RTCStatsReportInternal aFinalStats);
+
+child: // parent -> child messages
+  async GetStats(nsString aPcIdFilter) returns (RTCStatsReportInternal[] stats);
+  async ClearStats();
+  async GetLog() returns (WebrtcGlobalLog logs);
+  async ClearLog();
+  async SetAecLogging(bool aEnable);
+  async SetDebugMode(int aLevel);
+#endif
+};
+
+} // end namespace net
+} // end namespace mozilla
+
+
diff --git a/dom/media/webrtc/PeerIdentity.cpp b/dom/media/webrtc/PeerIdentity.cpp
new file mode 100644
index 0000000000..d36fea5ea3
--- /dev/null
+++ b/dom/media/webrtc/PeerIdentity.cpp
@@ -0,0 +1,80 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-
+ * vim: sw=2 ts=2 sts=2 expandtab
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "PeerIdentity.h"
+
+#include "mozilla/DebugOnly.h"
+#include "nsCOMPtr.h"
+#include "nsIIDNService.h"
+#include "nsNetCID.h"
+#include "nsServiceManagerUtils.h"
+
+namespace mozilla {
+
+bool PeerIdentity::Equals(const PeerIdentity& aOther) const {
+  return Equals(aOther.mPeerIdentity);
+}
+
+bool PeerIdentity::Equals(const nsAString& aOtherString) const {
+  nsString user;
+  GetUser(mPeerIdentity, user);
+  nsString otherUser;
+  GetUser(aOtherString, otherUser);
+  if (user != otherUser) {
+    return false;
+  }
+
+  nsString host;
+  GetHost(mPeerIdentity, host);
+  nsString otherHost;
+  GetHost(aOtherString, otherHost);
+
+  nsresult rv;
+  nsCOMPtr<nsIIDNService> idnService =
+      do_GetService("@mozilla.org/network/idn-service;1", &rv);
+  if (NS_WARN_IF(NS_FAILED(rv))) {
+    return host == otherHost;
+  }
+
+  nsCString normHost;
+  GetNormalizedHost(idnService, host, normHost);
+  nsCString normOtherHost;
+  GetNormalizedHost(idnService, otherHost, normOtherHost);
+  return normHost == normOtherHost;
+}
+
+/* static */
+void PeerIdentity::GetUser(const nsAString& aPeerIdentity, nsAString& aUser) {
+  int32_t at = aPeerIdentity.FindChar('@');
+  if (at >= 0) {
+    aUser = Substring(aPeerIdentity, 0, at);
+  } else {
+    aUser.Truncate();
+  }
+}
+
+/* static */
+void PeerIdentity::GetHost(const nsAString& aPeerIdentity, nsAString& aHost) {
+  int32_t at = aPeerIdentity.FindChar('@');
+  if (at >= 0) {
+    aHost = Substring(aPeerIdentity, at + 1);
+  } else {
+    aHost = aPeerIdentity;
+  }
+}
+
+/* static */
+void PeerIdentity::GetNormalizedHost(const nsCOMPtr<nsIIDNService>& aIdnService,
+                                     const nsAString& aHost,
+                                     nsACString& aNormalizedHost) {
+  const nsCString chost = NS_ConvertUTF16toUTF8(aHost);
+  DebugOnly<nsresult> rv =
+      aIdnService->ConvertUTF8toACE(chost, aNormalizedHost);
+  NS_WARNING_ASSERTION(NS_SUCCEEDED(rv),
+                       "Failed to convert UTF-8 host to ASCII");
+}
+
+} /* namespace mozilla */
diff --git a/dom/media/webrtc/PeerIdentity.h b/dom/media/webrtc/PeerIdentity.h
new file mode 100644
index 0000000000..64364d3359
--- /dev/null
+++ b/dom/media/webrtc/PeerIdentity.h
@@ -0,0 +1,70 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-
+ * vim: sw=2 ts=2 sts=2 expandtab
+ * This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef PeerIdentity_h
+#define PeerIdentity_h
+
+#include "nsISupportsImpl.h"
+#include "nsString.h"
+
+template <class T>
+class nsCOMPtr;
+class nsIIDNService;
+
+namespace mozilla {
+
+/**
+ * This class implements the identifier used in WebRTC identity.  Peers are
+ * identified using a string in the form [<user>@]<domain>, for instance,
+ * "user@example.com'. The (optional) user portion is a site-controlled string
+ * containing any character other than '@'.  The domain portion is a valid IDN
+ * domain name and is compared accordingly.
+ *
+ * See:
+ * http://tools.ietf.org/html/draft-ietf-rtcweb-security-arch-09#section-5.6.5.3.3.1
+ */
+class PeerIdentity final {
+ public:
+  NS_INLINE_DECL_REFCOUNTING(PeerIdentity)
+
+  explicit PeerIdentity(const nsAString& aPeerIdentity)
+      : mPeerIdentity(aPeerIdentity) {}
+
+  bool Equals(const PeerIdentity& aOther) const;
+  bool Equals(const nsAString& aOtherString) const;
+  const nsString& ToString() const { return mPeerIdentity; }
+
+ private:
+  ~PeerIdentity() = default;
+  static void GetUser(const nsAString& aPeerIdentity, nsAString& aUser);
+  static void GetHost(const nsAString& aPeerIdentity, nsAString& aHost);
+
+  static void GetNormalizedHost(const nsCOMPtr<nsIIDNService>& aIdnService,
+                                const nsAString& aHost,
+                                nsACString& aNormalizedHost);
+
+  nsString mPeerIdentity;
+};
+
+inline bool operator==(const PeerIdentity& aOne, const PeerIdentity& aTwo) {
+  return aOne.Equals(aTwo);
+}
+
+inline bool operator==(const PeerIdentity& aOne, const nsAString& aString) {
+  return aOne.Equals(aString);
+}
+
+inline bool operator!=(const PeerIdentity& aOne, const PeerIdentity& aTwo) {
+  return !aOne.Equals(aTwo);
+}
+
+inline bool operator!=(const PeerIdentity& aOne, const nsAString& aString) {
+  return !aOne.Equals(aString);
+}
+
+} /* namespace mozilla */
+
+#endif /* PeerIdentity_h */
diff --git a/dom/media/webrtc/RTCCertificate.cpp b/dom/media/webrtc/RTCCertificate.cpp
new file mode 100644
index 0000000000..69b6b0f563
--- /dev/null
+++ b/dom/media/webrtc/RTCCertificate.cpp
@@ -0,0 +1,438 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "mozilla/dom/RTCCertificate.h"
+
+#include <cstdio>
+#include <cstring>
+#include <memory>
+#include <new>
+#include <utility>
+#include "ErrorList.h"
+#include "MainThreadUtils.h"
+#include "cert.h"
+#include "cryptohi.h"
+#include "js/StructuredClone.h"
+#include "js/TypeDecls.h"
+#include "js/Value.h"
+#include "keyhi.h"
+#include "mozilla/ErrorResult.h"
+#include "mozilla/MacroForEach.h"
+#include "mozilla/OwningNonNull.h"
+#include "mozilla/UniquePtr.h"
+#include "mozilla/dom/BindingDeclarations.h"
+#include "mozilla/dom/CryptoBuffer.h"
+#include "mozilla/dom/CryptoKey.h"
+#include "mozilla/dom/KeyAlgorithmBinding.h"
+#include "mozilla/dom/KeyAlgorithmProxy.h"
+#include "mozilla/dom/Promise.h"
+#include "mozilla/dom/RTCCertificateBinding.h"
+#include "mozilla/dom/StructuredCloneHolder.h"
+#include "mozilla/dom/SubtleCryptoBinding.h"
+#include "mozilla/dom/UnionTypes.h"
+#include "mozilla/dom/WebCryptoCommon.h"
+#include "mozilla/dom/WebCryptoTask.h"
+#include "mozilla/fallible.h"
+#include "nsDebug.h"
+#include "nsError.h"
+#include "nsLiteralString.h"
+#include "nsStringFlags.h"
+#include "nsStringFwd.h"
+#include "nsTLiteralString.h"
+#include "pk11pub.h"
+#include "plarena.h"
+#include "secasn1.h"
+#include "secasn1t.h"
+#include "seccomon.h"
+#include "secmodt.h"
+#include "secoid.h"
+#include "secoidt.h"
+#include "transport/dtlsidentity.h"
+#include "xpcpublic.h"
+
+namespace mozilla::dom {
+
+#define RTCCERTIFICATE_SC_VERSION 0x00000001
+
+NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE(RTCCertificate, mGlobal)
+NS_IMPL_CYCLE_COLLECTING_ADDREF(RTCCertificate)
+NS_IMPL_CYCLE_COLLECTING_RELEASE(RTCCertificate)
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(RTCCertificate)
+  NS_WRAPPERCACHE_INTERFACE_MAP_ENTRY
+  NS_INTERFACE_MAP_ENTRY(nsISupports)
+NS_INTERFACE_MAP_END
+
+// Note: explicit casts necessary to avoid
+//       warning C4307: '*' : integral constant overflow
+#define ONE_DAY                                 \
+  PRTime(PR_USEC_PER_SEC) * PRTime(60)  /*sec*/ \
+      * PRTime(60) /*min*/ * PRTime(24) /*hours*/
+#define EXPIRATION_DEFAULT ONE_DAY* PRTime(30)
+#define EXPIRATION_SLACK ONE_DAY
+#define EXPIRATION_MAX ONE_DAY* PRTime(365) /*year*/
+
+const size_t RTCCertificateCommonNameLength = 16;
+const size_t RTCCertificateMinRsaSize = 1024;
+
+class GenerateRTCCertificateTask : public GenerateAsymmetricKeyTask {
+ public:
+  GenerateRTCCertificateTask(nsIGlobalObject* aGlobal, JSContext* aCx,
+                             const ObjectOrString& aAlgorithm,
+                             const Sequence<nsString>& aKeyUsages,
+                             PRTime aExpires)
+      : GenerateAsymmetricKeyTask(aGlobal, aCx, aAlgorithm, true, aKeyUsages),
+        mExpires(aExpires),
+        mAuthType(ssl_kea_null),
+        mCertificate(nullptr),
+        mSignatureAlg(SEC_OID_UNKNOWN) {
+    if (NS_FAILED(mEarlyRv)) {
+      // webrtc-pc says to throw NotSupportedError if we have passed "an
+      // algorithm that the user agent cannot or will not use to generate a
+      // certificate". This catches these cases.
+      mEarlyRv = NS_ERROR_DOM_NOT_SUPPORTED_ERR;
+    }
+  }
+
+ private:
+  PRTime mExpires;
+  SSLKEAType mAuthType;
+  UniqueCERTCertificate mCertificate;
+  SECOidTag mSignatureAlg;
+
+  static CERTName* GenerateRandomName(PK11SlotInfo* aSlot) {
+    uint8_t randomName[RTCCertificateCommonNameLength];
+    SECStatus rv =
+        PK11_GenerateRandomOnSlot(aSlot, randomName, sizeof(randomName));
+    if (rv != SECSuccess) {
+      return nullptr;
+    }
+
+    char buf[sizeof(randomName) * 2 + 4];
+    strncpy(buf, "CN=", 4);
+    for (size_t i = 0; i < sizeof(randomName); ++i) {
+      snprintf(&buf[i * 2 + 3], 3, "%.2x", randomName[i]);
+    }
+    buf[sizeof(buf) - 1] = '\0';
+
+    return CERT_AsciiToName(buf);
+  }
+
+  nsresult GenerateCertificate() {
+    UniquePK11SlotInfo slot(PK11_GetInternalSlot());
+    MOZ_ASSERT(slot.get());
+
+    UniqueCERTName subjectName(GenerateRandomName(slot.get()));
+    if (!subjectName) {
+      return NS_ERROR_DOM_UNKNOWN_ERR;
+    }
+
+    UniqueSECKEYPublicKey publicKey(mKeyPair->mPublicKey->GetPublicKey());
+    UniqueCERTSubjectPublicKeyInfo spki(
+        SECKEY_CreateSubjectPublicKeyInfo(publicKey.get()));
+    if (!spki) {
+      return NS_ERROR_DOM_UNKNOWN_ERR;
+    }
+
+    UniqueCERTCertificateRequest certreq(
+        CERT_CreateCertificateRequest(subjectName.get(), spki.get(), nullptr));
+    if (!certreq) {
+      return NS_ERROR_DOM_UNKNOWN_ERR;
+    }
+
+    PRTime now = PR_Now();
+    PRTime notBefore = now - EXPIRATION_SLACK;
+    mExpires += now;
+
+    UniqueCERTValidity validity(CERT_CreateValidity(notBefore, mExpires));
+    if (!validity) {
+      return NS_ERROR_DOM_UNKNOWN_ERR;
+    }
+
+    unsigned long serial;
+    // Note: This serial in principle could collide, but it's unlikely, and we
+    // don't expect anyone to be validating certificates anyway.
+    SECStatus rv = PK11_GenerateRandomOnSlot(
+        slot.get(), reinterpret_cast<unsigned char*>(&serial), sizeof(serial));
+    if (rv != SECSuccess) {
+      return NS_ERROR_DOM_UNKNOWN_ERR;
+    }
+
+    // NB: CERTCertificates created with CERT_CreateCertificate are not safe to
+    // use with other NSS functions like CERT_DupCertificate.  The strategy
+    // here is to create a tbsCertificate ("to-be-signed certificate"), encode
+    // it, and sign it, resulting in a signed DER certificate that can be
+    // decoded into a CERTCertificate.
+    UniqueCERTCertificate tbsCertificate(CERT_CreateCertificate(
+        serial, subjectName.get(), validity.get(), certreq.get()));
+    if (!tbsCertificate) {
+      return NS_ERROR_DOM_UNKNOWN_ERR;
+    }
+
+    MOZ_ASSERT(mSignatureAlg != SEC_OID_UNKNOWN);
+    PLArenaPool* arena = tbsCertificate->arena;
+
+    rv = SECOID_SetAlgorithmID(arena, &tbsCertificate->signature, mSignatureAlg,
+                               nullptr);
+    if (rv != SECSuccess) {
+      return NS_ERROR_DOM_UNKNOWN_ERR;
+    }
+
+    // Set version to X509v3.
+    *(tbsCertificate->version.data) = SEC_CERTIFICATE_VERSION_3;
+    tbsCertificate->version.len = 1;
+
+    SECItem innerDER = {siBuffer, nullptr, 0};
+    if (!SEC_ASN1EncodeItem(arena, &innerDER, tbsCertificate.get(),
+                            SEC_ASN1_GET(CERT_CertificateTemplate))) {
+      return NS_ERROR_DOM_UNKNOWN_ERR;
+    }
+
+    SECItem* certDer = PORT_ArenaZNew(arena, SECItem);
+    if (!certDer) {
+      return NS_ERROR_DOM_UNKNOWN_ERR;
+    }
+
+    UniqueSECKEYPrivateKey privateKey(mKeyPair->mPrivateKey->GetPrivateKey());
+    rv = SEC_DerSignData(arena, certDer, innerDER.data, innerDER.len,
+                         privateKey.get(), mSignatureAlg);
+    if (rv != SECSuccess) {
+      return NS_ERROR_DOM_UNKNOWN_ERR;
+    }
+
+    mCertificate.reset(CERT_NewTempCertificate(CERT_GetDefaultCertDB(), certDer,
+                                               nullptr, false, true));
+    if (!mCertificate) {
+      return NS_ERROR_DOM_UNKNOWN_ERR;
+    }
+    return NS_OK;
+  }
+
+  nsresult BeforeCrypto() override {
+    if (mAlgName.EqualsLiteral(WEBCRYPTO_ALG_RSASSA_PKCS1)) {
+      // Double check that size is OK.
+      auto sz = static_cast<size_t>(mRsaParams.keySizeInBits);
+      if (sz < RTCCertificateMinRsaSize) {
+        return NS_ERROR_DOM_NOT_SUPPORTED_ERR;
+      }
+
+      KeyAlgorithmProxy& alg = mKeyPair->mPublicKey->Algorithm();
+      if (alg.mType != KeyAlgorithmProxy::RSA ||
+          !alg.mRsa.mHash.mName.EqualsLiteral(WEBCRYPTO_ALG_SHA256)) {
+        return NS_ERROR_DOM_NOT_SUPPORTED_ERR;
+      }
+
+      mSignatureAlg = SEC_OID_PKCS1_SHA256_WITH_RSA_ENCRYPTION;
+      mAuthType = ssl_kea_rsa;
+
+    } else if (mAlgName.EqualsLiteral(WEBCRYPTO_ALG_ECDSA)) {
+      // We only support good curves in WebCrypto.
+      // If that ever changes, check that a good one was chosen.
+
+      mSignatureAlg = SEC_OID_ANSIX962_ECDSA_SHA256_SIGNATURE;
+      mAuthType = ssl_kea_ecdh;
+    } else {
+      return NS_ERROR_DOM_NOT_SUPPORTED_ERR;
+    }
+    return NS_OK;
+  }
+
+  nsresult DoCrypto() override {
+    nsresult rv = GenerateAsymmetricKeyTask::DoCrypto();
+    NS_ENSURE_SUCCESS(rv, rv);
+
+    rv = GenerateCertificate();
+    NS_ENSURE_SUCCESS(rv, rv);
+
+    return NS_OK;
+  }
+
+  virtual void Resolve() override {
+    // Make copies of the private key and certificate, otherwise, when this
+    // object is deleted, the structures they reference will be deleted too.
+    UniqueSECKEYPrivateKey key = mKeyPair->mPrivateKey->GetPrivateKey();
+    CERTCertificate* cert = CERT_DupCertificate(mCertificate.get());
+    RefPtr<RTCCertificate> result =
+        new RTCCertificate(mResultPromise->GetParentObject(), key.release(),
+                           cert, mAuthType, mExpires);
+    mResultPromise->MaybeResolve(result);
+  }
+};
+
+static PRTime ReadExpires(JSContext* aCx, const ObjectOrString& aOptions,
+                          ErrorResult& aRv) {
+  // This conversion might fail, but we don't really care; use the default.
+  // If this isn't an object, or it doesn't coerce into the right type,
+  // then we won't get the |expires| value.  Either will be caught later.
+  RTCCertificateExpiration expiration;
+  if (!aOptions.IsObject()) {
+    return EXPIRATION_DEFAULT;
+  }
+  JS::Rooted<JS::Value> value(aCx, JS::ObjectValue(*aOptions.GetAsObject()));
+  if (!expiration.Init(aCx, value)) {
+    aRv.NoteJSContextException(aCx);
+    return 0;
+  }
+
+  if (!expiration.mExpires.WasPassed()) {
+    return EXPIRATION_DEFAULT;
+  }
+  static const uint64_t max =
+      static_cast<uint64_t>(EXPIRATION_MAX / PR_USEC_PER_MSEC);
+  if (expiration.mExpires.Value() > max) {
+    return EXPIRATION_MAX;
+  }
+  return static_cast<PRTime>(expiration.mExpires.Value() * PR_USEC_PER_MSEC);
+}
+
+already_AddRefed<Promise> RTCCertificate::GenerateCertificate(
+    const GlobalObject& aGlobal, const ObjectOrString& aOptions,
+    ErrorResult& aRv, JS::Compartment* aCompartment) {
+  nsIGlobalObject* global = xpc::NativeGlobal(aGlobal.Get());
+  RefPtr<Promise> p = Promise::Create(global, aRv);
+  if (aRv.Failed()) {
+    return nullptr;
+  }
+  Sequence<nsString> usages;
+  if (!usages.AppendElement(u"sign"_ns, fallible)) {
+    aRv.Throw(NS_ERROR_OUT_OF_MEMORY);
+    return nullptr;
+  }
+
+  PRTime expires = ReadExpires(aGlobal.Context(), aOptions, aRv);
+  if (aRv.Failed()) {
+    return nullptr;
+  }
+  RefPtr<WebCryptoTask> task = new GenerateRTCCertificateTask(
+      global, aGlobal.Context(), aOptions, usages, expires);
+  task->DispatchWithPromise(p);
+  return p.forget();
+}
+
+RTCCertificate::RTCCertificate(nsIGlobalObject* aGlobal)
+    : mGlobal(aGlobal),
+      mPrivateKey(nullptr),
+      mCertificate(nullptr),
+      mAuthType(ssl_kea_null),
+      mExpires(0) {}
+
+RTCCertificate::RTCCertificate(nsIGlobalObject* aGlobal,
+                               SECKEYPrivateKey* aPrivateKey,
+                               CERTCertificate* aCertificate,
+                               SSLKEAType aAuthType, PRTime aExpires)
+    : mGlobal(aGlobal),
+      mPrivateKey(aPrivateKey),
+      mCertificate(aCertificate),
+      mAuthType(aAuthType),
+      mExpires(aExpires) {}
+
+RefPtr<DtlsIdentity> RTCCertificate::CreateDtlsIdentity() const {
+  if (!mPrivateKey || !mCertificate) {
+    return nullptr;
+  }
+  UniqueSECKEYPrivateKey key(SECKEY_CopyPrivateKey(mPrivateKey.get()));
+  UniqueCERTCertificate cert(CERT_DupCertificate(mCertificate.get()));
+  RefPtr<DtlsIdentity> id =
+      new DtlsIdentity(std::move(key), std::move(cert), mAuthType);
+  return id;
+}
+
+JSObject* RTCCertificate::WrapObject(JSContext* aCx,
+                                     JS::Handle<JSObject*> aGivenProto) {
+  return RTCCertificate_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+bool RTCCertificate::WritePrivateKey(JSStructuredCloneWriter* aWriter) const {
+  JsonWebKey jwk;
+  nsresult rv = CryptoKey::PrivateKeyToJwk(mPrivateKey.get(), jwk);
+  if (NS_FAILED(rv)) {
+    return false;
+  }
+  nsString json;
+  if (!jwk.ToJSON(json)) {
+    return false;
+  }
+  return StructuredCloneHolder::WriteString(aWriter, json);
+}
+
+bool RTCCertificate::WriteCertificate(JSStructuredCloneWriter* aWriter) const {
+  UniqueCERTCertificateList certs(CERT_CertListFromCert(mCertificate.get()));
+  if (!certs || certs->len <= 0) {
+    return false;
+  }
+  if (!JS_WriteUint32Pair(aWriter, certs->certs[0].len, 0)) {
+    return false;
+  }
+  return JS_WriteBytes(aWriter, certs->certs[0].data, certs->certs[0].len);
+}
+
+bool RTCCertificate::WriteStructuredClone(
+    JSContext* aCx, JSStructuredCloneWriter* aWriter) const {
+  if (!mPrivateKey || !mCertificate) {
+    return false;
+  }
+
+  return JS_WriteUint32Pair(aWriter, RTCCERTIFICATE_SC_VERSION, mAuthType) &&
+         JS_WriteUint32Pair(aWriter, (mExpires >> 32) & 0xffffffff,
+                            mExpires & 0xffffffff) &&
+         WritePrivateKey(aWriter) && WriteCertificate(aWriter);
+}
+
+bool RTCCertificate::ReadPrivateKey(JSStructuredCloneReader* aReader) {
+  nsString json;
+  if (!StructuredCloneHolder::ReadString(aReader, json)) {
+    return false;
+  }
+  JsonWebKey jwk;
+  if (!jwk.Init(json)) {
+    return false;
+  }
+  mPrivateKey = CryptoKey::PrivateKeyFromJwk(jwk);
+  return !!mPrivateKey;
+}
+
+bool RTCCertificate::ReadCertificate(JSStructuredCloneReader* aReader) {
+  CryptoBuffer cert;
+  if (!ReadBuffer(aReader, cert) || cert.Length() == 0) {
+    return false;
+  }
+
+  SECItem der = {siBuffer, cert.Elements(),
+                 static_cast<unsigned int>(cert.Length())};
+  mCertificate.reset(CERT_NewTempCertificate(CERT_GetDefaultCertDB(), &der,
+                                             nullptr, true, true));
+  return !!mCertificate;
+}
+
+// static
+already_AddRefed<RTCCertificate> RTCCertificate::ReadStructuredClone(
+    JSContext* aCx, nsIGlobalObject* aGlobal,
+    JSStructuredCloneReader* aReader) {
+  if (!NS_IsMainThread()) {
+    // These objects are mainthread-only.
+    return nullptr;
+  }
+  uint32_t version, authType;
+  if (!JS_ReadUint32Pair(aReader, &version, &authType) ||
+      version != RTCCERTIFICATE_SC_VERSION) {
+    return nullptr;
+  }
+  RefPtr<RTCCertificate> cert = new RTCCertificate(aGlobal);
+  cert->mAuthType = static_cast<SSLKEAType>(authType);
+
+  uint32_t high, low;
+  if (!JS_ReadUint32Pair(aReader, &high, &low)) {
+    return nullptr;
+  }
+  cert->mExpires = static_cast<PRTime>(high) << 32 | low;
+
+  if (!cert->ReadPrivateKey(aReader) || !cert->ReadCertificate(aReader)) {
+    return nullptr;
+  }
+
+  return cert.forget();
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webrtc/RTCCertificate.h b/dom/media/webrtc/RTCCertificate.h
new file mode 100644
index 0000000000..a5cc6bde32
--- /dev/null
+++ b/dom/media/webrtc/RTCCertificate.h
@@ -0,0 +1,98 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef mozilla_dom_RTCCertificate_h
+#define mozilla_dom_RTCCertificate_h
+
+#include <cstdint>
+#include "ScopedNSSTypes.h"
+#include "certt.h"
+#include "js/RootingAPI.h"
+#include "keythi.h"
+#include "mozilla/AlreadyAddRefed.h"
+#include "mozilla/Assertions.h"
+#include "mozilla/RefPtr.h"
+#include "nsCycleCollectionParticipant.h"
+#include "nsIGlobalObject.h"
+#include "nsISupports.h"
+#include "nsWrapperCache.h"
+#include "prtime.h"
+#include "sslt.h"
+
+class JSObject;
+struct JSContext;
+struct JSStructuredCloneReader;
+struct JSStructuredCloneWriter;
+
+namespace JS {
+class Compartment;
+}
+
+namespace mozilla {
+class DtlsIdentity;
+class ErrorResult;
+
+namespace dom {
+
+class GlobalObject;
+class ObjectOrString;
+class Promise;
+
+class RTCCertificate final : public nsISupports, public nsWrapperCache {
+ public:
+  NS_DECL_CYCLE_COLLECTING_ISUPPORTS
+  NS_DECL_CYCLE_COLLECTION_WRAPPERCACHE_CLASS(RTCCertificate)
+
+  // WebIDL method that implements RTCPeerConnection.generateCertificate.
+  static already_AddRefed<Promise> GenerateCertificate(
+      const GlobalObject& aGlobal, const ObjectOrString& aOptions,
+      ErrorResult& aRv, JS::Compartment* aCompartment = nullptr);
+
+  explicit RTCCertificate(nsIGlobalObject* aGlobal);
+  RTCCertificate(nsIGlobalObject* aGlobal, SECKEYPrivateKey* aPrivateKey,
+                 CERTCertificate* aCertificate, SSLKEAType aAuthType,
+                 PRTime aExpires);
+
+  nsIGlobalObject* GetParentObject() const { return mGlobal; }
+  virtual JSObject* WrapObject(JSContext* aCx,
+                               JS::Handle<JSObject*> aGivenProto) override;
+
+  // WebIDL expires attribute.  Note: JS dates are milliseconds since epoch;
+  // NSPR PRTime is in microseconds since the same epoch.
+  uint64_t Expires() const { return mExpires / PR_USEC_PER_MSEC; }
+
+  // Accessors for use by PeerConnectionImpl.
+  RefPtr<DtlsIdentity> CreateDtlsIdentity() const;
+  const UniqueCERTCertificate& Certificate() const { return mCertificate; }
+
+  // Structured clone methods
+  bool WriteStructuredClone(JSContext* aCx,
+                            JSStructuredCloneWriter* aWriter) const;
+  static already_AddRefed<RTCCertificate> ReadStructuredClone(
+      JSContext* aCx, nsIGlobalObject* aGlobal,
+      JSStructuredCloneReader* aReader);
+
+ private:
+  ~RTCCertificate() = default;
+  void operator=(const RTCCertificate&) = delete;
+  RTCCertificate(const RTCCertificate&) = delete;
+
+  bool ReadCertificate(JSStructuredCloneReader* aReader);
+  bool ReadPrivateKey(JSStructuredCloneReader* aReader);
+  bool WriteCertificate(JSStructuredCloneWriter* aWriter) const;
+  bool WritePrivateKey(JSStructuredCloneWriter* aWriter) const;
+
+  RefPtr<nsIGlobalObject> mGlobal;
+  UniqueSECKEYPrivateKey mPrivateKey;
+  UniqueCERTCertificate mCertificate;
+  SSLKEAType mAuthType;
+  PRTime mExpires;
+};
+
+}  // namespace dom
+}  // namespace mozilla
+
+#endif  // mozilla_dom_RTCCertificate_h
diff --git a/dom/media/webrtc/RTCIdentityProviderRegistrar.cpp b/dom/media/webrtc/RTCIdentityProviderRegistrar.cpp
new file mode 100644
index 0000000000..44e4c0d775
--- /dev/null
+++ b/dom/media/webrtc/RTCIdentityProviderRegistrar.cpp
@@ -0,0 +1,70 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "RTCIdentityProviderRegistrar.h"
+#include "mozilla/Attributes.h"
+#include "nsCycleCollectionParticipant.h"
+
+namespace mozilla::dom {
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(RTCIdentityProviderRegistrar)
+  NS_WRAPPERCACHE_INTERFACE_MAP_ENTRY
+  NS_INTERFACE_MAP_ENTRY(nsISupports)
+NS_INTERFACE_MAP_END
+
+NS_IMPL_CYCLE_COLLECTING_ADDREF(RTCIdentityProviderRegistrar)
+NS_IMPL_CYCLE_COLLECTING_RELEASE(RTCIdentityProviderRegistrar)
+
+NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE(RTCIdentityProviderRegistrar, mGlobal,
+                                      mGenerateAssertionCallback,
+                                      mValidateAssertionCallback)
+
+RTCIdentityProviderRegistrar::RTCIdentityProviderRegistrar(
+    nsIGlobalObject* aGlobal)
+    : mGlobal(aGlobal),
+      mGenerateAssertionCallback(nullptr),
+      mValidateAssertionCallback(nullptr) {}
+
+RTCIdentityProviderRegistrar::~RTCIdentityProviderRegistrar() = default;
+
+nsIGlobalObject* RTCIdentityProviderRegistrar::GetParentObject() const {
+  return mGlobal;
+}
+
+JSObject* RTCIdentityProviderRegistrar::WrapObject(
+    JSContext* aCx, JS::Handle<JSObject*> aGivenProto) {
+  return RTCIdentityProviderRegistrar_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+void RTCIdentityProviderRegistrar::Register(const RTCIdentityProvider& aIdp) {
+  mGenerateAssertionCallback = aIdp.mGenerateAssertion;
+  mValidateAssertionCallback = aIdp.mValidateAssertion;
+}
+
+bool RTCIdentityProviderRegistrar::HasIdp() const {
+  return mGenerateAssertionCallback && mValidateAssertionCallback;
+}
+
+already_AddRefed<Promise> RTCIdentityProviderRegistrar::GenerateAssertion(
+    const nsAString& aContents, const nsAString& aOrigin,
+    const RTCIdentityProviderOptions& aOptions, ErrorResult& aRv) {
+  if (!mGenerateAssertionCallback) {
+    aRv.Throw(NS_ERROR_NOT_INITIALIZED);
+    return nullptr;
+  }
+  RefPtr<GenerateAssertionCallback> callback(mGenerateAssertionCallback);
+  return callback->Call(aContents, aOrigin, aOptions, aRv);
+}
+already_AddRefed<Promise> RTCIdentityProviderRegistrar::ValidateAssertion(
+    const nsAString& aAssertion, const nsAString& aOrigin, ErrorResult& aRv) {
+  if (!mValidateAssertionCallback) {
+    aRv.Throw(NS_ERROR_NOT_INITIALIZED);
+    return nullptr;
+  }
+  RefPtr<ValidateAssertionCallback> callback(mValidateAssertionCallback);
+  return callback->Call(aAssertion, aOrigin, aRv);
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webrtc/RTCIdentityProviderRegistrar.h b/dom/media/webrtc/RTCIdentityProviderRegistrar.h
new file mode 100644
index 0000000000..0510471f17
--- /dev/null
+++ b/dom/media/webrtc/RTCIdentityProviderRegistrar.h
@@ -0,0 +1,59 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef RTCIDENTITYPROVIDER_H_
+#define RTCIDENTITYPROVIDER_H_
+
+#include "mozilla/RefPtr.h"
+#include "nsCOMPtr.h"
+#include "nsISupportsImpl.h"
+#include "nsIGlobalObject.h"
+#include "nsWrapperCache.h"
+#include "mozilla/Attributes.h"
+#include "mozilla/dom/Promise.h"
+#include "mozilla/dom/BindingDeclarations.h"
+#include "mozilla/dom/RTCIdentityProviderBinding.h"
+
+namespace mozilla::dom {
+
+struct RTCIdentityProvider;
+
+class RTCIdentityProviderRegistrar final : public nsISupports,
+                                           public nsWrapperCache {
+ public:
+  NS_DECL_CYCLE_COLLECTING_ISUPPORTS
+  NS_DECL_CYCLE_COLLECTION_WRAPPERCACHE_CLASS(RTCIdentityProviderRegistrar)
+
+  explicit RTCIdentityProviderRegistrar(nsIGlobalObject* aGlobal);
+
+  // As required
+  nsIGlobalObject* GetParentObject() const;
+  virtual JSObject* WrapObject(JSContext* aCx,
+                               JS::Handle<JSObject*> aGivenProto) override;
+
+  // setter and checker
+  void Register(const RTCIdentityProvider& aIdp);
+  bool HasIdp() const;
+
+  MOZ_CAN_RUN_SCRIPT
+  already_AddRefed<Promise> GenerateAssertion(
+      const nsAString& aContents, const nsAString& aOrigin,
+      const RTCIdentityProviderOptions& aOptions, ErrorResult& aRv);
+  MOZ_CAN_RUN_SCRIPT
+  already_AddRefed<Promise> ValidateAssertion(const nsAString& assertion,
+                                              const nsAString& origin,
+                                              ErrorResult& aRv);
+
+ private:
+  ~RTCIdentityProviderRegistrar();
+
+  nsCOMPtr<nsIGlobalObject> mGlobal;
+  RefPtr<GenerateAssertionCallback> mGenerateAssertionCallback;
+  RefPtr<ValidateAssertionCallback> mValidateAssertionCallback;
+};
+
+}  // namespace mozilla::dom
+
+#endif /* RTCIDENTITYPROVIDER_H_ */
diff --git a/dom/media/webrtc/SineWaveGenerator.h b/dom/media/webrtc/SineWaveGenerator.h
new file mode 100644
index 0000000000..73bab53f19
--- /dev/null
+++ b/dom/media/webrtc/SineWaveGenerator.h
@@ -0,0 +1,58 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef SINEWAVEGENERATOR_H_
+#define SINEWAVEGENERATOR_H_
+
+#include "MediaSegment.h"
+#include "prtime.h"
+
+namespace mozilla {
+
+// generate 1k sine wave per second
+template <typename Sample>
+class SineWaveGenerator {
+  static_assert(std::is_same<Sample, int16_t>::value ||
+                std::is_same<Sample, float>::value);
+
+ public:
+  static const int bytesPerSample = sizeof(Sample);
+  static const int millisecondsPerSecond = PR_MSEC_PER_SEC;
+  static constexpr float twopi = 2 * M_PI;
+
+  /* If more than 1 channel, generated samples are interleaved. */
+  SineWaveGenerator(uint32_t aSampleRate, uint32_t aFrequency)
+      : mPhase(0.), mPhaseIncrement(twopi * aFrequency / aSampleRate) {}
+
+  // NOTE: only safely called from a single thread (MTG callback)
+  void generate(Sample* aBuffer, TrackTicks aFrameCount,
+                uint32_t aChannelCount = 1) {
+    while (aFrameCount--) {
+      Sample value = sin(mPhase) * Amplitude();
+      for (uint32_t channel = 0; channel < aChannelCount; channel++) {
+        *aBuffer++ = value;
+      }
+      mPhase += mPhaseIncrement;
+      if (mPhase > twopi) {
+        mPhase -= twopi;
+      }
+    }
+  }
+
+  static float Amplitude() {
+    // Set volume to -20db.
+    if (std::is_same<Sample, int16_t>::value) {
+      return 3276.8;  // 32768.0 * 10^(-20/20) = 3276.8
+    }
+    return 0.1f;  // 1.0 * 10^(-20/20) = 0.1
+  }
+
+ private:
+  double mPhase;
+  const double mPhaseIncrement;
+};
+
+}  // namespace mozilla
+
+#endif /* SINEWAVEGENERATOR_H_ */
diff --git a/dom/media/webrtc/WebrtcGlobal.h b/dom/media/webrtc/WebrtcGlobal.h
new file mode 100644
index 0000000000..dac51dcf6b
--- /dev/null
+++ b/dom/media/webrtc/WebrtcGlobal.h
@@ -0,0 +1,506 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _WEBRTC_GLOBAL_H_
+#define _WEBRTC_GLOBAL_H_
+
+#include "WebrtcIPCTraits.h"
+#include "ipc/EnumSerializer.h"
+#include "ipc/IPCMessageUtilsSpecializations.h"
+#include "mozilla/dom/BindingDeclarations.h"
+#include "mozilla/dom/RTCDataChannelBinding.h"
+#include "mozilla/dom/RTCStatsReportBinding.h"
+#include "mozilla/UniquePtr.h"
+
+typedef mozilla::dom::RTCStatsReportInternal StatsReport;
+typedef nsTArray<mozilla::UniquePtr<StatsReport>> RTCReports;
+typedef mozilla::dom::Sequence<nsString> WebrtcGlobalLog;
+
+namespace mozilla {
+namespace dom {
+// Calls aFunction with all public members of aStats.
+// Typical usage would have aFunction take a parameter pack.
+// To avoid inconsistencies, this should be the only explicit list of the
+// public RTCStatscollection members in C++.
+template <typename Collection, typename Function>
+static auto ForAllPublicRTCStatsCollectionMembers(Collection& aStats,
+                                                  Function aFunction) {
+  static_assert(std::is_same_v<typename std::remove_const<Collection>::type,
+                               RTCStatsCollection>,
+                "aStats must be a const or non-const RTCStatsCollection");
+  return aFunction(
+      aStats.mInboundRtpStreamStats, aStats.mOutboundRtpStreamStats,
+      aStats.mRemoteInboundRtpStreamStats, aStats.mRemoteOutboundRtpStreamStats,
+      aStats.mMediaSourceStats, aStats.mPeerConnectionStats,
+      aStats.mRtpContributingSourceStats, aStats.mIceCandidatePairStats,
+      aStats.mIceCandidateStats, aStats.mTrickledIceCandidateStats,
+      aStats.mDataChannelStats, aStats.mCodecStats);
+}
+
+// Calls aFunction with all members of aStats, including internal ones.
+// Typical usage would have aFunction take a parameter pack.
+// To avoid inconsistencies, this should be the only explicit list of the
+// internal RTCStatscollection members in C++.
+template <typename Collection, typename Function>
+static auto ForAllRTCStatsCollectionMembers(Collection& aStats,
+                                            Function aFunction) {
+  static_assert(std::is_same_v<typename std::remove_const<Collection>::type,
+                               RTCStatsCollection>,
+                "aStats must be a const or non-const RTCStatsCollection");
+  return ForAllPublicRTCStatsCollectionMembers(aStats, [&](auto&... aMember) {
+    return aFunction(aMember..., aStats.mRawLocalCandidates,
+                     aStats.mRawRemoteCandidates, aStats.mVideoFrameHistories,
+                     aStats.mBandwidthEstimations);
+  });
+}
+}  // namespace dom
+}  // namespace mozilla
+
+namespace IPC {
+
+template <>
+struct ParamTraits<mozilla::dom::RTCStatsType>
+    : public ContiguousEnumSerializer<mozilla::dom::RTCStatsType,
+                                      mozilla::dom::RTCStatsType::Codec,
+                                      mozilla::dom::RTCStatsType::EndGuard_> {};
+
+template <>
+struct ParamTraits<mozilla::dom::RTCStatsIceCandidatePairState>
+    : public ContiguousEnumSerializer<
+          mozilla::dom::RTCStatsIceCandidatePairState,
+          mozilla::dom::RTCStatsIceCandidatePairState::Frozen,
+          mozilla::dom::RTCStatsIceCandidatePairState::EndGuard_> {};
+
+template <>
+struct ParamTraits<mozilla::dom::RTCIceCandidateType>
+    : public ContiguousEnumSerializer<
+          mozilla::dom::RTCIceCandidateType,
+          mozilla::dom::RTCIceCandidateType::Host,
+          mozilla::dom::RTCIceCandidateType::EndGuard_> {};
+
+template <>
+struct ParamTraits<mozilla::dom::RTCBundlePolicy>
+    : public ContiguousEnumSerializer<
+          mozilla::dom::RTCBundlePolicy,
+          mozilla::dom::RTCBundlePolicy::Balanced,
+          mozilla::dom::RTCBundlePolicy::EndGuard_> {};
+
+DEFINE_IPC_SERIALIZER_WITH_FIELDS(mozilla::dom::RTCIceServerInternal, mUrls,
+                                  mCredentialProvided, mUserNameProvided);
+
+DEFINE_IPC_SERIALIZER_WITH_FIELDS(mozilla::dom::RTCConfigurationInternal,
+                                  mBundlePolicy, mCertificatesProvided,
+                                  mIceServers, mIceTransportPolicy,
+                                  mPeerIdentityProvided, mSdpSemantics);
+
+DEFINE_IPC_SERIALIZER_WITH_FIELDS(mozilla::dom::RTCSdpParsingErrorInternal,
+                                  mLineNumber, mError);
+
+DEFINE_IPC_SERIALIZER_WITH_FIELDS(mozilla::dom::RTCSdpHistoryEntryInternal,
+                                  mTimestamp, mIsLocal, mSdp, mErrors);
+
+template <>
+struct ParamTraits<mozilla::dom::RTCStatsCollection> {
+  static void Write(MessageWriter* aWriter,
+                    const mozilla::dom::RTCStatsCollection& aParam) {
+    mozilla::dom::ForAllRTCStatsCollectionMembers(
+        aParam,
+        [&](const auto&... aMember) { WriteParams(aWriter, aMember...); });
+  }
+
+  static bool Read(MessageReader* aReader,
+                   mozilla::dom::RTCStatsCollection* aResult) {
+    return mozilla::dom::ForAllRTCStatsCollectionMembers(
+        *aResult,
+        [&](auto&... aMember) { return ReadParams(aReader, aMember...); });
+  }
+};
+
+DEFINE_IPC_SERIALIZER_WITH_SUPER_CLASS_AND_FIELDS(
+    mozilla::dom::RTCStatsReportInternal, mozilla::dom::RTCStatsCollection,
+    mClosed, mSdpHistory, mPcid, mBrowserId, mTimestamp, mCallDurationMs,
+    mIceRestarts, mIceRollbacks, mOfferer, mConfiguration);
+
+typedef mozilla::dom::RTCStats RTCStats;
+
+static void WriteRTCStats(MessageWriter* aWriter, const RTCStats& aParam) {
+  // RTCStats base class
+  WriteParam(aWriter, aParam.mId);
+  WriteParam(aWriter, aParam.mTimestamp);
+  WriteParam(aWriter, aParam.mType);
+}
+
+static bool ReadRTCStats(MessageReader* aReader, RTCStats* aResult) {
+  // RTCStats base class
+  if (!ReadParam(aReader, &(aResult->mId)) ||
+      !ReadParam(aReader, &(aResult->mTimestamp)) ||
+      !ReadParam(aReader, &(aResult->mType))) {
+    return false;
+  }
+
+  return true;
+}
+
+template <>
+struct ParamTraits<mozilla::dom::RTCIceCandidatePairStats> {
+  typedef mozilla::dom::RTCIceCandidatePairStats paramType;
+
+  static void Write(MessageWriter* aWriter, const paramType& aParam) {
+    WriteParam(aWriter, aParam.mTransportId);
+    WriteParam(aWriter, aParam.mLocalCandidateId);
+    WriteParam(aWriter, aParam.mPriority);
+    WriteParam(aWriter, aParam.mNominated);
+    WriteParam(aWriter, aParam.mWritable);
+    WriteParam(aWriter, aParam.mReadable);
+    WriteParam(aWriter, aParam.mRemoteCandidateId);
+    WriteParam(aWriter, aParam.mSelected);
+    WriteParam(aWriter, aParam.mComponentId);
+    WriteParam(aWriter, aParam.mState);
+    WriteParam(aWriter, aParam.mBytesSent);
+    WriteParam(aWriter, aParam.mBytesReceived);
+    WriteParam(aWriter, aParam.mLastPacketSentTimestamp);
+    WriteParam(aWriter, aParam.mLastPacketReceivedTimestamp);
+    WriteRTCStats(aWriter, aParam);
+  }
+
+  static bool Read(MessageReader* aReader, paramType* aResult) {
+    if (!ReadParam(aReader, &(aResult->mTransportId)) ||
+        !ReadParam(aReader, &(aResult->mLocalCandidateId)) ||
+        !ReadParam(aReader, &(aResult->mPriority)) ||
+        !ReadParam(aReader, &(aResult->mNominated)) ||
+        !ReadParam(aReader, &(aResult->mWritable)) ||
+        !ReadParam(aReader, &(aResult->mReadable)) ||
+        !ReadParam(aReader, &(aResult->mRemoteCandidateId)) ||
+        !ReadParam(aReader, &(aResult->mSelected)) ||
+        !ReadParam(aReader, &(aResult->mComponentId)) ||
+        !ReadParam(aReader, &(aResult->mState)) ||
+        !ReadParam(aReader, &(aResult->mBytesSent)) ||
+        !ReadParam(aReader, &(aResult->mBytesReceived)) ||
+        !ReadParam(aReader, &(aResult->mLastPacketSentTimestamp)) ||
+        !ReadParam(aReader, &(aResult->mLastPacketReceivedTimestamp)) ||
+        !ReadRTCStats(aReader, aResult)) {
+      return false;
+    }
+
+    return true;
+  }
+};
+
+template <>
+struct ParamTraits<mozilla::dom::RTCIceCandidateStats> {
+  typedef mozilla::dom::RTCIceCandidateStats paramType;
+
+  static void Write(MessageWriter* aWriter, const paramType& aParam) {
+    WriteParam(aWriter, aParam.mCandidateType);
+    WriteParam(aWriter, aParam.mPriority);
+    WriteParam(aWriter, aParam.mTransportId);
+    WriteParam(aWriter, aParam.mAddress);
+    WriteParam(aWriter, aParam.mRelayProtocol);
+    WriteParam(aWriter, aParam.mPort);
+    WriteParam(aWriter, aParam.mProtocol);
+    WriteParam(aWriter, aParam.mProxied);
+    WriteRTCStats(aWriter, aParam);
+  }
+
+  static bool Read(MessageReader* aReader, paramType* aResult) {
+    if (!ReadParam(aReader, &(aResult->mCandidateType)) ||
+        !ReadParam(aReader, &(aResult->mPriority)) ||
+        !ReadParam(aReader, &(aResult->mTransportId)) ||
+        !ReadParam(aReader, &(aResult->mAddress)) ||
+        !ReadParam(aReader, &(aResult->mRelayProtocol)) ||
+        !ReadParam(aReader, &(aResult->mPort)) ||
+        !ReadParam(aReader, &(aResult->mProtocol)) ||
+        !ReadParam(aReader, &(aResult->mProxied)) ||
+        !ReadRTCStats(aReader, aResult)) {
+      return false;
+    }
+
+    return true;
+  }
+};
+
+static void WriteRTCRtpStreamStats(
+    MessageWriter* aWriter, const mozilla::dom::RTCRtpStreamStats& aParam) {
+  WriteParam(aWriter, aParam.mSsrc);
+  WriteParam(aWriter, aParam.mMediaType);
+  WriteParam(aWriter, aParam.mKind);
+  WriteParam(aWriter, aParam.mTransportId);
+  WriteParam(aWriter, aParam.mCodecId);
+  WriteRTCStats(aWriter, aParam);
+}
+
+static bool ReadRTCRtpStreamStats(MessageReader* aReader,
+                                  mozilla::dom::RTCRtpStreamStats* aResult) {
+  return ReadParam(aReader, &(aResult->mSsrc)) &&
+         ReadParam(aReader, &(aResult->mMediaType)) &&
+         ReadParam(aReader, &(aResult->mKind)) &&
+         ReadParam(aReader, &(aResult->mTransportId)) &&
+         ReadParam(aReader, &(aResult->mCodecId)) &&
+         ReadRTCStats(aReader, aResult);
+}
+
+static void WriteRTCReceivedRtpStreamStats(
+    MessageWriter* aWriter,
+    const mozilla::dom::RTCReceivedRtpStreamStats& aParam) {
+  WriteParam(aWriter, aParam.mPacketsReceived);
+  WriteParam(aWriter, aParam.mPacketsLost);
+  WriteParam(aWriter, aParam.mJitter);
+  WriteParam(aWriter, aParam.mDiscardedPackets);
+  WriteParam(aWriter, aParam.mPacketsDiscarded);
+  WriteRTCRtpStreamStats(aWriter, aParam);
+}
+
+static bool ReadRTCReceivedRtpStreamStats(
+    MessageReader* aReader, mozilla::dom::RTCReceivedRtpStreamStats* aResult) {
+  return ReadParam(aReader, &(aResult->mPacketsReceived)) &&
+         ReadParam(aReader, &(aResult->mPacketsLost)) &&
+         ReadParam(aReader, &(aResult->mJitter)) &&
+         ReadParam(aReader, &(aResult->mDiscardedPackets)) &&
+         ReadParam(aReader, &(aResult->mPacketsDiscarded)) &&
+         ReadRTCRtpStreamStats(aReader, aResult);
+}
+
+template <>
+struct ParamTraits<mozilla::dom::RTCInboundRtpStreamStats> {
+  typedef mozilla::dom::RTCInboundRtpStreamStats paramType;
+
+  static void Write(MessageWriter* aWriter, const paramType& aParam) {
+    WriteParam(aWriter, aParam.mTrackIdentifier);
+    WriteParam(aWriter, aParam.mRemoteId);
+    WriteParam(aWriter, aParam.mFramesDecoded);
+    WriteParam(aWriter, aParam.mFramesDropped);
+    WriteParam(aWriter, aParam.mFrameWidth);
+    WriteParam(aWriter, aParam.mFrameHeight);
+    WriteParam(aWriter, aParam.mFramesPerSecond);
+    WriteParam(aWriter, aParam.mQpSum);
+    WriteParam(aWriter, aParam.mTotalDecodeTime);
+    WriteParam(aWriter, aParam.mTotalInterFrameDelay);
+    WriteParam(aWriter, aParam.mTotalSquaredInterFrameDelay);
+    WriteParam(aWriter, aParam.mLastPacketReceivedTimestamp);
+    WriteParam(aWriter, aParam.mHeaderBytesReceived);
+    WriteParam(aWriter, aParam.mFecPacketsReceived);
+    WriteParam(aWriter, aParam.mFecPacketsDiscarded);
+    WriteParam(aWriter, aParam.mBytesReceived);
+    WriteParam(aWriter, aParam.mNackCount);
+    WriteParam(aWriter, aParam.mFirCount);
+    WriteParam(aWriter, aParam.mPliCount);
+    WriteParam(aWriter, aParam.mTotalProcessingDelay);
+    // Always missing from libwebrtc stats
+    // WriteParam(aWriter, aParam.mEstimatedPlayoutTimestamp);
+    WriteParam(aWriter, aParam.mFramesReceived);
+    WriteParam(aWriter, aParam.mJitterBufferDelay);
+    WriteParam(aWriter, aParam.mJitterBufferEmittedCount);
+    WriteParam(aWriter, aParam.mTotalSamplesReceived);
+    WriteParam(aWriter, aParam.mConcealedSamples);
+    WriteParam(aWriter, aParam.mSilentConcealedSamples);
+    WriteParam(aWriter, aParam.mConcealmentEvents);
+    WriteParam(aWriter, aParam.mInsertedSamplesForDeceleration);
+    WriteParam(aWriter, aParam.mRemovedSamplesForAcceleration);
+    WriteParam(aWriter, aParam.mAudioLevel);
+    WriteParam(aWriter, aParam.mTotalAudioEnergy);
+    WriteParam(aWriter, aParam.mTotalSamplesDuration);
+    WriteRTCReceivedRtpStreamStats(aWriter, aParam);
+  }
+
+  static bool Read(MessageReader* aReader, paramType* aResult) {
+    return ReadParam(aReader, &(aResult->mTrackIdentifier)) &&
+           ReadParam(aReader, &(aResult->mRemoteId)) &&
+           ReadParam(aReader, &(aResult->mFramesDecoded)) &&
+           ReadParam(aReader, &(aResult->mFramesDropped)) &&
+           ReadParam(aReader, &(aResult->mFrameWidth)) &&
+           ReadParam(aReader, &(aResult->mFrameHeight)) &&
+           ReadParam(aReader, &(aResult->mFramesPerSecond)) &&
+           ReadParam(aReader, &(aResult->mQpSum)) &&
+           ReadParam(aReader, &(aResult->mTotalDecodeTime)) &&
+           ReadParam(aReader, &(aResult->mTotalInterFrameDelay)) &&
+           ReadParam(aReader, &(aResult->mTotalSquaredInterFrameDelay)) &&
+           ReadParam(aReader, &(aResult->mLastPacketReceivedTimestamp)) &&
+           ReadParam(aReader, &(aResult->mHeaderBytesReceived)) &&
+           ReadParam(aReader, &(aResult->mFecPacketsReceived)) &&
+           ReadParam(aReader, &(aResult->mFecPacketsDiscarded)) &&
+           ReadParam(aReader, &(aResult->mBytesReceived)) &&
+           ReadParam(aReader, &(aResult->mNackCount)) &&
+           ReadParam(aReader, &(aResult->mFirCount)) &&
+           ReadParam(aReader, &(aResult->mPliCount)) &&
+           ReadParam(aReader, &(aResult->mTotalProcessingDelay)) &&
+           // Always missing from libwebrtc
+           // ReadParam(aReader, &(aResult->mEstimatedPlayoutTimestamp)) &&
+           ReadParam(aReader, &(aResult->mFramesReceived)) &&
+           ReadParam(aReader, &(aResult->mJitterBufferDelay)) &&
+           ReadParam(aReader, &(aResult->mJitterBufferEmittedCount)) &&
+           ReadParam(aReader, &(aResult->mTotalSamplesReceived)) &&
+           ReadParam(aReader, &(aResult->mConcealedSamples)) &&
+           ReadParam(aReader, &(aResult->mSilentConcealedSamples)) &&
+           ReadParam(aReader, &(aResult->mConcealmentEvents)) &&
+           ReadParam(aReader, &(aResult->mInsertedSamplesForDeceleration)) &&
+           ReadParam(aReader, &(aResult->mRemovedSamplesForAcceleration)) &&
+           ReadParam(aReader, &(aResult->mAudioLevel)) &&
+           ReadParam(aReader, &(aResult->mTotalAudioEnergy)) &&
+           ReadParam(aReader, &(aResult->mTotalSamplesDuration)) &&
+           ReadRTCReceivedRtpStreamStats(aReader, aResult);
+  }
+};
+
+static void WriteRTCSentRtpStreamStats(
+    MessageWriter* aWriter, const mozilla::dom::RTCSentRtpStreamStats& aParam) {
+  WriteParam(aWriter, aParam.mPacketsSent);
+  WriteParam(aWriter, aParam.mBytesSent);
+  WriteRTCRtpStreamStats(aWriter, aParam);
+}
+
+static bool ReadRTCSentRtpStreamStats(
+    MessageReader* aReader, mozilla::dom::RTCSentRtpStreamStats* aResult) {
+  return ReadParam(aReader, &(aResult->mPacketsSent)) &&
+         ReadParam(aReader, &(aResult->mBytesSent)) &&
+         ReadRTCRtpStreamStats(aReader, aResult);
+}
+
+template <>
+struct ParamTraits<mozilla::dom::RTCOutboundRtpStreamStats> {
+  typedef mozilla::dom::RTCOutboundRtpStreamStats paramType;
+
+  static void Write(MessageWriter* aWriter, const paramType& aParam) {
+    WriteParam(aWriter, aParam.mRemoteId);
+    WriteParam(aWriter, aParam.mFramesEncoded);
+    WriteParam(aWriter, aParam.mQpSum);
+    WriteParam(aWriter, aParam.mNackCount);
+    WriteParam(aWriter, aParam.mFirCount);
+    WriteParam(aWriter, aParam.mPliCount);
+    WriteParam(aWriter, aParam.mHeaderBytesSent);
+    WriteParam(aWriter, aParam.mRetransmittedPacketsSent);
+    WriteParam(aWriter, aParam.mRetransmittedBytesSent);
+    WriteParam(aWriter, aParam.mTotalEncodedBytesTarget);
+    WriteParam(aWriter, aParam.mFrameWidth);
+    WriteParam(aWriter, aParam.mFrameHeight);
+    WriteParam(aWriter, aParam.mFramesSent);
+    WriteParam(aWriter, aParam.mHugeFramesSent);
+    WriteParam(aWriter, aParam.mTotalEncodeTime);
+    WriteRTCSentRtpStreamStats(aWriter, aParam);
+  }
+
+  static bool Read(MessageReader* aReader, paramType* aResult) {
+    return ReadParam(aReader, &(aResult->mRemoteId)) &&
+           ReadParam(aReader, &(aResult->mFramesEncoded)) &&
+           ReadParam(aReader, &(aResult->mQpSum)) &&
+           ReadParam(aReader, &(aResult->mNackCount)) &&
+           ReadParam(aReader, &(aResult->mFirCount)) &&
+           ReadParam(aReader, &(aResult->mPliCount)) &&
+           ReadParam(aReader, &(aResult->mHeaderBytesSent)) &&
+           ReadParam(aReader, &(aResult->mRetransmittedPacketsSent)) &&
+           ReadParam(aReader, &(aResult->mRetransmittedBytesSent)) &&
+           ReadParam(aReader, &(aResult->mTotalEncodedBytesTarget)) &&
+           ReadParam(aReader, &(aResult->mFrameWidth)) &&
+           ReadParam(aReader, &(aResult->mFrameHeight)) &&
+           ReadParam(aReader, &(aResult->mFramesSent)) &&
+           ReadParam(aReader, &(aResult->mHugeFramesSent)) &&
+           ReadParam(aReader, &(aResult->mTotalEncodeTime)) &&
+           ReadRTCSentRtpStreamStats(aReader, aResult);
+  }
+};
+
+template <>
+struct ParamTraits<mozilla::dom::RTCRemoteInboundRtpStreamStats> {
+  typedef mozilla::dom::RTCRemoteInboundRtpStreamStats paramType;
+
+  static void Write(MessageWriter* aWriter, const paramType& aParam) {
+    WriteParam(aWriter, aParam.mLocalId);
+    WriteParam(aWriter, aParam.mRoundTripTime);
+    WriteParam(aWriter, aParam.mTotalRoundTripTime);
+    WriteParam(aWriter, aParam.mFractionLost);
+    WriteParam(aWriter, aParam.mRoundTripTimeMeasurements);
+    WriteRTCReceivedRtpStreamStats(aWriter, aParam);
+  }
+
+  static bool Read(MessageReader* aReader, paramType* aResult) {
+    return ReadParam(aReader, &(aResult->mLocalId)) &&
+           ReadParam(aReader, &(aResult->mRoundTripTime)) &&
+           ReadParam(aReader, &(aResult->mTotalRoundTripTime)) &&
+           ReadParam(aReader, &(aResult->mFractionLost)) &&
+           ReadParam(aReader, &(aResult->mRoundTripTimeMeasurements)) &&
+           ReadRTCReceivedRtpStreamStats(aReader, aResult);
+  }
+};
+
+template <>
+struct ParamTraits<mozilla::dom::RTCRemoteOutboundRtpStreamStats> {
+  typedef mozilla::dom::RTCRemoteOutboundRtpStreamStats paramType;
+
+  static void Write(MessageWriter* aWriter, const paramType& aParam) {
+    WriteParam(aWriter, aParam.mLocalId);
+    WriteParam(aWriter, aParam.mRemoteTimestamp);
+    WriteRTCSentRtpStreamStats(aWriter, aParam);
+  }
+
+  static bool Read(MessageReader* aReader, paramType* aResult) {
+    return ReadParam(aReader, &(aResult->mLocalId)) &&
+           ReadParam(aReader, &(aResult->mRemoteTimestamp)) &&
+           ReadRTCSentRtpStreamStats(aReader, aResult);
+  }
+};
+
+DEFINE_IPC_SERIALIZER_WITH_FIELDS(mozilla::dom::RTCMediaSourceStats, mId,
+                                  mTimestamp, mType, mTrackIdentifier, mKind);
+
+template <>
+struct ParamTraits<mozilla::dom::RTCRTPContributingSourceStats> {
+  typedef mozilla::dom::RTCRTPContributingSourceStats paramType;
+
+  static void Write(MessageWriter* aWriter, const paramType& aParam) {
+    WriteParam(aWriter, aParam.mContributorSsrc);
+    WriteParam(aWriter, aParam.mInboundRtpStreamId);
+    WriteRTCStats(aWriter, aParam);
+  }
+
+  static bool Read(MessageReader* aReader, paramType* aResult) {
+    if (!ReadParam(aReader, &(aResult->mContributorSsrc)) ||
+        !ReadParam(aReader, &(aResult->mInboundRtpStreamId)) ||
+        !ReadRTCStats(aReader, aResult)) {
+      return false;
+    }
+    return true;
+  }
+};
+
+DEFINE_IPC_SERIALIZER_WITH_FIELDS(mozilla::dom::RTCPeerConnectionStats, mId,
+                                  mTimestamp, mType, mDataChannelsOpened,
+                                  mDataChannelsClosed);
+
+DEFINE_IPC_SERIALIZER_WITH_FIELDS(
+    mozilla::dom::RTCVideoFrameHistoryEntryInternal, mWidth, mHeight,
+    mRotationAngle, mFirstFrameTimestamp, mLastFrameTimestamp,
+    mConsecutiveFrames, mLocalSsrc, mRemoteSsrc);
+
+DEFINE_IPC_SERIALIZER_WITH_FIELDS(mozilla::dom::RTCVideoFrameHistoryInternal,
+                                  mTrackIdentifier, mEntries);
+
+DEFINE_IPC_SERIALIZER_WITH_FIELDS(mozilla::dom::RTCBandwidthEstimationInternal,
+                                  mTrackIdentifier, mSendBandwidthBps,
+                                  mMaxPaddingBps, mReceiveBandwidthBps,
+                                  mPacerDelayMs, mRttMs);
+
+DEFINE_IPC_SERIALIZER_WITH_FIELDS(mozilla::dom::RTCDataChannelStats, mId,
+                                  mTimestamp, mType, mLabel, mProtocol,
+                                  mDataChannelIdentifier, mState, mMessagesSent,
+                                  mBytesSent, mMessagesReceived, mBytesReceived)
+
+template <>
+struct ParamTraits<mozilla::dom::RTCDataChannelState>
+    : public ContiguousEnumSerializer<
+          mozilla::dom::RTCDataChannelState,
+          mozilla::dom::RTCDataChannelState::Connecting,
+          mozilla::dom::RTCDataChannelState::EndGuard_> {};
+
+DEFINE_IPC_SERIALIZER_WITH_FIELDS(mozilla::dom::RTCCodecStats, mTimestamp,
+                                  mType, mId, mPayloadType, mCodecType,
+                                  mTransportId, mMimeType, mClockRate,
+                                  mChannels, mSdpFmtpLine)
+
+template <>
+struct ParamTraits<mozilla::dom::RTCCodecType>
+    : public ContiguousEnumSerializer<mozilla::dom::RTCCodecType,
+                                      mozilla::dom::RTCCodecType::Encode,
+                                      mozilla::dom::RTCCodecType::EndGuard_> {};
+}  // namespace IPC
+
+#endif  // _WEBRTC_GLOBAL_H_
diff --git a/dom/media/webrtc/WebrtcIPCTraits.h b/dom/media/webrtc/WebrtcIPCTraits.h
new file mode 100644
index 0000000000..b076745608
--- /dev/null
+++ b/dom/media/webrtc/WebrtcIPCTraits.h
@@ -0,0 +1,89 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _WEBRTC_IPC_TRAITS_H_
+#define _WEBRTC_IPC_TRAITS_H_
+
+#include "ipc/EnumSerializer.h"
+#include "ipc/IPCMessageUtils.h"
+#include "ipc/IPCMessageUtilsSpecializations.h"
+#include "mozilla/dom/BindingDeclarations.h"
+#include "mozilla/dom/RTCConfigurationBinding.h"
+#include "mozilla/media/webrtc/WebrtcGlobal.h"
+#include "mozilla/dom/CandidateInfo.h"
+#include "mozilla/MacroForEach.h"
+#include "transport/dtlsidentity.h"
+#include <vector>
+
+namespace mozilla {
+typedef std::vector<std::string> StringVector;
+}
+
+namespace IPC {
+
+template <>
+struct ParamTraits<mozilla::dom::OwningStringOrStringSequence> {
+  typedef mozilla::dom::OwningStringOrStringSequence paramType;
+
+  // Ugh. OwningStringOrStringSequence already has this enum, but it is
+  // private generated code. So we have to re-create it.
+  enum Type { kUninitialized, kString, kStringSequence };
+
+  static void Write(MessageWriter* aWriter, const paramType& aParam) {
+    if (aParam.IsString()) {
+      aWriter->WriteInt16(kString);
+      WriteParam(aWriter, aParam.GetAsString());
+    } else if (aParam.IsStringSequence()) {
+      aWriter->WriteInt16(kStringSequence);
+      WriteParam(aWriter, aParam.GetAsStringSequence());
+    } else {
+      aWriter->WriteInt16(kUninitialized);
+    }
+  }
+
+  static bool Read(MessageReader* aReader, paramType* aResult) {
+    int16_t type;
+    if (!aReader->ReadInt16(&type)) {
+      return false;
+    }
+
+    switch (type) {
+      case kUninitialized:
+        aResult->Uninit();
+        return true;
+      case kString:
+        return ReadParam(aReader, &aResult->SetAsString());
+      case kStringSequence:
+        return ReadParam(aReader, &aResult->SetAsStringSequence());
+    }
+
+    return false;
+  }
+};
+
+template <typename T>
+struct WebidlEnumSerializer
+    : public ContiguousEnumSerializer<T, T(0), T::EndGuard_> {};
+
+template <>
+struct ParamTraits<mozilla::dom::RTCIceCredentialType>
+    : public WebidlEnumSerializer<mozilla::dom::RTCIceCredentialType> {};
+
+template <>
+struct ParamTraits<mozilla::dom::RTCIceTransportPolicy>
+    : public WebidlEnumSerializer<mozilla::dom::RTCIceTransportPolicy> {};
+
+DEFINE_IPC_SERIALIZER_WITH_FIELDS(mozilla::dom::RTCIceServer, mCredential,
+                                  mCredentialType, mUrl, mUrls, mUsername)
+
+DEFINE_IPC_SERIALIZER_WITH_FIELDS(mozilla::CandidateInfo, mCandidate, mUfrag,
+                                  mDefaultHostRtp, mDefaultPortRtp,
+                                  mDefaultHostRtcp, mDefaultPortRtcp,
+                                  mMDNSAddress, mActualAddress)
+
+DEFINE_IPC_SERIALIZER_WITH_FIELDS(mozilla::DtlsDigest, algorithm_, value_)
+
+}  // namespace IPC
+
+#endif  // _WEBRTC_IPC_TRAITS_H_
diff --git a/dom/media/webrtc/common/CandidateInfo.h b/dom/media/webrtc/common/CandidateInfo.h
new file mode 100644
index 0000000000..eb5b2ea299
--- /dev/null
+++ b/dom/media/webrtc/common/CandidateInfo.h
@@ -0,0 +1,27 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _CANDIDATE_INFO_H__
+#define _CANDIDATE_INFO_H__
+
+#include <string>
+#include <cstdint>
+
+namespace mozilla {
+
+// This is used both by IPDL code, and by signaling code.
+struct CandidateInfo {
+  std::string mCandidate;
+  std::string mMDNSAddress;
+  std::string mActualAddress;
+  std::string mUfrag;
+  std::string mDefaultHostRtp;
+  uint16_t mDefaultPortRtp = 0;
+  std::string mDefaultHostRtcp;
+  uint16_t mDefaultPortRtcp = 0;
+};
+
+}  // namespace mozilla
+
+#endif  //_CANDIDATE_INFO_H__
diff --git a/dom/media/webrtc/common/CommonTypes.h b/dom/media/webrtc/common/CommonTypes.h
new file mode 100644
index 0000000000..e9c9ffdefc
--- /dev/null
+++ b/dom/media/webrtc/common/CommonTypes.h
@@ -0,0 +1,52 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#pragma once
+
+#include <string>
+
+namespace csf {
+
+namespace ProviderStateEnum {
+enum ProviderState {
+  Ready,
+  Registering,
+  AwaitingIpAddress,
+  FetchingDeviceConfig,
+  Idle,
+  RecoveryPending,
+  Connected
+};
+const std::string toString(ProviderState);
+}  // namespace ProviderStateEnum
+namespace LoginErrorStatusEnum {
+enum LoginErrorStatus {
+  Ok,                             // No Error
+  Unknown,                        // Unknown Error
+  NoCallManagerConfigured,        // No Primary or Backup Call Manager
+  NoDevicesFound,                 // No devices
+  NoCsfDevicesFound,              // Devices but none of type CSF
+  PhoneConfigGenError,            // Could not generate phone config
+  SipProfileGenError,             // Could not build SIP profile
+  ConfigNotSet,                   // Config not set before calling login()
+  CreateConfigProviderFailed,     // Could not create ConfigProvider
+  CreateSoftPhoneProviderFailed,  // Could not create SoftPhoneProvider
+  MissingUsername,                // Username argument missing,
+  ManualLogout,                   // logout() has been called
+  LoggedInElseWhere,  // Another process has the mutex indicating it is logged
+                      // in
+  AuthenticationFailure,  // Authentication failure (probably bad password, but
+                          // best not to say for sure)
+  CtiCouldNotConnect,     // Could not connect to CTI service
+  InvalidServerSearchList
+};
+const std::string toString(LoginErrorStatus);
+}  // namespace LoginErrorStatusEnum
+
+namespace ErrorCodeEnum {
+enum ErrorCode { Ok, Unknown, InvalidState, InvalidArgument };
+const std::string toString(ErrorCode);
+}  // namespace ErrorCodeEnum
+
+}  // namespace csf
diff --git a/dom/media/webrtc/common/EncodingConstraints.h b/dom/media/webrtc/common/EncodingConstraints.h
new file mode 100644
index 0000000000..0bfa7f9f75
--- /dev/null
+++ b/dom/media/webrtc/common/EncodingConstraints.h
@@ -0,0 +1,58 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _ENCODING_CONSTRAINTS_H_
+#define _ENCODING_CONSTRAINTS_H_
+
+#include <algorithm>
+#include "mozilla/Maybe.h"
+
+namespace mozilla {
+class EncodingConstraints {
+ public:
+  EncodingConstraints()
+      : maxWidth(0),
+        maxHeight(0),
+        maxFs(0),
+        maxBr(0),
+        maxPps(0),
+        maxMbps(0),
+        maxCpb(0),
+        maxDpb(0),
+        scaleDownBy(1.0) {}
+
+  bool operator==(const EncodingConstraints& constraints) const {
+    return maxWidth == constraints.maxWidth &&
+           maxHeight == constraints.maxHeight && maxFps == constraints.maxFps &&
+           maxFs == constraints.maxFs && maxBr == constraints.maxBr &&
+           maxPps == constraints.maxPps && maxMbps == constraints.maxMbps &&
+           maxCpb == constraints.maxCpb && maxDpb == constraints.maxDpb &&
+           scaleDownBy == constraints.scaleDownBy;
+  }
+
+  /**
+   * This returns true if the constraints affecting resolution are equal.
+   */
+  bool ResolutionEquals(const EncodingConstraints& constraints) const {
+    return maxWidth == constraints.maxWidth &&
+           maxHeight == constraints.maxHeight && maxFs == constraints.maxFs &&
+           scaleDownBy == constraints.scaleDownBy;
+  }
+
+  uint32_t maxWidth;
+  uint32_t maxHeight;
+  Maybe<double> maxFps;
+  uint32_t maxFs;
+  uint32_t maxBr;
+  uint32_t maxPps;
+  uint32_t maxMbps;    // macroblocks per second
+  uint32_t maxCpb;     // coded picture buffer size
+  uint32_t maxDpb;     // decoded picture buffer size
+  double scaleDownBy;  // To preserve resolution
+};
+}  // namespace mozilla
+
+#endif  // _ENCODING_CONSTRAINTS_H_
diff --git a/dom/media/webrtc/common/MediaEngineWrapper.h b/dom/media/webrtc/common/MediaEngineWrapper.h
new file mode 100644
index 0000000000..ba48ad57d8
--- /dev/null
+++ b/dom/media/webrtc/common/MediaEngineWrapper.h
@@ -0,0 +1,32 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef MEDIA_ENGINE_WRAPPER_H_
+#define MEDIA_ENGINE_WRAPPER_H_
+
+#include <mozilla/Scoped.h>
+
+namespace mozilla {
+/**
+ * A Custom scoped template to release a resoure of Type T
+ * with a function of Type F
+ * ScopedCustomReleasePtr<webrtc::VoENetwork> ptr =
+ *               webrtc::VoENetwork->GetInterface(voiceEngine);
+ *
+ */
+template <typename T>
+struct ScopedCustomReleaseTraits0 {
+  typedef T* type;
+  static T* empty() { return nullptr; }
+  static void release(T* ptr) {
+    if (ptr) {
+      (ptr)->Release();
+    }
+  }
+};
+
+SCOPED_TEMPLATE(ScopedCustomReleasePtr, ScopedCustomReleaseTraits0)
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/common/NullDeleter.h b/dom/media/webrtc/common/NullDeleter.h
new file mode 100644
index 0000000000..76326c197e
--- /dev/null
+++ b/dom/media/webrtc/common/NullDeleter.h
@@ -0,0 +1,14 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#pragma once
+
+/*
+ * Helper class to allow smart pointers to stack objects to be constructed for
+ * ease of unit testing. Recycled here to help expose a shared_ptr interface to
+ * objects which are really raw pointers.
+ */
+struct null_deleter {
+  void operator()(void const*) const {}
+};
diff --git a/dom/media/webrtc/common/NullTransport.h b/dom/media/webrtc/common/NullTransport.h
new file mode 100644
index 0000000000..a9b270c6d6
--- /dev/null
+++ b/dom/media/webrtc/common/NullTransport.h
@@ -0,0 +1,56 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef NULL_TRANSPORT_H_
+#define NULL_TRANSPORT_H_
+
+#include "mozilla/Attributes.h"
+
+#include "api/call/transport.h"
+
+namespace mozilla {
+
+/**
+ * NullTransport is registered as ExternalTransport to throw away data
+ */
+class NullTransport : public webrtc::Transport {
+ public:
+  virtual bool SendRtp(const uint8_t* packet, size_t length,
+                       const webrtc::PacketOptions& options) override {
+    (void)packet;
+    (void)length;
+    (void)options;
+    return true;
+  }
+
+  virtual bool SendRtcp(const uint8_t* packet, size_t length) override {
+    (void)packet;
+    (void)length;
+    return true;
+  }
+#if 0
+  virtual int SendPacket(int channel, const void *data, size_t len)
+  {
+    (void) channel; (void) data;
+    return len;
+  }
+
+  virtual int SendRTCPPacket(int channel, const void *data, size_t len)
+  {
+    (void) channel; (void) data;
+    return len;
+  }
+#endif
+  NullTransport() {}
+
+  virtual ~NullTransport() {}
+
+ private:
+  NullTransport(const NullTransport& other) = delete;
+  void operator=(const NullTransport& other) = delete;
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/common/Wrapper.h b/dom/media/webrtc/common/Wrapper.h
new file mode 100644
index 0000000000..69c7406b07
--- /dev/null
+++ b/dom/media/webrtc/common/Wrapper.h
@@ -0,0 +1,157 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#pragma once
+
+/*
+ * Wrapper - Helper class for wrapper objects.
+ *
+ * This helps to construct a shared_ptr object which wraps access to an
+ * underlying handle. (The handle could be a pointer to some low-level type, a
+ * conventional C handle, an int ID, a GUID, etc.)
+ *
+ * Usage:
+ *   To obtain a FooPtr from a foo_handle_t, call
+ *     FooPtr Foo::wrap(foo_handle_t);
+ *
+ * To implement Foo using Wrapper, Foo needs to include this macro in its class
+ * definition:
+ *   CSF_DECLARE_WRAP(Foo, foo_handle_t);
+ * It also needs to include this in the cpp file, to provide the wrap()
+ * implementation and define the static Wrapper.
+ *   CSF_IMPLEMENT_WRAP(Foo, foo_handle_t);
+ * These are all declared in common/Wrapper.h - Foo.h needs to include this
+ * too.
+ * The client needs to declare Foo(foo_handle_t) as private, and provide a
+ * suitable implementation, as well as implementing wrappers for any other
+ * functions to be exposed.
+ * The client needs to implement ~Foo() to perform any cleanup as usual.
+ *
+ * wrap() will always return the same FooPtr for a given foo_handle_t, it will
+ * not construct additional objects if a suitable one already exists.
+ * changeHandle() is used in rare cases where the underlying handle is changed,
+ *                but the wrapper object is intended to remain.  This is the
+ *                case for the "fake" CC_DPCall generated on
+ *                CC_DPLine::CreateCall(), where the correct IDPCall* is
+ *                provided later.
+ * reset() is a cleanup step to wipe the handle map and allow memory to be
+ * reclaimed.
+ *
+ * Future enhancements:
+ * - For now, objects remain in the map forever.  Better would be to add a
+ *   releaseHandle() function which would allow the map to be emptied as
+ *   underlying handles expired.  While we can't force the client to give up
+ *   its shared_ptr<Foo> objects, we can remove our own copy, for instance on a
+ *   call ended event.
+ */
+
+#include <map>
+#include "prlock.h"
+#include "mozilla/Assertions.h"
+
+/*
+ * Wrapper has its own autolock class because the instances are declared
+ * statically and mozilla::Mutex will not work properly when instantiated
+ * in a static constructor.
+ */
+
+class LockNSPR {
+ public:
+  LockNSPR() : lock_(nullptr) {
+    lock_ = PR_NewLock();
+    MOZ_ASSERT(lock_);
+  }
+  ~LockNSPR() { PR_DestroyLock(lock_); }
+
+  void Acquire() { PR_Lock(lock_); }
+
+  void Release() { PR_Unlock(lock_); }
+
+ private:
+  PRLock* lock_;
+};
+
+class AutoLockNSPR {
+ public:
+  explicit AutoLockNSPR(LockNSPR& lock) : lock_(lock) { lock_.Acquire(); }
+  ~AutoLockNSPR() { lock_.Release(); }
+
+ private:
+  LockNSPR& lock_;
+};
+
+template <class T>
+class Wrapper {
+ private:
+  typedef std::map<typename T::Handle, typename T::Ptr> HandleMapType;
+  HandleMapType handleMap;
+  LockNSPR handleMapMutex;
+
+ public:
+  Wrapper() {}
+
+  typename T::Ptr wrap(typename T::Handle handle) {
+    AutoLockNSPR lock(handleMapMutex);
+    typename HandleMapType::iterator it = handleMap.find(handle);
+    if (it != handleMap.end()) {
+      return it->second;
+    } else {
+      typename T::Ptr p(new T(handle));
+      handleMap[handle] = p;
+      return p;
+    }
+  }
+
+  bool changeHandle(typename T::Handle oldHandle,
+                    typename T::Handle newHandle) {
+    AutoLockNSPR lock(handleMapMutex);
+    typename HandleMapType::iterator it = handleMap.find(oldHandle);
+    if (it != handleMap.end()) {
+      typename T::Ptr p = it->second;
+      handleMap.erase(it);
+      handleMap[newHandle] = p;
+      return true;
+    } else {
+      return false;
+    }
+  }
+
+  bool release(typename T::Handle handle) {
+    AutoLockNSPR lock(handleMapMutex);
+    typename HandleMapType::iterator it = handleMap.find(handle);
+    if (it != handleMap.end()) {
+      handleMap.erase(it);
+      return true;
+    } else {
+      return false;
+    }
+  }
+
+  void reset() {
+    AutoLockNSPR lock(handleMapMutex);
+    handleMap.clear();
+  }
+};
+
+#define CSF_DECLARE_WRAP(classname, handletype)  \
+ public:                                         \
+  static classname##Ptr wrap(handletype handle); \
+  static void reset();                           \
+  static void release(handletype handle);        \
+                                                 \
+ private:                                        \
+  friend class Wrapper<classname>;               \
+  typedef classname##Ptr Ptr;                    \
+  typedef handletype Handle;                     \
+  static Wrapper<classname>& getWrapper() {      \
+    static Wrapper<classname> wrapper;           \
+    return wrapper;                              \
+  }
+
+#define CSF_IMPLEMENT_WRAP(classname, handletype)     \
+  classname##Ptr classname::wrap(handletype handle) { \
+    return getWrapper().wrap(handle);                 \
+  }                                                   \
+  void classname::reset() { getWrapper().reset(); }   \
+  void classname::release(handletype handle) { getWrapper().release(handle); }
diff --git a/dom/media/webrtc/common/YuvStamper.cpp b/dom/media/webrtc/common/YuvStamper.cpp
new file mode 100644
index 0000000000..b0c8426983
--- /dev/null
+++ b/dom/media/webrtc/common/YuvStamper.cpp
@@ -0,0 +1,394 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifdef HAVE_NETINET_IN_H
+#  include <netinet/in.h>
+#elif defined XP_WIN
+#  include <winsock2.h>
+#endif
+#include <string.h>
+
+#include "nspr.h"
+#include "YuvStamper.h"
+#include "mozilla/Sprintf.h"
+
+typedef uint32_t UINT4;  // Needed for r_crc32() call
+extern "C" {
+#include "r_crc32.h"
+}
+
+namespace mozilla {
+
+#define ON_5 0x20
+#define ON_4 0x10
+#define ON_3 0x08
+#define ON_2 0x04
+#define ON_1 0x02
+#define ON_0 0x01
+
+/*
+  0, 0, 1, 1, 0, 0,
+  0, 1, 0, 0, 1, 0,
+  1, 0, 0, 0, 0, 1,
+  1, 0, 0, 0, 0, 1,
+  1, 0, 0, 0, 0, 1,
+  0, 1, 0, 0, 1, 0,
+  0, 0, 1, 1, 0, 0
+*/
+static unsigned char DIGIT_0[] = {ON_3 | ON_2, ON_4 | ON_1, ON_5 | ON_0,
+                                  ON_5 | ON_0, ON_5 | ON_0, ON_4 | ON_1,
+                                  ON_3 | ON_2};
+
+/*
+  0, 0, 0, 1, 0, 0,
+  0, 0, 0, 1, 0, 0,
+  0, 0, 0, 1, 0, 0,
+  0, 0, 0, 1, 0, 0,
+  0, 0, 0, 1, 0, 0,
+  0, 0, 0, 1, 0, 0,
+  0, 0, 0, 1, 0, 0,
+*/
+static unsigned char DIGIT_1[] = {ON_2, ON_2, ON_2, ON_2, ON_2, ON_2, ON_2};
+
+/*
+  1, 1, 1, 1, 1, 0,
+  0, 0, 0, 0, 0, 1,
+  0, 0, 0, 0, 0, 1,
+  0, 1, 1, 1, 1, 0,
+  1, 0, 0, 0, 0, 0,
+  1, 0, 0, 0, 0, 0,
+  0, 1, 1, 1, 1, 1,
+*/
+static unsigned char DIGIT_2[] = {
+    ON_5 | ON_4 | ON_3 | ON_2 | ON_1, ON_0, ON_0,
+    ON_4 | ON_3 | ON_2 | ON_1,        ON_5, ON_5,
+    ON_4 | ON_3 | ON_2 | ON_1 | ON_0,
+};
+
+/*
+  1, 1, 1, 1, 1, 0,
+  0, 0, 0, 0, 0, 1,
+  0, 0, 0, 0, 0, 1,
+  0, 1, 1, 1, 1, 1,
+  0, 0, 0, 0, 0, 1,
+  0, 0, 0, 0, 0, 1,
+  1, 1, 1, 1, 1, 0,
+*/
+static unsigned char DIGIT_3[] = {
+    ON_5 | ON_4 | ON_3 | ON_2 | ON_1, ON_0, ON_0,
+    ON_4 | ON_3 | ON_2 | ON_1 | ON_0, ON_0, ON_0,
+    ON_5 | ON_4 | ON_3 | ON_2 | ON_1,
+};
+
+/*
+  0, 1, 0, 0, 0, 1,
+  0, 1, 0, 0, 0, 1,
+  0, 1, 0, 0, 0, 1,
+  0, 1, 1, 1, 1, 1,
+  0, 0, 0, 0, 0, 1,
+  0, 0, 0, 0, 0, 1,
+  0, 0, 0, 0, 0, 1
+*/
+static unsigned char DIGIT_4[] = {
+    ON_4 | ON_0, ON_4 | ON_0, ON_4 | ON_0, ON_4 | ON_3 | ON_2 | ON_1 | ON_0,
+    ON_0,        ON_0,        ON_0,
+};
+
+/*
+  0, 1, 1, 1, 1, 1,
+  1, 0, 0, 0, 0, 0,
+  1, 0, 0, 0, 0, 0,
+  0, 1, 1, 1, 1, 0,
+  0, 0, 0, 0, 0, 1,
+  0, 0, 0, 0, 0, 1,
+  1, 1, 1, 1, 1, 0,
+*/
+static unsigned char DIGIT_5[] = {
+    ON_4 | ON_3 | ON_2 | ON_1 | ON_0, ON_5, ON_5,
+    ON_4 | ON_3 | ON_2 | ON_1,        ON_0, ON_0,
+    ON_5 | ON_4 | ON_3 | ON_2 | ON_1,
+};
+
+/*
+  0, 1, 1, 1, 1, 1,
+  1, 0, 0, 0, 0, 0,
+  1, 0, 0, 0, 0, 0,
+  1, 1, 1, 1, 1, 0,
+  1, 0, 0, 0, 0, 1,
+  1, 0, 0, 0, 0, 1,
+  0, 1, 1, 1, 1, 0,
+*/
+static unsigned char DIGIT_6[] = {
+    ON_4 | ON_3 | ON_2 | ON_1 | ON_0, ON_5,        ON_5,
+    ON_4 | ON_3 | ON_2 | ON_1,        ON_5 | ON_0, ON_5 | ON_0,
+    ON_4 | ON_3 | ON_2 | ON_1,
+};
+
+/*
+  1, 1, 1, 1, 1, 1,
+  0, 0, 0, 0, 0, 1,
+  0, 0, 0, 0, 1, 0,
+  0, 0, 0, 1, 0, 0,
+  0, 0, 1, 0, 0, 0,
+  0, 1, 0, 0, 0, 0,
+  1, 0, 0, 0, 0, 0
+*/
+static unsigned char DIGIT_7[] = {ON_5 | ON_4 | ON_3 | ON_2 | ON_1 | ON_0,
+                                  ON_0,
+                                  ON_1,
+                                  ON_2,
+                                  ON_3,
+                                  ON_4,
+                                  ON_5};
+
+/*
+  0, 1, 1, 1, 1, 1,
+  1, 0, 0, 0, 0, 1,
+  1, 0, 0, 0, 0, 1,
+  0, 1, 1, 1, 1, 0,
+  1, 0, 0, 0, 0, 1,
+  1, 0, 0, 0, 0, 1,
+  0, 1, 1, 1, 1, 0
+*/
+static unsigned char DIGIT_8[] = {
+    ON_4 | ON_3 | ON_2 | ON_1, ON_5 | ON_0, ON_5 | ON_0,
+    ON_4 | ON_3 | ON_2 | ON_1, ON_5 | ON_0, ON_5 | ON_0,
+    ON_4 | ON_3 | ON_2 | ON_1,
+};
+
+/*
+  0, 1, 1, 1, 1, 1,
+  1, 0, 0, 0, 0, 1,
+  1, 0, 0, 0, 0, 1,
+  0, 1, 1, 1, 1, 1,
+  0, 0, 0, 0, 0, 1,
+  0, 0, 0, 0, 0, 1,
+  0, 1, 1, 1, 1, 0
+*/
+static unsigned char DIGIT_9[] = {
+    ON_4 | ON_3 | ON_2 | ON_1 | ON_0, ON_5 | ON_0, ON_5 | ON_0,
+    ON_4 | ON_3 | ON_2 | ON_1 | ON_0, ON_0,        ON_0,
+    ON_4 | ON_3 | ON_2 | ON_1,
+};
+
+static unsigned char* DIGITS[] = {DIGIT_0, DIGIT_1, DIGIT_2, DIGIT_3, DIGIT_4,
+                                  DIGIT_5, DIGIT_6, DIGIT_7, DIGIT_8, DIGIT_9};
+
+YuvStamper::YuvStamper(unsigned char* pYData, uint32_t width, uint32_t height,
+                       uint32_t stride, uint32_t x, uint32_t y,
+                       unsigned char symbol_width, unsigned char symbol_height)
+    : pYData(pYData),
+      mStride(stride),
+      mWidth(width),
+      mHeight(height),
+      mSymbolWidth(symbol_width),
+      mSymbolHeight(symbol_height),
+      mCursor(x, y) {}
+
+bool YuvStamper::Encode(uint32_t width, uint32_t height, uint32_t stride,
+                        unsigned char* pYData, unsigned char* pMsg,
+                        size_t msg_len, uint32_t x, uint32_t y) {
+  YuvStamper stamper(pYData, width, height, stride, x, y, sBitSize, sBitSize);
+
+  // Reserve space for a checksum.
+  if (stamper.Capacity() < 8 * (msg_len + sizeof(uint32_t))) {
+    return false;
+  }
+
+  bool ok = false;
+  uint32_t crc;
+  unsigned char* pCrc = reinterpret_cast<unsigned char*>(&crc);
+  r_crc32(reinterpret_cast<char*>(pMsg), (int)msg_len, &crc);
+  crc = htonl(crc);
+
+  while (msg_len-- > 0) {
+    if (!stamper.Write8(*pMsg++)) {
+      return false;
+    }
+  }
+
+  // Add checksum after the message.
+  ok = stamper.Write8(*pCrc++) && stamper.Write8(*pCrc++) &&
+       stamper.Write8(*pCrc++) && stamper.Write8(*pCrc++);
+
+  return ok;
+}
+
+bool YuvStamper::Decode(uint32_t width, uint32_t height, uint32_t stride,
+                        unsigned char* pYData, unsigned char* pMsg,
+                        size_t msg_len, uint32_t x, uint32_t y) {
+  YuvStamper stamper(pYData, width, height, stride, x, y, sBitSize, sBitSize);
+
+  unsigned char* ptr = pMsg;
+  size_t len = msg_len;
+  uint32_t crc, msg_crc;
+  unsigned char* pCrc = reinterpret_cast<unsigned char*>(&crc);
+
+  // Account for space reserved for the checksum
+  if (stamper.Capacity() < 8 * (len + sizeof(uint32_t))) {
+    return false;
+  }
+
+  while (len-- > 0) {
+    if (!stamper.Read8(*ptr++)) {
+      return false;
+    }
+  }
+
+  if (!(stamper.Read8(*pCrc++) && stamper.Read8(*pCrc++) &&
+        stamper.Read8(*pCrc++) && stamper.Read8(*pCrc++))) {
+    return false;
+  }
+
+  r_crc32(reinterpret_cast<char*>(pMsg), (int)msg_len, &msg_crc);
+  return crc == htonl(msg_crc);
+}
+
+inline uint32_t YuvStamper::Capacity() {
+  // Enforce at least a symbol width and height offset from outer edges.
+  if (mCursor.y + mSymbolHeight > mHeight) {
+    return 0;
+  }
+
+  if (mCursor.x + mSymbolWidth > mWidth && !AdvanceCursor()) {
+    return 0;
+  }
+
+  // Normalize frame integral to mSymbolWidth x mSymbolHeight
+  uint32_t width = mWidth / mSymbolWidth;
+  uint32_t height = mHeight / mSymbolHeight;
+  uint32_t x = mCursor.x / mSymbolWidth;
+  uint32_t y = mCursor.y / mSymbolHeight;
+
+  return (width * height - width * y) - x;
+}
+
+bool YuvStamper::Write8(unsigned char value) {
+  // Encode MSB to LSB.
+  unsigned char mask = 0x80;
+  while (mask) {
+    if (!WriteBit(!!(value & mask))) {
+      return false;
+    }
+    mask >>= 1;
+  }
+  return true;
+}
+
+bool YuvStamper::WriteBit(bool one) {
+  // A bit is mapped to a mSymbolWidth x mSymbolHeight square of luma data
+  // points. Don't use ternary op.:
+  // https://bugzilla.mozilla.org/show_bug.cgi?id=1001708
+  unsigned char value;
+  if (one)
+    value = sYOn;
+  else
+    value = sYOff;
+
+  for (uint32_t y = 0; y < mSymbolHeight; y++) {
+    for (uint32_t x = 0; x < mSymbolWidth; x++) {
+      *(pYData + (mCursor.x + x) + ((mCursor.y + y) * mStride)) = value;
+    }
+  }
+
+  return AdvanceCursor();
+}
+
+bool YuvStamper::AdvanceCursor() {
+  mCursor.x += mSymbolWidth;
+  if (mCursor.x + mSymbolWidth > mWidth) {
+    // move to the start of the next row if possible.
+    mCursor.y += mSymbolHeight;
+    if (mCursor.y + mSymbolHeight > mHeight) {
+      // end of frame, do not advance
+      mCursor.y -= mSymbolHeight;
+      mCursor.x -= mSymbolWidth;
+      return false;
+    } else {
+      mCursor.x = 0;
+    }
+  }
+
+  return true;
+}
+
+bool YuvStamper::Read8(unsigned char& value) {
+  unsigned char octet = 0;
+  unsigned char bit = 0;
+
+  for (int i = 8; i > 0; --i) {
+    if (!ReadBit(bit)) {
+      return false;
+    }
+    octet <<= 1;
+    octet |= bit;
+  }
+
+  value = octet;
+  return true;
+}
+
+bool YuvStamper::ReadBit(unsigned char& bit) {
+  uint32_t sum = 0;
+  for (uint32_t y = 0; y < mSymbolHeight; y++) {
+    for (uint32_t x = 0; x < mSymbolWidth; x++) {
+      sum += *(pYData + mStride * (mCursor.y + y) + mCursor.x + x);
+    }
+  }
+
+  // apply threshold to collected bit square
+  bit = (sum > (sBitThreshold * mSymbolWidth * mSymbolHeight)) ? 1 : 0;
+  return AdvanceCursor();
+}
+
+bool YuvStamper::WriteDigits(uint32_t value) {
+  char buf[20];
+  SprintfLiteral(buf, "%.5u", value);
+  size_t size = strlen(buf);
+
+  if (Capacity() < size) {
+    return false;
+  }
+
+  for (size_t i = 0; i < size; ++i) {
+    if (!WriteDigit(buf[i] - '0')) return false;
+    if (!AdvanceCursor()) {
+      return false;
+    }
+  }
+
+  return true;
+}
+
+bool YuvStamper::WriteDigit(unsigned char digit) {
+  if (digit > sizeof(DIGITS) / sizeof(DIGITS[0])) return false;
+
+  unsigned char* dig = DIGITS[digit];
+  for (uint32_t row = 0; row < sDigitHeight; ++row) {
+    unsigned char mask = 0x01 << (sDigitWidth - 1);
+    for (uint32_t col = 0; col < sDigitWidth; ++col, mask >>= 1) {
+      if (dig[row] & mask) {
+        for (uint32_t xx = 0; xx < sPixelSize; ++xx) {
+          for (uint32_t yy = 0; yy < sPixelSize; ++yy) {
+            WritePixel(pYData, mCursor.x + (col * sPixelSize) + xx,
+                       mCursor.y + (row * sPixelSize) + yy);
+          }
+        }
+      }
+    }
+  }
+
+  return true;
+}
+
+void YuvStamper::WritePixel(unsigned char* data, uint32_t x, uint32_t y) {
+  unsigned char* ptr = &data[y * mStride + x];
+  // Don't use ternary op.: https://bugzilla.mozilla.org/show_bug.cgi?id=1001708
+  if (*ptr > sLumaThreshold)
+    *ptr = sLumaMin;
+  else
+    *ptr = sLumaMax;
+}
+
+}  // namespace mozilla.
diff --git a/dom/media/webrtc/common/YuvStamper.h b/dom/media/webrtc/common/YuvStamper.h
new file mode 100644
index 0000000000..f355055cd9
--- /dev/null
+++ b/dom/media/webrtc/common/YuvStamper.h
@@ -0,0 +1,77 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef YUV_STAMPER_H_
+#define YUV_STAMPER_H_
+
+#include <cstdint>
+
+namespace mozilla {
+
+class YuvStamper {
+ public:
+  bool WriteDigits(uint32_t value);
+
+  template <typename T>
+  static bool Write(uint32_t width, uint32_t height, uint32_t stride,
+                    unsigned char* pYData, const T& value, uint32_t x = 0,
+                    uint32_t y = 0) {
+    YuvStamper stamper(pYData, width, height, stride, x, y,
+                       (sDigitWidth + sInterDigit) * sPixelSize,
+                       (sDigitHeight + sInterLine) * sPixelSize);
+    return stamper.WriteDigits(value);
+  }
+
+  static bool Encode(uint32_t width, uint32_t height, uint32_t stride,
+                     unsigned char* pYData, unsigned char* pMsg, size_t msg_len,
+                     uint32_t x = 0, uint32_t y = 0);
+
+  static bool Decode(uint32_t width, uint32_t height, uint32_t stride,
+                     unsigned char* pYData, unsigned char* pMsg, size_t msg_len,
+                     uint32_t x = 0, uint32_t y = 0);
+
+ private:
+  YuvStamper(unsigned char* pYData, uint32_t width, uint32_t height,
+             uint32_t stride, uint32_t x, uint32_t y,
+             unsigned char symbol_width, unsigned char symbol_height);
+
+  bool WriteDigit(unsigned char digit);
+  void WritePixel(unsigned char* data, uint32_t x, uint32_t y);
+  uint32_t Capacity();
+  bool AdvanceCursor();
+  bool WriteBit(bool one);
+  bool Write8(unsigned char value);
+  bool ReadBit(unsigned char& value);
+  bool Read8(unsigned char& bit);
+
+  const static unsigned char sPixelSize = 3;
+  const static unsigned char sDigitWidth = 6;
+  const static unsigned char sDigitHeight = 7;
+  const static unsigned char sInterDigit = 1;
+  const static unsigned char sInterLine = 1;
+  const static uint32_t sBitSize = 4;
+  const static uint32_t sBitThreshold = 60;
+  const static unsigned char sYOn = 0x80;
+  const static unsigned char sYOff = 0;
+  const static unsigned char sLumaThreshold = 96;
+  const static unsigned char sLumaMin = 16;
+  const static unsigned char sLumaMax = 235;
+
+  unsigned char* pYData;
+  uint32_t mStride;
+  uint32_t mWidth;
+  uint32_t mHeight;
+  unsigned char mSymbolWidth;
+  unsigned char mSymbolHeight;
+
+  struct Cursor {
+    Cursor(uint32_t x, uint32_t y) : x(x), y(y) {}
+    uint32_t x;
+    uint32_t y;
+  } mCursor;
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/common/browser_logging/CSFLog.cpp b/dom/media/webrtc/common/browser_logging/CSFLog.cpp
new file mode 100644
index 0000000000..d58f5726a5
--- /dev/null
+++ b/dom/media/webrtc/common/browser_logging/CSFLog.cpp
@@ -0,0 +1,85 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include <stdio.h>
+#include <string.h>
+#include <stdarg.h>
+
+#include "CSFLog.h"
+
+#include <map>
+#include "prrwlock.h"
+#include "prthread.h"
+#include "nsThreadUtils.h"
+
+#include "mozilla/Logging.h"
+#include "mozilla/Sprintf.h"
+
+mozilla::LazyLogModule gSignalingLog("signaling");
+
+void CSFLogV(CSFLogLevel priority, const char* sourceFile, int sourceLine,
+             const char* tag, const char* format, va_list args) {
+#ifdef STDOUT_LOGGING
+  printf("%s\n:", tag);
+  vprintf(format, args);
+#else
+
+  mozilla::LogLevel level =
+      static_cast<mozilla::LogLevel>(static_cast<unsigned int>(priority));
+
+  // Skip doing any of this work if we're not logging the indicated level...
+  if (!MOZ_LOG_TEST(gSignalingLog, level)) {
+    return;
+  }
+
+  // Trim the path component from the filename
+  const char* lastSlash = sourceFile;
+  while (*sourceFile) {
+    if (*sourceFile == '/' || *sourceFile == '\\') {
+      lastSlash = sourceFile;
+    }
+    sourceFile++;
+  }
+  sourceFile = lastSlash;
+  if (*sourceFile == '/' || *sourceFile == '\\') {
+    sourceFile++;
+  }
+
+#  define MAX_MESSAGE_LENGTH 1024
+  char message[MAX_MESSAGE_LENGTH];
+
+  const char* threadName = NULL;
+
+  // Check if we're the main thread...
+  if (NS_IsMainThread()) {
+    threadName = "main";
+  } else {
+    threadName = PR_GetThreadName(PR_GetCurrentThread());
+  }
+
+  // If we can't find it anywhere, use a blank string
+  if (!threadName) {
+    threadName = "";
+  }
+
+  VsprintfLiteral(message, format, args);
+  MOZ_LOG(
+      gSignalingLog, level,
+      ("[%s|%s] %s:%d: %s", threadName, tag, sourceFile, sourceLine, message));
+#endif
+}
+
+void CSFLog(CSFLogLevel priority, const char* sourceFile, int sourceLine,
+            const char* tag, const char* format, ...) {
+  va_list ap;
+  va_start(ap, format);
+
+  CSFLogV(priority, sourceFile, sourceLine, tag, format, ap);
+  va_end(ap);
+}
+
+int CSFLogTestLevel(CSFLogLevel priority) {
+  return MOZ_LOG_TEST(gSignalingLog, static_cast<mozilla::LogLevel>(
+                                         static_cast<unsigned int>(priority)));
+}
diff --git a/dom/media/webrtc/common/browser_logging/CSFLog.h b/dom/media/webrtc/common/browser_logging/CSFLog.h
new file mode 100644
index 0000000000..eb46b37cc3
--- /dev/null
+++ b/dom/media/webrtc/common/browser_logging/CSFLog.h
@@ -0,0 +1,58 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef CSFLOG_H
+#define CSFLOG_H
+
+#include <stdarg.h>
+
+typedef enum {
+  CSF_LOG_ERROR = 1,
+  CSF_LOG_WARNING,
+  CSF_LOG_INFO,
+  CSF_LOG_DEBUG,
+  CSF_LOG_VERBOSE,
+} CSFLogLevel;
+
+#define CSFLogError(tag, format, ...) \
+  CSFLog(CSF_LOG_ERROR, __FILE__, __LINE__, tag, format, ##__VA_ARGS__)
+#define CSFLogErrorV(tag, format, va_list_arg) \
+  CSFLogV(CSF_LOG_ERROR, __FILE__, __LINE__, tag, format, va_list_arg)
+#define CSFLogWarn(tag, format, ...) \
+  CSFLog(CSF_LOG_WARNING, __FILE__, __LINE__, tag, format, ##__VA_ARGS__)
+#define CSFLogWarnV(tag, format, va_list_arg) \
+  CSFLogV(CSF_LOG_WARNING, __FILE__, __LINE__, tag, format, va_list_arg)
+#define CSFLogInfo(tag, format, ...) \
+  CSFLog(CSF_LOG_INFO, __FILE__, __LINE__, tag, format, ##__VA_ARGS__)
+#define CSFLogInfoV(tag, format, va_list_arg) \
+  CSFLogV(CSF_LOG_INFO, __FILE__, __LINE__, tag, format, va_list_arg)
+#define CSFLogDebug(tag, format, ...) \
+  CSFLog(CSF_LOG_DEBUG, __FILE__, __LINE__, tag, format, ##__VA_ARGS__)
+#define CSFLogDebugV(tag, format, va_list_arg) \
+  CSFLogV(CSF_LOG_DEBUG, __FILE__, __LINE__, tag, format, va_list_arg)
+#define CSFLogVerbose(tag, format, ...) \
+  CSFLog(CSF_LOG_VERBOSE, __FILE__, __LINE__, tag, format, ##__VA_ARGS__)
+#define CSFLogVerboseV(tag, format, va_list_arg) \
+  CSFLogV(CSF_LOG_VERBOSE, __FILE__, __LINE__, tag, format, va_list_arg)
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+void CSFLog(CSFLogLevel priority, const char* sourceFile, int sourceLine,
+            const char* tag, const char* format, ...)
+#ifdef __GNUC__
+    __attribute__((format(printf, 5, 6)))
+#endif
+    ;
+
+void CSFLogV(CSFLogLevel priority, const char* sourceFile, int sourceLine,
+             const char* tag, const char* format, va_list args);
+
+int CSFLogTestLevel(CSFLogLevel priority);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif
diff --git a/dom/media/webrtc/common/browser_logging/WebRtcLog.cpp b/dom/media/webrtc/common/browser_logging/WebRtcLog.cpp
new file mode 100644
index 0000000000..0f3769604e
--- /dev/null
+++ b/dom/media/webrtc/common/browser_logging/WebRtcLog.cpp
@@ -0,0 +1,162 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "WebRtcLog.h"
+
+#include "mozilla/Logging.h"
+#include "mozilla/StaticPtr.h"
+#include "prenv.h"
+#include "rtc_base/logging.h"
+
+#include "nscore.h"
+#include "nsString.h"
+#include "nsXULAppAPI.h"
+#include "mozilla/Preferences.h"
+
+#include "nsIFile.h"
+#include "nsDirectoryServiceUtils.h"
+#include "nsDirectoryServiceDefs.h"
+#include "nsNativeCharsetUtils.h"
+
+using mozilla::LogLevel;
+
+static mozilla::LazyLogModule sWebRtcLog("webrtc_trace");
+static mozilla::LazyLogModule sLogAEC("AEC");
+
+class LogSinkImpl : public rtc::LogSink {
+ public:
+  LogSinkImpl() {}
+
+ private:
+  void OnLogMessage(const std::string& message) override {
+    MOZ_LOG(sWebRtcLog, LogLevel::Debug, ("%s", message.data()));
+  }
+};
+
+// For RTC_LOG()
+static mozilla::StaticAutoPtr<LogSinkImpl> sSink;
+
+void GetWebRtcLogPrefs() {
+  rtc::LogMessage::set_aec_debug_size(
+      mozilla::Preferences::GetUint("media.webrtc.debug.aec_dump_max_size"));
+}
+
+mozilla::LogLevel CheckOverrides() {
+  mozilla::LogModule* log_info = sWebRtcLog;
+  mozilla::LogLevel log_level = log_info->Level();
+
+  log_info = sLogAEC;
+  if (sLogAEC && (log_info->Level() != mozilla::LogLevel::Disabled)) {
+    rtc::LogMessage::set_aec_debug(true);
+  }
+
+  return log_level;
+}
+
+void ConfigWebRtcLog(mozilla::LogLevel level) {
+  rtc::LoggingSeverity log_level;
+  switch (level) {
+    case mozilla::LogLevel::Verbose:
+      log_level = rtc::LoggingSeverity::LS_VERBOSE;
+      break;
+    case mozilla::LogLevel::Debug:
+    case mozilla::LogLevel::Info:
+      log_level = rtc::LoggingSeverity::LS_INFO;
+      break;
+    case mozilla::LogLevel::Warning:
+      log_level = rtc::LoggingSeverity::LS_WARNING;
+      break;
+    case mozilla::LogLevel::Error:
+      log_level = rtc::LoggingSeverity::LS_ERROR;
+      break;
+    case mozilla::LogLevel::Disabled:
+      log_level = rtc::LoggingSeverity::LS_NONE;
+      break;
+    default:
+      MOZ_ASSERT(false);
+      break;
+  }
+  rtc::LogMessage::LogToDebug(log_level);
+  if (level != mozilla::LogLevel::Disabled) {
+    // always capture LOG(...) << ... logging in webrtc.org code to nspr logs
+    if (!sSink) {
+      sSink = new LogSinkImpl();
+      rtc::LogMessage::AddLogToStream(sSink, log_level);
+      // it's ok if this leaks to program end
+    }
+  } else if (sSink) {
+    rtc::LogMessage::RemoveLogToStream(sSink);
+    sSink = nullptr;
+  }
+}
+
+void StartWebRtcLog(mozilla::LogLevel log_level) {
+  if (log_level == mozilla::LogLevel::Disabled) {
+    return;
+  }
+
+  GetWebRtcLogPrefs();
+  mozilla::LogLevel level = CheckOverrides();
+
+  ConfigWebRtcLog(level);
+}
+
+void EnableWebRtcLog() {
+  GetWebRtcLogPrefs();
+  mozilla::LogLevel level = CheckOverrides();
+  ConfigWebRtcLog(level);
+}
+
+// Called when we destroy the singletons from PeerConnectionCtx or if the
+// user changes logging in about:webrtc
+void StopWebRtcLog() {
+  if (sSink) {
+    rtc::LogMessage::RemoveLogToStream(sSink);
+    sSink = nullptr;
+  }
+}
+
+nsCString ConfigAecLog() {
+  nsCString aecLogDir;
+  if (rtc::LogMessage::aec_debug()) {
+    return ""_ns;
+  }
+#if defined(ANDROID)
+  const char* default_tmp_dir = "/dev/null";
+  aecLogDir.Assign(default_tmp_dir);
+#else
+  nsCOMPtr<nsIFile> tempDir;
+  nsresult rv = NS_GetSpecialDirectory(NS_OS_TEMP_DIR, getter_AddRefs(tempDir));
+  if (NS_SUCCEEDED(rv)) {
+#  ifdef XP_WIN
+    // WebRTC wants a path encoded in the native charset, not UTF-8.
+    nsAutoString temp;
+    tempDir->GetPath(temp);
+    NS_CopyUnicodeToNative(temp, aecLogDir);
+#  else
+    tempDir->GetNativePath(aecLogDir);
+#  endif
+  }
+#endif
+  rtc::LogMessage::set_aec_debug_filename(aecLogDir.get());
+
+  return aecLogDir;
+}
+
+nsCString StartAecLog() {
+  nsCString aecLogDir;
+  if (rtc::LogMessage::aec_debug()) {
+    return ""_ns;
+  }
+
+  GetWebRtcLogPrefs();
+  CheckOverrides();
+  aecLogDir = ConfigAecLog();
+
+  rtc::LogMessage::set_aec_debug(true);
+
+  return aecLogDir;
+}
+
+void StopAecLog() { rtc::LogMessage::set_aec_debug(false); }
diff --git a/dom/media/webrtc/common/browser_logging/WebRtcLog.h b/dom/media/webrtc/common/browser_logging/WebRtcLog.h
new file mode 100644
index 0000000000..b933ff43a5
--- /dev/null
+++ b/dom/media/webrtc/common/browser_logging/WebRtcLog.h
@@ -0,0 +1,17 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef WEBRTCLOG_H_
+#define WEBRTCLOG_H_
+
+#include "mozilla/Logging.h"
+#include "nsStringFwd.h"
+
+nsCString StartAecLog();
+void StopAecLog();
+void EnableWebRtcLog();
+void StartWebRtcLog(mozilla::LogLevel level);
+void StopWebRtcLog();
+
+#endif
diff --git a/dom/media/webrtc/common/csf_common.h b/dom/media/webrtc/common/csf_common.h
new file mode 100644
index 0000000000..8b01047553
--- /dev/null
+++ b/dom/media/webrtc/common/csf_common.h
@@ -0,0 +1,90 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _CSF_COMMON_E58E5677_950A_424c_B6C2_CA180092E6A2_H
+#define _CSF_COMMON_E58E5677_950A_424c_B6C2_CA180092E6A2_H
+
+#include <assert.h>
+#include <memory>
+#include <vector>
+#include <stdlib.h>
+
+/*
+
+This header file defines:
+
+csf_countof
+csf_sprintf
+csf_vsprintf
+
+*/
+
+/*
+  General security tip: Ensure that "format" is never a user-defined string.
+  Format should ALWAYS be something that's built into your code, not user
+  supplied. For example: never write:
+
+  csf_sprintf(buffer, csf_countof(buffer), pUserSuppliedString);
+
+  Instead write:
+
+  csf_sprintf(buffer, csf_countof(buffer), "%s", pUserSuppliedString);
+
+*/
+
+#ifdef WIN32
+#  if !defined(_countof)
+#    if !defined(__cplusplus)
+#      define _countof(_Array) (sizeof(_Array) / sizeof(_Array[0]))
+#    else
+extern "C++" {
+template <typename _CountofType, size_t _SizeOfArray>
+char (*_csf_countof_helper(_CountofType (&_Array)[_SizeOfArray]))[_SizeOfArray];
+#      define _countof(_Array) sizeof(*_csf_countof_helper(_Array))
+}
+#    endif
+#  endif
+#else
+#  define _countof(_Array) (sizeof(_Array) / sizeof(_Array[0]))
+#endif
+// csf_countof
+
+#define csf_countof(anArray) _countof(anArray)
+
+// csf_sprintf
+
+#ifdef _WIN32
+// Unlike snprintf, sprintf_s guarantees that the buffer will be null-terminated
+// (unless the buffer size is zero).
+#  define csf_sprintf(/* char* */ buffer,                                   \
+                      /* size_t */ sizeOfBufferInCharsInclNullTerm,         \
+                      /* const char * */ format, ...)                       \
+    _snprintf_s(buffer, sizeOfBufferInCharsInclNullTerm, _TRUNCATE, format, \
+                __VA_ARGS__)
+#else
+#  define csf_sprintf(/* char */ buffer,                                    \
+                      /* size_t */ sizeOfBufferInCharsInclNullTerm,         \
+                      /* const char * */ format, ...)                       \
+    snprintf(buffer, sizeOfBufferInCharsInclNullTerm, format, __VA_ARGS__); \
+    buffer[sizeOfBufferInCharsInclNullTerm - 1] = '\0'
+#endif
+
+// csf_vsprintf
+
+#ifdef _WIN32
+#  define csf_vsprintf(/* char* */ buffer,                                  \
+                       /* size_t */ sizeOfBufferInCharsInclNullTerm,        \
+                       /* const char * */ format, /* va_list */ vaList)     \
+    vsnprintf_s(buffer, sizeOfBufferInCharsInclNullTerm, _TRUNCATE, format, \
+                vaList);                                                    \
+    buffer[sizeOfBufferInCharsInclNullTerm - 1] = '\0'
+#else
+#  define csf_vsprintf(/* char */ buffer,                               \
+                       /* size_t */ sizeOfBufferInCharsInclNullTerm,    \
+                       /* const char * */ format, /* va_list */ vaList) \
+    vsprintf(buffer, format, vaList);                                   \
+    buffer[sizeOfBufferInCharsInclNullTerm - 1] = '\0'
+#endif
+
+#endif
diff --git a/dom/media/webrtc/common/moz.build b/dom/media/webrtc/common/moz.build
new file mode 100644
index 0000000000..7605d27835
--- /dev/null
+++ b/dom/media/webrtc/common/moz.build
@@ -0,0 +1,23 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+include("/dom/media/webrtc/third_party_build/webrtc.mozbuild")
+
+EXPORTS.mozilla.dom += ["CandidateInfo.h"]
+
+LOCAL_INCLUDES += [
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr",
+    "/third_party/libwebrtc",
+    "/third_party/libwebrtc/third_party/abseil-cpp",
+]
+
+UNIFIED_SOURCES += [
+    "browser_logging/CSFLog.cpp",
+    "browser_logging/WebRtcLog.cpp",
+    "time_profiling/timecard.c",
+    "YuvStamper.cpp",
+]
+
+FINAL_LIBRARY = "xul"
diff --git a/dom/media/webrtc/common/time_profiling/timecard.c b/dom/media/webrtc/common/time_profiling/timecard.c
new file mode 100644
index 0000000000..c0218cb92d
--- /dev/null
+++ b/dom/media/webrtc/common/time_profiling/timecard.c
@@ -0,0 +1,112 @@
+/* -*- Mode: C; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include <stdio.h>
+#include <string.h>
+#include "timecard.h"
+#include "mozilla/mozalloc.h"
+
+Timecard* create_timecard() {
+  Timecard* tc = moz_xcalloc(1, sizeof(Timecard));
+  tc->entries_allocated = TIMECARD_INITIAL_TABLE_SIZE;
+  tc->entries = moz_xcalloc(tc->entries_allocated, sizeof(TimecardEntry));
+  tc->start_time = PR_Now();
+  return tc;
+}
+
+void destroy_timecard(Timecard* tc) {
+  free(tc->entries);
+  free(tc);
+}
+
+void stamp_timecard(Timecard* tc, const char* event, const char* file,
+                    unsigned int line, const char* function) {
+  TimecardEntry* entry = NULL;
+
+  /* Trim the path component from the filename */
+  const char* last_slash = file;
+  while (*file) {
+    if (*file == '/' || *file == '\\') {
+      last_slash = file;
+    }
+    file++;
+  }
+  file = last_slash;
+  if (*file == '/' || *file == '\\') {
+    file++;
+  }
+
+  /* Ensure there is enough space left in the entries list */
+  if (tc->curr_entry == tc->entries_allocated) {
+    tc->entries_allocated *= 2;
+    tc->entries = moz_xrealloc(tc->entries,
+                               tc->entries_allocated * sizeof(TimecardEntry));
+  }
+
+  /* Record the data into the timecard entry */
+  entry = &tc->entries[tc->curr_entry];
+  entry->timestamp = PR_Now();
+  entry->event = event;
+  entry->file = file;
+  entry->line = line;
+  entry->function = function;
+  tc->curr_entry++;
+}
+
+void print_timecard(Timecard* tc) {
+  size_t i;
+  TimecardEntry* entry;
+  size_t event_width = 5;
+  size_t file_width = 4;
+  size_t function_width = 8;
+  size_t line_width;
+  PRTime offset, delta;
+
+  for (i = 0; i < tc->curr_entry; i++) {
+    entry = &tc->entries[i];
+    if (strlen(entry->event) > event_width) {
+      event_width = strlen(entry->event);
+    }
+    if (strlen(entry->file) > file_width) {
+      file_width = strlen(entry->file);
+    }
+    if (strlen(entry->function) > function_width) {
+      function_width = strlen(entry->function);
+    }
+  }
+
+  printf("\nTimecard created %4ld.%6.6ld\n\n",
+         (long)(tc->start_time / PR_USEC_PER_SEC),
+         (long)(tc->start_time % PR_USEC_PER_SEC));
+
+  line_width =
+      1 + 11 + 11 + event_width + file_width + 6 + function_width + (4 * 3);
+
+  printf(" %-11s | %-11s | %-*s | %-*s | %-*s\n", "Timestamp", "Delta",
+         (int)event_width, "Event", (int)file_width + 6, "File",
+         (int)function_width, "Function");
+
+  for (i = 0; i <= line_width; i++) {
+    printf("=");
+  }
+  printf("\n");
+
+  for (i = 0; i < tc->curr_entry; i++) {
+    entry = &tc->entries[i];
+    offset = entry->timestamp - tc->start_time;
+    if (i > 0) {
+      delta = entry->timestamp - tc->entries[i - 1].timestamp;
+    } else {
+      delta = entry->timestamp - tc->start_time;
+    }
+    printf(" %4ld.%6.6ld | %4ld.%6.6ld | %-*s | %*s:%-5d | %-*s\n",
+           (long)(offset / PR_USEC_PER_SEC), (long)(offset % PR_USEC_PER_SEC),
+           (long)(delta / PR_USEC_PER_SEC), (long)(delta % PR_USEC_PER_SEC),
+           (int)event_width, entry->event, (int)file_width, entry->file,
+           entry->line, (int)function_width, entry->function);
+  }
+  printf("\n");
+}
diff --git a/dom/media/webrtc/common/time_profiling/timecard.h b/dom/media/webrtc/common/time_profiling/timecard.h
new file mode 100644
index 0000000000..38d4a8d1ec
--- /dev/null
+++ b/dom/media/webrtc/common/time_profiling/timecard.h
@@ -0,0 +1,74 @@
+/* -*- Mode: C; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef timecard_h__
+#define timecard_h__
+
+#include <stdlib.h>
+#include "prtime.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#define STAMP_TIMECARD(card, event)                                      \
+  do {                                                                   \
+    if (card) {                                                          \
+      stamp_timecard((card), (event), __FILE__, __LINE__, __FUNCTION__); \
+    }                                                                    \
+  } while (0)
+
+#define TIMECARD_INITIAL_TABLE_SIZE 16
+
+/*
+ * The "const char *" members of this structure point to static strings.
+ * We do not own them, and should not attempt to deallocate them.
+ */
+
+typedef struct {
+  PRTime timestamp;
+  const char* event;
+  const char* file;
+  unsigned int line;
+  const char* function;
+} TimecardEntry;
+
+typedef struct Timecard {
+  size_t curr_entry;
+  size_t entries_allocated;
+  TimecardEntry* entries;
+  PRTime start_time;
+} Timecard;
+
+/**
+ * Creates a new Timecard structure for tracking events.
+ */
+Timecard* create_timecard();
+
+/**
+ * Frees the memory associated with a timecard. After returning, the
+ * timecard pointed to by tc is no longer valid.
+ */
+void destroy_timecard(Timecard* tc);
+
+/**
+ * Records a new event in the indicated timecard. This should not be
+ * called directly; code should instead use the STAMP_TIMECARD macro,
+ * above.
+ */
+void stamp_timecard(Timecard* tc, const char* event, const char* file,
+                    unsigned int line, const char* function);
+
+/**
+ * Formats and outputs the contents of a timecard onto stdout.
+ */
+void print_timecard(Timecard* tc);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif
diff --git a/dom/media/webrtc/jsapi/MediaTransportHandler.cpp b/dom/media/webrtc/jsapi/MediaTransportHandler.cpp
new file mode 100644
index 0000000000..3ee95f36f6
--- /dev/null
+++ b/dom/media/webrtc/jsapi/MediaTransportHandler.cpp
@@ -0,0 +1,1727 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "MediaTransportHandler.h"
+#include "MediaTransportHandlerIPC.h"
+#include "transport/nricemediastream.h"
+#include "transport/nriceresolver.h"
+#include "transport/transportflow.h"
+#include "transport/transportlayerice.h"
+#include "transport/transportlayerdtls.h"
+#include "transport/transportlayersrtp.h"
+
+// Config stuff
+#include "mozilla/dom/RTCConfigurationBinding.h"
+#include "mozilla/Preferences.h"
+#include "mozilla/StaticPrefs_network.h"
+
+// Parsing STUN/TURN URIs
+#include "nsIURI.h"
+#include "nsNetUtil.h"
+#include "nsURLHelper.h"
+#include "nsIURLParser.h"
+
+// Logging stuff
+#include "common/browser_logging/CSFLog.h"
+
+// For fetching ICE logging
+#include "transport/rlogconnector.h"
+
+// DTLS
+#include "sdp/SdpAttribute.h"
+
+#include "transport/runnable_utils.h"
+
+#include "mozilla/Algorithm.h"
+#include "mozilla/Telemetry.h"
+
+#include "mozilla/dom/RTCStatsReportBinding.h"
+
+#include "nss.h"                // For NSS_NoDB_Init
+#include "mozilla/PublicSSL.h"  // For psm::InitializeCipherSuite
+
+#include "nsISocketTransportService.h"
+#include "nsDNSService2.h"
+
+#include <string>
+#include <vector>
+#include <map>
+
+#ifdef MOZ_GECKO_PROFILER
+#  include "mozilla/ProfilerMarkers.h"
+
+#  define MEDIA_TRANSPORT_HANDLER_PACKET_RECEIVED(aPacket)              \
+    PROFILER_MARKER_TEXT("WebRTC Packet Received", MEDIA_RT, {},        \
+                         ProfilerString8View::WrapNullTerminatedString( \
+                             PacketTypeToString((aPacket).type())));
+#else
+#  define MEDIA_TRANSPORT_HANDLER_PACKET_RECEIVED(aPacket)
+#endif
+
+namespace mozilla {
+
+static const char* mthLogTag = "MediaTransportHandler";
+#ifdef LOGTAG
+#  undef LOGTAG
+#endif
+#define LOGTAG mthLogTag
+
+class MediaTransportHandlerSTS : public MediaTransportHandler,
+                                 public sigslot::has_slots<> {
+ public:
+  explicit MediaTransportHandlerSTS(nsISerialEventTarget* aCallbackThread);
+
+  RefPtr<IceLogPromise> GetIceLog(const nsCString& aPattern) override;
+  void ClearIceLog() override;
+  void EnterPrivateMode() override;
+  void ExitPrivateMode() override;
+
+  void CreateIceCtx(const std::string& aName) override;
+
+  nsresult SetIceConfig(const nsTArray<dom::RTCIceServer>& aIceServers,
+                        dom::RTCIceTransportPolicy aIcePolicy) override;
+
+  // We will probably be able to move the proxy lookup stuff into
+  // this class once we move mtransport to its own process.
+  void SetProxyConfig(NrSocketProxyConfig&& aProxyConfig) override;
+
+  void EnsureProvisionalTransport(const std::string& aTransportId,
+                                  const std::string& aUfrag,
+                                  const std::string& aPwd,
+                                  int aComponentCount) override;
+
+  void SetTargetForDefaultLocalAddressLookup(const std::string& aTargetIp,
+                                             uint16_t aTargetPort) override;
+
+  // We set default-route-only as late as possible because it depends on what
+  // capture permissions have been granted on the window, which could easily
+  // change between Init (ie; when the PC is created) and StartIceGathering
+  // (ie; when we set the local description).
+  void StartIceGathering(bool aDefaultRouteOnly, bool aObfuscateHostAddresses,
+                         // This will go away once mtransport moves to its
+                         // own process, because we won't need to get this
+                         // via IPC anymore
+                         const nsTArray<NrIceStunAddr>& aStunAddrs) override;
+
+  void ActivateTransport(
+      const std::string& aTransportId, const std::string& aLocalUfrag,
+      const std::string& aLocalPwd, size_t aComponentCount,
+      const std::string& aUfrag, const std::string& aPassword,
+      const nsTArray<uint8_t>& aKeyDer, const nsTArray<uint8_t>& aCertDer,
+      SSLKEAType aAuthType, bool aDtlsClient, const DtlsDigestList& aDigests,
+      bool aPrivacyRequested) override;
+
+  void RemoveTransportsExcept(
+      const std::set<std::string>& aTransportIds) override;
+
+  void StartIceChecks(bool aIsControlling,
+                      const std::vector<std::string>& aIceOptions) override;
+
+  void AddIceCandidate(const std::string& aTransportId,
+                       const std::string& aCandidate, const std::string& aUfrag,
+                       const std::string& aObfuscatedAddress) override;
+
+  void UpdateNetworkState(bool aOnline) override;
+
+  void SendPacket(const std::string& aTransportId,
+                  MediaPacket&& aPacket) override;
+
+  RefPtr<dom::RTCStatsPromise> GetIceStats(const std::string& aTransportId,
+                                           DOMHighResTimeStamp aNow) override;
+
+  void Shutdown();
+
+ private:
+  void Destroy() override;
+  void Destroy_s();
+  void DestroyFinal();
+  void Shutdown_s();
+  RefPtr<TransportFlow> CreateTransportFlow(
+      const std::string& aTransportId, bool aIsRtcp,
+      const RefPtr<DtlsIdentity>& aDtlsIdentity, bool aDtlsClient,
+      const DtlsDigestList& aDigests, bool aPrivacyRequested);
+
+  struct Transport {
+    RefPtr<TransportFlow> mFlow;
+    RefPtr<TransportFlow> mRtcpFlow;
+  };
+
+  using MediaTransportHandler::OnAlpnNegotiated;
+  using MediaTransportHandler::OnCandidate;
+  using MediaTransportHandler::OnConnectionStateChange;
+  using MediaTransportHandler::OnEncryptedSending;
+  using MediaTransportHandler::OnGatheringStateChange;
+  using MediaTransportHandler::OnPacketReceived;
+  using MediaTransportHandler::OnRtcpStateChange;
+  using MediaTransportHandler::OnStateChange;
+
+  void OnGatheringStateChange(NrIceCtx* aIceCtx,
+                              NrIceCtx::GatheringState aState);
+  void OnConnectionStateChange(NrIceCtx* aIceCtx,
+                               NrIceCtx::ConnectionState aState);
+  void OnCandidateFound(NrIceMediaStream* aStream,
+                        const std::string& aCandidate,
+                        const std::string& aUfrag, const std::string& aMDNSAddr,
+                        const std::string& aActualAddr);
+  void OnStateChange(TransportLayer* aLayer, TransportLayer::State);
+  void OnRtcpStateChange(TransportLayer* aLayer, TransportLayer::State);
+  void PacketReceived(TransportLayer* aLayer, MediaPacket& aPacket);
+  void EncryptedPacketSending(TransportLayer* aLayer, MediaPacket& aPacket);
+  RefPtr<TransportFlow> GetTransportFlow(const std::string& aTransportId,
+                                         bool aIsRtcp) const;
+  void GetIceStats(const NrIceMediaStream& aStream, DOMHighResTimeStamp aNow,
+                   dom::RTCStatsCollection* aStats) const;
+
+  virtual ~MediaTransportHandlerSTS() = default;
+  nsCOMPtr<nsISerialEventTarget> mStsThread;
+  RefPtr<NrIceCtx> mIceCtx;
+  RefPtr<NrIceResolver> mDNSResolver;
+  std::map<std::string, Transport> mTransports;
+  bool mObfuscateHostAddresses = false;
+  bool mTurnDisabled = false;
+  uint32_t mMinDtlsVersion = 0;
+  uint32_t mMaxDtlsVersion = 0;
+  bool mForceNoHost = false;
+  Maybe<NrIceCtx::NatSimulatorConfig> mNatConfig;
+
+  std::set<std::string> mSignaledAddresses;
+
+  // Init can only be done on main, but we want this to be usable on any thread
+  using InitPromise = MozPromise<bool, std::string, false>;
+  RefPtr<InitPromise> mInitPromise;
+};
+
+/* static */
+already_AddRefed<MediaTransportHandler> MediaTransportHandler::Create(
+    nsISerialEventTarget* aCallbackThread) {
+  RefPtr<MediaTransportHandler> result;
+  if (XRE_IsContentProcess() &&
+      Preferences::GetBool("media.peerconnection.mtransport_process") &&
+      StaticPrefs::network_process_enabled()) {
+    result = new MediaTransportHandlerIPC(aCallbackThread);
+  } else {
+    result = new MediaTransportHandlerSTS(aCallbackThread);
+  }
+  result->Initialize();
+  return result.forget();
+}
+
+class STSShutdownHandler : public nsISTSShutdownObserver {
+ public:
+  NS_DECL_ISUPPORTS
+
+  // Lazy singleton
+  static RefPtr<STSShutdownHandler>& Instance() {
+    MOZ_ASSERT(NS_IsMainThread());
+    static RefPtr<STSShutdownHandler> sHandler(new STSShutdownHandler);
+    return sHandler;
+  }
+
+  void Shutdown() {
+    MOZ_ASSERT(NS_IsMainThread());
+    for (const auto& handler : mHandlers) {
+      handler->Shutdown();
+    }
+    mHandlers.clear();
+  }
+
+  STSShutdownHandler() {
+    CSFLogDebug(LOGTAG, "%s", __func__);
+    nsresult res;
+    nsCOMPtr<nsISocketTransportService> sts =
+        do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &res);
+    MOZ_RELEASE_ASSERT(NS_SUCCEEDED(res));
+    MOZ_RELEASE_ASSERT(sts);
+    sts->AddShutdownObserver(this);
+  }
+
+  NS_IMETHOD Observe() override {
+    CSFLogDebug(LOGTAG, "%s", __func__);
+    Shutdown();
+    nsresult res;
+    nsCOMPtr<nsISocketTransportService> sts =
+        do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &res);
+    MOZ_RELEASE_ASSERT(NS_SUCCEEDED(res));
+    MOZ_RELEASE_ASSERT(sts);
+    sts->RemoveShutdownObserver(this);
+    Instance() = nullptr;
+    return NS_OK;
+  }
+
+  void Register(MediaTransportHandlerSTS* aHandler) {
+    MOZ_ASSERT(NS_IsMainThread());
+    mHandlers.insert(aHandler);
+  }
+
+  void Deregister(MediaTransportHandlerSTS* aHandler) {
+    MOZ_ASSERT(NS_IsMainThread());
+    mHandlers.erase(aHandler);
+  }
+
+ private:
+  virtual ~STSShutdownHandler() = default;
+
+  // Raw ptrs, registered on init, deregistered on destruction, all on main
+  std::set<MediaTransportHandlerSTS*> mHandlers;
+};
+
+NS_IMPL_ISUPPORTS(STSShutdownHandler, nsISTSShutdownObserver);
+
+MediaTransportHandlerSTS::MediaTransportHandlerSTS(
+    nsISerialEventTarget* aCallbackThread)
+    : MediaTransportHandler(aCallbackThread) {
+  nsresult rv;
+  mStsThread = do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &rv);
+  if (!mStsThread) {
+    MOZ_CRASH();
+  }
+
+  RLogConnector::CreateInstance();
+
+  CSFLogDebug(LOGTAG, "%s done %p", __func__, this);
+
+  // We do not set up mDNSService here, because we are not running on main (we
+  // use PBackground), and the DNS service asserts.
+}
+
+static NrIceCtx::Policy toNrIcePolicy(dom::RTCIceTransportPolicy aPolicy) {
+  switch (aPolicy) {
+    case dom::RTCIceTransportPolicy::Relay:
+      return NrIceCtx::ICE_POLICY_RELAY;
+    case dom::RTCIceTransportPolicy::All:
+      return NrIceCtx::ICE_POLICY_ALL;
+    default:
+      MOZ_CRASH();
+  }
+  return NrIceCtx::ICE_POLICY_ALL;
+}
+
+// list of known acceptable ports for webrtc
+int16_t gGoodWebrtcPortList[] = {
+    53,    // Some deplyoments use DNS port to punch through overzealous NATs
+    3478,  // stun or turn
+    5349,  // stuns or turns
+    0,     // Sentinel value: This MUST be zero
+};
+
+static nsresult addNrIceServer(const nsString& aIceUrl,
+                               const dom::RTCIceServer& aIceServer,
+                               std::vector<NrIceStunServer>* aStunServersOut,
+                               std::vector<NrIceTurnServer>* aTurnServersOut) {
+  // Without STUN/TURN handlers, NS_NewURI returns nsSimpleURI rather than
+  // nsStandardURL. To parse STUN/TURN URI's to spec
+  // http://tools.ietf.org/html/draft-nandakumar-rtcweb-stun-uri-02#section-3
+  // http://tools.ietf.org/html/draft-petithuguenin-behave-turn-uri-03#section-3
+  // we parse out the query-string, and use ParseAuthority() on the rest
+  RefPtr<nsIURI> url;
+  nsresult rv = NS_NewURI(getter_AddRefs(url), aIceUrl);
+  NS_ENSURE_SUCCESS(rv, rv);
+  bool isStun = url->SchemeIs("stun");
+  bool isStuns = url->SchemeIs("stuns");
+  bool isTurn = url->SchemeIs("turn");
+  bool isTurns = url->SchemeIs("turns");
+  if (!(isStun || isStuns || isTurn || isTurns)) {
+    return NS_ERROR_FAILURE;
+  }
+  if (isStuns) {
+    return NS_OK;  // TODO: Support STUNS (Bug 1056934)
+  }
+
+  nsAutoCString spec;
+  rv = url->GetSpec(spec);
+  NS_ENSURE_SUCCESS(rv, rv);
+
+  // TODO(jib@mozilla.com): Revisit once nsURI supports STUN/TURN (Bug 833509)
+  int32_t port;
+  nsAutoCString host;
+  nsAutoCString transport;
+  {
+    uint32_t hostPos;
+    int32_t hostLen;
+    nsAutoCString path;
+    rv = url->GetPathQueryRef(path);
+    NS_ENSURE_SUCCESS(rv, rv);
+
+    // Tolerate query-string + parse 'transport=[udp|tcp]' by hand.
+    int32_t questionmark = path.FindChar('?');
+    if (questionmark >= 0) {
+      const nsCString match = "transport="_ns;
+
+      for (int32_t i = questionmark, endPos; i >= 0; i = endPos) {
+        endPos = path.FindCharInSet("&", i + 1);
+        const nsDependentCSubstring fieldvaluepair =
+            Substring(path, i + 1, endPos);
+        if (StringBeginsWith(fieldvaluepair, match)) {
+          transport = Substring(fieldvaluepair, match.Length());
+          ToLowerCase(transport);
+        }
+      }
+      path.SetLength(questionmark);
+    }
+
+    rv = net_GetAuthURLParser()->ParseAuthority(
+        path.get(), static_cast<int>(path.Length()), nullptr, nullptr, nullptr,
+        nullptr, &hostPos, &hostLen, &port);
+    NS_ENSURE_SUCCESS(rv, rv);
+    if (!hostLen) {
+      return NS_ERROR_FAILURE;
+    }
+    if (hostPos > 1) {
+      /* The username was removed */
+      return NS_ERROR_FAILURE;
+    }
+    path.Mid(host, hostPos, hostLen);
+    // Strip off brackets around IPv6 literals
+    host.Trim("[]");
+  }
+  if (port == -1) port = (isStuns || isTurns) ? 5349 : 3478;
+
+  // First check the known good ports for webrtc
+  bool goodPort = false;
+  for (int i = 0; !goodPort && gGoodWebrtcPortList[i]; i++) {
+    if (port == gGoodWebrtcPortList[i]) {
+      goodPort = true;
+    }
+  }
+
+  // if not in the list of known good ports for webrtc, check
+  // the generic block list using NS_CheckPortSafety.
+  if (!goodPort) {
+    rv = NS_CheckPortSafety(port, nullptr);
+    NS_ENSURE_SUCCESS(rv, rv);
+  }
+
+  if (isStuns || isTurns) {
+    // Should we barf if transport is set to udp or something?
+    transport = kNrIceTransportTls;
+  }
+
+  if (transport.IsEmpty()) {
+    transport = kNrIceTransportUdp;
+  }
+
+  if (isTurn || isTurns) {
+    std::string pwd(
+        NS_ConvertUTF16toUTF8(aIceServer.mCredential.Value()).get());
+    std::string username(
+        NS_ConvertUTF16toUTF8(aIceServer.mUsername.Value()).get());
+
+    std::vector<unsigned char> password(pwd.begin(), pwd.end());
+
+    UniquePtr<NrIceTurnServer> server(NrIceTurnServer::Create(
+        host.get(), port, username, password, transport.get()));
+    if (!server) {
+      return NS_ERROR_FAILURE;
+    }
+    if (server->HasFqdn()) {
+      // Add an IPv4 entry, then an IPv6 entry
+      aTurnServersOut->push_back(*server);
+      server->SetUseIPv6IfFqdn();
+    }
+    aTurnServersOut->emplace_back(std::move(*server));
+  } else {
+    UniquePtr<NrIceStunServer> server(
+        NrIceStunServer::Create(host.get(), port, transport.get()));
+    if (!server) {
+      return NS_ERROR_FAILURE;
+    }
+    if (server->HasFqdn()) {
+      // Add an IPv4 entry, then an IPv6 entry
+      aStunServersOut->push_back(*server);
+      server->SetUseIPv6IfFqdn();
+    }
+    aStunServersOut->emplace_back(std::move(*server));
+  }
+  return NS_OK;
+}
+
+/* static */
+nsresult MediaTransportHandler::ConvertIceServers(
+    const nsTArray<dom::RTCIceServer>& aIceServers,
+    std::vector<NrIceStunServer>* aStunServers,
+    std::vector<NrIceTurnServer>* aTurnServers) {
+  for (const auto& iceServer : aIceServers) {
+    NS_ENSURE_STATE(iceServer.mUrls.WasPassed());
+    NS_ENSURE_STATE(iceServer.mUrls.Value().IsStringSequence());
+    for (const auto& iceUrl : iceServer.mUrls.Value().GetAsStringSequence()) {
+      nsresult rv =
+          addNrIceServer(iceUrl, iceServer, aStunServers, aTurnServers);
+      if (NS_FAILED(rv)) {
+        CSFLogError(LOGTAG, "%s: invalid STUN/TURN server: %s", __FUNCTION__,
+                    NS_ConvertUTF16toUTF8(iceUrl).get());
+        return rv;
+      }
+    }
+  }
+
+  return NS_OK;
+}
+
+static NrIceCtx::GlobalConfig GetGlobalConfig() {
+  NrIceCtx::GlobalConfig config;
+  config.mAllowLinkLocal =
+      Preferences::GetBool("media.peerconnection.ice.link_local", false);
+  config.mAllowLoopback =
+      Preferences::GetBool("media.peerconnection.ice.loopback", false);
+  config.mTcpEnabled =
+      Preferences::GetBool("media.peerconnection.ice.tcp", false);
+  config.mStunClientMaxTransmits = Preferences::GetInt(
+      "media.peerconnection.ice.stun_client_maximum_transmits",
+      config.mStunClientMaxTransmits);
+  config.mTrickleIceGracePeriod =
+      Preferences::GetInt("media.peerconnection.ice.trickle_grace_period",
+                          config.mTrickleIceGracePeriod);
+  config.mIceTcpSoSockCount = Preferences::GetInt(
+      "media.peerconnection.ice.tcp_so_sock_count", config.mIceTcpSoSockCount);
+  config.mIceTcpListenBacklog =
+      Preferences::GetInt("media.peerconnection.ice.tcp_listen_backlog",
+                          config.mIceTcpListenBacklog);
+  (void)Preferences::GetCString("media.peerconnection.ice.force_interface",
+                                config.mForceNetInterface);
+  return config;
+}
+
+static Maybe<NrIceCtx::NatSimulatorConfig> GetNatConfig() {
+  bool block_tcp = Preferences::GetBool(
+      "media.peerconnection.nat_simulator.block_tcp", false);
+  bool block_udp = Preferences::GetBool(
+      "media.peerconnection.nat_simulator.block_udp", false);
+  bool block_tls = Preferences::GetBool(
+      "media.peerconnection.nat_simulator.block_tls", false);
+  int error_code_for_drop = Preferences::GetInt(
+      "media.peerconnection.nat_simulator.error_code_for_drop", 0);
+  nsAutoCString mapping_type;
+  (void)Preferences::GetCString(
+      "media.peerconnection.nat_simulator.mapping_type", mapping_type);
+  nsAutoCString filtering_type;
+  (void)Preferences::GetCString(
+      "media.peerconnection.nat_simulator.filtering_type", filtering_type);
+  nsAutoCString redirect_address;
+  (void)Preferences::GetCString(
+      "media.peerconnection.nat_simulator.redirect_address", redirect_address);
+  nsAutoCString redirect_targets;
+  (void)Preferences::GetCString(
+      "media.peerconnection.nat_simulator.redirect_targets", redirect_targets);
+
+  if (block_udp || block_tcp || block_tls || !mapping_type.IsEmpty() ||
+      !filtering_type.IsEmpty() || !redirect_address.IsEmpty()) {
+    CSFLogDebug(LOGTAG, "NAT filtering type: %s", filtering_type.get());
+    CSFLogDebug(LOGTAG, "NAT mapping type: %s", mapping_type.get());
+    NrIceCtx::NatSimulatorConfig natConfig;
+    natConfig.mBlockUdp = block_udp;
+    natConfig.mBlockTcp = block_tcp;
+    natConfig.mBlockTls = block_tls;
+    natConfig.mErrorCodeForDrop = error_code_for_drop;
+    natConfig.mFilteringType = filtering_type;
+    natConfig.mMappingType = mapping_type;
+    if (redirect_address.Length()) {
+      CSFLogDebug(LOGTAG, "Redirect address: %s", redirect_address.get());
+      CSFLogDebug(LOGTAG, "Redirect targets: %s", redirect_targets.get());
+      natConfig.mRedirectAddress = redirect_address;
+      std::stringstream str(redirect_targets.Data());
+      std::string target;
+      while (getline(str, target, ',')) {
+        CSFLogDebug(LOGTAG, "Adding target: %s", target.c_str());
+        natConfig.mRedirectTargets.AppendElement(target);
+      }
+    }
+    return Some(natConfig);
+  }
+  return Nothing();
+}
+
+void MediaTransportHandlerSTS::CreateIceCtx(const std::string& aName) {
+  mInitPromise = InvokeAsync(
+      GetMainThreadSerialEventTarget(), __func__,
+      [=, self = RefPtr<MediaTransportHandlerSTS>(this)]() {
+        CSFLogDebug(LOGTAG, "%s starting", __func__);
+        if (!NSS_IsInitialized()) {
+          if (NSS_NoDB_Init(nullptr) != SECSuccess) {
+            MOZ_CRASH();
+            return InitPromise::CreateAndReject("NSS_NoDB_Init failed",
+                                                __func__);
+          }
+
+          if (NS_FAILED(mozilla::psm::InitializeCipherSuite())) {
+            MOZ_CRASH();
+            return InitPromise::CreateAndReject("InitializeCipherSuite failed",
+                                                __func__);
+          }
+
+          mozilla::psm::DisableMD5();
+        }
+
+        static bool globalInitDone = false;
+        if (!globalInitDone) {
+          // Ensure the DNS service is initted for the first time on main
+          DebugOnly<RefPtr<nsIDNSService>> dnsService =
+              RefPtr<nsIDNSService>(nsDNSService::GetXPCOMSingleton());
+          MOZ_ASSERT(dnsService.value);
+          mStsThread->Dispatch(
+              WrapRunnableNM(&NrIceCtx::InitializeGlobals, GetGlobalConfig()),
+              NS_DISPATCH_NORMAL);
+          globalInitDone = true;
+        }
+
+        // Give us a way to globally turn off TURN support
+        mTurnDisabled =
+            Preferences::GetBool("media.peerconnection.turn.disable", false);
+        // We are reading these here, because when we setup the DTLS transport
+        // we are on the wrong thread to read prefs
+        mMinDtlsVersion =
+            Preferences::GetUint("media.peerconnection.dtls.version.min");
+        mMaxDtlsVersion =
+            Preferences::GetUint("media.peerconnection.dtls.version.max");
+        mForceNoHost =
+            Preferences::GetBool("media.peerconnection.ice.no_host", false);
+        mNatConfig = GetNatConfig();
+
+        MOZ_RELEASE_ASSERT(STSShutdownHandler::Instance());
+        STSShutdownHandler::Instance()->Register(this);
+
+        return InvokeAsync(
+            mStsThread, __func__,
+            [=, self = RefPtr<MediaTransportHandlerSTS>(this)]() {
+              mIceCtx = NrIceCtx::Create(aName);
+              if (!mIceCtx) {
+                return InitPromise::CreateAndReject("NrIceCtx::Create failed",
+                                                    __func__);
+              }
+
+              mIceCtx->SignalGatheringStateChange.connect(
+                  this, &MediaTransportHandlerSTS::OnGatheringStateChange);
+              mIceCtx->SignalConnectionStateChange.connect(
+                  this, &MediaTransportHandlerSTS::OnConnectionStateChange);
+
+              mDNSResolver = new NrIceResolver;
+              nsresult rv;
+              if (NS_FAILED(rv = mDNSResolver->Init())) {
+                CSFLogError(LOGTAG, "%s: Failed to initialize dns resolver",
+                            __FUNCTION__);
+                return InitPromise::CreateAndReject(
+                    "Failed to initialize dns resolver", __func__);
+              }
+              if (NS_FAILED(rv = mIceCtx->SetResolver(
+                                mDNSResolver->AllocateResolver()))) {
+                CSFLogError(LOGTAG, "%s: Failed to get dns resolver",
+                            __FUNCTION__);
+                return InitPromise::CreateAndReject(
+                    "Failed to get dns resolver", __func__);
+              }
+
+              CSFLogDebug(LOGTAG, "%s done", __func__);
+              return InitPromise::CreateAndResolve(true, __func__);
+            });
+      });
+}
+
+nsresult MediaTransportHandlerSTS::SetIceConfig(
+    const nsTArray<dom::RTCIceServer>& aIceServers,
+    dom::RTCIceTransportPolicy aIcePolicy) {
+  // We rely on getting an error when this happens, so do it up front.
+  std::vector<NrIceStunServer> stunServers;
+  std::vector<NrIceTurnServer> turnServers;
+  nsresult rv = ConvertIceServers(aIceServers, &stunServers, &turnServers);
+  if (NS_FAILED(rv)) {
+    return rv;
+  }
+
+  MOZ_RELEASE_ASSERT(mInitPromise);
+
+  mInitPromise->Then(
+      mStsThread, __func__,
+      [=, self = RefPtr<MediaTransportHandlerSTS>(this)]() {
+        if (!mIceCtx) {
+          CSFLogError(LOGTAG, "%s: mIceCtx is null", __FUNCTION__);
+          return;
+        }
+        NrIceCtx::Config config;
+        config.mPolicy = toNrIcePolicy(aIcePolicy);
+        if (config.mPolicy == NrIceCtx::ICE_POLICY_ALL && mForceNoHost) {
+          config.mPolicy = NrIceCtx::ICE_POLICY_NO_HOST;
+        }
+        config.mNatSimulatorConfig = mNatConfig;
+
+        nsresult rv;
+
+        if (NS_FAILED(rv = mIceCtx->SetStunServers(stunServers))) {
+          CSFLogError(LOGTAG, "%s: Failed to set stun servers", __FUNCTION__);
+          return;
+        }
+        if (!mTurnDisabled) {
+          if (NS_FAILED(rv = mIceCtx->SetTurnServers(turnServers))) {
+            CSFLogError(LOGTAG, "%s: Failed to set turn servers", __FUNCTION__);
+            return;
+          }
+        } else if (!turnServers.empty()) {
+          CSFLogError(LOGTAG, "%s: Setting turn servers disabled",
+                      __FUNCTION__);
+        }
+        if (NS_FAILED(rv = mIceCtx->SetIceConfig(config))) {
+          CSFLogError(LOGTAG, "%s: Failed to set config", __FUNCTION__);
+        }
+      });
+
+  return NS_OK;
+}
+
+void MediaTransportHandlerSTS::Shutdown() {
+  CSFLogDebug(LOGTAG, "%s", __func__);
+  MOZ_ASSERT(NS_IsMainThread());
+  mStsThread->Dispatch(NewNonOwningRunnableMethod(
+      __func__, this, &MediaTransportHandlerSTS::Shutdown_s));
+}
+
+void MediaTransportHandlerSTS::Shutdown_s() {
+  CSFLogDebug(LOGTAG, "%s", __func__);
+  disconnect_all();
+  // Clear the transports before destroying the ice ctx so that
+  // the close_notify alerts have a chance to be sent as the
+  // TransportFlow destructors execute.
+  mTransports.clear();
+  if (mIceCtx) {
+    NrIceStats stats = mIceCtx->Destroy();
+    CSFLogDebug(LOGTAG,
+                "Ice Telemetry: stun (retransmits: %d)"
+                "   turn (401s: %d   403s: %d   438s: %d)",
+                stats.stun_retransmits, stats.turn_401s, stats.turn_403s,
+                stats.turn_438s);
+  }
+  mIceCtx = nullptr;
+  mDNSResolver = nullptr;
+}
+
+void MediaTransportHandlerSTS::Destroy() {
+  CSFLogDebug(LOGTAG, "%s %p", __func__, this);
+  // Our "destruction tour" starts on main, because we need to deregister.
+  if (!NS_IsMainThread()) {
+    GetMainThreadSerialEventTarget()->Dispatch(
+        NewNonOwningRunnableMethod("MediaTransportHandlerSTS::Destroy", this,
+                                   &MediaTransportHandlerSTS::Destroy));
+    return;
+  }
+
+  MOZ_ASSERT(NS_IsMainThread());
+  if (STSShutdownHandler::Instance()) {
+    STSShutdownHandler::Instance()->Deregister(this);
+    Shutdown();
+  }
+
+  // mIceCtx still has a reference to us via sigslot! We must dispach to STS,
+  // and clean up there. However, by the time _that_ happens, we may have
+  // dispatched a signal callback to mCallbackThread, so we have to dispatch
+  // the final destruction to mCallbackThread.
+  nsresult rv = mStsThread->Dispatch(
+      NewNonOwningRunnableMethod("MediaTransportHandlerSTS::Destroy_s", this,
+                                 &MediaTransportHandlerSTS::Destroy_s));
+  if (NS_WARN_IF(NS_FAILED(rv))) {
+    CSFLogError(LOGTAG,
+                "Unable to dispatch to STS: why has the XPCOM shutdown handler "
+                "not been invoked?");
+    delete this;
+  }
+}
+
+void MediaTransportHandlerSTS::Destroy_s() {
+  if (mCallbackThread && !mCallbackThread->IsOnCurrentThread()) {
+    nsresult rv = mCallbackThread->Dispatch(NewNonOwningRunnableMethod(
+        __func__, this, &MediaTransportHandlerSTS::DestroyFinal));
+    if (NS_SUCCEEDED(rv)) {
+      return;
+    }
+  }
+
+  DestroyFinal();
+}
+
+void MediaTransportHandlerSTS::DestroyFinal() { delete this; }
+
+void MediaTransportHandlerSTS::SetProxyConfig(
+    NrSocketProxyConfig&& aProxyConfig) {
+  MOZ_RELEASE_ASSERT(mInitPromise);
+
+  mInitPromise->Then(
+      mStsThread, __func__,
+      [this, self = RefPtr<MediaTransportHandlerSTS>(this),
+       aProxyConfig = std::move(aProxyConfig)]() mutable {
+        if (!mIceCtx) {
+          return;  // Probably due to XPCOM shutdown
+        }
+
+        mIceCtx->SetProxyConfig(std::move(aProxyConfig));
+      },
+      [](const std::string& aError) {});
+}
+
+void MediaTransportHandlerSTS::EnsureProvisionalTransport(
+    const std::string& aTransportId, const std::string& aUfrag,
+    const std::string& aPwd, int aComponentCount) {
+  MOZ_RELEASE_ASSERT(mInitPromise);
+
+  mInitPromise->Then(
+      mStsThread, __func__,
+      [=, self = RefPtr<MediaTransportHandlerSTS>(this)]() {
+        if (!mIceCtx) {
+          return;  // Probably due to XPCOM shutdown
+        }
+
+        RefPtr<NrIceMediaStream> stream(mIceCtx->GetStream(aTransportId));
+        if (!stream) {
+          CSFLogDebug(LOGTAG, "%s: Creating ICE media stream=%s components=%d",
+                      mIceCtx->name().c_str(), aTransportId.c_str(),
+                      aComponentCount);
+
+          std::ostringstream os;
+          os << mIceCtx->name() << " transport-id=" << aTransportId;
+          stream =
+              mIceCtx->CreateStream(aTransportId, os.str(), aComponentCount);
+
+          if (!stream) {
+            CSFLogError(LOGTAG, "Failed to create ICE stream.");
+            return;
+          }
+
+          stream->SignalCandidate.connect(
+              this, &MediaTransportHandlerSTS::OnCandidateFound);
+        }
+
+        // Begins an ICE restart if this stream has a different ufrag/pwd
+        stream->SetIceCredentials(aUfrag, aPwd);
+
+        // Make sure there's an entry in mTransports
+        mTransports[aTransportId];
+      },
+      [](const std::string& aError) {});
+}
+
+void MediaTransportHandlerSTS::ActivateTransport(
+    const std::string& aTransportId, const std::string& aLocalUfrag,
+    const std::string& aLocalPwd, size_t aComponentCount,
+    const std::string& aUfrag, const std::string& aPassword,
+    const nsTArray<uint8_t>& aKeyDer, const nsTArray<uint8_t>& aCertDer,
+    SSLKEAType aAuthType, bool aDtlsClient, const DtlsDigestList& aDigests,
+    bool aPrivacyRequested) {
+  MOZ_RELEASE_ASSERT(mInitPromise);
+
+  mInitPromise->Then(
+      mStsThread, __func__,
+      [=, keyDer = aKeyDer.Clone(), certDer = aCertDer.Clone(),
+       self = RefPtr<MediaTransportHandlerSTS>(this)]() {
+        if (!mIceCtx) {
+          return;  // Probably due to XPCOM shutdown
+        }
+
+        MOZ_ASSERT(aComponentCount);
+        RefPtr<DtlsIdentity> dtlsIdentity(
+            DtlsIdentity::Deserialize(keyDer, certDer, aAuthType));
+        if (!dtlsIdentity) {
+          MOZ_ASSERT(false);
+          return;
+        }
+
+        RefPtr<NrIceMediaStream> stream(mIceCtx->GetStream(aTransportId));
+        if (!stream) {
+          MOZ_ASSERT(false);
+          return;
+        }
+
+        CSFLogDebug(LOGTAG, "%s: Activating ICE media stream=%s components=%u",
+                    mIceCtx->name().c_str(), aTransportId.c_str(),
+                    static_cast<unsigned>(aComponentCount));
+
+        std::vector<std::string> attrs;
+        attrs.reserve(2 /* ufrag + pwd */);
+        attrs.push_back("ice-ufrag:" + aUfrag);
+        attrs.push_back("ice-pwd:" + aPassword);
+
+        // If we started an ICE restart in EnsureProvisionalTransport, this is
+        // where we decide whether to commit or rollback.
+        nsresult rv = stream->ConnectToPeer(aLocalUfrag, aLocalPwd, attrs);
+        if (NS_FAILED(rv)) {
+          CSFLogError(LOGTAG, "Couldn't parse ICE attributes, rv=%u",
+                      static_cast<unsigned>(rv));
+          MOZ_ASSERT(false);
+          return;
+        }
+
+        Transport transport = mTransports[aTransportId];
+        if (!transport.mFlow) {
+          transport.mFlow =
+              CreateTransportFlow(aTransportId, false, dtlsIdentity,
+                                  aDtlsClient, aDigests, aPrivacyRequested);
+          if (!transport.mFlow) {
+            return;
+          }
+          TransportLayer* dtls =
+              transport.mFlow->GetLayer(TransportLayerDtls::ID());
+          dtls->SignalStateChange.connect(
+              this, &MediaTransportHandlerSTS::OnStateChange);
+          if (aComponentCount < 2) {
+            dtls->SignalStateChange.connect(
+                this, &MediaTransportHandlerSTS::OnRtcpStateChange);
+          }
+        }
+
+        if (aComponentCount == 2) {
+          if (!transport.mRtcpFlow) {
+            transport.mRtcpFlow =
+                CreateTransportFlow(aTransportId, true, dtlsIdentity,
+                                    aDtlsClient, aDigests, aPrivacyRequested);
+            if (!transport.mRtcpFlow) {
+              return;
+            }
+            TransportLayer* dtls =
+                transport.mRtcpFlow->GetLayer(TransportLayerDtls::ID());
+            dtls->SignalStateChange.connect(
+                this, &MediaTransportHandlerSTS::OnRtcpStateChange);
+          }
+        } else {
+          transport.mRtcpFlow = nullptr;
+          // components are 1-indexed
+          stream->DisableComponent(2);
+        }
+
+        mTransports[aTransportId] = transport;
+      },
+      [](const std::string& aError) {});
+}
+
+void MediaTransportHandlerSTS::SetTargetForDefaultLocalAddressLookup(
+    const std::string& aTargetIp, uint16_t aTargetPort) {
+  MOZ_RELEASE_ASSERT(mInitPromise);
+
+  mInitPromise->Then(
+      mStsThread, __func__,
+      [=, self = RefPtr<MediaTransportHandlerSTS>(this)]() {
+        if (!mIceCtx) {
+          return;  // Probably due to XPCOM shutdown
+        }
+
+        mIceCtx->SetTargetForDefaultLocalAddressLookup(aTargetIp, aTargetPort);
+      },
+      [](const std::string& aError) {});
+}
+
+void MediaTransportHandlerSTS::StartIceGathering(
+    bool aDefaultRouteOnly, bool aObfuscateHostAddresses,
+    const nsTArray<NrIceStunAddr>& aStunAddrs) {
+  MOZ_RELEASE_ASSERT(mInitPromise);
+
+  mInitPromise->Then(
+      mStsThread, __func__,
+      [=, stunAddrs = aStunAddrs.Clone(),
+       self = RefPtr<MediaTransportHandlerSTS>(this)]() {
+        if (!mIceCtx) {
+          return;  // Probably due to XPCOM shutdown
+        }
+
+        mObfuscateHostAddresses = aObfuscateHostAddresses;
+
+        // Belt and suspenders - in e10s mode, the call below to SetStunAddrs
+        // needs to have the proper flags set on ice ctx.  For non-e10s,
+        // setting those flags happens in StartGathering.  We could probably
+        // just set them here, and only do it here.
+        mIceCtx->SetCtxFlags(aDefaultRouteOnly);
+
+        if (stunAddrs.Length()) {
+          mIceCtx->SetStunAddrs(stunAddrs);
+        }
+
+        // Start gathering, but only if there are streams
+        if (!mIceCtx->GetStreams().empty()) {
+          mIceCtx->StartGathering(aDefaultRouteOnly, aObfuscateHostAddresses);
+        }
+      },
+      [](const std::string& aError) {});
+}
+
+void MediaTransportHandlerSTS::StartIceChecks(
+    bool aIsControlling, const std::vector<std::string>& aIceOptions) {
+  MOZ_RELEASE_ASSERT(mInitPromise);
+
+  mInitPromise->Then(
+      mStsThread, __func__,
+      [=, self = RefPtr<MediaTransportHandlerSTS>(this)]() {
+        if (!mIceCtx) {
+          return;  // Probably due to XPCOM shutdown
+        }
+
+        nsresult rv = mIceCtx->ParseGlobalAttributes(aIceOptions);
+        if (NS_FAILED(rv)) {
+          CSFLogError(LOGTAG, "%s: couldn't parse global parameters",
+                      __FUNCTION__);
+          return;
+        }
+
+        rv = mIceCtx->SetControlling(aIsControlling ? NrIceCtx::ICE_CONTROLLING
+                                                    : NrIceCtx::ICE_CONTROLLED);
+        if (NS_FAILED(rv)) {
+          CSFLogError(LOGTAG, "%s: couldn't set controlling to %d",
+                      __FUNCTION__, aIsControlling);
+          return;
+        }
+
+        rv = mIceCtx->StartChecks();
+        if (NS_FAILED(rv)) {
+          CSFLogError(LOGTAG, "%s: couldn't start checks", __FUNCTION__);
+          return;
+        }
+      },
+      [](const std::string& aError) {});
+}
+
+void TokenizeCandidate(const std::string& aCandidate,
+                       std::vector<std::string>& aTokens) {
+  aTokens.clear();
+
+  std::istringstream iss(aCandidate);
+  std::string token;
+  while (std::getline(iss, token, ' ')) {
+    aTokens.push_back(token);
+  }
+}
+
+void MediaTransportHandlerSTS::AddIceCandidate(
+    const std::string& aTransportId, const std::string& aCandidate,
+    const std::string& aUfrag, const std::string& aObfuscatedAddress) {
+  MOZ_RELEASE_ASSERT(mInitPromise);
+
+  mInitPromise->Then(
+      mStsThread, __func__,
+      [=, self = RefPtr<MediaTransportHandlerSTS>(this)]() {
+        if (!mIceCtx) {
+          return;  // Probably due to XPCOM shutdown
+        }
+
+        std::vector<std::string> tokens;
+        TokenizeCandidate(aCandidate, tokens);
+
+        RefPtr<NrIceMediaStream> stream(mIceCtx->GetStream(aTransportId));
+        if (!stream) {
+          CSFLogError(LOGTAG,
+                      "No ICE stream for candidate with transport id %s: %s",
+                      aTransportId.c_str(), aCandidate.c_str());
+          return;
+        }
+
+        nsresult rv = stream->ParseTrickleCandidate(aCandidate, aUfrag,
+                                                    aObfuscatedAddress);
+        if (NS_SUCCEEDED(rv)) {
+          // If the address is not obfuscated, we want to track it as
+          // explicitly signaled so that we know it is fine to reveal
+          // the address later on.
+          if (mObfuscateHostAddresses && tokens.size() > 4 &&
+              aObfuscatedAddress.empty()) {
+            mSignaledAddresses.insert(tokens[4]);
+          }
+        } else {
+          CSFLogError(LOGTAG,
+                      "Couldn't process ICE candidate with transport id %s: "
+                      "%s",
+                      aTransportId.c_str(), aCandidate.c_str());
+        }
+      },
+      [](const std::string& aError) {});
+}
+
+void MediaTransportHandlerSTS::UpdateNetworkState(bool aOnline) {
+  MOZ_RELEASE_ASSERT(mInitPromise);
+
+  mInitPromise->Then(
+      mStsThread, __func__,
+      [=, self = RefPtr<MediaTransportHandlerSTS>(this)]() {
+        if (!mIceCtx) {
+          return;  // Probably due to XPCOM shutdown
+        }
+
+        mIceCtx->UpdateNetworkState(aOnline);
+      },
+      [](const std::string& aError) {});
+}
+
+void MediaTransportHandlerSTS::RemoveTransportsExcept(
+    const std::set<std::string>& aTransportIds) {
+  MOZ_RELEASE_ASSERT(mInitPromise);
+
+  mInitPromise->Then(
+      mStsThread, __func__,
+      [=, self = RefPtr<MediaTransportHandlerSTS>(this)]() {
+        if (!mIceCtx) {
+          return;  // Probably due to XPCOM shutdown
+        }
+
+        for (auto it = mTransports.begin(); it != mTransports.end();) {
+          const std::string transportId(it->first);
+          if (!aTransportIds.count(transportId)) {
+            if (it->second.mFlow) {
+              OnStateChange(transportId, TransportLayer::TS_NONE);
+              OnRtcpStateChange(transportId, TransportLayer::TS_NONE);
+            }
+            // Erase the transport before destroying the ice stream so that
+            // the close_notify alerts have a chance to be sent as the
+            // TransportFlow destructors execute.
+            it = mTransports.erase(it);
+            // We're already on the STS thread, but the TransportFlow
+            // destructor executed when mTransports.erase(it) is called
+            // above dispatches the call to DestroyFinal to the STS thread. If
+            // we don't also dispatch the call to destroy the NrIceMediaStream
+            // to the STS thread, it will tear down the NrIceMediaStream
+            // before the TransportFlow is destroyed.  Without a valid
+            // NrIceMediaStream the close_notify alert cannot be sent.
+            mStsThread->Dispatch(NS_NewRunnableFunction(
+                __func__, [iceCtx = RefPtr<NrIceCtx>(mIceCtx), transportId] {
+                  iceCtx->DestroyStream(transportId);
+                }));
+          } else {
+            MOZ_ASSERT(it->second.mFlow);
+            ++it;
+          }
+        }
+      },
+      [](const std::string& aError) {});
+}
+
+void MediaTransportHandlerSTS::SendPacket(const std::string& aTransportId,
+                                          MediaPacket&& aPacket) {
+  MOZ_RELEASE_ASSERT(mInitPromise);
+
+  mInitPromise->Then(
+      mStsThread, __func__,
+      [this, self = RefPtr<MediaTransportHandlerSTS>(this), aTransportId,
+       aPacket = std::move(aPacket)]() mutable {
+        if (!mIceCtx) {
+          return;  // Probably due to XPCOM shutdown
+        }
+
+        MOZ_ASSERT(aPacket.type() != MediaPacket::UNCLASSIFIED);
+        RefPtr<TransportFlow> flow =
+            GetTransportFlow(aTransportId, aPacket.type() == MediaPacket::RTCP);
+
+        if (!flow) {
+          CSFLogError(LOGTAG,
+                      "%s: No such transport flow (%s) for outgoing packet",
+                      mIceCtx->name().c_str(), aTransportId.c_str());
+          return;
+        }
+
+        TransportLayer* layer = nullptr;
+        switch (aPacket.type()) {
+          case MediaPacket::SCTP:
+            layer = flow->GetLayer(TransportLayerDtls::ID());
+            break;
+          case MediaPacket::RTP:
+          case MediaPacket::RTCP:
+            layer = flow->GetLayer(TransportLayerSrtp::ID());
+            break;
+          default:
+            // Maybe it would be useful to allow the injection of other packet
+            // types for testing?
+            MOZ_ASSERT(false);
+            return;
+        }
+
+        MOZ_ASSERT(layer);
+
+        if (layer->SendPacket(aPacket) < 0) {
+          CSFLogError(LOGTAG, "%s: Transport flow (%s) failed to send packet",
+                      mIceCtx->name().c_str(), aTransportId.c_str());
+        }
+      },
+      [](const std::string& aError) {});
+}
+
+TransportLayer::State MediaTransportHandler::GetState(
+    const std::string& aTransportId, bool aRtcp) const {
+  // TODO Bug 1520692: we should allow Datachannel to connect without
+  // DTLS SRTP keys
+  if (mCallbackThread) {
+    MOZ_ASSERT(mCallbackThread->IsOnCurrentThread());
+  }
+
+  const std::map<std::string, TransportLayer::State>* cache = nullptr;
+  if (aRtcp) {
+    cache = &mRtcpStateCache;
+  } else {
+    cache = &mStateCache;
+  }
+
+  auto it = cache->find(aTransportId);
+  if (it != cache->end()) {
+    return it->second;
+  }
+  return TransportLayer::TS_NONE;
+}
+
+void MediaTransportHandler::OnCandidate(const std::string& aTransportId,
+                                        const CandidateInfo& aCandidateInfo) {
+  if (mCallbackThread && !mCallbackThread->IsOnCurrentThread()) {
+    mCallbackThread->Dispatch(
+        // This is being called from sigslot, which does not hold a strong ref.
+        WrapRunnable(this, &MediaTransportHandler::OnCandidate, aTransportId,
+                     aCandidateInfo),
+        NS_DISPATCH_NORMAL);
+    return;
+  }
+
+  SignalCandidate(aTransportId, aCandidateInfo);
+}
+
+void MediaTransportHandler::OnAlpnNegotiated(const std::string& aAlpn) {
+  if (mCallbackThread && !mCallbackThread->IsOnCurrentThread()) {
+    mCallbackThread->Dispatch(
+        // This is being called from sigslot, which does not hold a strong ref.
+        WrapRunnable(this, &MediaTransportHandler::OnAlpnNegotiated, aAlpn),
+        NS_DISPATCH_NORMAL);
+    return;
+  }
+
+  const bool privacyRequested = aAlpn == "c-webrtc";
+  SignalAlpnNegotiated(aAlpn, privacyRequested);
+}
+
+void MediaTransportHandler::OnGatheringStateChange(
+    dom::RTCIceGatheringState aState) {
+  if (mCallbackThread && !mCallbackThread->IsOnCurrentThread()) {
+    mCallbackThread->Dispatch(
+        // This is being called from sigslot, which does not hold a strong ref.
+        WrapRunnable(this, &MediaTransportHandler::OnGatheringStateChange,
+                     aState),
+        NS_DISPATCH_NORMAL);
+    return;
+  }
+
+  SignalGatheringStateChange(aState);
+}
+
+void MediaTransportHandler::OnConnectionStateChange(
+    dom::RTCIceConnectionState aState) {
+  if (mCallbackThread && !mCallbackThread->IsOnCurrentThread()) {
+    mCallbackThread->Dispatch(
+        // This is being called from sigslot, which does not hold a strong ref.
+        WrapRunnable(this, &MediaTransportHandler::OnConnectionStateChange,
+                     aState),
+        NS_DISPATCH_NORMAL);
+    return;
+  }
+
+  SignalConnectionStateChange(aState);
+}
+
+void MediaTransportHandler::OnPacketReceived(const std::string& aTransportId,
+                                             const MediaPacket& aPacket) {
+  if (mCallbackThread && !mCallbackThread->IsOnCurrentThread()) {
+    mCallbackThread->Dispatch(
+        // This is being called from sigslot, which does not hold a strong ref.
+        WrapRunnable(this, &MediaTransportHandler::OnPacketReceived,
+                     aTransportId, aPacket.Clone()),
+        NS_DISPATCH_NORMAL);
+    return;
+  }
+
+  SignalPacketReceived(aTransportId, aPacket);
+}
+
+void MediaTransportHandler::OnEncryptedSending(const std::string& aTransportId,
+                                               const MediaPacket& aPacket) {
+  if (mCallbackThread && !mCallbackThread->IsOnCurrentThread()) {
+    mCallbackThread->Dispatch(
+        // This is being called from sigslot, which does not hold a strong ref.
+        WrapRunnable(this, &MediaTransportHandler::OnEncryptedSending,
+                     aTransportId, aPacket.Clone()),
+        NS_DISPATCH_NORMAL);
+    return;
+  }
+
+  SignalEncryptedSending(aTransportId, aPacket);
+}
+
+void MediaTransportHandler::OnStateChange(const std::string& aTransportId,
+                                          TransportLayer::State aState) {
+  if (mCallbackThread && !mCallbackThread->IsOnCurrentThread()) {
+    mCallbackThread->Dispatch(
+        // This is being called from sigslot, which does not hold a strong ref.
+        WrapRunnable(this, &MediaTransportHandler::OnStateChange, aTransportId,
+                     aState),
+        NS_DISPATCH_NORMAL);
+    return;
+  }
+
+  if (aState == TransportLayer::TS_NONE) {
+    mStateCache.erase(aTransportId);
+  } else {
+    mStateCache[aTransportId] = aState;
+  }
+  SignalStateChange(aTransportId, aState);
+}
+
+void MediaTransportHandler::OnRtcpStateChange(const std::string& aTransportId,
+                                              TransportLayer::State aState) {
+  if (mCallbackThread && !mCallbackThread->IsOnCurrentThread()) {
+    mCallbackThread->Dispatch(
+        // This is being called from sigslot, which does not hold a strong ref.
+        WrapRunnable(this, &MediaTransportHandler::OnRtcpStateChange,
+                     aTransportId, aState),
+        NS_DISPATCH_NORMAL);
+    return;
+  }
+
+  if (aState == TransportLayer::TS_NONE) {
+    mRtcpStateCache.erase(aTransportId);
+  } else {
+    mRtcpStateCache[aTransportId] = aState;
+  }
+  SignalRtcpStateChange(aTransportId, aState);
+}
+
+RefPtr<dom::RTCStatsPromise> MediaTransportHandlerSTS::GetIceStats(
+    const std::string& aTransportId, DOMHighResTimeStamp aNow) {
+  MOZ_RELEASE_ASSERT(mInitPromise);
+
+  return mInitPromise->Then(mStsThread, __func__, [=, self = RefPtr(this)]() {
+    UniquePtr<dom::RTCStatsCollection> stats(new dom::RTCStatsCollection);
+    if (mIceCtx) {
+      for (const auto& stream : mIceCtx->GetStreams()) {
+        if (aTransportId.empty() || aTransportId == stream->GetId()) {
+          GetIceStats(*stream, aNow, stats.get());
+        }
+      }
+    }
+    return dom::RTCStatsPromise::CreateAndResolve(std::move(stats), __func__);
+  });
+}
+
+RefPtr<MediaTransportHandler::IceLogPromise>
+MediaTransportHandlerSTS::GetIceLog(const nsCString& aPattern) {
+  return InvokeAsync(
+      mStsThread, __func__, [=, self = RefPtr<MediaTransportHandlerSTS>(this)] {
+        dom::Sequence<nsString> converted;
+        RLogConnector* logs = RLogConnector::GetInstance();
+        std::deque<std::string> result;
+        // Might not exist yet.
+        if (logs) {
+          logs->Filter(aPattern.get(), 0, &result);
+        }
+        /// XXX(Bug 1631386) Check if we should reject the promise instead of
+        /// crashing in an OOM situation.
+        if (!converted.SetCapacity(result.size(), fallible)) {
+          mozalloc_handle_oom(sizeof(nsString) * result.size());
+        }
+        for (auto& line : result) {
+          // Cannot fail, SetCapacity was called before.
+          (void)converted.AppendElement(NS_ConvertUTF8toUTF16(line.c_str()),
+                                        fallible);
+        }
+        return IceLogPromise::CreateAndResolve(std::move(converted), __func__);
+      });
+}
+
+void MediaTransportHandlerSTS::ClearIceLog() {
+  if (!mStsThread->IsOnCurrentThread()) {
+    mStsThread->Dispatch(WrapRunnable(RefPtr<MediaTransportHandlerSTS>(this),
+                                      &MediaTransportHandlerSTS::ClearIceLog),
+                         NS_DISPATCH_NORMAL);
+    return;
+  }
+
+  RLogConnector* logs = RLogConnector::GetInstance();
+  if (logs) {
+    logs->Clear();
+  }
+}
+
+void MediaTransportHandlerSTS::EnterPrivateMode() {
+  if (!mStsThread->IsOnCurrentThread()) {
+    mStsThread->Dispatch(
+        WrapRunnable(RefPtr<MediaTransportHandlerSTS>(this),
+                     &MediaTransportHandlerSTS::EnterPrivateMode),
+        NS_DISPATCH_NORMAL);
+    return;
+  }
+
+  RLogConnector::GetInstance()->EnterPrivateMode();
+}
+
+void MediaTransportHandlerSTS::ExitPrivateMode() {
+  if (!mStsThread->IsOnCurrentThread()) {
+    mStsThread->Dispatch(
+        WrapRunnable(RefPtr<MediaTransportHandlerSTS>(this),
+                     &MediaTransportHandlerSTS::ExitPrivateMode),
+        NS_DISPATCH_NORMAL);
+    return;
+  }
+
+  auto* log = RLogConnector::GetInstance();
+  MOZ_ASSERT(log);
+  if (log) {
+    log->ExitPrivateMode();
+  }
+}
+
+static void ToRTCIceCandidateStats(
+    const std::vector<NrIceCandidate>& candidates,
+    dom::RTCStatsType candidateType, const nsString& transportId,
+    DOMHighResTimeStamp now, dom::RTCStatsCollection* stats,
+    bool obfuscateHostAddresses,
+    const std::set<std::string>& signaledAddresses) {
+  MOZ_ASSERT(stats);
+  for (const auto& candidate : candidates) {
+    dom::RTCIceCandidateStats cand;
+    cand.mType.Construct(candidateType);
+    NS_ConvertASCIItoUTF16 codeword(candidate.codeword.c_str());
+    cand.mTransportId.Construct(transportId);
+    cand.mId.Construct(codeword);
+    cand.mTimestamp.Construct(now);
+    cand.mCandidateType.Construct(dom::RTCIceCandidateType(candidate.type));
+    cand.mPriority.Construct(candidate.priority);
+    // https://tools.ietf.org/html/draft-ietf-rtcweb-mdns-ice-candidates-03#section-3.3.1
+    // This obfuscates the address with the mDNS address if one exists
+    if (!candidate.mdns_addr.empty()) {
+      cand.mAddress.Construct(
+          NS_ConvertASCIItoUTF16(candidate.mdns_addr.c_str()));
+    } else if (obfuscateHostAddresses &&
+               candidate.type == NrIceCandidate::ICE_PEER_REFLEXIVE &&
+               signaledAddresses.find(candidate.cand_addr.host) ==
+                   signaledAddresses.end()) {
+      cand.mAddress.Construct(NS_ConvertASCIItoUTF16("(redacted)"));
+    } else {
+      cand.mAddress.Construct(
+          NS_ConvertASCIItoUTF16(candidate.cand_addr.host.c_str()));
+    }
+    cand.mPort.Construct(candidate.cand_addr.port);
+    cand.mProtocol.Construct(
+        NS_ConvertASCIItoUTF16(candidate.cand_addr.transport.c_str()));
+    if (candidateType == dom::RTCStatsType::Local_candidate &&
+        dom::RTCIceCandidateType(candidate.type) ==
+            dom::RTCIceCandidateType::Relay) {
+      cand.mRelayProtocol.Construct(
+          NS_ConvertASCIItoUTF16(candidate.local_addr.transport.c_str()));
+    }
+    cand.mProxied.Construct(NS_ConvertASCIItoUTF16(
+        candidate.is_proxied ? "proxied" : "non-proxied"));
+    if (!stats->mIceCandidateStats.AppendElement(cand, fallible)) {
+      // XXX(Bug 1632090) Instead of extending the array 1-by-1 (which might
+      // involve multiple reallocations) and potentially crashing here,
+      // SetCapacity could be called outside the loop once.
+      mozalloc_handle_oom(0);
+    }
+    if (candidate.trickled) {
+      if (!stats->mTrickledIceCandidateStats.AppendElement(cand, fallible)) {
+        mozalloc_handle_oom(0);
+      }
+    }
+  }
+}
+
+void MediaTransportHandlerSTS::GetIceStats(
+    const NrIceMediaStream& aStream, DOMHighResTimeStamp aNow,
+    dom::RTCStatsCollection* aStats) const {
+  MOZ_ASSERT(mStsThread->IsOnCurrentThread());
+
+  NS_ConvertASCIItoUTF16 transportId(aStream.GetId().c_str());
+
+  std::vector<NrIceCandidatePair> candPairs;
+  nsresult res = aStream.GetCandidatePairs(&candPairs);
+  if (NS_FAILED(res)) {
+    CSFLogError(LOGTAG,
+                "%s: Error getting candidate pairs for transport id \"%s\"",
+                __FUNCTION__, aStream.GetId().c_str());
+    return;
+  }
+
+  for (auto& candPair : candPairs) {
+    NS_ConvertASCIItoUTF16 codeword(candPair.codeword.c_str());
+    NS_ConvertASCIItoUTF16 localCodeword(candPair.local.codeword.c_str());
+    NS_ConvertASCIItoUTF16 remoteCodeword(candPair.remote.codeword.c_str());
+    // Only expose candidate-pair statistics to chrome, until we've thought
+    // through the implications of exposing it to content.
+
+    dom::RTCIceCandidatePairStats s;
+    s.mId.Construct(codeword);
+    s.mTransportId.Construct(transportId);
+    s.mTimestamp.Construct(aNow);
+    s.mType.Construct(dom::RTCStatsType::Candidate_pair);
+    s.mLocalCandidateId.Construct(localCodeword);
+    s.mRemoteCandidateId.Construct(remoteCodeword);
+    s.mNominated.Construct(candPair.nominated);
+    s.mWritable.Construct(candPair.writable);
+    s.mReadable.Construct(candPair.readable);
+    s.mPriority.Construct(candPair.priority);
+    s.mSelected.Construct(candPair.selected);
+    s.mBytesSent.Construct(candPair.bytes_sent);
+    s.mBytesReceived.Construct(candPair.bytes_recvd);
+    s.mLastPacketSentTimestamp.Construct(candPair.ms_since_last_send);
+    s.mLastPacketReceivedTimestamp.Construct(candPair.ms_since_last_recv);
+    s.mState.Construct(dom::RTCStatsIceCandidatePairState(candPair.state));
+    s.mComponentId.Construct(candPair.component_id);
+    if (!aStats->mIceCandidatePairStats.AppendElement(s, fallible)) {
+      // XXX(Bug 1632090) Instead of extending the array 1-by-1 (which might
+      // involve multiple reallocations) and potentially crashing here,
+      // SetCapacity could be called outside the loop once.
+      mozalloc_handle_oom(0);
+    }
+  }
+
+  std::vector<NrIceCandidate> candidates;
+  if (NS_SUCCEEDED(aStream.GetLocalCandidates(&candidates))) {
+    ToRTCIceCandidateStats(candidates, dom::RTCStatsType::Local_candidate,
+                           transportId, aNow, aStats, mObfuscateHostAddresses,
+                           mSignaledAddresses);
+    // add the local candidates unparsed string to a sequence
+    for (const auto& candidate : candidates) {
+      if (!aStats->mRawLocalCandidates.AppendElement(
+              NS_ConvertASCIItoUTF16(candidate.label.c_str()), fallible)) {
+        // XXX(Bug 1632090) Instead of extending the array 1-by-1 (which might
+        // involve multiple reallocations) and potentially crashing here,
+        // SetCapacity could be called outside the loop once.
+        mozalloc_handle_oom(0);
+      }
+    }
+  }
+  candidates.clear();
+
+  if (NS_SUCCEEDED(aStream.GetRemoteCandidates(&candidates))) {
+    ToRTCIceCandidateStats(candidates, dom::RTCStatsType::Remote_candidate,
+                           transportId, aNow, aStats, mObfuscateHostAddresses,
+                           mSignaledAddresses);
+    // add the remote candidates unparsed string to a sequence
+    for (const auto& candidate : candidates) {
+      if (!aStats->mRawRemoteCandidates.AppendElement(
+              NS_ConvertASCIItoUTF16(candidate.label.c_str()), fallible)) {
+        // XXX(Bug 1632090) Instead of extending the array 1-by-1 (which might
+        // involve multiple reallocations) and potentially crashing here,
+        // SetCapacity could be called outside the loop once.
+        mozalloc_handle_oom(0);
+      }
+    }
+  }
+}
+
+RefPtr<TransportFlow> MediaTransportHandlerSTS::GetTransportFlow(
+    const std::string& aTransportId, bool aIsRtcp) const {
+  auto it = mTransports.find(aTransportId);
+  if (it == mTransports.end()) {
+    return nullptr;
+  }
+
+  if (aIsRtcp) {
+    return it->second.mRtcpFlow ? it->second.mRtcpFlow : it->second.mFlow;
+    ;
+  }
+
+  return it->second.mFlow;
+}
+
+RefPtr<TransportFlow> MediaTransportHandlerSTS::CreateTransportFlow(
+    const std::string& aTransportId, bool aIsRtcp,
+    const RefPtr<DtlsIdentity>& aDtlsIdentity, bool aDtlsClient,
+    const DtlsDigestList& aDigests, bool aPrivacyRequested) {
+  nsresult rv;
+  RefPtr<TransportFlow> flow = new TransportFlow(aTransportId);
+
+  // The media streams are made on STS so we need to defer setup.
+  auto ice = MakeUnique<TransportLayerIce>();
+  auto dtls = MakeUnique<TransportLayerDtls>();
+  auto srtp = MakeUnique<TransportLayerSrtp>(*dtls);
+  dtls->SetRole(aDtlsClient ? TransportLayerDtls::CLIENT
+                            : TransportLayerDtls::SERVER);
+
+  dtls->SetIdentity(aDtlsIdentity);
+
+  dtls->SetMinMaxVersion(
+      static_cast<TransportLayerDtls::Version>(mMinDtlsVersion),
+      static_cast<TransportLayerDtls::Version>(mMaxDtlsVersion));
+
+  for (const auto& digest : aDigests) {
+    rv = dtls->SetVerificationDigest(digest);
+    if (NS_FAILED(rv)) {
+      CSFLogError(LOGTAG, "Could not set fingerprint");
+      return nullptr;
+    }
+  }
+
+  std::vector<uint16_t> srtpCiphers =
+      TransportLayerDtls::GetDefaultSrtpCiphers();
+
+  rv = dtls->SetSrtpCiphers(srtpCiphers);
+  if (NS_FAILED(rv)) {
+    CSFLogError(LOGTAG, "Couldn't set SRTP ciphers");
+    return nullptr;
+  }
+
+  // Always permits negotiation of the confidential mode.
+  // Only allow non-confidential (which is an allowed default),
+  // if we aren't confidential.
+  std::set<std::string> alpn = {"c-webrtc"};
+  std::string alpnDefault;
+  if (!aPrivacyRequested) {
+    alpnDefault = "webrtc";
+    alpn.insert(alpnDefault);
+  }
+  rv = dtls->SetAlpn(alpn, alpnDefault);
+  if (NS_FAILED(rv)) {
+    CSFLogError(LOGTAG, "Couldn't set ALPN");
+    return nullptr;
+  }
+
+  ice->SetParameters(mIceCtx->GetStream(aTransportId), aIsRtcp ? 2 : 1);
+  NS_ENSURE_SUCCESS(ice->Init(), nullptr);
+  NS_ENSURE_SUCCESS(dtls->Init(), nullptr);
+  NS_ENSURE_SUCCESS(srtp->Init(), nullptr);
+  dtls->Chain(ice.get());
+  srtp->Chain(ice.get());
+
+  dtls->SignalPacketReceived.connect(this,
+                                     &MediaTransportHandlerSTS::PacketReceived);
+  srtp->SignalPacketReceived.connect(this,
+                                     &MediaTransportHandlerSTS::PacketReceived);
+  ice->SignalPacketSending.connect(
+      this, &MediaTransportHandlerSTS::EncryptedPacketSending);
+  flow->PushLayer(ice.release());
+  flow->PushLayer(dtls.release());
+  flow->PushLayer(srtp.release());
+  return flow;
+}
+
+static mozilla::dom::RTCIceGatheringState toDomIceGatheringState(
+    NrIceCtx::GatheringState aState) {
+  switch (aState) {
+    case NrIceCtx::ICE_CTX_GATHER_INIT:
+      return dom::RTCIceGatheringState::New;
+    case NrIceCtx::ICE_CTX_GATHER_STARTED:
+      return dom::RTCIceGatheringState::Gathering;
+    case NrIceCtx::ICE_CTX_GATHER_COMPLETE:
+      return dom::RTCIceGatheringState::Complete;
+  }
+  MOZ_CRASH();
+}
+
+void MediaTransportHandlerSTS::OnGatheringStateChange(
+    NrIceCtx* aIceCtx, NrIceCtx::GatheringState aState) {
+  OnGatheringStateChange(toDomIceGatheringState(aState));
+}
+
+static mozilla::dom::RTCIceConnectionState toDomIceConnectionState(
+    NrIceCtx::ConnectionState aState) {
+  switch (aState) {
+    case NrIceCtx::ICE_CTX_INIT:
+      return dom::RTCIceConnectionState::New;
+    case NrIceCtx::ICE_CTX_CHECKING:
+      return dom::RTCIceConnectionState::Checking;
+    case NrIceCtx::ICE_CTX_CONNECTED:
+      return dom::RTCIceConnectionState::Connected;
+    case NrIceCtx::ICE_CTX_COMPLETED:
+      return dom::RTCIceConnectionState::Completed;
+    case NrIceCtx::ICE_CTX_FAILED:
+      return dom::RTCIceConnectionState::Failed;
+    case NrIceCtx::ICE_CTX_DISCONNECTED:
+      return dom::RTCIceConnectionState::Disconnected;
+    case NrIceCtx::ICE_CTX_CLOSED:
+      return dom::RTCIceConnectionState::Closed;
+  }
+  MOZ_CRASH();
+}
+
+void MediaTransportHandlerSTS::OnConnectionStateChange(
+    NrIceCtx* aIceCtx, NrIceCtx::ConnectionState aState) {
+  OnConnectionStateChange(toDomIceConnectionState(aState));
+}
+
+// The stuff below here will eventually go into the MediaTransportChild class
+void MediaTransportHandlerSTS::OnCandidateFound(
+    NrIceMediaStream* aStream, const std::string& aCandidate,
+    const std::string& aUfrag, const std::string& aMDNSAddr,
+    const std::string& aActualAddr) {
+  CandidateInfo info;
+  info.mCandidate = aCandidate;
+  MOZ_ASSERT(!aUfrag.empty());
+  info.mUfrag = aUfrag;
+  NrIceCandidate defaultRtpCandidate;
+  NrIceCandidate defaultRtcpCandidate;
+  nsresult rv = aStream->GetDefaultCandidate(1, &defaultRtpCandidate);
+  if (NS_SUCCEEDED(rv)) {
+    if (!defaultRtpCandidate.mdns_addr.empty()) {
+      info.mDefaultHostRtp = "0.0.0.0";
+      info.mDefaultPortRtp = 9;
+    } else {
+      info.mDefaultHostRtp = defaultRtpCandidate.cand_addr.host;
+      info.mDefaultPortRtp = defaultRtpCandidate.cand_addr.port;
+    }
+  } else {
+    CSFLogError(LOGTAG,
+                "%s: GetDefaultCandidates failed for transport id %s, "
+                "res=%u",
+                __FUNCTION__, aStream->GetId().c_str(),
+                static_cast<unsigned>(rv));
+  }
+
+  // Optional; component won't exist if doing rtcp-mux
+  if (NS_SUCCEEDED(aStream->GetDefaultCandidate(2, &defaultRtcpCandidate))) {
+    if (!defaultRtcpCandidate.mdns_addr.empty()) {
+      info.mDefaultHostRtcp = defaultRtcpCandidate.mdns_addr;
+    } else {
+      info.mDefaultHostRtcp = defaultRtcpCandidate.cand_addr.host;
+    }
+    info.mDefaultPortRtcp = defaultRtcpCandidate.cand_addr.port;
+  }
+
+  info.mMDNSAddress = aMDNSAddr;
+  info.mActualAddress = aActualAddr;
+
+  OnCandidate(aStream->GetId(), info);
+}
+
+void MediaTransportHandlerSTS::OnStateChange(TransportLayer* aLayer,
+                                             TransportLayer::State aState) {
+  if (aState == TransportLayer::TS_OPEN) {
+    MOZ_ASSERT(aLayer->id() == TransportLayerDtls::ID());
+    TransportLayerDtls* dtlsLayer = static_cast<TransportLayerDtls*>(aLayer);
+    OnAlpnNegotiated(dtlsLayer->GetNegotiatedAlpn());
+  }
+
+  // DTLS state indicates the readiness of the transport as a whole, because
+  // SRTP uses the keys from the DTLS handshake.
+  MediaTransportHandler::OnStateChange(aLayer->flow_id(), aState);
+}
+
+void MediaTransportHandlerSTS::OnRtcpStateChange(TransportLayer* aLayer,
+                                                 TransportLayer::State aState) {
+  MediaTransportHandler::OnRtcpStateChange(aLayer->flow_id(), aState);
+}
+
+constexpr static const char* PacketTypeToString(MediaPacket::Type type) {
+  switch (type) {
+    case MediaPacket::Type::UNCLASSIFIED:
+      return "UNCLASSIFIED";
+    case MediaPacket::Type::SRTP:
+      return "SRTP";
+    case MediaPacket::Type::SRTCP:
+      return "SRTCP";
+    case MediaPacket::Type::DTLS:
+      return "DTLS";
+    case MediaPacket::Type::RTP:
+      return "RTP";
+    case MediaPacket::Type::RTCP:
+      return "RTCP";
+    case MediaPacket::Type::SCTP:
+      return "SCTP";
+    default:
+      MOZ_ASSERT(false, "unreached");
+      return "";
+  }
+}
+
+void MediaTransportHandlerSTS::PacketReceived(TransportLayer* aLayer,
+                                              MediaPacket& aPacket) {
+  MEDIA_TRANSPORT_HANDLER_PACKET_RECEIVED(aPacket);
+  OnPacketReceived(aLayer->flow_id(), aPacket);
+}
+
+void MediaTransportHandlerSTS::EncryptedPacketSending(TransportLayer* aLayer,
+                                                      MediaPacket& aPacket) {
+  OnEncryptedSending(aLayer->flow_id(), aPacket);
+}
+
+}  // namespace mozilla
+
+#undef MEDIA_TRANSPORT_HANDLER_PACKET_RECEIVED
diff --git a/dom/media/webrtc/jsapi/MediaTransportHandler.h b/dom/media/webrtc/jsapi/MediaTransportHandler.h
new file mode 100644
index 0000000000..a776cb6fd7
--- /dev/null
+++ b/dom/media/webrtc/jsapi/MediaTransportHandler.h
@@ -0,0 +1,167 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _MTRANSPORTHANDLER_H__
+#define _MTRANSPORTHANDLER_H__
+
+#include "mozilla/RefPtr.h"
+#include "nsISupportsImpl.h"
+#include "transport/sigslot.h"
+#include "transport/transportlayer.h"  // Need the State enum
+#include "transport/dtlsidentity.h"    // For DtlsDigest
+#include "mozilla/dom/RTCPeerConnectionBinding.h"
+#include "mozilla/dom/RTCConfigurationBinding.h"
+#include "transport/nricectx.h"  // Need some enums
+#include "common/CandidateInfo.h"
+#include "transport/nr_socket_proxy_config.h"
+#include "RTCStatsReport.h"
+
+#include "nsString.h"
+
+#include <string>
+#include <set>
+#include <vector>
+
+namespace mozilla {
+class DtlsIdentity;
+class NrIceCtx;
+class NrIceMediaStream;
+class NrIceResolver;
+class TransportFlow;
+class RTCStatsQuery;
+
+namespace dom {
+struct RTCStatsReportInternal;
+}
+
+class MediaTransportHandler {
+ public:
+  // Creates either a MediaTransportHandlerSTS or a MediaTransportHandlerIPC,
+  // as appropriate. If you want signals to fire on a specific thread, pass
+  // the event target here, otherwise they will fire on whatever is convenient.
+  // Note: This also determines what thread the state cache is updated on!
+  // Don't call GetState on any other thread!
+  static already_AddRefed<MediaTransportHandler> Create(
+      nsISerialEventTarget* aCallbackThread);
+
+  explicit MediaTransportHandler(nsISerialEventTarget* aCallbackThread)
+      : mCallbackThread(aCallbackThread) {}
+
+  // Exposed so we can synchronously validate ICE servers from PeerConnection
+  static nsresult ConvertIceServers(
+      const nsTArray<dom::RTCIceServer>& aIceServers,
+      std::vector<NrIceStunServer>* aStunServers,
+      std::vector<NrIceTurnServer>* aTurnServers);
+
+  typedef MozPromise<dom::Sequence<nsString>, nsresult, true> IceLogPromise;
+
+  virtual void Initialize() {}
+
+  // There's a wrinkle here; the ICE logging is not separated out by
+  // MediaTransportHandler. These are a little more like static methods, but
+  // to avoid needing yet another IPC interface, we bolt them on here.
+  virtual RefPtr<IceLogPromise> GetIceLog(const nsCString& aPattern) = 0;
+  virtual void ClearIceLog() = 0;
+  virtual void EnterPrivateMode() = 0;
+  virtual void ExitPrivateMode() = 0;
+
+  virtual void CreateIceCtx(const std::string& aName) = 0;
+
+  virtual nsresult SetIceConfig(const nsTArray<dom::RTCIceServer>& aIceServers,
+                                dom::RTCIceTransportPolicy aIcePolicy) = 0;
+
+  // We will probably be able to move the proxy lookup stuff into
+  // this class once we move mtransport to its own process.
+  virtual void SetProxyConfig(NrSocketProxyConfig&& aProxyConfig) = 0;
+
+  virtual void EnsureProvisionalTransport(const std::string& aTransportId,
+                                          const std::string& aLocalUfrag,
+                                          const std::string& aLocalPwd,
+                                          int aComponentCount) = 0;
+
+  virtual void SetTargetForDefaultLocalAddressLookup(
+      const std::string& aTargetIp, uint16_t aTargetPort) = 0;
+
+  // We set default-route-only as late as possible because it depends on what
+  // capture permissions have been granted on the window, which could easily
+  // change between Init (ie; when the PC is created) and StartIceGathering
+  // (ie; when we set the local description).
+  virtual void StartIceGathering(bool aDefaultRouteOnly,
+                                 bool aObfuscateHostAddresses,
+                                 // TODO: It probably makes sense to look
+                                 // this up internally
+                                 const nsTArray<NrIceStunAddr>& aStunAddrs) = 0;
+
+  virtual void ActivateTransport(
+      const std::string& aTransportId, const std::string& aLocalUfrag,
+      const std::string& aLocalPwd, size_t aComponentCount,
+      const std::string& aUfrag, const std::string& aPassword,
+      const nsTArray<uint8_t>& aKeyDer, const nsTArray<uint8_t>& aCertDer,
+      SSLKEAType aAuthType, bool aDtlsClient, const DtlsDigestList& aDigests,
+      bool aPrivacyRequested) = 0;
+
+  virtual void RemoveTransportsExcept(
+      const std::set<std::string>& aTransportIds) = 0;
+
+  virtual void StartIceChecks(bool aIsControlling,
+                              const std::vector<std::string>& aIceOptions) = 0;
+
+  virtual void SendPacket(const std::string& aTransportId,
+                          MediaPacket&& aPacket) = 0;
+
+  virtual void AddIceCandidate(const std::string& aTransportId,
+                               const std::string& aCandidate,
+                               const std::string& aUFrag,
+                               const std::string& aObfuscatedAddress) = 0;
+
+  virtual void UpdateNetworkState(bool aOnline) = 0;
+
+  virtual RefPtr<dom::RTCStatsPromise> GetIceStats(
+      const std::string& aTransportId, DOMHighResTimeStamp aNow) = 0;
+
+  sigslot::signal2<const std::string&, const CandidateInfo&> SignalCandidate;
+  sigslot::signal2<const std::string&, bool> SignalAlpnNegotiated;
+  sigslot::signal1<dom::RTCIceGatheringState> SignalGatheringStateChange;
+  sigslot::signal1<dom::RTCIceConnectionState> SignalConnectionStateChange;
+
+  sigslot::signal2<const std::string&, const MediaPacket&> SignalPacketReceived;
+  sigslot::signal2<const std::string&, const MediaPacket&>
+      SignalEncryptedSending;
+  sigslot::signal2<const std::string&, TransportLayer::State> SignalStateChange;
+  sigslot::signal2<const std::string&, TransportLayer::State>
+      SignalRtcpStateChange;
+
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING_WITH_DESTROY(MediaTransportHandler,
+                                                     Destroy())
+
+  TransportLayer::State GetState(const std::string& aTransportId,
+                                 bool aRtcp) const;
+
+ protected:
+  void OnCandidate(const std::string& aTransportId,
+                   const CandidateInfo& aCandidateInfo);
+  void OnAlpnNegotiated(const std::string& aAlpn);
+  void OnGatheringStateChange(dom::RTCIceGatheringState aState);
+  void OnConnectionStateChange(dom::RTCIceConnectionState aState);
+  void OnPacketReceived(const std::string& aTransportId,
+                        const MediaPacket& aPacket);
+  void OnEncryptedSending(const std::string& aTransportId,
+                          const MediaPacket& aPacket);
+  void OnStateChange(const std::string& aTransportId,
+                     TransportLayer::State aState);
+  void OnRtcpStateChange(const std::string& aTransportId,
+                         TransportLayer::State aState);
+  virtual void Destroy() = 0;
+  virtual ~MediaTransportHandler() = default;
+  std::map<std::string, TransportLayer::State> mStateCache;
+  std::map<std::string, TransportLayer::State> mRtcpStateCache;
+  RefPtr<nsISerialEventTarget> mCallbackThread;
+};
+
+void TokenizeCandidate(const std::string& aCandidate,
+                       std::vector<std::string>& aTokens);
+
+}  // namespace mozilla
+
+#endif  //_MTRANSPORTHANDLER_H__
diff --git a/dom/media/webrtc/jsapi/MediaTransportHandlerIPC.cpp b/dom/media/webrtc/jsapi/MediaTransportHandlerIPC.cpp
new file mode 100644
index 0000000000..6369d229ee
--- /dev/null
+++ b/dom/media/webrtc/jsapi/MediaTransportHandlerIPC.cpp
@@ -0,0 +1,414 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "MediaTransportHandlerIPC.h"
+#include "mozilla/dom/MediaTransportChild.h"
+#include "nsThreadUtils.h"
+#include "mozilla/net/SocketProcessBridgeChild.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/ipc/BackgroundChild.h"
+#include "mozilla/ipc/PBackgroundChild.h"
+#include "common/browser_logging/CSFLog.h"
+
+namespace mozilla {
+
+static const char* mthipcLogTag = "MediaTransportHandler";
+#ifdef LOGTAG
+#  undef LOGTAG
+#endif
+#define LOGTAG mthipcLogTag
+
+MediaTransportHandlerIPC::MediaTransportHandlerIPC(
+    nsISerialEventTarget* aCallbackThread)
+    : MediaTransportHandler(aCallbackThread) {}
+
+void MediaTransportHandlerIPC::Initialize() {
+  mInitPromise = net::SocketProcessBridgeChild::GetSocketProcessBridge()->Then(
+      mCallbackThread, __func__,
+      [this, self = RefPtr<MediaTransportHandlerIPC>(this)](
+          const RefPtr<net::SocketProcessBridgeChild>& aBridge) {
+        ipc::PBackgroundChild* actor =
+            ipc::BackgroundChild::GetOrCreateSocketActorForCurrentThread();
+        // An actor that can't send is possible if the socket process has
+        // crashed but hasn't been reconnected properly. See
+        // SocketProcessBridgeChild::ActorDestroy for more info.
+        if (!actor || !actor->CanSend()) {
+          NS_WARNING(
+              "MediaTransportHandlerIPC async init failed! Webrtc networking "
+              "will not work!");
+          return InitPromise::CreateAndReject(
+              nsCString("GetOrCreateSocketActorForCurrentThread failed!"),
+              __func__);
+        }
+        MediaTransportChild* child = new MediaTransportChild(this);
+        // PBackgroungChild owns mChild! When it is done with it,
+        // mChild will let us know it it going away.
+        mChild = actor->SendPMediaTransportConstructor(child);
+        CSFLogDebug(LOGTAG, "%s Init done", __func__);
+        return InitPromise::CreateAndResolve(true, __func__);
+      },
+      [=](const nsCString& aError) {
+        CSFLogError(LOGTAG,
+                    "MediaTransportHandlerIPC async init failed! Webrtc "
+                    "networking will not work! Error was %s",
+                    aError.get());
+        NS_WARNING(
+            "MediaTransportHandlerIPC async init failed! Webrtc networking "
+            "will not work!");
+        return InitPromise::CreateAndReject(aError, __func__);
+      });
+}
+
+RefPtr<MediaTransportHandler::IceLogPromise>
+MediaTransportHandlerIPC::GetIceLog(const nsCString& aPattern) {
+  return mInitPromise->Then(
+      mCallbackThread, __func__,
+      [=, self = RefPtr<MediaTransportHandlerIPC>(this)](bool /* dummy */) {
+        if (!mChild) {
+          return IceLogPromise::CreateAndReject(NS_ERROR_FAILURE, __func__);
+        }
+        // Compiler has trouble deducing the return type here for some reason,
+        // so we use a temp variable as a hint.
+        // SendGetIceLog _almost_ returns an IceLogPromise; the reject value
+        // differs (ipc::ResponseRejectReason vs nsresult) so we need to
+        // convert.
+        RefPtr<IceLogPromise> promise = mChild->SendGetIceLog(aPattern)->Then(
+            mCallbackThread, __func__,
+            [](WebrtcGlobalLog&& aLogLines) {
+              return IceLogPromise::CreateAndResolve(std::move(aLogLines),
+                                                     __func__);
+            },
+            [](ipc::ResponseRejectReason aReason) {
+              return IceLogPromise::CreateAndReject(NS_ERROR_FAILURE, __func__);
+            });
+        return promise;
+      },
+      [](const nsCString& aError) {
+        return IceLogPromise::CreateAndReject(NS_ERROR_FAILURE, __func__);
+      });
+}
+
+void MediaTransportHandlerIPC::ClearIceLog() {
+  mInitPromise->Then(
+      mCallbackThread, __func__,
+      [=, self = RefPtr<MediaTransportHandlerIPC>(this)](bool /*dummy*/) {
+        if (mChild) {
+          mChild->SendClearIceLog();
+        }
+      },
+      [](const nsCString& aError) {});
+}
+
+void MediaTransportHandlerIPC::EnterPrivateMode() {
+  mInitPromise->Then(
+      mCallbackThread, __func__,
+      [=, self = RefPtr<MediaTransportHandlerIPC>(this)](bool /*dummy*/) {
+        if (mChild) {
+          mChild->SendEnterPrivateMode();
+        }
+      },
+      [](const nsCString& aError) {});
+}
+
+void MediaTransportHandlerIPC::ExitPrivateMode() {
+  mInitPromise->Then(
+      mCallbackThread, __func__,
+      [=, self = RefPtr<MediaTransportHandlerIPC>(this)](bool /*dummy*/) {
+        if (mChild) {
+          mChild->SendExitPrivateMode();
+        }
+      },
+      [](const nsCString& aError) {});
+}
+
+void MediaTransportHandlerIPC::CreateIceCtx(const std::string& aName) {
+  CSFLogDebug(LOGTAG, "MediaTransportHandlerIPC::CreateIceCtx start");
+
+  mInitPromise->Then(
+      mCallbackThread, __func__,
+      [=, self = RefPtr<MediaTransportHandlerIPC>(this)](bool /*dummy*/) {
+        if (mChild) {
+          CSFLogDebug(LOGTAG, "%s starting", __func__);
+          if (NS_WARN_IF(!mChild->SendCreateIceCtx(aName))) {
+            CSFLogError(LOGTAG, "%s failed!", __func__);
+          }
+        }
+      },
+      [](const nsCString& aError) {});
+}
+
+nsresult MediaTransportHandlerIPC::SetIceConfig(
+    const nsTArray<dom::RTCIceServer>& aIceServers,
+    dom::RTCIceTransportPolicy aIcePolicy) {
+  // Run some validation on this side of the IPC boundary so we can return
+  // errors synchronously. We don't actually use the results. It might make
+  // sense to move this check to PeerConnection and have this API take the
+  // converted form, but we would need to write IPC serialization code for
+  // the NrIce*Server types.
+  std::vector<NrIceStunServer> stunServers;
+  std::vector<NrIceTurnServer> turnServers;
+  nsresult rv = ConvertIceServers(aIceServers, &stunServers, &turnServers);
+  if (NS_FAILED(rv)) {
+    return rv;
+  }
+
+  mInitPromise->Then(
+      mCallbackThread, __func__,
+      [=, iceServers = aIceServers.Clone(),
+       self = RefPtr<MediaTransportHandlerIPC>(this)](bool /*dummy*/) {
+        if (mChild) {
+          if (NS_WARN_IF(!mChild->SendSetIceConfig(std::move(iceServers),
+                                                   aIcePolicy))) {
+            CSFLogError(LOGTAG, "%s failed!", __func__);
+          }
+        }
+      },
+      [](const nsCString& aError) {});
+
+  return NS_OK;
+}
+
+void MediaTransportHandlerIPC::Destroy() {
+  if (mChild) {
+    MediaTransportChild::Send__delete__(mChild);
+    mChild = nullptr;
+  }
+  delete this;
+}
+
+// We will probably be able to move the proxy lookup stuff into
+// this class once we move mtransport to its own process.
+void MediaTransportHandlerIPC::SetProxyConfig(
+    NrSocketProxyConfig&& aProxyConfig) {
+  mInitPromise->Then(
+      mCallbackThread, __func__,
+      [aProxyConfig = std::move(aProxyConfig), this,
+       self = RefPtr<MediaTransportHandlerIPC>(this)](bool /*dummy*/) mutable {
+        if (mChild) {
+          mChild->SendSetProxyConfig(aProxyConfig.GetConfig());
+        }
+      },
+      [](const nsCString& aError) {});
+}
+
+void MediaTransportHandlerIPC::EnsureProvisionalTransport(
+    const std::string& aTransportId, const std::string& aLocalUfrag,
+    const std::string& aLocalPwd, int aComponentCount) {
+  mInitPromise->Then(
+      mCallbackThread, __func__,
+      [=, self = RefPtr<MediaTransportHandlerIPC>(this)](bool /*dummy*/) {
+        if (mChild) {
+          mChild->SendEnsureProvisionalTransport(aTransportId, aLocalUfrag,
+                                                 aLocalPwd, aComponentCount);
+        }
+      },
+      [](const nsCString& aError) {});
+}
+
+void MediaTransportHandlerIPC::SetTargetForDefaultLocalAddressLookup(
+    const std::string& aTargetIp, uint16_t aTargetPort) {
+  mInitPromise->Then(
+      mCallbackThread, __func__,
+      [=, self = RefPtr<MediaTransportHandlerIPC>(this)](bool /*dummy*/) {
+        if (mChild) {
+          mChild->SendSetTargetForDefaultLocalAddressLookup(aTargetIp,
+                                                            aTargetPort);
+        }
+      },
+      [](const nsCString& aError) {});
+}
+
+// We set default-route-only as late as possible because it depends on what
+// capture permissions have been granted on the window, which could easily
+// change between Init (ie; when the PC is created) and StartIceGathering
+// (ie; when we set the local description).
+void MediaTransportHandlerIPC::StartIceGathering(
+    bool aDefaultRouteOnly, bool aObfuscateHostAddresses,
+    // TODO(bug 1522205): It probably makes sense to look this up internally
+    const nsTArray<NrIceStunAddr>& aStunAddrs) {
+  mInitPromise->Then(
+      mCallbackThread, __func__,
+      [=, stunAddrs = aStunAddrs.Clone(),
+       self = RefPtr<MediaTransportHandlerIPC>(this)](bool /*dummy*/) {
+        if (mChild) {
+          mChild->SendStartIceGathering(aDefaultRouteOnly,
+                                        aObfuscateHostAddresses, stunAddrs);
+        }
+      },
+      [](const nsCString& aError) {});
+}
+
+void MediaTransportHandlerIPC::ActivateTransport(
+    const std::string& aTransportId, const std::string& aLocalUfrag,
+    const std::string& aLocalPwd, size_t aComponentCount,
+    const std::string& aUfrag, const std::string& aPassword,
+    const nsTArray<uint8_t>& aKeyDer, const nsTArray<uint8_t>& aCertDer,
+    SSLKEAType aAuthType, bool aDtlsClient, const DtlsDigestList& aDigests,
+    bool aPrivacyRequested) {
+  mInitPromise->Then(
+      mCallbackThread, __func__,
+      [=, keyDer = aKeyDer.Clone(), certDer = aCertDer.Clone(),
+       self = RefPtr<MediaTransportHandlerIPC>(this)](bool /*dummy*/) {
+        if (mChild) {
+          mChild->SendActivateTransport(aTransportId, aLocalUfrag, aLocalPwd,
+                                        aComponentCount, aUfrag, aPassword,
+                                        keyDer, certDer, aAuthType, aDtlsClient,
+                                        aDigests, aPrivacyRequested);
+        }
+      },
+      [](const nsCString& aError) {});
+}
+
+void MediaTransportHandlerIPC::RemoveTransportsExcept(
+    const std::set<std::string>& aTransportIds) {
+  std::vector<std::string> transportIds(aTransportIds.begin(),
+                                        aTransportIds.end());
+  mInitPromise->Then(
+      mCallbackThread, __func__,
+      [=, self = RefPtr<MediaTransportHandlerIPC>(this)](bool /*dummy*/) {
+        if (mChild) {
+          mChild->SendRemoveTransportsExcept(transportIds);
+        }
+      },
+      [](const nsCString& aError) {});
+}
+
+void MediaTransportHandlerIPC::StartIceChecks(
+    bool aIsControlling, const std::vector<std::string>& aIceOptions) {
+  mInitPromise->Then(
+      mCallbackThread, __func__,
+      [=, self = RefPtr<MediaTransportHandlerIPC>(this)](bool /*dummy*/) {
+        if (mChild) {
+          mChild->SendStartIceChecks(aIsControlling, aIceOptions);
+        }
+      },
+      [](const nsCString& aError) {});
+}
+
+void MediaTransportHandlerIPC::SendPacket(const std::string& aTransportId,
+                                          MediaPacket&& aPacket) {
+  mInitPromise->Then(
+      mCallbackThread, __func__,
+      [this, self = RefPtr<MediaTransportHandlerIPC>(this), aTransportId,
+       aPacket = std::move(aPacket)](bool /*dummy*/) mutable {
+        if (mChild) {
+          mChild->SendSendPacket(aTransportId, aPacket);
+        }
+      },
+      [](const nsCString& aError) {});
+}
+
+void MediaTransportHandlerIPC::AddIceCandidate(
+    const std::string& aTransportId, const std::string& aCandidate,
+    const std::string& aUfrag, const std::string& aObfuscatedAddress) {
+  mInitPromise->Then(
+      mCallbackThread, __func__,
+      [=, self = RefPtr<MediaTransportHandlerIPC>(this)](bool /*dummy*/) {
+        if (mChild) {
+          mChild->SendAddIceCandidate(aTransportId, aCandidate, aUfrag,
+                                      aObfuscatedAddress);
+        }
+      },
+      [](const nsCString& aError) {});
+}
+
+void MediaTransportHandlerIPC::UpdateNetworkState(bool aOnline) {
+  mInitPromise->Then(
+      mCallbackThread, __func__,
+      [=, self = RefPtr<MediaTransportHandlerIPC>(this)](bool /*dummy*/) {
+        if (mChild) {
+          mChild->SendUpdateNetworkState(aOnline);
+        }
+      },
+      [](const nsCString& aError) {});
+}
+
+RefPtr<dom::RTCStatsPromise> MediaTransportHandlerIPC::GetIceStats(
+    const std::string& aTransportId, DOMHighResTimeStamp aNow) {
+  using IPCPromise = dom::PMediaTransportChild::GetIceStatsPromise;
+  return mInitPromise
+      ->Then(mCallbackThread, __func__,
+             [aTransportId, aNow, this, self = RefPtr(this)](
+                 const InitPromise::ResolveOrRejectValue& aValue) {
+               if (aValue.IsReject()) {
+                 return IPCPromise::CreateAndResolve(
+                     MakeUnique<dom::RTCStatsCollection>(),
+                     "MediaTransportHandlerIPC::GetIceStats_1");
+               }
+               if (!mChild) {
+                 return IPCPromise::CreateAndResolve(
+                     MakeUnique<dom::RTCStatsCollection>(),
+                     "MediaTransportHandlerIPC::GetIceStats_1");
+               }
+               return mChild->SendGetIceStats(aTransportId, aNow);
+             })
+      ->Then(mCallbackThread, __func__,
+             [](IPCPromise::ResolveOrRejectValue&& aValue) {
+               if (aValue.IsReject()) {
+                 return dom::RTCStatsPromise::CreateAndResolve(
+                     MakeUnique<dom::RTCStatsCollection>(),
+                     "MediaTransportHandlerIPC::GetIceStats_2");
+               }
+               return dom::RTCStatsPromise::CreateAndResolve(
+                   std::move(aValue.ResolveValue()),
+                   "MediaTransportHandlerIPC::GetIceStats_2");
+             });
+}
+
+MediaTransportChild::MediaTransportChild(MediaTransportHandlerIPC* aUser)
+    : mUser(aUser) {}
+
+MediaTransportChild::~MediaTransportChild() { mUser->mChild = nullptr; }
+
+mozilla::ipc::IPCResult MediaTransportChild::RecvOnCandidate(
+    const string& transportId, const CandidateInfo& candidateInfo) {
+  mUser->OnCandidate(transportId, candidateInfo);
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportChild::RecvOnAlpnNegotiated(
+    const string& alpn) {
+  mUser->OnAlpnNegotiated(alpn);
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportChild::RecvOnGatheringStateChange(
+    const int& state) {
+  mUser->OnGatheringStateChange(static_cast<dom::RTCIceGatheringState>(state));
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportChild::RecvOnConnectionStateChange(
+    const int& state) {
+  mUser->OnConnectionStateChange(
+      static_cast<dom::RTCIceConnectionState>(state));
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportChild::RecvOnPacketReceived(
+    const string& transportId, const MediaPacket& packet) {
+  mUser->OnPacketReceived(transportId, packet);
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportChild::RecvOnEncryptedSending(
+    const string& transportId, const MediaPacket& packet) {
+  mUser->OnEncryptedSending(transportId, packet);
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportChild::RecvOnStateChange(
+    const string& transportId, const int& state) {
+  mUser->OnStateChange(transportId, static_cast<TransportLayer::State>(state));
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportChild::RecvOnRtcpStateChange(
+    const string& transportId, const int& state) {
+  mUser->OnRtcpStateChange(transportId,
+                           static_cast<TransportLayer::State>(state));
+  return ipc::IPCResult::Ok();
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/jsapi/MediaTransportHandlerIPC.h b/dom/media/webrtc/jsapi/MediaTransportHandlerIPC.h
new file mode 100644
index 0000000000..6b285dee2a
--- /dev/null
+++ b/dom/media/webrtc/jsapi/MediaTransportHandlerIPC.h
@@ -0,0 +1,96 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _MTRANSPORTHANDLER_IPC_H__
+#define _MTRANSPORTHANDLER_IPC_H__
+
+#include "jsapi/MediaTransportHandler.h"
+#include "mozilla/dom/PMediaTransportChild.h"
+
+namespace mozilla {
+
+class MediaTransportChild;
+
+// Implementation of MediaTransportHandler that uses IPC (PMediaTransport) to
+// talk to mtransport on another process.
+class MediaTransportHandlerIPC final : public MediaTransportHandler {
+ public:
+  explicit MediaTransportHandlerIPC(nsISerialEventTarget* aCallbackThread);
+  void Initialize() override;
+  RefPtr<IceLogPromise> GetIceLog(const nsCString& aPattern) override;
+  void ClearIceLog() override;
+  void EnterPrivateMode() override;
+  void ExitPrivateMode() override;
+
+  void CreateIceCtx(const std::string& aName) override;
+
+  nsresult SetIceConfig(const nsTArray<dom::RTCIceServer>& aIceServers,
+                        dom::RTCIceTransportPolicy aIcePolicy) override;
+
+  // We will probably be able to move the proxy lookup stuff into
+  // this class once we move mtransport to its own process.
+  void SetProxyConfig(NrSocketProxyConfig&& aProxyConfig) override;
+
+  void EnsureProvisionalTransport(const std::string& aTransportId,
+                                  const std::string& aLocalUfrag,
+                                  const std::string& aLocalPwd,
+                                  int aComponentCount) override;
+
+  void SetTargetForDefaultLocalAddressLookup(const std::string& aTargetIp,
+                                             uint16_t aTargetPort) override;
+
+  // We set default-route-only as late as possible because it depends on what
+  // capture permissions have been granted on the window, which could easily
+  // change between Init (ie; when the PC is created) and StartIceGathering
+  // (ie; when we set the local description).
+  void StartIceGathering(bool aDefaultRouteOnly, bool aObfuscateHostAddresses,
+                         // TODO: It probably makes sense to look
+                         // this up internally
+                         const nsTArray<NrIceStunAddr>& aStunAddrs) override;
+
+  void ActivateTransport(
+      const std::string& aTransportId, const std::string& aLocalUfrag,
+      const std::string& aLocalPwd, size_t aComponentCount,
+      const std::string& aUfrag, const std::string& aPassword,
+      const nsTArray<uint8_t>& aKeyDer, const nsTArray<uint8_t>& aCertDer,
+      SSLKEAType aAuthType, bool aDtlsClient, const DtlsDigestList& aDigests,
+      bool aPrivacyRequested) override;
+
+  void RemoveTransportsExcept(
+      const std::set<std::string>& aTransportIds) override;
+
+  void StartIceChecks(bool aIsControlling,
+                      const std::vector<std::string>& aIceOptions) override;
+
+  void SendPacket(const std::string& aTransportId,
+                  MediaPacket&& aPacket) override;
+
+  void AddIceCandidate(const std::string& aTransportId,
+                       const std::string& aCandidate, const std::string& aUfrag,
+                       const std::string& aObfuscatedAddress) override;
+
+  void UpdateNetworkState(bool aOnline) override;
+
+  RefPtr<dom::RTCStatsPromise> GetIceStats(const std::string& aTransportId,
+                                           DOMHighResTimeStamp aNow) override;
+
+ private:
+  friend class MediaTransportChild;
+  void Destroy() override;
+
+  // We do not own this; it will tell us when it is going away.
+  dom::PMediaTransportChild* mChild = nullptr;
+
+  // |mChild| can only be initted asynchronously, |mInitPromise| resolves
+  // when that happens. The |Then| calls make it convenient to dispatch API
+  // calls to main, which is a bonus.
+  // Init promise is not exclusive; this lets us call |Then| on it for every
+  // API call we get, instead of creating another promise each time.
+  typedef MozPromise<bool, nsCString, false> InitPromise;
+  RefPtr<InitPromise> mInitPromise;
+};
+
+}  // namespace mozilla
+
+#endif  //_MTRANSPORTHANDLER_IPC_H__
diff --git a/dom/media/webrtc/jsapi/MediaTransportParent.cpp b/dom/media/webrtc/jsapi/MediaTransportParent.cpp
new file mode 100644
index 0000000000..d0f90df8c5
--- /dev/null
+++ b/dom/media/webrtc/jsapi/MediaTransportParent.cpp
@@ -0,0 +1,240 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "mozilla/dom/MediaTransportParent.h"
+#include "jsapi/MediaTransportHandler.h"
+
+#include "transport/sigslot.h"
+#include "common/browser_logging/CSFLog.h"
+
+namespace mozilla {
+
+// Deals with the MediaTransportHandler interface, so MediaTransportParent
+// doesn't have to..
+class MediaTransportParent::Impl : public sigslot::has_slots<> {
+ public:
+  explicit Impl(MediaTransportParent* aParent)
+      : mHandler(MediaTransportHandler::Create(GetCurrentSerialEventTarget())),
+        mParent(aParent) {
+    mHandler->SignalCandidate.connect(this,
+                                      &MediaTransportParent::Impl::OnCandidate);
+    mHandler->SignalAlpnNegotiated.connect(
+        this, &MediaTransportParent::Impl::OnAlpnNegotiated);
+    mHandler->SignalGatheringStateChange.connect(
+        this, &MediaTransportParent::Impl::OnGatheringStateChange);
+    mHandler->SignalConnectionStateChange.connect(
+        this, &MediaTransportParent::Impl::OnConnectionStateChange);
+    mHandler->SignalPacketReceived.connect(
+        this, &MediaTransportParent::Impl::OnPacketReceived);
+    mHandler->SignalEncryptedSending.connect(
+        this, &MediaTransportParent::Impl::OnEncryptedSending);
+    mHandler->SignalStateChange.connect(
+        this, &MediaTransportParent::Impl::OnStateChange);
+    mHandler->SignalRtcpStateChange.connect(
+        this, &MediaTransportParent::Impl::OnRtcpStateChange);
+  }
+
+  virtual ~Impl() {
+    disconnect_all();
+    mHandler = nullptr;
+  }
+
+  void OnCandidate(const std::string& aTransportId,
+                   const CandidateInfo& aCandidateInfo) {
+    NS_ENSURE_TRUE_VOID(mParent->SendOnCandidate(aTransportId, aCandidateInfo));
+  }
+
+  void OnAlpnNegotiated(const std::string& aAlpn, bool aPrivacyRequested) {
+    NS_ENSURE_TRUE_VOID(mParent->SendOnAlpnNegotiated(aAlpn));
+  }
+
+  void OnGatheringStateChange(dom::RTCIceGatheringState aState) {
+    NS_ENSURE_TRUE_VOID(
+        mParent->SendOnGatheringStateChange(static_cast<int>(aState)));
+  }
+
+  void OnConnectionStateChange(dom::RTCIceConnectionState aState) {
+    NS_ENSURE_TRUE_VOID(
+        mParent->SendOnConnectionStateChange(static_cast<int>(aState)));
+  }
+
+  void OnPacketReceived(const std::string& aTransportId,
+                        const MediaPacket& aPacket) {
+    NS_ENSURE_TRUE_VOID(mParent->SendOnPacketReceived(aTransportId, aPacket));
+  }
+
+  void OnEncryptedSending(const std::string& aTransportId,
+                          const MediaPacket& aPacket) {
+    NS_ENSURE_TRUE_VOID(mParent->SendOnEncryptedSending(aTransportId, aPacket));
+  }
+
+  void OnStateChange(const std::string& aTransportId,
+                     TransportLayer::State aState) {
+    NS_ENSURE_TRUE_VOID(mParent->SendOnStateChange(aTransportId, aState));
+  }
+
+  void OnRtcpStateChange(const std::string& aTransportId,
+                         TransportLayer::State aState) {
+    NS_ENSURE_TRUE_VOID(mParent->SendOnRtcpStateChange(aTransportId, aState));
+  }
+
+  RefPtr<MediaTransportHandler> mHandler;
+
+ private:
+  MediaTransportParent* mParent;
+};
+
+MediaTransportParent::MediaTransportParent() : mImpl(new Impl(this)) {}
+
+MediaTransportParent::~MediaTransportParent() {}
+
+mozilla::ipc::IPCResult MediaTransportParent::RecvGetIceLog(
+    const nsCString& pattern, GetIceLogResolver&& aResolve) {
+  mImpl->mHandler->GetIceLog(pattern)->Then(
+      GetCurrentSerialEventTarget(), __func__,
+      // IPDL doesn't give us a reject function, so we cannot reject async, so
+      // we are forced to resolve with an empty result. Laaaaaaame.
+      [aResolve = std::move(aResolve)](
+          MediaTransportHandler::IceLogPromise::ResolveOrRejectValue&&
+              aResult) mutable {
+        WebrtcGlobalLog logLines;
+        if (aResult.IsResolve()) {
+          logLines = std::move(aResult.ResolveValue());
+        }
+        aResolve(logLines);
+      });
+
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportParent::RecvClearIceLog() {
+  mImpl->mHandler->ClearIceLog();
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportParent::RecvEnterPrivateMode() {
+  mImpl->mHandler->EnterPrivateMode();
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportParent::RecvExitPrivateMode() {
+  mImpl->mHandler->ExitPrivateMode();
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportParent::RecvCreateIceCtx(
+    const string& name) {
+  mImpl->mHandler->CreateIceCtx(name);
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportParent::RecvSetIceConfig(
+    nsTArray<RTCIceServer>&& iceServers,
+    const RTCIceTransportPolicy& icePolicy) {
+  nsresult rv = mImpl->mHandler->SetIceConfig(iceServers, icePolicy);
+  if (NS_FAILED(rv)) {
+    return ipc::IPCResult::Fail(WrapNotNull(this), __func__,
+                                "MediaTransportHandler::SetIceConfig failed");
+  }
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportParent::RecvSetProxyConfig(
+    const net::WebrtcProxyConfig& aProxyConfig) {
+  mImpl->mHandler->SetProxyConfig(NrSocketProxyConfig(aProxyConfig));
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportParent::RecvEnsureProvisionalTransport(
+    const string& transportId, const string& localUfrag, const string& localPwd,
+    const int& componentCount) {
+  mImpl->mHandler->EnsureProvisionalTransport(transportId, localUfrag, localPwd,
+                                              componentCount);
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult
+MediaTransportParent::RecvSetTargetForDefaultLocalAddressLookup(
+    const std::string& targetIp, uint16_t targetPort) {
+  mImpl->mHandler->SetTargetForDefaultLocalAddressLookup(targetIp, targetPort);
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportParent::RecvStartIceGathering(
+    const bool& defaultRouteOnly, const bool& obfuscateHostAddresses,
+    const net::NrIceStunAddrArray& stunAddrs) {
+  mImpl->mHandler->StartIceGathering(defaultRouteOnly, obfuscateHostAddresses,
+                                     stunAddrs);
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportParent::RecvActivateTransport(
+    const string& transportId, const string& localUfrag, const string& localPwd,
+    const int& componentCount, const string& remoteUfrag,
+    const string& remotePwd, nsTArray<uint8_t>&& keyDer,
+    nsTArray<uint8_t>&& certDer, const int& authType, const bool& dtlsClient,
+    const DtlsDigestList& digests, const bool& privacyRequested) {
+  mImpl->mHandler->ActivateTransport(
+      transportId, localUfrag, localPwd, componentCount, remoteUfrag, remotePwd,
+      keyDer, certDer, static_cast<SSLKEAType>(authType), dtlsClient, digests,
+      privacyRequested);
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportParent::RecvRemoveTransportsExcept(
+    const StringVector& transportIds) {
+  std::set<std::string> ids(transportIds.begin(), transportIds.end());
+  mImpl->mHandler->RemoveTransportsExcept(ids);
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportParent::RecvStartIceChecks(
+    const bool& isControlling, const StringVector& iceOptions) {
+  mImpl->mHandler->StartIceChecks(isControlling, iceOptions);
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportParent::RecvSendPacket(
+    const string& transportId, MediaPacket&& packet) {
+  mImpl->mHandler->SendPacket(transportId, std::move(packet));
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportParent::RecvAddIceCandidate(
+    const string& transportId, const string& candidate, const string& ufrag,
+    const string& obfuscatedAddr) {
+  mImpl->mHandler->AddIceCandidate(transportId, candidate, ufrag,
+                                   obfuscatedAddr);
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportParent::RecvUpdateNetworkState(
+    const bool& online) {
+  mImpl->mHandler->UpdateNetworkState(online);
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportParent::RecvGetIceStats(
+    const string& transportId, const double& now,
+    GetIceStatsResolver&& aResolve) {
+  mImpl->mHandler->GetIceStats(transportId, now)
+      ->Then(
+          GetCurrentSerialEventTarget(), __func__,
+          // IPDL doesn't give us a reject function, so we cannot reject async,
+          // so we are forced to resolve with an unmodified result. Laaaaaaame.
+          [aResolve = std::move(aResolve)](
+              dom::RTCStatsPromise::ResolveOrRejectValue&& aResult) {
+            if (aResult.IsResolve()) {
+              aResolve(aResult.ResolveValue());
+            } else {
+              aResolve(MakeUnique<dom::RTCStatsCollection>());
+            }
+          });
+
+  return ipc::IPCResult::Ok();
+}
+
+void MediaTransportParent::ActorDestroy(ActorDestroyReason aWhy) {}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/jsapi/PacketDumper.cpp b/dom/media/webrtc/jsapi/PacketDumper.cpp
new file mode 100644
index 0000000000..2b62e0806e
--- /dev/null
+++ b/dom/media/webrtc/jsapi/PacketDumper.cpp
@@ -0,0 +1,124 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "jsapi/PacketDumper.h"
+#include "jsapi/PeerConnectionImpl.h"
+#include "mozilla/media/MediaUtils.h"  // NewRunnableFrom
+#include "nsThreadUtils.h"             // NS_DispatchToMainThread
+
+namespace mozilla {
+
+/* static */
+RefPtr<PacketDumper> PacketDumper::GetPacketDumper(
+    const std::string& aPcHandle) {
+  MOZ_ASSERT(NS_IsMainThread());
+  PeerConnectionWrapper pcw(aPcHandle);
+  if (pcw.impl()) {
+    return pcw.impl()->GetPacketDumper();
+  }
+
+  return new PacketDumper("");
+}
+
+PacketDumper::PacketDumper(const std::string& aPcHandle)
+    : mPcHandle(aPcHandle),
+      mPacketDumpEnabled(false),
+      mPacketDumpFlagsMutex("Packet dump flags mutex") {}
+
+void PacketDumper::Dump(size_t aLevel, dom::mozPacketDumpType aType,
+                        bool aSending, const void* aData, size_t aSize) {
+  // Optimization; avoids making a copy of the buffer, but we need to lock a
+  // mutex and check the flags. Could be optimized further, if we really want to
+  if (!ShouldDumpPacket(aLevel, aType, aSending)) {
+    return;
+  }
+
+  UniquePtr<uint8_t[]> ownedPacket = MakeUnique<uint8_t[]>(aSize);
+  memcpy(ownedPacket.get(), aData, aSize);
+
+  RefPtr<Runnable> dumpRunnable = media::NewRunnableFrom(std::bind(
+      [this, self = RefPtr<PacketDumper>(this), aLevel, aType, aSending,
+       aSize](UniquePtr<uint8_t[]>& aPacket) -> nsresult {
+        // Check again; packet dump might have been disabled since the dispatch
+        if (ShouldDumpPacket(aLevel, aType, aSending)) {
+          PeerConnectionWrapper pcw(mPcHandle);
+          RefPtr<PeerConnectionImpl> pc = pcw.impl();
+          if (pc) {
+            pc->DumpPacket_m(aLevel, aType, aSending, aPacket, aSize);
+          }
+        }
+        return NS_OK;
+      },
+      std::move(ownedPacket)));
+
+  NS_DispatchToMainThread(dumpRunnable);
+}
+
+nsresult PacketDumper::EnablePacketDump(unsigned long aLevel,
+                                        dom::mozPacketDumpType aType,
+                                        bool aSending) {
+  mPacketDumpEnabled = true;
+  std::vector<unsigned>* packetDumpFlags;
+  if (aSending) {
+    packetDumpFlags = &mSendPacketDumpFlags;
+  } else {
+    packetDumpFlags = &mRecvPacketDumpFlags;
+  }
+
+  unsigned flag = 1 << (unsigned)aType;
+
+  MutexAutoLock lock(mPacketDumpFlagsMutex);
+  if (aLevel >= packetDumpFlags->size()) {
+    packetDumpFlags->resize(aLevel + 1);
+  }
+
+  (*packetDumpFlags)[aLevel] |= flag;
+  return NS_OK;
+}
+
+nsresult PacketDumper::DisablePacketDump(unsigned long aLevel,
+                                         dom::mozPacketDumpType aType,
+                                         bool aSending) {
+  std::vector<unsigned>* packetDumpFlags;
+  if (aSending) {
+    packetDumpFlags = &mSendPacketDumpFlags;
+  } else {
+    packetDumpFlags = &mRecvPacketDumpFlags;
+  }
+
+  unsigned flag = 1 << (unsigned)aType;
+
+  MutexAutoLock lock(mPacketDumpFlagsMutex);
+  if (aLevel < packetDumpFlags->size()) {
+    (*packetDumpFlags)[aLevel] &= ~flag;
+  }
+
+  return NS_OK;
+}
+
+bool PacketDumper::ShouldDumpPacket(size_t aLevel, dom::mozPacketDumpType aType,
+                                    bool aSending) const {
+  if (!mPacketDumpEnabled) {
+    return false;
+  }
+
+  MutexAutoLock lock(mPacketDumpFlagsMutex);
+
+  const std::vector<unsigned>* packetDumpFlags;
+
+  if (aSending) {
+    packetDumpFlags = &mSendPacketDumpFlags;
+  } else {
+    packetDumpFlags = &mRecvPacketDumpFlags;
+  }
+
+  if (aLevel < packetDumpFlags->size()) {
+    unsigned flag = 1 << (unsigned)aType;
+    return flag & packetDumpFlags->at(aLevel);
+  }
+
+  return false;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/jsapi/PacketDumper.h b/dom/media/webrtc/jsapi/PacketDumper.h
new file mode 100644
index 0000000000..e998b3871f
--- /dev/null
+++ b/dom/media/webrtc/jsapi/PacketDumper.h
@@ -0,0 +1,52 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _PACKET_DUMPER_H_
+#define _PACKET_DUMPER_H_
+
+#include "nsISupportsImpl.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/dom/RTCPeerConnectionBinding.h"
+
+#include <vector>
+
+namespace mozilla {
+class PeerConnectionImpl;
+
+class PacketDumper {
+ public:
+  static RefPtr<PacketDumper> GetPacketDumper(const std::string& aPcHandle);
+
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(PacketDumper)
+
+  PacketDumper(const PacketDumper&) = delete;
+  PacketDumper& operator=(const PacketDumper&) = delete;
+
+  void Dump(size_t aLevel, dom::mozPacketDumpType aType, bool aSending,
+            const void* aData, size_t aSize);
+
+  nsresult EnablePacketDump(unsigned long aLevel, dom::mozPacketDumpType aType,
+                            bool aSending);
+
+  nsresult DisablePacketDump(unsigned long aLevel, dom::mozPacketDumpType aType,
+                             bool aSending);
+
+ private:
+  friend class PeerConnectionImpl;
+  explicit PacketDumper(const std::string& aPcHandle);
+  ~PacketDumper() = default;
+  bool ShouldDumpPacket(size_t aLevel, dom::mozPacketDumpType aType,
+                        bool aSending) const;
+
+  // This class is not cycle-collected, so it cannot hold onto a strong ref
+  const std::string mPcHandle;
+  std::vector<unsigned> mSendPacketDumpFlags;
+  std::vector<unsigned> mRecvPacketDumpFlags;
+  Atomic<bool> mPacketDumpEnabled;
+  mutable Mutex mPacketDumpFlagsMutex;
+};
+
+}  // namespace mozilla
+
+#endif  // _PACKET_DUMPER_H_
diff --git a/dom/media/webrtc/jsapi/PeerConnectionCtx.cpp b/dom/media/webrtc/jsapi/PeerConnectionCtx.cpp
new file mode 100644
index 0000000000..9a8f27fb59
--- /dev/null
+++ b/dom/media/webrtc/jsapi/PeerConnectionCtx.cpp
@@ -0,0 +1,650 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "PeerConnectionCtx.h"
+
+#include "WebrtcGlobalStatsHistory.h"
+#include "api/audio/audio_mixer.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "call/audio_state.h"
+#include "common/browser_logging/CSFLog.h"
+#include "common/browser_logging/WebRtcLog.h"
+#include "gmp-video-decode.h"  // GMP_API_VIDEO_DECODER
+#include "gmp-video-encode.h"  // GMP_API_VIDEO_ENCODER
+#include "libwebrtcglue/CallWorkerThread.h"
+#include "modules/audio_device/include/fake_audio_device.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/include/aec_dump.h"
+#include "mozilla/UniquePtr.h"
+#include "mozilla/dom/RTCPeerConnectionBinding.h"
+#include "mozilla/Preferences.h"
+#include "mozilla/Services.h"
+#include "mozilla/StaticPtr.h"
+#include "mozilla/Telemetry.h"
+#include "mozilla/Types.h"
+#include "mozilla/dom/RTCStatsReportBinding.h"
+#include "nsCRTGlue.h"
+#include "nsIIOService.h"
+#include "nsIObserver.h"
+#include "nsIObserverService.h"
+#include "nsNetCID.h"               // NS_SOCKETTRANSPORTSERVICE_CONTRACTID
+#include "nsServiceManagerUtils.h"  // do_GetService
+#include "PeerConnectionImpl.h"
+#include "prcvar.h"
+#include "transport/runnable_utils.h"
+#include "WebrtcGlobalChild.h"
+#include "WebrtcGlobalInformation.h"
+
+static const char* pccLogTag = "PeerConnectionCtx";
+#ifdef LOGTAG
+#  undef LOGTAG
+#endif
+#define LOGTAG pccLogTag
+
+using namespace webrtc;
+
+namespace {
+class DummyAudioMixer : public AudioMixer {
+ public:
+  bool AddSource(Source*) override { return true; }
+  void RemoveSource(Source*) override {}
+  void Mix(size_t, AudioFrame*) override { MOZ_CRASH("Unexpected call"); }
+};
+
+class DummyAudioProcessing : public AudioProcessing {
+ public:
+  int Initialize() override {
+    MOZ_CRASH("Unexpected call");
+    return kNoError;
+  }
+  int Initialize(const ProcessingConfig&) override { return Initialize(); }
+  void ApplyConfig(const Config&) override { MOZ_CRASH("Unexpected call"); }
+  int proc_sample_rate_hz() const override {
+    MOZ_CRASH("Unexpected call");
+    return 0;
+  }
+  int proc_split_sample_rate_hz() const override {
+    MOZ_CRASH("Unexpected call");
+    return 0;
+  }
+  size_t num_input_channels() const override {
+    MOZ_CRASH("Unexpected call");
+    return 0;
+  }
+  size_t num_proc_channels() const override {
+    MOZ_CRASH("Unexpected call");
+    return 0;
+  }
+  size_t num_output_channels() const override {
+    MOZ_CRASH("Unexpected call");
+    return 0;
+  }
+  size_t num_reverse_channels() const override {
+    MOZ_CRASH("Unexpected call");
+    return 0;
+  }
+  void set_output_will_be_muted(bool) override { MOZ_CRASH("Unexpected call"); }
+  void SetRuntimeSetting(RuntimeSetting) override {
+    MOZ_CRASH("Unexpected call");
+  }
+  bool PostRuntimeSetting(RuntimeSetting setting) override { return false; }
+  int ProcessStream(const int16_t* const, const StreamConfig&,
+                    const StreamConfig&, int16_t* const) override {
+    MOZ_CRASH("Unexpected call");
+    return kNoError;
+  }
+  int ProcessStream(const float* const*, const StreamConfig&,
+                    const StreamConfig&, float* const*) override {
+    MOZ_CRASH("Unexpected call");
+    return kNoError;
+  }
+  int ProcessReverseStream(const int16_t* const, const StreamConfig&,
+                           const StreamConfig&, int16_t* const) override {
+    MOZ_CRASH("Unexpected call");
+    return kNoError;
+  }
+  int ProcessReverseStream(const float* const*, const StreamConfig&,
+                           const StreamConfig&, float* const*) override {
+    MOZ_CRASH("Unexpected call");
+    return kNoError;
+  }
+  int AnalyzeReverseStream(const float* const*, const StreamConfig&) override {
+    MOZ_CRASH("Unexpected call");
+    return kNoError;
+  }
+  bool GetLinearAecOutput(
+      rtc::ArrayView<std::array<float, 160>>) const override {
+    MOZ_CRASH("Unexpected call");
+    return false;
+  }
+  void set_stream_analog_level(int) override { MOZ_CRASH("Unexpected call"); }
+  int recommended_stream_analog_level() const override {
+    MOZ_CRASH("Unexpected call");
+    return -1;
+  }
+  int set_stream_delay_ms(int) override {
+    MOZ_CRASH("Unexpected call");
+    return kNoError;
+  }
+  int stream_delay_ms() const override {
+    MOZ_CRASH("Unexpected call");
+    return 0;
+  }
+  void set_stream_key_pressed(bool) override { MOZ_CRASH("Unexpected call"); }
+  bool CreateAndAttachAecDump(absl::string_view, int64_t,
+                              rtc::TaskQueue*) override {
+    MOZ_CRASH("Unexpected call");
+    return false;
+  }
+  bool CreateAndAttachAecDump(FILE*, int64_t, rtc::TaskQueue*) override {
+    MOZ_CRASH("Unexpected call");
+    return false;
+  }
+  void AttachAecDump(std::unique_ptr<AecDump>) override {
+    MOZ_CRASH("Unexpected call");
+  }
+  void DetachAecDump() override { MOZ_CRASH("Unexpected call"); }
+  AudioProcessingStats GetStatistics() override {
+    return AudioProcessingStats();
+  }
+  AudioProcessingStats GetStatistics(bool) override { return GetStatistics(); }
+  AudioProcessing::Config GetConfig() const override {
+    MOZ_CRASH("Unexpected call");
+    return Config();
+  }
+};
+}  // namespace
+
+namespace mozilla {
+
+using namespace dom;
+
+SharedWebrtcState::SharedWebrtcState(
+    RefPtr<AbstractThread> aCallWorkerThread,
+    webrtc::AudioState::Config&& aAudioStateConfig,
+    RefPtr<webrtc::AudioDecoderFactory> aAudioDecoderFactory,
+    UniquePtr<webrtc::WebRtcKeyValueConfig> aTrials)
+    : mCallWorkerThread(std::move(aCallWorkerThread)),
+      mAudioStateConfig(std::move(aAudioStateConfig)),
+      mAudioDecoderFactory(std::move(aAudioDecoderFactory)),
+      mTrials(std::move(aTrials)) {}
+
+SharedWebrtcState::~SharedWebrtcState() = default;
+
+class PeerConnectionCtxObserver : public nsIObserver {
+ public:
+  NS_DECL_ISUPPORTS
+
+  PeerConnectionCtxObserver() {}
+
+  void Init() {
+    nsCOMPtr<nsIObserverService> observerService =
+        services::GetObserverService();
+    if (!observerService) return;
+
+    nsresult rv = NS_OK;
+
+    rv = observerService->AddObserver(this, NS_XPCOM_WILL_SHUTDOWN_OBSERVER_ID,
+                                      false);
+    MOZ_ALWAYS_SUCCEEDS(rv);
+    rv = observerService->AddObserver(this, NS_IOSERVICE_OFFLINE_STATUS_TOPIC,
+                                      false);
+    MOZ_ALWAYS_SUCCEEDS(rv);
+    (void)rv;
+  }
+
+  NS_IMETHOD Observe(nsISupports* aSubject, const char* aTopic,
+                     const char16_t* aData) override {
+    if (strcmp(aTopic, NS_XPCOM_WILL_SHUTDOWN_OBSERVER_ID) == 0) {
+      CSFLogDebug(LOGTAG, "Shutting down PeerConnectionCtx");
+      PeerConnectionCtx::Destroy();
+
+      nsCOMPtr<nsIObserverService> observerService =
+          services::GetObserverService();
+      if (!observerService) return NS_ERROR_FAILURE;
+
+      nsresult rv = observerService->RemoveObserver(
+          this, NS_IOSERVICE_OFFLINE_STATUS_TOPIC);
+      MOZ_ALWAYS_SUCCEEDS(rv);
+      rv = observerService->RemoveObserver(this,
+                                           NS_XPCOM_WILL_SHUTDOWN_OBSERVER_ID);
+      MOZ_ALWAYS_SUCCEEDS(rv);
+
+      // Make sure we're not deleted while still inside ::Observe()
+      RefPtr<PeerConnectionCtxObserver> kungFuDeathGrip(this);
+      PeerConnectionCtx::gPeerConnectionCtxObserver = nullptr;
+    }
+    if (strcmp(aTopic, NS_IOSERVICE_OFFLINE_STATUS_TOPIC) == 0) {
+      if (NS_strcmp(aData, u"" NS_IOSERVICE_OFFLINE) == 0) {
+        CSFLogDebug(LOGTAG, "Updating network state to offline");
+        PeerConnectionCtx::UpdateNetworkState(false);
+      } else if (NS_strcmp(aData, u"" NS_IOSERVICE_ONLINE) == 0) {
+        CSFLogDebug(LOGTAG, "Updating network state to online");
+        PeerConnectionCtx::UpdateNetworkState(true);
+      } else {
+        CSFLogDebug(LOGTAG, "Received unsupported network state event");
+        MOZ_CRASH();
+      }
+    }
+    return NS_OK;
+  }
+
+ private:
+  virtual ~PeerConnectionCtxObserver() {
+    nsCOMPtr<nsIObserverService> observerService =
+        services::GetObserverService();
+    if (observerService) {
+      observerService->RemoveObserver(this, NS_IOSERVICE_OFFLINE_STATUS_TOPIC);
+      observerService->RemoveObserver(this, NS_XPCOM_WILL_SHUTDOWN_OBSERVER_ID);
+    }
+  }
+};
+
+NS_IMPL_ISUPPORTS(PeerConnectionCtxObserver, nsIObserver);
+
+PeerConnectionCtx* PeerConnectionCtx::gInstance;
+StaticRefPtr<PeerConnectionCtxObserver>
+    PeerConnectionCtx::gPeerConnectionCtxObserver;
+
+nsresult PeerConnectionCtx::InitializeGlobal() {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  nsresult res;
+
+  if (!gInstance) {
+    CSFLogDebug(LOGTAG, "Creating PeerConnectionCtx");
+    PeerConnectionCtx* ctx = new PeerConnectionCtx();
+
+    res = ctx->Initialize();
+    PR_ASSERT(NS_SUCCEEDED(res));
+    if (!NS_SUCCEEDED(res)) return res;
+
+    gInstance = ctx;
+
+    if (!PeerConnectionCtx::gPeerConnectionCtxObserver) {
+      PeerConnectionCtx::gPeerConnectionCtxObserver =
+          new PeerConnectionCtxObserver();
+      PeerConnectionCtx::gPeerConnectionCtxObserver->Init();
+    }
+  }
+
+  EnableWebRtcLog();
+  return NS_OK;
+}
+
+PeerConnectionCtx* PeerConnectionCtx::GetInstance() {
+  MOZ_ASSERT(gInstance);
+  return gInstance;
+}
+
+bool PeerConnectionCtx::isActive() { return gInstance; }
+
+void PeerConnectionCtx::Destroy() {
+  CSFLogDebug(LOGTAG, "%s", __FUNCTION__);
+
+  if (gInstance) {
+    // Null out gInstance first, so PeerConnectionImpl doesn't try to use it
+    // in Cleanup.
+    auto* instance = gInstance;
+    gInstance = nullptr;
+    instance->Cleanup();
+    delete instance;
+  }
+
+  StopWebRtcLog();
+}
+
+template <typename T>
+static void RecordCommonRtpTelemetry(const T& list, const T& lastList,
+                                     const bool isRemote) {
+  using namespace Telemetry;
+  for (const auto& s : list) {
+    const bool isAudio = s.mKind.Find(u"audio") != -1;
+    if (s.mPacketsLost.WasPassed() && s.mPacketsReceived.WasPassed()) {
+      if (const uint64_t total =
+              s.mPacketsLost.Value() + s.mPacketsReceived.Value()) {
+        HistogramID id =
+            isRemote ? (isAudio ? WEBRTC_AUDIO_QUALITY_OUTBOUND_PACKETLOSS_RATE
+                                : WEBRTC_VIDEO_QUALITY_OUTBOUND_PACKETLOSS_RATE)
+                     : (isAudio ? WEBRTC_AUDIO_QUALITY_INBOUND_PACKETLOSS_RATE
+                                : WEBRTC_VIDEO_QUALITY_INBOUND_PACKETLOSS_RATE);
+        Accumulate(id, (s.mPacketsLost.Value() * 1000) / total);
+      }
+    }
+    if (s.mJitter.WasPassed()) {
+      HistogramID id = isRemote
+                           ? (isAudio ? WEBRTC_AUDIO_QUALITY_OUTBOUND_JITTER
+                                      : WEBRTC_VIDEO_QUALITY_OUTBOUND_JITTER)
+                           : (isAudio ? WEBRTC_AUDIO_QUALITY_INBOUND_JITTER
+                                      : WEBRTC_VIDEO_QUALITY_INBOUND_JITTER);
+      Accumulate(id, s.mJitter.Value() * 1000);
+    }
+  }
+}
+
+// Telemetry reporting every second after start of first call.
+// The threading model around the media pipelines is weird:
+// - The pipelines are containers,
+// - containers that are only safe on main thread, with members only safe on
+//   STS,
+// - hence the there and back again approach.
+
+void PeerConnectionCtx::DeliverStats(
+    UniquePtr<dom::RTCStatsReportInternal>&& aReport) {
+  using namespace Telemetry;
+
+  // First, get reports from a second ago, if any, for calculations below
+  UniquePtr<dom::RTCStatsReportInternal> lastReport;
+  {
+    auto i = mLastReports.find(aReport->mPcid);
+    if (i != mLastReports.end()) {
+      lastReport = std::move(i->second);
+    } else {
+      lastReport = MakeUnique<dom::RTCStatsReportInternal>();
+    }
+  }
+  // Record Telemetery
+  RecordCommonRtpTelemetry(aReport->mInboundRtpStreamStats,
+                           lastReport->mInboundRtpStreamStats, false);
+  // Record bandwidth telemetry
+  for (const auto& s : aReport->mInboundRtpStreamStats) {
+    if (s.mBytesReceived.WasPassed()) {
+      const bool isAudio = s.mKind.Find(u"audio") != -1;
+      for (const auto& lastS : lastReport->mInboundRtpStreamStats) {
+        if (lastS.mId == s.mId) {
+          int32_t deltaMs = s.mTimestamp.Value() - lastS.mTimestamp.Value();
+          // In theory we're called every second, so delta *should* be in that
+          // range. Small deltas could cause errors due to division
+          if (deltaMs < 500 || deltaMs > 60000 ||
+              !lastS.mBytesReceived.WasPassed()) {
+            break;
+          }
+          HistogramID id = isAudio
+                               ? WEBRTC_AUDIO_QUALITY_INBOUND_BANDWIDTH_KBITS
+                               : WEBRTC_VIDEO_QUALITY_INBOUND_BANDWIDTH_KBITS;
+          // We could accumulate values until enough time has passed
+          // and then Accumulate() but this isn't that important
+          Accumulate(
+              id,
+              ((s.mBytesReceived.Value() - lastS.mBytesReceived.Value()) * 8) /
+                  deltaMs);
+          break;
+        }
+      }
+    }
+  }
+  RecordCommonRtpTelemetry(aReport->mRemoteInboundRtpStreamStats,
+                           lastReport->mRemoteInboundRtpStreamStats, true);
+  for (const auto& s : aReport->mRemoteInboundRtpStreamStats) {
+    if (s.mRoundTripTime.WasPassed()) {
+      const bool isAudio = s.mKind.Find(u"audio") != -1;
+      HistogramID id = isAudio ? WEBRTC_AUDIO_QUALITY_OUTBOUND_RTT
+                               : WEBRTC_VIDEO_QUALITY_OUTBOUND_RTT;
+      Accumulate(id, s.mRoundTripTime.Value() * 1000);
+    }
+  }
+
+  mLastReports[aReport->mPcid] = std::move(aReport);
+}
+
+void PeerConnectionCtx::EverySecondTelemetryCallback_m(nsITimer* timer,
+                                                       void* closure) {
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_ASSERT(PeerConnectionCtx::isActive());
+
+  for (auto& idAndPc : GetInstance()->mPeerConnections) {
+    if (!idAndPc.second->IsClosed()) {
+      idAndPc.second->GetStats(nullptr, true)
+          ->Then(
+              GetMainThreadSerialEventTarget(), __func__,
+              [=](UniquePtr<dom::RTCStatsReportInternal>&& aReport) {
+                if (PeerConnectionCtx::isActive()) {
+                  PeerConnectionCtx::GetInstance()->DeliverStats(
+                      std::move(aReport));
+                }
+              },
+              [=](nsresult aError) {});
+      idAndPc.second->CollectConduitTelemetryData();
+    }
+  }
+}
+
+void PeerConnectionCtx::UpdateNetworkState(bool online) {
+  auto ctx = GetInstance();
+  if (ctx->mPeerConnections.empty()) {
+    return;
+  }
+  for (auto pc : ctx->mPeerConnections) {
+    pc.second->UpdateNetworkState(online);
+  }
+}
+
+SharedWebrtcState* PeerConnectionCtx::GetSharedWebrtcState() const {
+  MOZ_ASSERT(NS_IsMainThread());
+  return mSharedWebrtcState;
+}
+
+void PeerConnectionCtx::RemovePeerConnection(const std::string& aKey) {
+  MOZ_ASSERT(NS_IsMainThread());
+  auto it = mPeerConnections.find(aKey);
+  if (it != mPeerConnections.end()) {
+    if (it->second->GetFinalStats() && !it->second->LongTermStatsIsDisabled()) {
+      WebrtcGlobalInformation::StashStats(*(it->second->GetFinalStats()));
+    }
+    nsAutoString pcId = NS_ConvertASCIItoUTF16(it->second->GetName().c_str());
+    if (XRE_IsContentProcess()) {
+      if (auto* child = WebrtcGlobalChild::Get(); child) {
+        auto pcId = NS_ConvertASCIItoUTF16(it->second->GetName().c_str());
+        child->SendPeerConnectionFinalStats(*(it->second->GetFinalStats()));
+        child->SendPeerConnectionDestroyed(pcId);
+      }
+    } else {
+      using Update = WebrtcGlobalInformation::PcTrackingUpdate;
+      auto update = Update::Remove(pcId);
+      auto finalStats =
+          MakeUnique<RTCStatsReportInternal>(*(it->second->GetFinalStats()));
+      WebrtcGlobalStatsHistory::Record(std::move(finalStats));
+      WebrtcGlobalInformation::PeerConnectionTracking(update);
+    }
+
+    mPeerConnections.erase(it);
+    if (mPeerConnections.empty()) {
+      mSharedWebrtcState = nullptr;
+      StopTelemetryTimer();
+    }
+  }
+}
+
+void PeerConnectionCtx::AddPeerConnection(const std::string& aKey,
+                                          PeerConnectionImpl* aPeerConnection) {
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_ASSERT(mPeerConnections.count(aKey) == 0,
+             "PeerConnection with this key should not already exist");
+  if (mPeerConnections.empty()) {
+    AudioState::Config audioStateConfig;
+    audioStateConfig.audio_mixer = new rtc::RefCountedObject<DummyAudioMixer>();
+    AudioProcessingBuilder audio_processing_builder;
+    audioStateConfig.audio_processing =
+        new rtc::RefCountedObject<DummyAudioProcessing>();
+    audioStateConfig.audio_device_module =
+        new rtc::RefCountedObject<FakeAudioDeviceModule>();
+
+    SharedThreadPoolWebRtcTaskQueueFactory taskQueueFactory;
+    constexpr bool supportTailDispatch = true;
+    // Note the NonBlocking DeletionPolicy!
+    // This task queue is passed into libwebrtc as a raw pointer.
+    // WebrtcCallWrapper guarantees that it outlives its webrtc::Call instance.
+    // Outside of libwebrtc we must use ref-counting to either the
+    // WebrtcCallWrapper or to the CallWorkerThread to keep it alive.
+    auto callWorkerThread =
+        WrapUnique(taskQueueFactory
+                       .CreateTaskQueueWrapper<DeletionPolicy::NonBlocking>(
+                           "CallWorker", supportTailDispatch,
+                           webrtc::TaskQueueFactory::Priority::NORMAL,
+                           MediaThreadType::WEBRTC_CALL_THREAD)
+                       .release());
+
+    UniquePtr<webrtc::WebRtcKeyValueConfig> trials =
+        WrapUnique(new NoTrialsConfig());
+
+    mSharedWebrtcState = MakeAndAddRef<SharedWebrtcState>(
+        new CallWorkerThread(std::move(callWorkerThread)),
+        std::move(audioStateConfig),
+        already_AddRefed(CreateBuiltinAudioDecoderFactory().release()),
+        std::move(trials));
+    StartTelemetryTimer();
+  }
+  auto pcId = NS_ConvertASCIItoUTF16(aPeerConnection->GetName().c_str());
+  if (XRE_IsContentProcess()) {
+    if (auto* child = WebrtcGlobalChild::Get(); child) {
+      child->SendPeerConnectionCreated(
+          pcId, aPeerConnection->LongTermStatsIsDisabled());
+    }
+  } else {
+    using Update = WebrtcGlobalInformation::PcTrackingUpdate;
+    auto update = Update::Add(pcId, aPeerConnection->LongTermStatsIsDisabled());
+    WebrtcGlobalInformation::PeerConnectionTracking(update);
+  }
+  mPeerConnections[aKey] = aPeerConnection;
+}
+
+PeerConnectionImpl* PeerConnectionCtx::GetPeerConnection(
+    const std::string& aKey) const {
+  MOZ_ASSERT(NS_IsMainThread());
+  auto iterator = mPeerConnections.find(aKey);
+  if (iterator == mPeerConnections.end()) {
+    return nullptr;
+  }
+  return iterator->second;
+}
+
+void PeerConnectionCtx::ClearClosedStats() {
+  for (auto& [id, pc] : mPeerConnections) {
+    Unused << id;
+    if (pc->IsClosed()) {
+      // Rare case
+      pc->DisableLongTermStats();
+    }
+  }
+}
+
+nsresult PeerConnectionCtx::Initialize() {
+  MOZ_ASSERT(NS_IsMainThread());
+  initGMP();
+  SdpRidAttributeList::kMaxRidLength =
+      webrtc::BaseRtpStringExtension::kMaxValueSizeBytes;
+
+  if (XRE_IsContentProcess()) {
+    WebrtcGlobalChild::Get();
+  }
+
+  return NS_OK;
+}
+
+nsresult PeerConnectionCtx::StartTelemetryTimer() {
+  return NS_NewTimerWithFuncCallback(getter_AddRefs(mTelemetryTimer),
+                                     EverySecondTelemetryCallback_m, this, 1000,
+                                     nsITimer::TYPE_REPEATING_PRECISE_CAN_SKIP,
+                                     "EverySecondTelemetryCallback_m");
+}
+
+void PeerConnectionCtx::StopTelemetryTimer() {
+  if (mTelemetryTimer) {
+    mTelemetryTimer->Cancel();
+    mTelemetryTimer = nullptr;
+  }
+}
+
+static void GMPReady_m() {
+  if (PeerConnectionCtx::isActive()) {
+    PeerConnectionCtx::GetInstance()->onGMPReady();
+  }
+};
+
+static void GMPReady() {
+  GetMainThreadSerialEventTarget()->Dispatch(WrapRunnableNM(&GMPReady_m),
+                                             NS_DISPATCH_NORMAL);
+};
+
+void PeerConnectionCtx::initGMP() {
+  mGMPService = do_GetService("@mozilla.org/gecko-media-plugin-service;1");
+
+  if (!mGMPService) {
+    CSFLogError(LOGTAG, "%s failed to get the gecko-media-plugin-service",
+                __FUNCTION__);
+    return;
+  }
+
+  nsCOMPtr<nsIThread> thread;
+  nsresult rv = mGMPService->GetThread(getter_AddRefs(thread));
+
+  if (NS_FAILED(rv)) {
+    mGMPService = nullptr;
+    CSFLogError(LOGTAG,
+                "%s failed to get the gecko-media-plugin thread, err=%u",
+                __FUNCTION__, static_cast<unsigned>(rv));
+    return;
+  }
+
+  // presumes that all GMP dir scans have been queued for the GMPThread
+  thread->Dispatch(WrapRunnableNM(&GMPReady), NS_DISPATCH_NORMAL);
+}
+
+nsresult PeerConnectionCtx::Cleanup() {
+  CSFLogDebug(LOGTAG, "%s", __FUNCTION__);
+  MOZ_ASSERT(NS_IsMainThread());
+
+  mQueuedJSEPOperations.Clear();
+  mGMPService = nullptr;
+  mTransportHandler = nullptr;
+  for (auto& [id, pc] : mPeerConnections) {
+    (void)id;
+    pc->Close();
+  }
+  mPeerConnections.clear();
+  mSharedWebrtcState = nullptr;
+  return NS_OK;
+}
+
+void PeerConnectionCtx::queueJSEPOperation(nsIRunnable* aOperation) {
+  mQueuedJSEPOperations.AppendElement(aOperation);
+}
+
+void PeerConnectionCtx::onGMPReady() {
+  mGMPReady = true;
+  for (size_t i = 0; i < mQueuedJSEPOperations.Length(); ++i) {
+    mQueuedJSEPOperations[i]->Run();
+  }
+  mQueuedJSEPOperations.Clear();
+}
+
+bool PeerConnectionCtx::gmpHasH264() {
+  if (!mGMPService) {
+    return false;
+  }
+
+  // XXX I'd prefer if this was all known ahead of time...
+
+  AutoTArray<nsCString, 1> tags;
+  tags.AppendElement("h264"_ns);
+
+  bool has_gmp;
+  nsresult rv;
+  rv = mGMPService->HasPluginForAPI(nsLiteralCString(GMP_API_VIDEO_ENCODER),
+                                    tags, &has_gmp);
+  if (NS_FAILED(rv) || !has_gmp) {
+    return false;
+  }
+
+  rv = mGMPService->HasPluginForAPI(nsLiteralCString(GMP_API_VIDEO_DECODER),
+                                    tags, &has_gmp);
+  if (NS_FAILED(rv) || !has_gmp) {
+    return false;
+  }
+
+  return true;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/jsapi/PeerConnectionCtx.h b/dom/media/webrtc/jsapi/PeerConnectionCtx.h
new file mode 100644
index 0000000000..fdd81f6406
--- /dev/null
+++ b/dom/media/webrtc/jsapi/PeerConnectionCtx.h
@@ -0,0 +1,194 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef peerconnectionctx_h___h__
+#define peerconnectionctx_h___h__
+
+#include <map>
+#include <string>
+
+#include "WebrtcGlobalChild.h"
+#include "api/field_trials_view.h"
+#include "api/scoped_refptr.h"
+#include "call/audio_state.h"
+#include "MediaTransportHandler.h"  // Mostly for IceLogPromise
+#include "mozIGeckoMediaPluginService.h"
+#include "mozilla/Attributes.h"
+#include "mozilla/StaticPtr.h"
+#include "nsIRunnable.h"
+#include "PeerConnectionImpl.h"
+
+namespace webrtc {
+class AudioDecoderFactory;
+
+// Used for testing in mediapipeline_unittest.cpp, MockCall.h
+class NoTrialsConfig : public FieldTrialsView {
+ public:
+  NoTrialsConfig() = default;
+  std::string Lookup(absl::string_view key) const override {
+    // Upstream added a new default field trial string for
+    // CongestionWindow, that we don't want.  In
+    // third_party/libwebrtc/rtc_base/experiments/rate_control_settings.cc
+    // they set kCongestionWindowDefaultFieldTrialString to
+    // "QueueSize:350,MinBitrate:30000,DropFrame:true". With QueueSize
+    // set, GoogCcNetworkController::UpdateCongestionWindowSize is
+    // called.  Because negative values are calculated in
+    // feedback_rtt, an assert fires when calculating data_window in
+    // GoogCcNetworkController::UpdateCongestionWindowSize.  We probably
+    // need to figure out why we're calculating negative feedback_rtt.
+    // See Bug 1780620.
+    if ("WebRTC-CongestionWindow" == key) {
+      return std::string("MinBitrate:30000,DropFrame:true");
+    }
+    return std::string();
+  }
+};
+}  // namespace webrtc
+
+namespace mozilla {
+class PeerConnectionCtxObserver;
+
+namespace dom {
+class WebrtcGlobalInformation;
+}
+
+/**
+ * Refcounted class containing state shared across all PeerConnections and all
+ * Call instances. Managed by PeerConnectionCtx, and kept around while there are
+ * registered peer connections.
+ */
+class SharedWebrtcState {
+ public:
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(SharedWebrtcState)
+
+  SharedWebrtcState(RefPtr<AbstractThread> aCallWorkerThread,
+                    webrtc::AudioState::Config&& aAudioStateConfig,
+                    RefPtr<webrtc::AudioDecoderFactory> aAudioDecoderFactory,
+                    UniquePtr<webrtc::FieldTrialsView> aTrials);
+
+  // A global Call worker thread shared between all Call instances. Implements
+  // AbstractThread for running tasks that call into a Call instance through its
+  // webrtc::TaskQueue member, and for using AbstractThread-specific higher
+  // order constructs like StateMirroring.
+  const RefPtr<AbstractThread> mCallWorkerThread;
+
+  // AudioState config containing dummy implementations of the audio stack,
+  // since we use our own audio stack instead. Shared across all Call instances.
+  const webrtc::AudioState::Config mAudioStateConfig;
+
+  // AudioDecoderFactory instance shared between calls, to limit the number of
+  // instances in large calls.
+  const RefPtr<webrtc::AudioDecoderFactory> mAudioDecoderFactory;
+
+  // Trials instance shared between calls, to limit the number of instances in
+  // large calls.
+  const UniquePtr<webrtc::FieldTrialsView> mTrials;
+
+ private:
+  virtual ~SharedWebrtcState();
+};
+
+// A class to hold some of the singleton objects we need:
+// * The global PeerConnectionImpl table and its associated lock.
+// * Stats report objects for PCs that are gone
+// * GMP related state
+// * Upstream webrtc state shared across all Calls (processing thread)
+class PeerConnectionCtx {
+ public:
+  static nsresult InitializeGlobal();
+  static PeerConnectionCtx* GetInstance();
+  static bool isActive();
+  static void Destroy();
+
+  bool isReady() {
+    // If mGMPService is not set, we aren't using GMP.
+    if (mGMPService) {
+      return mGMPReady;
+    }
+    return true;
+  }
+
+  void queueJSEPOperation(nsIRunnable* aJSEPOperation);
+  void onGMPReady();
+
+  bool gmpHasH264();
+
+  static void UpdateNetworkState(bool online);
+
+  RefPtr<MediaTransportHandler> GetTransportHandler() const {
+    return mTransportHandler;
+  }
+
+  SharedWebrtcState* GetSharedWebrtcState() const;
+
+  void RemovePeerConnection(const std::string& aKey);
+  void AddPeerConnection(const std::string& aKey,
+                         PeerConnectionImpl* aPeerConnection);
+  PeerConnectionImpl* GetPeerConnection(const std::string& aKey) const;
+  template <typename Function>
+  void ForEachPeerConnection(Function&& aFunction) const {
+    MOZ_ASSERT(NS_IsMainThread());
+    for (const auto& pair : mPeerConnections) {
+      aFunction(pair.second);
+    }
+  }
+
+  void ClearClosedStats();
+
+ private:
+  std::map<const std::string, PeerConnectionImpl*> mPeerConnections;
+
+  PeerConnectionCtx()
+      : mGMPReady(false),
+        mTransportHandler(
+            MediaTransportHandler::Create(GetMainThreadSerialEventTarget())) {}
+
+  // This is a singleton, so don't copy construct it, etc.
+  PeerConnectionCtx(const PeerConnectionCtx& other) = delete;
+  void operator=(const PeerConnectionCtx& other) = delete;
+  virtual ~PeerConnectionCtx() = default;
+
+  nsresult Initialize();
+  nsresult StartTelemetryTimer();
+  void StopTelemetryTimer();
+  nsresult Cleanup();
+
+  void initGMP();
+
+  static void EverySecondTelemetryCallback_m(nsITimer* timer, void*);
+
+  nsCOMPtr<nsITimer> mTelemetryTimer;
+
+ private:
+  void DeliverStats(UniquePtr<dom::RTCStatsReportInternal>&& aReport);
+
+  std::map<nsString, UniquePtr<dom::RTCStatsReportInternal>> mLastReports;
+  // We cannot form offers/answers properly until the Gecko Media Plugin stuff
+  // has been initted, which is a complicated mess of thread dispatches,
+  // including sync dispatches to main. So, we need to be able to queue up
+  // offer creation (or SetRemote, when we're the answerer) until all of this is
+  // ready to go, since blocking on this init is just begging for deadlock.
+  nsCOMPtr<mozIGeckoMediaPluginService> mGMPService;
+  bool mGMPReady;
+  nsTArray<nsCOMPtr<nsIRunnable>> mQueuedJSEPOperations;
+
+  // Not initted, just for ICE logging stuff
+  RefPtr<MediaTransportHandler> mTransportHandler;
+
+  // State used by libwebrtc that needs to be shared across all PeerConnections
+  // and all Call instances. Set while there is at least one peer connection
+  // registered. CallWrappers can hold a ref to this object to be sure members
+  // are alive long enough.
+  RefPtr<SharedWebrtcState> mSharedWebrtcState;
+
+  static PeerConnectionCtx* gInstance;
+
+ public:
+  static mozilla::StaticRefPtr<mozilla::PeerConnectionCtxObserver>
+      gPeerConnectionCtxObserver;
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/jsapi/PeerConnectionImpl.cpp b/dom/media/webrtc/jsapi/PeerConnectionImpl.cpp
new file mode 100644
index 0000000000..567b682b2a
--- /dev/null
+++ b/dom/media/webrtc/jsapi/PeerConnectionImpl.cpp
@@ -0,0 +1,4640 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include <cstdlib>
+#include <cerrno>
+#include <deque>
+#include <set>
+#include <sstream>
+#include <vector>
+
+#include "common/browser_logging/CSFLog.h"
+#include "base/histogram.h"
+#include "common/time_profiling/timecard.h"
+
+#include "jsapi.h"
+#include "nspr.h"
+#include "nss.h"
+#include "pk11pub.h"
+
+#include "nsNetCID.h"
+#include "nsIIDNService.h"
+#include "nsILoadContext.h"
+#include "nsEffectiveTLDService.h"
+#include "nsServiceManagerUtils.h"
+#include "nsThreadUtils.h"
+#include "nsIPrefService.h"
+#include "nsIPrefBranch.h"
+#include "nsProxyRelease.h"
+#include "prtime.h"
+
+#include "libwebrtcglue/AudioConduit.h"
+#include "libwebrtcglue/VideoConduit.h"
+#include "libwebrtcglue/WebrtcCallWrapper.h"
+#include "MediaTrackGraph.h"
+#include "transport/runnable_utils.h"
+#include "IPeerConnection.h"
+#include "PeerConnectionCtx.h"
+#include "PeerConnectionImpl.h"
+#include "RemoteTrackSource.h"
+#include "nsDOMDataChannelDeclarations.h"
+#include "transport/dtlsidentity.h"
+#include "sdp/SdpAttribute.h"
+
+#include "jsep/JsepTrack.h"
+#include "jsep/JsepSession.h"
+#include "jsep/JsepSessionImpl.h"
+
+#include "transportbridge/MediaPipeline.h"
+#include "jsapi/RTCRtpReceiver.h"
+#include "jsapi/RTCRtpSender.h"
+
+#include "mozilla/IntegerPrintfMacros.h"
+#include "mozilla/Sprintf.h"
+
+#ifdef XP_WIN
+// We need to undef the MS macro for Document::CreateEvent
+#  ifdef CreateEvent
+#    undef CreateEvent
+#  endif
+#endif  // XP_WIN
+
+#include "mozilla/dom/Document.h"
+#include "nsGlobalWindow.h"
+#include "nsDOMDataChannel.h"
+#include "mozilla/dom/Location.h"
+#include "mozilla/dom/Promise.h"
+#include "mozilla/NullPrincipal.h"
+#include "mozilla/TimeStamp.h"
+#include "mozilla/Telemetry.h"
+#include "mozilla/Preferences.h"
+#include "mozilla/PublicSSL.h"
+#include "nsXULAppAPI.h"
+#include "nsContentUtils.h"
+#include "nsDOMJSUtils.h"
+#include "nsPrintfCString.h"
+#include "nsURLHelper.h"
+#include "nsNetUtil.h"
+#include "js/ArrayBuffer.h"    // JS::NewArrayBufferWithContents
+#include "js/GCAnnotations.h"  // JS_HAZ_ROOTED
+#include "js/RootingAPI.h"     // JS::{{,Mutable}Handle,Rooted}
+#include "mozilla/PeerIdentity.h"
+#include "mozilla/dom/RTCCertificate.h"
+#include "mozilla/dom/RTCSctpTransportBinding.h"  // RTCSctpTransportState
+#include "mozilla/dom/RTCDtlsTransportBinding.h"  // RTCDtlsTransportState
+#include "mozilla/dom/RTCRtpReceiverBinding.h"
+#include "mozilla/dom/RTCRtpSenderBinding.h"
+#include "mozilla/dom/RTCStatsReportBinding.h"
+#include "mozilla/dom/RTCPeerConnectionBinding.h"
+#include "mozilla/dom/PeerConnectionImplBinding.h"
+#include "mozilla/dom/RTCDataChannelBinding.h"
+#include "mozilla/dom/PluginCrashedEvent.h"
+#include "MediaStreamTrack.h"
+#include "AudioStreamTrack.h"
+#include "VideoStreamTrack.h"
+#include "nsIScriptGlobalObject.h"
+#include "DOMMediaStream.h"
+#include "WebrtcGlobalInformation.h"
+#include "mozilla/dom/Event.h"
+#include "mozilla/EventDispatcher.h"
+#include "mozilla/net/DataChannelProtocol.h"
+#include "MediaManager.h"
+
+#include "transport/nr_socket_proxy_config.h"
+#include "RTCSctpTransport.h"
+#include "RTCDtlsTransport.h"
+#include "jsep/JsepTransport.h"
+
+#include "nsILoadInfo.h"
+#include "nsIPrincipal.h"
+#include "mozilla/LoadInfo.h"
+#include "nsIProxiedChannel.h"
+
+#include "mozilla/dom/BrowserChild.h"
+#include "mozilla/net/WebrtcProxyConfig.h"
+
+#ifdef XP_WIN
+// We need to undef the MS macro again in case the windows include file
+// got imported after we included mozilla/dom/Document.h
+#  ifdef CreateEvent
+#    undef CreateEvent
+#  endif
+#endif  // XP_WIN
+
+#include "MediaSegment.h"
+
+#ifdef USE_FAKE_PCOBSERVER
+#  include "FakePCObserver.h"
+#else
+#  include "mozilla/dom/PeerConnectionObserverBinding.h"
+#endif
+#include "mozilla/dom/PeerConnectionObserverEnumsBinding.h"
+
+#define ICE_PARSING \
+  "In RTCConfiguration passed to RTCPeerConnection constructor"
+
+using namespace mozilla;
+using namespace mozilla::dom;
+
+typedef PCObserverString ObString;
+
+static const char* pciLogTag = "PeerConnectionImpl";
+#ifdef LOGTAG
+#  undef LOGTAG
+#endif
+#define LOGTAG pciLogTag
+
+static mozilla::LazyLogModule logModuleInfo("signaling");
+
+// Getting exceptions back down from PCObserver is generally not harmful.
+namespace {
+// This is a terrible hack.  The problem is that SuppressException is not
+// inline, and we link this file without libxul in some cases (e.g. for our test
+// setup).  So we can't use ErrorResult or IgnoredErrorResult because those call
+// SuppressException...  And we can't use FastErrorResult because we can't
+// include BindingUtils.h, because our linking is completely broken. Use
+// BaseErrorResult directly.  Please do not let me see _anyone_ doing this
+// without really careful review from someone who knows what they are doing.
+class JSErrorResult : public binding_danger::TErrorResult<
+                          binding_danger::JustAssertCleanupPolicy> {
+ public:
+  ~JSErrorResult() { SuppressException(); }
+} JS_HAZ_ROOTED;
+
+// The WrapRunnable() macros copy passed-in args and passes them to the function
+// later on the other thread. ErrorResult cannot be passed like this because it
+// disallows copy-semantics.
+//
+// This WrappableJSErrorResult hack solves this by not actually copying the
+// ErrorResult, but creating a new one instead, which works because we don't
+// care about the result.
+//
+// Since this is for JS-calls, these can only be dispatched to the main thread.
+
+class WrappableJSErrorResult {
+ public:
+  WrappableJSErrorResult() : mRv(MakeUnique<JSErrorResult>()), isCopy(false) {}
+  WrappableJSErrorResult(const WrappableJSErrorResult& other)
+      : mRv(MakeUnique<JSErrorResult>()), isCopy(true) {}
+  ~WrappableJSErrorResult() {
+    if (isCopy) {
+      MOZ_ASSERT(NS_IsMainThread());
+    }
+  }
+  operator ErrorResult&() { return *mRv; }
+
+ private:
+  mozilla::UniquePtr<JSErrorResult> mRv;
+  bool isCopy;
+} JS_HAZ_ROOTED;
+
+}  // namespace
+
+static nsresult InitNSSInContent() {
+  NS_ENSURE_TRUE(NS_IsMainThread(), NS_ERROR_NOT_SAME_THREAD);
+
+  if (!XRE_IsContentProcess()) {
+    MOZ_ASSERT_UNREACHABLE("Must be called in content process");
+    return NS_ERROR_FAILURE;
+  }
+
+  static bool nssStarted = false;
+  if (nssStarted) {
+    return NS_OK;
+  }
+
+  if (NSS_NoDB_Init(nullptr) != SECSuccess) {
+    CSFLogError(LOGTAG, "NSS_NoDB_Init failed.");
+    return NS_ERROR_FAILURE;
+  }
+
+  if (NS_FAILED(mozilla::psm::InitializeCipherSuite())) {
+    CSFLogError(LOGTAG, "Fail to set up nss cipher suite.");
+    return NS_ERROR_FAILURE;
+  }
+
+  mozilla::psm::DisableMD5();
+
+  nssStarted = true;
+
+  return NS_OK;
+}
+
+namespace mozilla {
+class DataChannel;
+}
+
+namespace mozilla {
+
+void PeerConnectionAutoTimer::RegisterConnection() { mRefCnt++; }
+
+void PeerConnectionAutoTimer::UnregisterConnection(bool aContainedAV) {
+  MOZ_ASSERT(mRefCnt);
+  mRefCnt--;
+  mUsedAV |= aContainedAV;
+  if (mRefCnt == 0) {
+    if (mUsedAV) {
+      Telemetry::Accumulate(
+          Telemetry::WEBRTC_AV_CALL_DURATION,
+          static_cast<uint32_t>((TimeStamp::Now() - mStart).ToSeconds()));
+    }
+    Telemetry::Accumulate(
+        Telemetry::WEBRTC_CALL_DURATION,
+        static_cast<uint32_t>((TimeStamp::Now() - mStart).ToSeconds()));
+  }
+}
+
+bool PeerConnectionAutoTimer::IsStopped() { return mRefCnt == 0; }
+
+NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE_CLASS(PeerConnectionImpl)
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(PeerConnectionImpl)
+  tmp->Close();
+  tmp->BreakCycles();
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mPCObserver, mWindow, mCertificate,
+                                  mSTSThread, mReceiveStreams, mOperations,
+                                  mSctpTransport, mKungFuDeathGrip)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK_PRESERVED_WRAPPER
+NS_IMPL_CYCLE_COLLECTION_UNLINK_END
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN(PeerConnectionImpl)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(
+      mPCObserver, mWindow, mCertificate, mSTSThread, mReceiveStreams,
+      mOperations, mTransceivers, mSctpTransport, mKungFuDeathGrip)
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
+
+NS_IMPL_CYCLE_COLLECTING_ADDREF(PeerConnectionImpl)
+NS_IMPL_CYCLE_COLLECTING_RELEASE(PeerConnectionImpl)
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(PeerConnectionImpl)
+  NS_WRAPPERCACHE_INTERFACE_MAP_ENTRY
+  NS_INTERFACE_MAP_ENTRY(nsISupports)
+NS_INTERFACE_MAP_END
+
+already_AddRefed<PeerConnectionImpl> PeerConnectionImpl::Constructor(
+    const dom::GlobalObject& aGlobal) {
+  RefPtr<PeerConnectionImpl> pc = new PeerConnectionImpl(&aGlobal);
+
+  CSFLogDebug(LOGTAG, "Created PeerConnection: %p", pc.get());
+
+  return pc.forget();
+}
+
+JSObject* PeerConnectionImpl::WrapObject(JSContext* aCx,
+                                         JS::Handle<JSObject*> aGivenProto) {
+  return PeerConnectionImpl_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+nsPIDOMWindowInner* PeerConnectionImpl::GetParentObject() const {
+  return mWindow;
+}
+
+bool PCUuidGenerator::Generate(std::string* idp) {
+  nsresult rv;
+
+  if (!mGenerator) {
+    mGenerator = do_GetService("@mozilla.org/uuid-generator;1", &rv);
+    if (NS_FAILED(rv)) {
+      return false;
+    }
+    if (!mGenerator) {
+      return false;
+    }
+  }
+
+  nsID id;
+  rv = mGenerator->GenerateUUIDInPlace(&id);
+  if (NS_FAILED(rv)) {
+    return false;
+  }
+  char buffer[NSID_LENGTH];
+  id.ToProvidedString(buffer);
+  idp->assign(buffer);
+
+  return true;
+}
+
+bool IsPrivateBrowsing(nsPIDOMWindowInner* aWindow) {
+  if (!aWindow) {
+    return false;
+  }
+
+  Document* doc = aWindow->GetExtantDoc();
+  if (!doc) {
+    return false;
+  }
+
+  nsILoadContext* loadContext = doc->GetLoadContext();
+  return loadContext && loadContext->UsePrivateBrowsing();
+}
+
+PeerConnectionImpl::PeerConnectionImpl(const GlobalObject* aGlobal)
+    : mTimeCard(MOZ_LOG_TEST(logModuleInfo, LogLevel::Error) ? create_timecard()
+                                                             : nullptr),
+      mJsConfiguration(),
+      mSignalingState(RTCSignalingState::Stable),
+      mIceConnectionState(RTCIceConnectionState::New),
+      mIceGatheringState(RTCIceGatheringState::New),
+      mConnectionState(RTCPeerConnectionState::New),
+      mWindow(do_QueryInterface(aGlobal ? aGlobal->GetAsSupports() : nullptr)),
+      mCertificate(nullptr),
+      mSTSThread(nullptr),
+      mForceIceTcp(false),
+      mTransportHandler(nullptr),
+      mUuidGen(MakeUnique<PCUuidGenerator>()),
+      mIceRestartCount(0),
+      mIceRollbackCount(0),
+      mHaveConfiguredCodecs(false),
+      mTrickle(true)  // TODO(ekr@rtfm.com): Use pref
+      ,
+      mPrivateWindow(false),
+      mActiveOnWindow(false),
+      mTimestampMaker(dom::RTCStatsTimestampMaker::Create(mWindow)),
+      mIdGenerator(new RTCStatsIdGenerator()),
+      listenPort(0),
+      connectPort(0),
+      connectStr(nullptr) {
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_ASSERT_IF(aGlobal, mWindow);
+  mKungFuDeathGrip = this;
+  if (aGlobal) {
+    if (IsPrivateBrowsing(mWindow)) {
+      mPrivateWindow = true;
+      mDisableLongTermStats = true;
+    }
+    mWindow->AddPeerConnection();
+    mActiveOnWindow = true;
+
+    if (mWindow->GetDocumentURI()) {
+      mWindow->GetDocumentURI()->GetAsciiHost(mHostname);
+      nsresult rv;
+      nsCOMPtr<nsIEffectiveTLDService> eTLDService(
+          do_GetService(NS_EFFECTIVETLDSERVICE_CONTRACTID, &rv));
+      if (eTLDService) {
+        Unused << eTLDService->GetBaseDomain(mWindow->GetDocumentURI(), 0,
+                                             mEffectiveTLDPlus1);
+      }
+
+      mRtxIsAllowed = !HostnameInPref(
+          "media.peerconnection.video.use_rtx.blocklist", mHostname);
+    }
+  }
+
+  if (!mUuidGen->Generate(&mHandle)) {
+    MOZ_CRASH();
+  }
+
+  CSFLogInfo(LOGTAG, "%s: PeerConnectionImpl constructor for %s", __FUNCTION__,
+             mHandle.c_str());
+  STAMP_TIMECARD(mTimeCard, "Constructor Completed");
+  mForceIceTcp =
+      Preferences::GetBool("media.peerconnection.ice.force_ice_tcp", false);
+  memset(mMaxReceiving, 0, sizeof(mMaxReceiving));
+  memset(mMaxSending, 0, sizeof(mMaxSending));
+  mJsConfiguration.mCertificatesProvided = false;
+  mJsConfiguration.mPeerIdentityProvided = false;
+}
+
+PeerConnectionImpl::~PeerConnectionImpl() {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  MOZ_ASSERT(!mTransportHandler,
+             "PeerConnection should either be closed, or not initted in the "
+             "first place.");
+
+  if (mTimeCard) {
+    STAMP_TIMECARD(mTimeCard, "Destructor Invoked");
+    STAMP_TIMECARD(mTimeCard, mHandle.c_str());
+    print_timecard(mTimeCard);
+    destroy_timecard(mTimeCard);
+    mTimeCard = nullptr;
+  }
+
+  CSFLogInfo(LOGTAG, "%s: PeerConnectionImpl destructor invoked for %s",
+             __FUNCTION__, mHandle.c_str());
+}
+
+nsresult PeerConnectionImpl::Initialize(PeerConnectionObserver& aObserver,
+                                        nsGlobalWindowInner* aWindow) {
+  nsresult res;
+
+  MOZ_ASSERT(NS_IsMainThread());
+
+  mPCObserver = &aObserver;
+
+  // Find the STS thread
+
+  mSTSThread = do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &res);
+  MOZ_ASSERT(mSTSThread);
+
+  // We do callback handling on STS instead of main to avoid media jank.
+  // Someday, we may have a dedicated thread for this.
+  mTransportHandler = MediaTransportHandler::Create(mSTSThread);
+  if (mPrivateWindow) {
+    mTransportHandler->EnterPrivateMode();
+  }
+
+  // Initialize NSS if we are in content process. For chrome process, NSS should
+  // already been initialized.
+  if (XRE_IsParentProcess()) {
+    // This code interferes with the C++ unit test startup code.
+    nsCOMPtr<nsISupports> nssDummy = do_GetService("@mozilla.org/psm;1", &res);
+    NS_ENSURE_SUCCESS(res, res);
+  } else {
+    NS_ENSURE_SUCCESS(res = InitNSSInContent(), res);
+  }
+
+  // Currently no standalone unit tests for DataChannel,
+  // which is the user of mWindow
+  MOZ_ASSERT(aWindow);
+  mWindow = aWindow;
+  NS_ENSURE_STATE(mWindow);
+
+  PRTime timestamp = PR_Now();
+  // Ok if we truncate this, but we want it to be large enough to reliably
+  // contain the location on the tests we run in CI.
+  char temp[256];
+
+  nsAutoCString locationCStr;
+
+  RefPtr<Location> location = mWindow->Location();
+  nsAutoString locationAStr;
+  res = location->ToString(locationAStr);
+  NS_ENSURE_SUCCESS(res, res);
+
+  CopyUTF16toUTF8(locationAStr, locationCStr);
+
+  SprintfLiteral(temp, "%s %" PRIu64 " (id=%" PRIu64 " url=%s)",
+                 mHandle.c_str(), static_cast<uint64_t>(timestamp),
+                 static_cast<uint64_t>(mWindow ? mWindow->WindowID() : 0),
+                 locationCStr.get() ? locationCStr.get() : "NULL");
+
+  mName = temp;
+
+  STAMP_TIMECARD(mTimeCard, "Initializing PC Ctx");
+  res = PeerConnectionCtx::InitializeGlobal();
+  NS_ENSURE_SUCCESS(res, res);
+
+  mTransportHandler->CreateIceCtx("PC:" + GetName());
+
+  mJsepSession =
+      MakeUnique<JsepSessionImpl>(mName, MakeUnique<PCUuidGenerator>());
+  mJsepSession->SetRtxIsAllowed(mRtxIsAllowed);
+
+  res = mJsepSession->Init();
+  if (NS_FAILED(res)) {
+    CSFLogError(LOGTAG, "%s: Couldn't init JSEP Session, res=%u", __FUNCTION__,
+                static_cast<unsigned>(res));
+    return res;
+  }
+
+  std::vector<UniquePtr<JsepCodecDescription>> preferredCodecs;
+  SetupPreferredCodecs(preferredCodecs);
+  mJsepSession->SetDefaultCodecs(preferredCodecs);
+
+  std::vector<RtpExtensionHeader> preferredHeaders;
+  SetupPreferredRtpExtensions(preferredHeaders);
+
+  for (const auto& header : preferredHeaders) {
+    mJsepSession->AddRtpExtension(header.mMediaType, header.extensionname,
+                                  header.direction);
+  }
+
+  if (XRE_IsContentProcess()) {
+    mStunAddrsRequest =
+        new net::StunAddrsRequestChild(new StunAddrsHandler(this));
+  }
+
+  // Initialize the media object.
+  mForceProxy = ShouldForceProxy();
+
+  // We put this here, in case we later want to set this based on a non-standard
+  // param in RTCConfiguration.
+  mAllowOldSetParameters = Preferences::GetBool(
+      "media.peerconnection.allow_old_setParameters", false);
+
+  // setup the stun local addresses IPC async call
+  InitLocalAddrs();
+
+  mSignalHandler = MakeUnique<SignalHandler>(this, mTransportHandler.get());
+
+  PeerConnectionCtx::GetInstance()->AddPeerConnection(mHandle, this);
+
+  return NS_OK;
+}
+
+void PeerConnectionImpl::Initialize(PeerConnectionObserver& aObserver,
+                                    nsGlobalWindowInner& aWindow,
+                                    ErrorResult& rv) {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  nsresult res = Initialize(aObserver, &aWindow);
+  if (NS_FAILED(res)) {
+    rv.Throw(res);
+    return;
+  }
+}
+
+void PeerConnectionImpl::SetCertificate(
+    mozilla::dom::RTCCertificate& aCertificate) {
+  PC_AUTO_ENTER_API_CALL_NO_CHECK();
+  MOZ_ASSERT(!mCertificate, "This can only be called once");
+  mCertificate = &aCertificate;
+
+  std::vector<uint8_t> fingerprint;
+  nsresult rv =
+      CalculateFingerprint(DtlsIdentity::DEFAULT_HASH_ALGORITHM, &fingerprint);
+  if (NS_FAILED(rv)) {
+    CSFLogError(LOGTAG, "%s: Couldn't calculate fingerprint, rv=%u",
+                __FUNCTION__, static_cast<unsigned>(rv));
+    mCertificate = nullptr;
+    return;
+  }
+  rv = mJsepSession->AddDtlsFingerprint(DtlsIdentity::DEFAULT_HASH_ALGORITHM,
+                                        fingerprint);
+  if (NS_FAILED(rv)) {
+    CSFLogError(LOGTAG, "%s: Couldn't set DTLS credentials, rv=%u",
+                __FUNCTION__, static_cast<unsigned>(rv));
+    mCertificate = nullptr;
+  }
+
+  if (mUncommittedJsepSession) {
+    Unused << mUncommittedJsepSession->AddDtlsFingerprint(
+        DtlsIdentity::DEFAULT_HASH_ALGORITHM, fingerprint);
+  }
+}
+
+const RefPtr<mozilla::dom::RTCCertificate>& PeerConnectionImpl::Certificate()
+    const {
+  PC_AUTO_ENTER_API_CALL_NO_CHECK();
+  return mCertificate;
+}
+
+RefPtr<DtlsIdentity> PeerConnectionImpl::Identity() const {
+  PC_AUTO_ENTER_API_CALL_NO_CHECK();
+  MOZ_ASSERT(mCertificate);
+  return mCertificate->CreateDtlsIdentity();
+}
+
+class CompareCodecPriority {
+ public:
+  void SetPreferredCodec(int32_t preferredCodec) {
+    // This pref really ought to be a string, preferably something like
+    // "H264" or "VP8" instead of a payload type.
+    // Bug 1101259.
+    std::ostringstream os;
+    os << preferredCodec;
+    mPreferredCodec = os.str();
+  }
+
+  bool operator()(const UniquePtr<JsepCodecDescription>& lhs,
+                  const UniquePtr<JsepCodecDescription>& rhs) const {
+    if (!mPreferredCodec.empty() && lhs->mDefaultPt == mPreferredCodec &&
+        rhs->mDefaultPt != mPreferredCodec) {
+      return true;
+    }
+
+    if (lhs->mStronglyPreferred && !rhs->mStronglyPreferred) {
+      return true;
+    }
+
+    return false;
+  }
+
+ private:
+  std::string mPreferredCodec;
+};
+
+class ConfigureCodec {
+ public:
+  explicit ConfigureCodec(nsCOMPtr<nsIPrefBranch>& branch)
+      : mHardwareH264Enabled(false),
+        mSoftwareH264Enabled(false),
+        mH264Enabled(false),
+        mVP9Enabled(true),
+        mVP9Preferred(false),
+        mH264Level(13),   // minimum suggested for WebRTC spec
+        mH264MaxBr(0),    // Unlimited
+        mH264MaxMbps(0),  // Unlimited
+        mVP8MaxFs(0),
+        mVP8MaxFr(0),
+        mUseTmmbr(false),
+        mUseRemb(false),
+        mUseTransportCC(false),
+        mUseAudioFec(false),
+        mRedUlpfecEnabled(false),
+        mDtmfEnabled(false) {
+    mSoftwareH264Enabled = PeerConnectionCtx::GetInstance()->gmpHasH264();
+
+    if (WebrtcVideoConduit::HasH264Hardware()) {
+      Telemetry::Accumulate(Telemetry::WEBRTC_HAS_H264_HARDWARE, true);
+      branch->GetBoolPref("media.webrtc.hw.h264.enabled",
+                          &mHardwareH264Enabled);
+    }
+
+    mH264Enabled = mHardwareH264Enabled || mSoftwareH264Enabled;
+    Telemetry::Accumulate(Telemetry::WEBRTC_SOFTWARE_H264_ENABLED,
+                          mSoftwareH264Enabled);
+    Telemetry::Accumulate(Telemetry::WEBRTC_HARDWARE_H264_ENABLED,
+                          mHardwareH264Enabled);
+    Telemetry::Accumulate(Telemetry::WEBRTC_H264_ENABLED, mH264Enabled);
+
+    branch->GetIntPref("media.navigator.video.h264.level", &mH264Level);
+    mH264Level &= 0xFF;
+
+    branch->GetIntPref("media.navigator.video.h264.max_br", &mH264MaxBr);
+
+    branch->GetIntPref("media.navigator.video.h264.max_mbps", &mH264MaxMbps);
+
+    branch->GetBoolPref("media.peerconnection.video.vp9_enabled", &mVP9Enabled);
+
+    branch->GetBoolPref("media.peerconnection.video.vp9_preferred",
+                        &mVP9Preferred);
+
+    branch->GetIntPref("media.navigator.video.max_fs", &mVP8MaxFs);
+    if (mVP8MaxFs <= 0) {
+      mVP8MaxFs = 12288;  // We must specify something other than 0
+    }
+
+    branch->GetIntPref("media.navigator.video.max_fr", &mVP8MaxFr);
+    if (mVP8MaxFr <= 0) {
+      mVP8MaxFr = 60;  // We must specify something other than 0
+    }
+
+    // TMMBR is enabled from a pref in about:config
+    branch->GetBoolPref("media.navigator.video.use_tmmbr", &mUseTmmbr);
+
+    // REMB is enabled by default, but can be disabled from about:config
+    branch->GetBoolPref("media.navigator.video.use_remb", &mUseRemb);
+
+    branch->GetBoolPref("media.navigator.video.use_transport_cc",
+                        &mUseTransportCC);
+
+    branch->GetBoolPref("media.navigator.audio.use_fec", &mUseAudioFec);
+
+    branch->GetBoolPref("media.navigator.video.red_ulpfec_enabled",
+                        &mRedUlpfecEnabled);
+
+    // media.peerconnection.dtmf.enabled controls both sdp generation for
+    // DTMF support as well as DTMF exposure to DOM
+    branch->GetBoolPref("media.peerconnection.dtmf.enabled", &mDtmfEnabled);
+  }
+
+  void operator()(UniquePtr<JsepCodecDescription>& codec) const {
+    switch (codec->Type()) {
+      case SdpMediaSection::kAudio: {
+        JsepAudioCodecDescription& audioCodec =
+            static_cast<JsepAudioCodecDescription&>(*codec);
+        if (audioCodec.mName == "opus") {
+          audioCodec.mFECEnabled = mUseAudioFec;
+        } else if (audioCodec.mName == "telephone-event") {
+          audioCodec.mEnabled = mDtmfEnabled;
+        }
+      } break;
+      case SdpMediaSection::kVideo: {
+        JsepVideoCodecDescription& videoCodec =
+            static_cast<JsepVideoCodecDescription&>(*codec);
+
+        if (videoCodec.mName == "H264") {
+          // Override level
+          videoCodec.mProfileLevelId &= 0xFFFF00;
+          videoCodec.mProfileLevelId |= mH264Level;
+
+          videoCodec.mConstraints.maxBr = mH264MaxBr;
+
+          videoCodec.mConstraints.maxMbps = mH264MaxMbps;
+
+          // Might disable it, but we set up other params anyway
+          videoCodec.mEnabled = mH264Enabled;
+
+          if (videoCodec.mPacketizationMode == 0 && !mSoftwareH264Enabled) {
+            // We're assuming packetization mode 0 is unsupported by
+            // hardware.
+            videoCodec.mEnabled = false;
+          }
+
+          if (mHardwareH264Enabled) {
+            videoCodec.mStronglyPreferred = true;
+          }
+        } else if (videoCodec.mName == "red") {
+          videoCodec.mEnabled = mRedUlpfecEnabled;
+        } else if (videoCodec.mName == "ulpfec") {
+          videoCodec.mEnabled = mRedUlpfecEnabled;
+        } else if (videoCodec.mName == "VP8" || videoCodec.mName == "VP9") {
+          if (videoCodec.mName == "VP9") {
+            if (!mVP9Enabled) {
+              videoCodec.mEnabled = false;
+              break;
+            }
+            if (mVP9Preferred) {
+              videoCodec.mStronglyPreferred = true;
+            }
+          }
+          videoCodec.mConstraints.maxFs = mVP8MaxFs;
+          videoCodec.mConstraints.maxFps = Some(mVP8MaxFr);
+        }
+
+        if (mUseTmmbr) {
+          videoCodec.EnableTmmbr();
+        }
+        if (mUseRemb) {
+          videoCodec.EnableRemb();
+        }
+        if (mUseTransportCC) {
+          videoCodec.EnableTransportCC();
+        }
+      } break;
+      case SdpMediaSection::kText:
+      case SdpMediaSection::kApplication:
+      case SdpMediaSection::kMessage: {
+      }  // Nothing to configure for these.
+    }
+  }
+
+ private:
+  bool mHardwareH264Enabled;
+  bool mSoftwareH264Enabled;
+  bool mH264Enabled;
+  bool mVP9Enabled;
+  bool mVP9Preferred;
+  int32_t mH264Level;
+  int32_t mH264MaxBr;
+  int32_t mH264MaxMbps;
+  int32_t mVP8MaxFs;
+  int32_t mVP8MaxFr;
+  bool mUseTmmbr;
+  bool mUseRemb;
+  bool mUseTransportCC;
+  bool mUseAudioFec;
+  bool mRedUlpfecEnabled;
+  bool mDtmfEnabled;
+};
+
+class ConfigureRedCodec {
+ public:
+  explicit ConfigureRedCodec(nsCOMPtr<nsIPrefBranch>& branch,
+                             std::vector<uint8_t>* redundantEncodings)
+      : mRedundantEncodings(redundantEncodings) {
+    // if we wanted to override or modify which encodings are considered
+    // for redundant encodings, we'd probably want to handle it here by
+    // checking prefs modifying the operator() code below
+  }
+
+  void operator()(UniquePtr<JsepCodecDescription>& codec) const {
+    if (codec->Type() == SdpMediaSection::kVideo && !codec->mEnabled) {
+      uint8_t pt = (uint8_t)strtoul(codec->mDefaultPt.c_str(), nullptr, 10);
+      // don't search for the codec payload type unless we have a valid
+      // conversion (non-zero)
+      if (pt != 0) {
+        std::vector<uint8_t>::iterator it = std::find(
+            mRedundantEncodings->begin(), mRedundantEncodings->end(), pt);
+        if (it != mRedundantEncodings->end()) {
+          mRedundantEncodings->erase(it);
+        }
+      }
+    }
+  }
+
+ private:
+  std::vector<uint8_t>* mRedundantEncodings;
+};
+
+nsresult PeerConnectionImpl::ConfigureJsepSessionCodecs() {
+  nsresult res;
+  nsCOMPtr<nsIPrefService> prefs =
+      do_GetService("@mozilla.org/preferences-service;1", &res);
+
+  if (NS_FAILED(res)) {
+    CSFLogError(LOGTAG, "%s: Couldn't get prefs service, res=%u", __FUNCTION__,
+                static_cast<unsigned>(res));
+    return res;
+  }
+
+  nsCOMPtr<nsIPrefBranch> branch = do_QueryInterface(prefs);
+  if (!branch) {
+    CSFLogError(LOGTAG, "%s: Couldn't get prefs branch", __FUNCTION__);
+    return NS_ERROR_FAILURE;
+  }
+
+  ConfigureCodec configurer(branch);
+  mJsepSession->ForEachCodec(configurer);
+
+  // if red codec is enabled, configure it for the other enabled codecs
+  for (auto& codec : mJsepSession->Codecs()) {
+    if (codec->mName == "red" && codec->mEnabled) {
+      JsepVideoCodecDescription* redCodec =
+          static_cast<JsepVideoCodecDescription*>(codec.get());
+      ConfigureRedCodec configureRed(branch, &(redCodec->mRedundantEncodings));
+      mJsepSession->ForEachCodec(configureRed);
+      break;
+    }
+  }
+
+  // We use this to sort the list of codecs once everything is configured
+  CompareCodecPriority comparator;
+
+  // Sort by priority
+  int32_t preferredCodec = 0;
+  branch->GetIntPref("media.navigator.video.preferred_codec", &preferredCodec);
+
+  if (preferredCodec) {
+    comparator.SetPreferredCodec(preferredCodec);
+  }
+
+  mJsepSession->SortCodecs(comparator);
+  return NS_OK;
+}
+
+// Data channels won't work without a window, so in order for the C++ unit
+// tests to work (it doesn't have a window available) we ifdef the following
+// two implementations.
+//
+// Note: 'media.peerconnection.sctp.force_maximum_message_size' changes
+// behaviour triggered by these parameters.
+NS_IMETHODIMP
+PeerConnectionImpl::EnsureDataConnection(uint16_t aLocalPort,
+                                         uint16_t aNumstreams,
+                                         uint32_t aMaxMessageSize,
+                                         bool aMMSSet) {
+  PC_AUTO_ENTER_API_CALL(false);
+
+  if (mDataConnection) {
+    CSFLogDebug(LOGTAG, "%s DataConnection already connected", __FUNCTION__);
+    mDataConnection->SetMaxMessageSize(aMMSSet, aMaxMessageSize);
+    return NS_OK;
+  }
+
+  nsCOMPtr<nsISerialEventTarget> target =
+      mWindow ? mWindow->EventTargetFor(TaskCategory::Other) : nullptr;
+  Maybe<uint64_t> mms = aMMSSet ? Some(aMaxMessageSize) : Nothing();
+  if (auto res = DataChannelConnection::Create(this, target, mTransportHandler,
+                                               aLocalPort, aNumstreams, mms)) {
+    mDataConnection = res.value();
+    CSFLogDebug(LOGTAG, "%s DataChannelConnection %p attached to %s",
+                __FUNCTION__, (void*)mDataConnection.get(), mHandle.c_str());
+    return NS_OK;
+  }
+  CSFLogError(LOGTAG, "%s DataConnection Create Failed", __FUNCTION__);
+  return NS_ERROR_FAILURE;
+}
+
+nsresult PeerConnectionImpl::GetDatachannelParameters(
+    uint32_t* channels, uint16_t* localport, uint16_t* remoteport,
+    uint32_t* remotemaxmessagesize, bool* mmsset, std::string* transportId,
+    bool* client) const {
+  // Clear, just in case we fail.
+  *channels = 0;
+  *localport = 0;
+  *remoteport = 0;
+  *remotemaxmessagesize = 0;
+  *mmsset = false;
+  transportId->clear();
+
+  Maybe<const JsepTransceiver> datachannelTransceiver =
+      mJsepSession->FindTransceiver([](const JsepTransceiver& aTransceiver) {
+        return aTransceiver.GetMediaType() == SdpMediaSection::kApplication;
+      });
+
+  if (!datachannelTransceiver ||
+      !datachannelTransceiver->mTransport.mComponents ||
+      !datachannelTransceiver->mSendTrack.GetNegotiatedDetails()) {
+    return NS_ERROR_FAILURE;
+  }
+
+  // This will release assert if there is no such index, and that's ok
+  const JsepTrackEncoding& encoding =
+      datachannelTransceiver->mSendTrack.GetNegotiatedDetails()->GetEncoding(0);
+
+  if (NS_WARN_IF(encoding.GetCodecs().empty())) {
+    CSFLogError(LOGTAG,
+                "%s: Negotiated m=application with no codec. "
+                "This is likely to be broken.",
+                __FUNCTION__);
+    return NS_ERROR_FAILURE;
+  }
+
+  for (const auto& codec : encoding.GetCodecs()) {
+    if (codec->Type() != SdpMediaSection::kApplication) {
+      CSFLogError(LOGTAG,
+                  "%s: Codec type for m=application was %u, this "
+                  "is a bug.",
+                  __FUNCTION__, static_cast<unsigned>(codec->Type()));
+      MOZ_ASSERT(false, "Codec for m=application was not \"application\"");
+      return NS_ERROR_FAILURE;
+    }
+
+    if (codec->mName != "webrtc-datachannel") {
+      CSFLogWarn(LOGTAG,
+                 "%s: Codec for m=application was not "
+                 "webrtc-datachannel (was instead %s). ",
+                 __FUNCTION__, codec->mName.c_str());
+      continue;
+    }
+
+    if (codec->mChannels) {
+      *channels = codec->mChannels;
+    } else {
+      *channels = WEBRTC_DATACHANNEL_STREAMS_DEFAULT;
+    }
+    const JsepApplicationCodecDescription* appCodec =
+        static_cast<const JsepApplicationCodecDescription*>(codec.get());
+    *localport = appCodec->mLocalPort;
+    *remoteport = appCodec->mRemotePort;
+    *remotemaxmessagesize = appCodec->mRemoteMaxMessageSize;
+    *mmsset = appCodec->mRemoteMMSSet;
+    MOZ_ASSERT(!datachannelTransceiver->mTransport.mTransportId.empty());
+    *transportId = datachannelTransceiver->mTransport.mTransportId;
+    *client = datachannelTransceiver->mTransport.mDtls->GetRole() ==
+              JsepDtlsTransport::kJsepDtlsClient;
+    return NS_OK;
+  }
+  return NS_ERROR_FAILURE;
+}
+
+nsresult PeerConnectionImpl::AddRtpTransceiverToJsepSession(
+    JsepTransceiver& transceiver) {
+  nsresult res = ConfigureJsepSessionCodecs();
+  if (NS_FAILED(res)) {
+    CSFLogError(LOGTAG, "Failed to configure codecs");
+    return res;
+  }
+
+  mJsepSession->AddTransceiver(transceiver);
+  return NS_OK;
+}
+
+static Maybe<SdpMediaSection::MediaType> ToSdpMediaType(
+    const nsAString& aKind) {
+  if (aKind.EqualsASCII("audio")) {
+    return Some(SdpMediaSection::MediaType::kAudio);
+  } else if (aKind.EqualsASCII("video")) {
+    return Some(SdpMediaSection::MediaType::kVideo);
+  }
+  return Nothing();
+}
+
+already_AddRefed<RTCRtpTransceiver> PeerConnectionImpl::AddTransceiver(
+    const dom::RTCRtpTransceiverInit& aInit, const nsAString& aKind,
+    dom::MediaStreamTrack* aSendTrack, bool aAddTrackMagic, ErrorResult& aRv) {
+  // Copy, because we might need to modify
+  RTCRtpTransceiverInit init(aInit);
+
+  Maybe<SdpMediaSection::MediaType> type = ToSdpMediaType(aKind);
+  if (NS_WARN_IF(!type.isSome())) {
+    MOZ_ASSERT(false, "Invalid media kind");
+    aRv = NS_ERROR_INVALID_ARG;
+    return nullptr;
+  }
+
+  JsepTransceiver jsepTransceiver(*type, *mUuidGen);
+  jsepTransceiver.SetRtxIsAllowed(mRtxIsAllowed);
+
+  // Do this last, since it is not possible to roll back.
+  nsresult rv = AddRtpTransceiverToJsepSession(jsepTransceiver);
+  if (NS_FAILED(rv)) {
+    CSFLogError(LOGTAG, "%s: AddRtpTransceiverToJsepSession failed, res=%u",
+                __FUNCTION__, static_cast<unsigned>(rv));
+    aRv = rv;
+    return nullptr;
+  }
+
+  auto& sendEncodings = init.mSendEncodings;
+
+  // CheckAndRectifyEncodings covers these six:
+  // If any encoding contains a rid member whose value does not conform to the
+  // grammar requirements specified in Section 10 of [RFC8851], throw a
+  // TypeError.
+
+  // If some but not all encodings contain a rid member, throw a TypeError.
+
+  // If any encoding contains a rid member whose value is the same as that of a
+  // rid contained in another encoding in sendEncodings, throw a TypeError.
+
+  // If kind is "audio", remove the scaleResolutionDownBy member from all
+  // encodings that contain one.
+
+  // If any encoding contains a scaleResolutionDownBy member whose value is
+  // less than 1.0, throw a RangeError.
+
+  // Verify that the value of each maxFramerate member in sendEncodings that is
+  // defined is greater than 0.0. If one of the maxFramerate values does not
+  // meet this requirement, throw a RangeError.
+  RTCRtpSender::CheckAndRectifyEncodings(sendEncodings,
+                                         *type == SdpMediaSection::kVideo, aRv);
+  if (aRv.Failed()) {
+    return nullptr;
+  }
+
+  // If any encoding contains a read-only parameter other than rid, throw an
+  // InvalidAccessError.
+  // NOTE: We don't support any additional read-only params right now. Also,
+  // spec shoehorns this in between checks that setParameters also performs
+  // (between the rid checks and the scaleResolutionDownBy checks).
+
+  // If any encoding contains a scaleResolutionDownBy member, then for each
+  // encoding without one, add a scaleResolutionDownBy member with the value
+  // 1.0.
+  for (const auto& constEncoding : sendEncodings) {
+    if (constEncoding.mScaleResolutionDownBy.WasPassed()) {
+      for (auto& encoding : sendEncodings) {
+        if (!encoding.mScaleResolutionDownBy.WasPassed()) {
+          encoding.mScaleResolutionDownBy.Construct(1.0f);
+        }
+      }
+      break;
+    }
+  }
+
+  // Let maxN be the maximum number of total simultaneous encodings the user
+  // agent may support for this kind, at minimum 1.This should be an optimistic
+  // number since the codec to be used is not known yet.
+  size_t maxN =
+      (*type == SdpMediaSection::kVideo) ? webrtc::kMaxSimulcastStreams : 1;
+
+  // If the number of encodings stored in sendEncodings exceeds maxN, then trim
+  // sendEncodings from the tail until its length is maxN.
+  // NOTE: Spec has this after all validation steps; even if there are elements
+  // that we will trim off, we still validate them.
+  if (sendEncodings.Length() > maxN) {
+    sendEncodings.TruncateLength(maxN);
+  }
+
+  // If kind is "video" and none of the encodings contain a
+  // scaleResolutionDownBy member, then for each encoding, add a
+  // scaleResolutionDownBy member with the value 2^(length of sendEncodings -
+  // encoding index - 1). This results in smaller-to-larger resolutions where
+  // the last encoding has no scaling applied to it, e.g. 4:2:1 if the length
+  // is 3.
+  // NOTE: The code above ensures that these are all set, or all unset, so we
+  // can just check the first one.
+  if (sendEncodings.Length() && *type == SdpMediaSection::kVideo &&
+      !sendEncodings[0].mScaleResolutionDownBy.WasPassed()) {
+    double scale = 1.0f;
+    for (auto it = sendEncodings.rbegin(); it != sendEncodings.rend(); ++it) {
+      it->mScaleResolutionDownBy.Construct(scale);
+      scale *= 2;
+    }
+  }
+
+  // If the number of encodings now stored in sendEncodings is 1, then remove
+  // any rid member from the lone entry.
+  if (sendEncodings.Length() == 1) {
+    sendEncodings[0].mRid.Reset();
+  }
+
+  RefPtr<RTCRtpTransceiver> transceiver = CreateTransceiver(
+      jsepTransceiver.GetUuid(),
+      jsepTransceiver.GetMediaType() == SdpMediaSection::kVideo, init,
+      aSendTrack, aAddTrackMagic, aRv);
+
+  if (aRv.Failed()) {
+    // Would be nice if we could peek at the rv without stealing it, so we
+    // could log...
+    CSFLogError(LOGTAG, "%s: failed", __FUNCTION__);
+    return nullptr;
+  }
+
+  mTransceivers.AppendElement(transceiver);
+  return transceiver.forget();
+}
+
+bool PeerConnectionImpl::CheckNegotiationNeeded() {
+  MOZ_ASSERT(mSignalingState == RTCSignalingState::Stable);
+  SyncToJsep();
+  return !mLocalIceCredentialsToReplace.empty() ||
+         mJsepSession->CheckNegotiationNeeded();
+}
+
+bool PeerConnectionImpl::CreatedSender(const dom::RTCRtpSender& aSender) const {
+  return aSender.IsMyPc(this);
+}
+
+nsresult PeerConnectionImpl::InitializeDataChannel() {
+  PC_AUTO_ENTER_API_CALL(false);
+  CSFLogDebug(LOGTAG, "%s", __FUNCTION__);
+
+  uint32_t channels = 0;
+  uint16_t localport = 0;
+  uint16_t remoteport = 0;
+  uint32_t remotemaxmessagesize = 0;
+  bool mmsset = false;
+  std::string transportId;
+  bool client = false;
+  nsresult rv = GetDatachannelParameters(&channels, &localport, &remoteport,
+                                         &remotemaxmessagesize, &mmsset,
+                                         &transportId, &client);
+
+  if (NS_FAILED(rv)) {
+    CSFLogDebug(LOGTAG, "%s: We did not negotiate datachannel", __FUNCTION__);
+    return NS_OK;
+  }
+
+  if (channels > MAX_NUM_STREAMS) {
+    channels = MAX_NUM_STREAMS;
+  }
+
+  rv = EnsureDataConnection(localport, channels, remotemaxmessagesize, mmsset);
+  if (NS_SUCCEEDED(rv)) {
+    if (mDataConnection->ConnectToTransport(transportId, client, localport,
+                                            remoteport)) {
+      return NS_OK;
+    }
+    // If we inited the DataConnection, call Destroy() before releasing it
+    mDataConnection->Destroy();
+  }
+  mDataConnection = nullptr;
+  return NS_ERROR_FAILURE;
+}
+
+already_AddRefed<nsDOMDataChannel> PeerConnectionImpl::CreateDataChannel(
+    const nsAString& aLabel, const nsAString& aProtocol, uint16_t aType,
+    bool ordered, uint16_t aMaxTime, uint16_t aMaxNum, bool aExternalNegotiated,
+    uint16_t aStream, ErrorResult& rv) {
+  RefPtr<nsDOMDataChannel> result;
+  rv = CreateDataChannel(aLabel, aProtocol, aType, ordered, aMaxTime, aMaxNum,
+                         aExternalNegotiated, aStream, getter_AddRefs(result));
+  return result.forget();
+}
+
+NS_IMETHODIMP
+PeerConnectionImpl::CreateDataChannel(
+    const nsAString& aLabel, const nsAString& aProtocol, uint16_t aType,
+    bool ordered, uint16_t aMaxTime, uint16_t aMaxNum, bool aExternalNegotiated,
+    uint16_t aStream, nsDOMDataChannel** aRetval) {
+  PC_AUTO_ENTER_API_CALL(false);
+  MOZ_ASSERT(aRetval);
+
+  RefPtr<DataChannel> dataChannel;
+  DataChannelConnection::Type theType =
+      static_cast<DataChannelConnection::Type>(aType);
+
+  nsresult rv = EnsureDataConnection(
+      WEBRTC_DATACHANNEL_PORT_DEFAULT, WEBRTC_DATACHANNEL_STREAMS_DEFAULT,
+      WEBRTC_DATACHANNEL_MAX_MESSAGE_SIZE_REMOTE_DEFAULT, false);
+  if (NS_FAILED(rv)) {
+    return rv;
+  }
+  dataChannel = mDataConnection->Open(
+      NS_ConvertUTF16toUTF8(aLabel), NS_ConvertUTF16toUTF8(aProtocol), theType,
+      ordered,
+      aType == DataChannelConnection::PARTIAL_RELIABLE_REXMIT
+          ? aMaxNum
+          : (aType == DataChannelConnection::PARTIAL_RELIABLE_TIMED ? aMaxTime
+                                                                    : 0),
+      nullptr, nullptr, aExternalNegotiated, aStream);
+  NS_ENSURE_TRUE(dataChannel, NS_ERROR_NOT_AVAILABLE);
+
+  CSFLogDebug(LOGTAG, "%s: making DOMDataChannel", __FUNCTION__);
+
+  Maybe<JsepTransceiver> dcTransceiver =
+      mJsepSession->FindTransceiver([](const JsepTransceiver& aTransceiver) {
+        return aTransceiver.GetMediaType() == SdpMediaSection::kApplication;
+      });
+
+  if (dcTransceiver) {
+    dcTransceiver->RestartDatachannelTransceiver();
+    mJsepSession->SetTransceiver(*dcTransceiver);
+  } else {
+    mJsepSession->AddTransceiver(
+        JsepTransceiver(SdpMediaSection::MediaType::kApplication, *mUuidGen));
+  }
+
+  RefPtr<nsDOMDataChannel> retval;
+  rv = NS_NewDOMDataChannel(dataChannel.forget(), mWindow,
+                            getter_AddRefs(retval));
+  if (NS_FAILED(rv)) {
+    return rv;
+  }
+  retval.forget(aRetval);
+  return NS_OK;
+}
+
+NS_IMPL_CYCLE_COLLECTION(PeerConnectionImpl::Operation, mPromise, mPc)
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(PeerConnectionImpl::Operation)
+  NS_INTERFACE_MAP_ENTRY(nsISupports)
+NS_INTERFACE_MAP_END
+NS_IMPL_CYCLE_COLLECTING_ADDREF(PeerConnectionImpl::Operation)
+NS_IMPL_CYCLE_COLLECTING_RELEASE(PeerConnectionImpl::Operation)
+
+PeerConnectionImpl::Operation::Operation(PeerConnectionImpl* aPc,
+                                         ErrorResult& aError)
+    : mPromise(aPc->MakePromise(aError)), mPc(aPc) {}
+
+PeerConnectionImpl::Operation::~Operation() = default;
+
+void PeerConnectionImpl::Operation::Call(ErrorResult& aError) {
+  RefPtr<dom::Promise> opPromise = CallImpl(aError);
+  if (aError.Failed()) {
+    return;
+  }
+  // Upon fulfillment or rejection of the promise returned by the operation,
+  // run the following steps:
+  // (NOTE: mPromise is p from https://w3c.github.io/webrtc-pc/#dfn-chain,
+  // and CallImpl() is what returns the promise for the operation itself)
+  opPromise->AppendNativeHandler(this);
+}
+
+void PeerConnectionImpl::Operation::ResolvedCallback(
+    JSContext* aCx, JS::Handle<JS::Value> aValue, ErrorResult& aRv) {
+  // If connection.[[IsClosed]] is true, abort these steps.
+  // (the spec wants p to never settle in this event)
+  if (!mPc->IsClosed()) {
+    // If the promise returned by operation was fulfilled with a
+    // value, fulfill p with that value.
+    mPromise->MaybeResolveWithClone(aCx, aValue);
+    // Upon fulfillment or rejection of p, execute the following
+    // steps:
+    // (Static analysis forces us to use a temporary)
+    RefPtr<PeerConnectionImpl> pc = mPc;
+    pc->RunNextOperation(aRv);
+  }
+}
+
+void PeerConnectionImpl::Operation::RejectedCallback(
+    JSContext* aCx, JS::Handle<JS::Value> aValue, ErrorResult& aRv) {
+  // If connection.[[IsClosed]] is true, abort these steps.
+  // (the spec wants p to never settle in this event)
+  if (!mPc->IsClosed()) {
+    // If the promise returned by operation was rejected with a
+    // value, reject p with that value.
+    mPromise->MaybeRejectWithClone(aCx, aValue);
+    // Upon fulfillment or rejection of p, execute the following
+    // steps:
+    // (Static analysis forces us to use a temporary)
+    RefPtr<PeerConnectionImpl> pc = mPc;
+    pc->RunNextOperation(aRv);
+  }
+}
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(PeerConnectionImpl::JSOperation,
+                                   PeerConnectionImpl::Operation, mOperation)
+
+NS_IMPL_ADDREF_INHERITED(PeerConnectionImpl::JSOperation,
+                         PeerConnectionImpl::Operation)
+NS_IMPL_RELEASE_INHERITED(PeerConnectionImpl::JSOperation,
+                          PeerConnectionImpl::Operation)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(PeerConnectionImpl::JSOperation)
+NS_INTERFACE_MAP_END_INHERITING(PeerConnectionImpl::Operation)
+
+PeerConnectionImpl::JSOperation::JSOperation(PeerConnectionImpl* aPc,
+                                             dom::ChainedOperation& aOp,
+                                             ErrorResult& aError)
+    : Operation(aPc, aError), mOperation(&aOp) {}
+
+RefPtr<dom::Promise> PeerConnectionImpl::JSOperation::CallImpl(
+    ErrorResult& aError) {
+  // Static analysis will not let us call this without a temporary :(
+  RefPtr<dom::ChainedOperation> op = mOperation;
+  return op->Call(aError);
+}
+
+already_AddRefed<dom::Promise> PeerConnectionImpl::Chain(
+    dom::ChainedOperation& aOperation, ErrorResult& aError) {
+  MOZ_RELEASE_ASSERT(!mChainingOperation);
+  mChainingOperation = true;
+  RefPtr<Operation> operation = new JSOperation(this, aOperation, aError);
+  if (aError.Failed()) {
+    return nullptr;
+  }
+  RefPtr<Promise> promise = Chain(operation, aError);
+  if (aError.Failed()) {
+    return nullptr;
+  }
+  mChainingOperation = false;
+  return promise.forget();
+}
+
+// This is kinda complicated, but it is what the spec requires us to do. The
+// core of what makes this complicated is the requirement that |aOperation| be
+// run _immediately_ (without any Promise.Then!) if the operations chain is
+// empty.
+already_AddRefed<dom::Promise> PeerConnectionImpl::Chain(
+    const RefPtr<Operation>& aOperation, ErrorResult& aError) {
+  // If connection.[[IsClosed]] is true, return a promise rejected with a newly
+  // created InvalidStateError.
+  if (IsClosed()) {
+    CSFLogDebug(LOGTAG, "%s:%d: Peer connection is closed", __FILE__, __LINE__);
+    RefPtr<dom::Promise> error = MakePromise(aError);
+    if (aError.Failed()) {
+      return nullptr;
+    }
+    error->MaybeRejectWithInvalidStateError("Peer connection is closed");
+    return error.forget();
+  }
+
+  // Append operation to [[Operations]].
+  mOperations.AppendElement(aOperation);
+
+  // If the length of [[Operations]] is exactly 1, execute operation.
+  if (mOperations.Length() == 1) {
+    aOperation->Call(aError);
+    if (aError.Failed()) {
+      return nullptr;
+    }
+  }
+
+  // This is the promise p from https://w3c.github.io/webrtc-pc/#dfn-chain
+  return do_AddRef(aOperation->GetPromise());
+}
+
+void PeerConnectionImpl::RunNextOperation(ErrorResult& aError) {
+  // If connection.[[IsClosed]] is true, abort these steps.
+  if (IsClosed()) {
+    return;
+  }
+
+  // Remove the first element of [[Operations]].
+  mOperations.RemoveElementAt(0);
+
+  // If [[Operations]] is non-empty, execute the operation represented by the
+  // first element of [[Operations]], and abort these steps.
+  if (mOperations.Length()) {
+    // Cannot call without a temporary :(
+    RefPtr<Operation> op = mOperations[0];
+    op->Call(aError);
+    return;
+  }
+
+  // If connection.[[UpdateNegotiationNeededFlagOnEmptyChain]] is false, abort
+  // these steps.
+  if (!mUpdateNegotiationNeededFlagOnEmptyChain) {
+    return;
+  }
+
+  // Set connection.[[UpdateNegotiationNeededFlagOnEmptyChain]] to false.
+  mUpdateNegotiationNeededFlagOnEmptyChain = false;
+  // Update the negotiation-needed flag for connection.
+  UpdateNegotiationNeeded();
+}
+
+void PeerConnectionImpl::SyncToJsep() {
+  for (const auto& transceiver : mTransceivers) {
+    transceiver->SyncToJsep(*mJsepSession);
+  }
+}
+
+void PeerConnectionImpl::SyncFromJsep() {
+  CSFLogDebug(LOGTAG, "%s", __FUNCTION__);
+  mJsepSession->ForEachTransceiver(
+      [this, self = RefPtr<PeerConnectionImpl>(this)](
+          const JsepTransceiver& jsepTransceiver) {
+        if (jsepTransceiver.GetMediaType() ==
+            SdpMediaSection::MediaType::kApplication) {
+          return;
+        }
+
+        CSFLogDebug(LOGTAG, "%s: Looking for match", __FUNCTION__);
+        RefPtr<RTCRtpTransceiver> transceiver;
+        for (auto& temp : mTransceivers) {
+          if (temp->GetJsepTransceiverId() == jsepTransceiver.GetUuid()) {
+            CSFLogDebug(LOGTAG, "%s: Found match", __FUNCTION__);
+            transceiver = temp;
+            break;
+          }
+        }
+
+        if (!transceiver) {
+          CSFLogDebug(LOGTAG, "%s: No match, making new", __FUNCTION__);
+          dom::RTCRtpTransceiverInit init;
+          init.mDirection = RTCRtpTransceiverDirection::Recvonly;
+          IgnoredErrorResult rv;
+          transceiver = CreateTransceiver(
+              jsepTransceiver.GetUuid(),
+              jsepTransceiver.GetMediaType() == SdpMediaSection::kVideo, init,
+              nullptr, false, rv);
+          if (NS_WARN_IF(rv.Failed())) {
+            MOZ_ASSERT(false);
+            return;
+          }
+          mTransceivers.AppendElement(transceiver);
+        }
+
+        CSFLogDebug(LOGTAG, "%s: Syncing transceiver", __FUNCTION__);
+        transceiver->SyncFromJsep(*mJsepSession);
+      });
+}
+
+already_AddRefed<dom::Promise> PeerConnectionImpl::MakePromise(
+    ErrorResult& aError) const {
+  nsCOMPtr<nsIGlobalObject> global = do_QueryInterface(mWindow);
+  return dom::Promise::Create(global, aError);
+}
+
+void PeerConnectionImpl::UpdateNegotiationNeeded() {
+  // If the length of connection.[[Operations]] is not 0, then set
+  // connection.[[UpdateNegotiationNeededFlagOnEmptyChain]] to true, and abort
+  // these steps.
+  if (mOperations.Length() != 0) {
+    mUpdateNegotiationNeededFlagOnEmptyChain = true;
+    return;
+  }
+
+  // Queue a task to run the following steps:
+  GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction(
+      __func__, [this, self = RefPtr<PeerConnectionImpl>(this)] {
+        // If connection.[[IsClosed]] is true, abort these steps.
+        if (IsClosed()) {
+          return;
+        }
+        // If the length of connection.[[Operations]] is not 0, then set
+        // connection.[[UpdateNegotiationNeededFlagOnEmptyChain]] to true, and
+        // abort these steps.
+        if (mOperations.Length()) {
+          mUpdateNegotiationNeededFlagOnEmptyChain = true;
+          return;
+        }
+        // If connection's signaling state is not "stable", abort these steps.
+        if (mSignalingState != RTCSignalingState::Stable) {
+          return;
+        }
+        // If the result of checking if negotiation is needed is false, clear
+        // the negotiation-needed flag by setting
+        // connection.[[NegotiationNeeded]] to false, and abort these steps.
+        if (!CheckNegotiationNeeded()) {
+          mNegotiationNeeded = false;
+          return;
+        }
+
+        // If connection.[[NegotiationNeeded]] is already true, abort these
+        // steps.
+        if (mNegotiationNeeded) {
+          return;
+        }
+
+        // Set connection.[[NegotiationNeeded]] to true.
+        mNegotiationNeeded = true;
+
+        // Fire an event named negotiationneeded at connection.
+        ErrorResult rv;
+        mPCObserver->FireNegotiationNeededEvent(rv);
+      }));
+}
+
+void PeerConnectionImpl::NotifyDataChannel(
+    already_AddRefed<DataChannel> aChannel) {
+  PC_AUTO_ENTER_API_CALL_NO_CHECK();
+
+  RefPtr<DataChannel> channel(aChannel);
+  MOZ_ASSERT(channel);
+  CSFLogDebug(LOGTAG, "%s: channel: %p", __FUNCTION__, channel.get());
+
+  RefPtr<nsDOMDataChannel> domchannel;
+  nsresult rv = NS_NewDOMDataChannel(channel.forget(), mWindow,
+                                     getter_AddRefs(domchannel));
+  NS_ENSURE_SUCCESS_VOID(rv);
+
+  JSErrorResult jrv;
+  mPCObserver->NotifyDataChannel(*domchannel, jrv);
+}
+
+void PeerConnectionImpl::NotifyDataChannelOpen(DataChannel*) {
+  mDataChannelsOpened++;
+}
+
+void PeerConnectionImpl::NotifyDataChannelClosed(DataChannel*) {
+  mDataChannelsClosed++;
+}
+
+void PeerConnectionImpl::NotifySctpConnected() {
+  if (!mSctpTransport) {
+    MOZ_ASSERT(false);
+    return;
+  }
+
+  mSctpTransport->UpdateState(RTCSctpTransportState::Connected);
+}
+
+void PeerConnectionImpl::NotifySctpClosed() {
+  if (!mSctpTransport) {
+    MOZ_ASSERT(false);
+    return;
+  }
+
+  mSctpTransport->UpdateState(RTCSctpTransportState::Closed);
+}
+
+NS_IMETHODIMP
+PeerConnectionImpl::CreateOffer(const RTCOfferOptions& aOptions) {
+  JsepOfferOptions options;
+  // convert the RTCOfferOptions to JsepOfferOptions
+  if (aOptions.mOfferToReceiveAudio.WasPassed()) {
+    options.mOfferToReceiveAudio =
+        mozilla::Some(size_t(aOptions.mOfferToReceiveAudio.Value()));
+  }
+
+  if (aOptions.mOfferToReceiveVideo.WasPassed()) {
+    options.mOfferToReceiveVideo =
+        mozilla::Some(size_t(aOptions.mOfferToReceiveVideo.Value()));
+  }
+
+  options.mIceRestart = mozilla::Some(aOptions.mIceRestart ||
+                                      !mLocalIceCredentialsToReplace.empty());
+
+  return CreateOffer(options);
+}
+
+static void DeferredCreateOffer(const std::string& aPcHandle,
+                                const JsepOfferOptions& aOptions) {
+  PeerConnectionWrapper wrapper(aPcHandle);
+
+  if (wrapper.impl()) {
+    if (!PeerConnectionCtx::GetInstance()->isReady()) {
+      MOZ_CRASH(
+          "Why is DeferredCreateOffer being executed when the "
+          "PeerConnectionCtx isn't ready?");
+    }
+    wrapper.impl()->CreateOffer(aOptions);
+  }
+}
+
+// Have to use unique_ptr because webidl enums are generated without a
+// copy c'tor.
+static std::unique_ptr<dom::PCErrorData> buildJSErrorData(
+    const JsepSession::Result& aResult, const std::string& aMessage) {
+  std::unique_ptr<dom::PCErrorData> result(new dom::PCErrorData);
+  result->mName = *aResult.mError;
+  result->mMessage = NS_ConvertASCIItoUTF16(aMessage.c_str());
+  return result;
+}
+
+// Used by unit tests and the IDL CreateOffer.
+NS_IMETHODIMP
+PeerConnectionImpl::CreateOffer(const JsepOfferOptions& aOptions) {
+  PC_AUTO_ENTER_API_CALL(true);
+
+  if (!PeerConnectionCtx::GetInstance()->isReady()) {
+    // Uh oh. We're not ready yet. Enqueue this operation.
+    PeerConnectionCtx::GetInstance()->queueJSEPOperation(
+        WrapRunnableNM(DeferredCreateOffer, mHandle, aOptions));
+    STAMP_TIMECARD(mTimeCard, "Deferring CreateOffer (not ready)");
+    return NS_OK;
+  }
+
+  CSFLogDebug(LOGTAG, "CreateOffer()");
+
+  nsresult nrv = ConfigureJsepSessionCodecs();
+  if (NS_FAILED(nrv)) {
+    CSFLogError(LOGTAG, "Failed to configure codecs");
+    return nrv;
+  }
+
+  STAMP_TIMECARD(mTimeCard, "Create Offer");
+
+  GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction(
+      __func__, [this, self = RefPtr<PeerConnectionImpl>(this), aOptions] {
+        std::string offer;
+
+        SyncToJsep();
+        UniquePtr<JsepSession> uncommittedJsepSession(mJsepSession->Clone());
+        JsepSession::Result result =
+            uncommittedJsepSession->CreateOffer(aOptions, &offer);
+        JSErrorResult rv;
+        if (result.mError.isSome()) {
+          std::string errorString = uncommittedJsepSession->GetLastError();
+
+          CSFLogError(LOGTAG, "%s: pc = %s, error = %s", __FUNCTION__,
+                      mHandle.c_str(), errorString.c_str());
+
+          mPCObserver->OnCreateOfferError(
+              *buildJSErrorData(result, errorString), rv);
+        } else {
+          mJsepSession = std::move(uncommittedJsepSession);
+          mPCObserver->OnCreateOfferSuccess(ObString(offer.c_str()), rv);
+        }
+      }));
+
+  return NS_OK;
+}
+
+NS_IMETHODIMP
+PeerConnectionImpl::CreateAnswer() {
+  PC_AUTO_ENTER_API_CALL(true);
+
+  CSFLogDebug(LOGTAG, "CreateAnswer()");
+
+  STAMP_TIMECARD(mTimeCard, "Create Answer");
+  // TODO(bug 1098015): Once RTCAnswerOptions is standardized, we'll need to
+  // add it as a param to CreateAnswer, and convert it here.
+  JsepAnswerOptions options;
+
+  GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction(
+      __func__, [this, self = RefPtr<PeerConnectionImpl>(this), options] {
+        std::string answer;
+        SyncToJsep();
+        UniquePtr<JsepSession> uncommittedJsepSession(mJsepSession->Clone());
+        JsepSession::Result result =
+            uncommittedJsepSession->CreateAnswer(options, &answer);
+        JSErrorResult rv;
+        if (result.mError.isSome()) {
+          std::string errorString = uncommittedJsepSession->GetLastError();
+
+          CSFLogError(LOGTAG, "%s: pc = %s, error = %s", __FUNCTION__,
+                      mHandle.c_str(), errorString.c_str());
+
+          mPCObserver->OnCreateAnswerError(
+              *buildJSErrorData(result, errorString), rv);
+        } else {
+          mJsepSession = std::move(uncommittedJsepSession);
+          mPCObserver->OnCreateAnswerSuccess(ObString(answer.c_str()), rv);
+        }
+      }));
+
+  return NS_OK;
+}
+
+dom::RTCSdpType ToDomSdpType(JsepSdpType aType) {
+  switch (aType) {
+    case kJsepSdpOffer:
+      return dom::RTCSdpType::Offer;
+    case kJsepSdpAnswer:
+      return dom::RTCSdpType::Answer;
+    case kJsepSdpPranswer:
+      return dom::RTCSdpType::Pranswer;
+    case kJsepSdpRollback:
+      return dom::RTCSdpType::Rollback;
+  }
+
+  MOZ_CRASH("Nonexistent JsepSdpType");
+}
+
+JsepSdpType ToJsepSdpType(dom::RTCSdpType aType) {
+  switch (aType) {
+    case dom::RTCSdpType::Offer:
+      return kJsepSdpOffer;
+    case dom::RTCSdpType::Pranswer:
+      return kJsepSdpPranswer;
+    case dom::RTCSdpType::Answer:
+      return kJsepSdpAnswer;
+    case dom::RTCSdpType::Rollback:
+      return kJsepSdpRollback;
+    case dom::RTCSdpType::EndGuard_:;
+  }
+
+  MOZ_CRASH("Nonexistent dom::RTCSdpType");
+}
+
+NS_IMETHODIMP
+PeerConnectionImpl::SetLocalDescription(int32_t aAction, const char* aSDP) {
+  PC_AUTO_ENTER_API_CALL(true);
+
+  if (!aSDP) {
+    CSFLogError(LOGTAG, "%s - aSDP is NULL", __FUNCTION__);
+    return NS_ERROR_FAILURE;
+  }
+
+  STAMP_TIMECARD(mTimeCard, "Set Local Description");
+
+  if (AnyLocalTrackHasPeerIdentity()) {
+    mRequestedPrivacy = Some(PrincipalPrivacy::Private);
+  }
+
+  mozilla::dom::RTCSdpHistoryEntryInternal sdpEntry;
+  sdpEntry.mIsLocal = true;
+  sdpEntry.mTimestamp = mTimestampMaker.GetNow().ToDom();
+  sdpEntry.mSdp = NS_ConvertASCIItoUTF16(aSDP);
+  auto appendHistory = [&]() {
+    if (!mSdpHistory.AppendElement(sdpEntry, fallible)) {
+      mozalloc_handle_oom(0);
+    }
+  };
+
+  mLocalRequestedSDP = aSDP;
+
+  SyncToJsep();
+
+  bool wasRestartingIce = mJsepSession->IsIceRestarting();
+  JsepSdpType sdpType;
+  switch (aAction) {
+    case IPeerConnection::kActionOffer:
+      sdpType = mozilla::kJsepSdpOffer;
+      break;
+    case IPeerConnection::kActionAnswer:
+      sdpType = mozilla::kJsepSdpAnswer;
+      break;
+    case IPeerConnection::kActionPRAnswer:
+      sdpType = mozilla::kJsepSdpPranswer;
+      break;
+    case IPeerConnection::kActionRollback:
+      sdpType = mozilla::kJsepSdpRollback;
+      break;
+    default:
+      MOZ_ASSERT(false);
+      appendHistory();
+      return NS_ERROR_FAILURE;
+  }
+  MOZ_ASSERT(!mUncommittedJsepSession);
+  mUncommittedJsepSession.reset(mJsepSession->Clone());
+  JsepSession::Result result =
+      mUncommittedJsepSession->SetLocalDescription(sdpType, mLocalRequestedSDP);
+  JSErrorResult rv;
+  if (result.mError.isSome()) {
+    std::string errorString = mUncommittedJsepSession->GetLastError();
+    mUncommittedJsepSession = nullptr;
+    CSFLogError(LOGTAG, "%s: pc = %s, error = %s", __FUNCTION__,
+                mHandle.c_str(), errorString.c_str());
+    mPCObserver->OnSetDescriptionError(*buildJSErrorData(result, errorString),
+                                       rv);
+    sdpEntry.mErrors = GetLastSdpParsingErrors();
+  } else {
+    if (wasRestartingIce) {
+      RecordIceRestartStatistics(sdpType);
+    }
+
+    mPCObserver->OnSetDescriptionSuccess(rv);
+  }
+
+  appendHistory();
+
+  if (rv.Failed()) {
+    return rv.StealNSResult();
+  }
+
+  return NS_OK;
+}
+
+static void DeferredSetRemote(const std::string& aPcHandle, int32_t aAction,
+                              const std::string& aSdp) {
+  PeerConnectionWrapper wrapper(aPcHandle);
+
+  if (wrapper.impl()) {
+    if (!PeerConnectionCtx::GetInstance()->isReady()) {
+      MOZ_CRASH(
+          "Why is DeferredSetRemote being executed when the "
+          "PeerConnectionCtx isn't ready?");
+    }
+    wrapper.impl()->SetRemoteDescription(aAction, aSdp.c_str());
+  }
+}
+
+NS_IMETHODIMP
+PeerConnectionImpl::SetRemoteDescription(int32_t action, const char* aSDP) {
+  PC_AUTO_ENTER_API_CALL(true);
+
+  if (!aSDP) {
+    CSFLogError(LOGTAG, "%s - aSDP is NULL", __FUNCTION__);
+    return NS_ERROR_FAILURE;
+  }
+
+  if (action == IPeerConnection::kActionOffer) {
+    if (!PeerConnectionCtx::GetInstance()->isReady()) {
+      // Uh oh. We're not ready yet. Enqueue this operation. (This must be a
+      // remote offer, or else we would not have gotten this far)
+      PeerConnectionCtx::GetInstance()->queueJSEPOperation(WrapRunnableNM(
+          DeferredSetRemote, mHandle, action, std::string(aSDP)));
+      STAMP_TIMECARD(mTimeCard, "Deferring SetRemote (not ready)");
+      return NS_OK;
+    }
+
+    nsresult nrv = ConfigureJsepSessionCodecs();
+    if (NS_FAILED(nrv)) {
+      CSFLogError(LOGTAG, "Failed to configure codecs");
+      return nrv;
+    }
+  }
+
+  STAMP_TIMECARD(mTimeCard, "Set Remote Description");
+
+  mozilla::dom::RTCSdpHistoryEntryInternal sdpEntry;
+  sdpEntry.mIsLocal = false;
+  sdpEntry.mTimestamp = mTimestampMaker.GetNow().ToDom();
+  sdpEntry.mSdp = NS_ConvertASCIItoUTF16(aSDP);
+  auto appendHistory = [&]() {
+    if (!mSdpHistory.AppendElement(sdpEntry, fallible)) {
+      mozalloc_handle_oom(0);
+    }
+  };
+
+  SyncToJsep();
+
+  mRemoteRequestedSDP = aSDP;
+  bool wasRestartingIce = mJsepSession->IsIceRestarting();
+  JsepSdpType sdpType;
+  switch (action) {
+    case IPeerConnection::kActionOffer:
+      sdpType = mozilla::kJsepSdpOffer;
+      break;
+    case IPeerConnection::kActionAnswer:
+      sdpType = mozilla::kJsepSdpAnswer;
+      break;
+    case IPeerConnection::kActionPRAnswer:
+      sdpType = mozilla::kJsepSdpPranswer;
+      break;
+    case IPeerConnection::kActionRollback:
+      sdpType = mozilla::kJsepSdpRollback;
+      break;
+    default:
+      MOZ_ASSERT(false);
+      return NS_ERROR_FAILURE;
+  }
+
+  MOZ_ASSERT(!mUncommittedJsepSession);
+  mUncommittedJsepSession.reset(mJsepSession->Clone());
+  JsepSession::Result result = mUncommittedJsepSession->SetRemoteDescription(
+      sdpType, mRemoteRequestedSDP);
+  JSErrorResult jrv;
+  if (result.mError.isSome()) {
+    std::string errorString = mUncommittedJsepSession->GetLastError();
+    mUncommittedJsepSession = nullptr;
+    sdpEntry.mErrors = GetLastSdpParsingErrors();
+    CSFLogError(LOGTAG, "%s: pc = %s, error = %s", __FUNCTION__,
+                mHandle.c_str(), errorString.c_str());
+    mPCObserver->OnSetDescriptionError(*buildJSErrorData(result, errorString),
+                                       jrv);
+  } else {
+    if (wasRestartingIce) {
+      RecordIceRestartStatistics(sdpType);
+    }
+
+    mPCObserver->OnSetDescriptionSuccess(jrv);
+  }
+
+  appendHistory();
+
+  if (jrv.Failed()) {
+    return jrv.StealNSResult();
+  }
+
+  return NS_OK;
+}
+
+already_AddRefed<dom::Promise> PeerConnectionImpl::GetStats(
+    MediaStreamTrack* aSelector) {
+  if (!mWindow) {
+    MOZ_CRASH("Cannot create a promise without a window!");
+  }
+
+  nsCOMPtr<nsIGlobalObject> global = do_QueryInterface(mWindow);
+  ErrorResult rv;
+  RefPtr<Promise> promise = Promise::Create(global, rv);
+  if (NS_WARN_IF(rv.Failed())) {
+    MOZ_CRASH("Failed to create a promise!");
+  }
+
+  if (!IsClosed()) {
+    GetStats(aSelector, false)
+        ->Then(
+            GetMainThreadSerialEventTarget(), __func__,
+            [promise, window = mWindow](
+                UniquePtr<dom::RTCStatsReportInternal>&& aReport) {
+              RefPtr<RTCStatsReport> report(new RTCStatsReport(window));
+              report->Incorporate(*aReport);
+              promise->MaybeResolve(std::move(report));
+            },
+            [promise, window = mWindow](nsresult aError) {
+              RefPtr<RTCStatsReport> report(new RTCStatsReport(window));
+              promise->MaybeResolve(std::move(report));
+            });
+  } else {
+    promise->MaybeReject(NS_ERROR_DOM_INVALID_STATE_ERR);
+  }
+
+  return promise.forget();
+}
+
+void PeerConnectionImpl::GetRemoteStreams(
+    nsTArray<RefPtr<DOMMediaStream>>& aStreamsOut) const {
+  aStreamsOut = mReceiveStreams.Clone();
+}
+
+NS_IMETHODIMP
+PeerConnectionImpl::AddIceCandidate(
+    const char* aCandidate, const char* aMid, const char* aUfrag,
+    const dom::Nullable<unsigned short>& aLevel) {
+  PC_AUTO_ENTER_API_CALL(true);
+
+  if (mForceIceTcp &&
+      std::string::npos != std::string(aCandidate).find(" UDP ")) {
+    CSFLogError(LOGTAG, "Blocking remote UDP candidate: %s", aCandidate);
+    return NS_OK;
+  }
+
+  STAMP_TIMECARD(mTimeCard, "Add Ice Candidate");
+
+  CSFLogDebug(LOGTAG, "AddIceCandidate: %s %s", aCandidate, aUfrag);
+
+  std::string transportId;
+  Maybe<unsigned short> level;
+  if (!aLevel.IsNull()) {
+    level = Some(aLevel.Value());
+  }
+  MOZ_DIAGNOSTIC_ASSERT(
+      !mUncommittedJsepSession,
+      "AddIceCandidate is chained, which means it should never "
+      "run while an sRD/sLD is in progress");
+  JsepSession::Result result = mJsepSession->AddRemoteIceCandidate(
+      aCandidate, aMid, level, aUfrag, &transportId);
+
+  if (!result.mError.isSome()) {
+    // We do not bother the MediaTransportHandler about this before
+    // offer/answer concludes.  Once offer/answer concludes, we will extract
+    // these candidates from the remote SDP.
+    if (mSignalingState == RTCSignalingState::Stable && !transportId.empty()) {
+      AddIceCandidate(aCandidate, transportId, aUfrag);
+      mRawTrickledCandidates.push_back(aCandidate);
+    }
+    // Spec says we queue a task for these updates
+    GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction(
+        __func__, [this, self = RefPtr<PeerConnectionImpl>(this)] {
+          if (IsClosed()) {
+            return;
+          }
+          mPendingRemoteDescription =
+              mJsepSession->GetRemoteDescription(kJsepDescriptionPending);
+          mCurrentRemoteDescription =
+              mJsepSession->GetRemoteDescription(kJsepDescriptionCurrent);
+          JSErrorResult rv;
+          mPCObserver->OnAddIceCandidateSuccess(rv);
+        }));
+  } else {
+    std::string errorString = mJsepSession->GetLastError();
+
+    CSFLogError(LOGTAG,
+                "Failed to incorporate remote candidate into SDP:"
+                " res = %u, candidate = %s, level = %i, error = %s",
+                static_cast<unsigned>(*result.mError), aCandidate,
+                level.valueOr(-1), errorString.c_str());
+
+    GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction(
+        __func__,
+        [this, self = RefPtr<PeerConnectionImpl>(this), errorString, result] {
+          if (IsClosed()) {
+            return;
+          }
+          JSErrorResult rv;
+          mPCObserver->OnAddIceCandidateError(
+              *buildJSErrorData(result, errorString), rv);
+        }));
+  }
+
+  return NS_OK;
+}
+
+NS_IMETHODIMP
+PeerConnectionImpl::CloseStreams() {
+  PC_AUTO_ENTER_API_CALL(false);
+
+  return NS_OK;
+}
+
+NS_IMETHODIMP
+PeerConnectionImpl::SetPeerIdentity(const nsAString& aPeerIdentity) {
+  PC_AUTO_ENTER_API_CALL(true);
+  MOZ_ASSERT(!aPeerIdentity.IsEmpty());
+
+  // once set, this can't be changed
+  if (mPeerIdentity) {
+    if (!mPeerIdentity->Equals(aPeerIdentity)) {
+      return NS_ERROR_FAILURE;
+    }
+  } else {
+    mPeerIdentity = new PeerIdentity(aPeerIdentity);
+    Document* doc = mWindow->GetExtantDoc();
+    if (!doc) {
+      CSFLogInfo(LOGTAG, "Can't update principal on streams; document gone");
+      return NS_ERROR_FAILURE;
+    }
+    for (const auto& transceiver : mTransceivers) {
+      transceiver->Sender()->GetPipeline()->UpdateSinkIdentity(
+          doc->NodePrincipal(), mPeerIdentity);
+    }
+  }
+  return NS_OK;
+}
+
+nsresult PeerConnectionImpl::OnAlpnNegotiated(bool aPrivacyRequested) {
+  PC_AUTO_ENTER_API_CALL(false);
+  MOZ_DIAGNOSTIC_ASSERT(!mRequestedPrivacy ||
+                        (*mRequestedPrivacy == PrincipalPrivacy::Private) ==
+                            aPrivacyRequested);
+
+  mRequestedPrivacy = Some(aPrivacyRequested ? PrincipalPrivacy::Private
+                                             : PrincipalPrivacy::NonPrivate);
+  // This updates the MediaPipelines with a private PrincipalHandle. Note that
+  // MediaPipelineReceive has its own AlpnNegotiated handler so it can get
+  // signaled off-main to drop data until it receives the new PrincipalHandle
+  // from us.
+  UpdateMediaPipelines();
+  return NS_OK;
+}
+
+void PeerConnectionImpl::OnDtlsStateChange(const std::string& aTransportId,
+                                           TransportLayer::State aState) {
+  auto it = mTransportIdToRTCDtlsTransport.find(aTransportId);
+  if (it != mTransportIdToRTCDtlsTransport.end()) {
+    it->second->UpdateState(aState);
+  }
+  UpdateConnectionState();
+}
+
+RTCPeerConnectionState PeerConnectionImpl::GetNewConnectionState() const {
+  // closed 	The RTCPeerConnection object's [[IsClosed]] slot is true.
+  if (IsClosed()) {
+    return RTCPeerConnectionState::Closed;
+  }
+
+  // Would use a bitset, but that requires lots of static_cast<size_t>
+  // Oh well.
+  std::set<RTCDtlsTransportState> statesFound;
+  for (const auto& [id, dtlsTransport] : mTransportIdToRTCDtlsTransport) {
+    Unused << id;
+    statesFound.insert(dtlsTransport->State());
+  }
+
+  // failed 	The previous state doesn't apply and any RTCIceTransports are
+  // in the "failed" state or any RTCDtlsTransports are in the "failed" state.
+  // NOTE: "any RTCIceTransports are in the failed state" is equivalent to
+  // mIceConnectionState == Failed
+  if (mIceConnectionState == RTCIceConnectionState::Failed ||
+      statesFound.count(RTCDtlsTransportState::Failed)) {
+    return RTCPeerConnectionState::Failed;
+  }
+
+  // disconnected 	None of the previous states apply and any
+  // RTCIceTransports are in the "disconnected" state.
+  // NOTE: "any RTCIceTransports are in the disconnected state" is equivalent to
+  // mIceConnectionState == Disconnected.
+  if (mIceConnectionState == RTCIceConnectionState::Disconnected) {
+    return RTCPeerConnectionState::Disconnected;
+  }
+
+  // new 	None of the previous states apply and all RTCIceTransports are
+  // in the "new" or "closed" state, and all RTCDtlsTransports are in the "new"
+  // or "closed" state, or there are no transports.
+  // NOTE: "all RTCIceTransports are in the new or closed state" is equivalent
+  // to mIceConnectionState == New.
+  if (mIceConnectionState == RTCIceConnectionState::New &&
+      !statesFound.count(RTCDtlsTransportState::Connecting) &&
+      !statesFound.count(RTCDtlsTransportState::Connected) &&
+      !statesFound.count(RTCDtlsTransportState::Failed)) {
+    return RTCPeerConnectionState::New;
+  }
+
+  // No transports
+  if (statesFound.empty()) {
+    return RTCPeerConnectionState::New;
+  }
+
+  // connecting 	None of the previous states apply and any
+  // RTCIceTransport is in the "new" or "checking" state or any
+  // RTCDtlsTransport is in the "new" or "connecting" state.
+  // NOTE: "None of the previous states apply and any RTCIceTransport is in the
+  // "new" or "checking" state" is equivalent to mIceConnectionState ==
+  // Checking.
+  if (mIceConnectionState == RTCIceConnectionState::Checking ||
+      statesFound.count(RTCDtlsTransportState::New) ||
+      statesFound.count(RTCDtlsTransportState::Connecting)) {
+    return RTCPeerConnectionState::Connecting;
+  }
+
+  // connected 	None of the previous states apply and all RTCIceTransports are
+  // in the "connected", "completed" or "closed" state, and all
+  // RTCDtlsTransports are in the "connected" or "closed" state.
+  // NOTE: "None of the previous states apply and all RTCIceTransports are in
+  // the "connected", "completed" or "closed" state" is equivalent to
+  // mIceConnectionState == Connected.
+  if (mIceConnectionState == RTCIceConnectionState::Connected &&
+      !statesFound.count(RTCDtlsTransportState::New) &&
+      !statesFound.count(RTCDtlsTransportState::Failed) &&
+      !statesFound.count(RTCDtlsTransportState::Connecting)) {
+    return RTCPeerConnectionState::Connected;
+  }
+
+  // !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
+  // THERE IS NO CATCH-ALL NONE-OF-THE-ABOVE IN THE SPEC! THIS IS REALLY BAD! !!
+  // !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
+
+  // Let's try to figure out how bad, precisely.
+  // Any one of these will cause us to bail above.
+  MOZ_ASSERT(mIceConnectionState != RTCIceConnectionState::Failed &&
+             mIceConnectionState != RTCIceConnectionState::Disconnected &&
+             mIceConnectionState != RTCIceConnectionState::Checking);
+  MOZ_ASSERT(!statesFound.count(RTCDtlsTransportState::New) &&
+             !statesFound.count(RTCDtlsTransportState::Connecting) &&
+             !statesFound.count(RTCDtlsTransportState::Failed));
+
+  // One of these must be set, or the empty() check would have failed above.
+  MOZ_ASSERT(statesFound.count(RTCDtlsTransportState::Connected) ||
+             statesFound.count(RTCDtlsTransportState::Closed));
+
+  // Here are our remaining possibilities:
+  // ICE connected, !statesFound.count(Connected), statesFound.count(Closed)
+  // ICE connected, statesFound.count(Connected), !statesFound.count(Closed)
+  // ICE connected, statesFound.count(Connected), statesFound.count(Closed)
+  //    All three of these would result in returning Connected above.
+
+  // ICE new, !statesFound.count(Connected), statesFound.count(Closed)
+  //    This results in returning New above. Whew.
+
+  // ICE new, statesFound.count(Connected), !statesFound.count(Closed)
+  // ICE new, statesFound.count(Connected), statesFound.count(Closed)
+  //    These would make it all the way here! Very weird state though, for all
+  //    ICE transports to be new/closed, but having a connected DTLS transport.
+  //    Handle this as a non-transition, just in case.
+  return mConnectionState;
+}
+
+void PeerConnectionImpl::UpdateConnectionState() {
+  auto newState = GetNewConnectionState();
+  if (newState != mConnectionState) {
+    CSFLogDebug(LOGTAG, "%s: %d -> %d (%p)", __FUNCTION__,
+                static_cast<int>(mConnectionState), static_cast<int>(newState),
+                this);
+    mConnectionState = newState;
+    if (mConnectionState != RTCPeerConnectionState::Closed) {
+      JSErrorResult jrv;
+      mPCObserver->OnStateChange(PCObserverStateType::ConnectionState, jrv);
+    }
+  }
+}
+
+void PeerConnectionImpl::OnMediaError(const std::string& aError) {
+  CSFLogError(LOGTAG, "Encountered media error! %s", aError.c_str());
+  // TODO: Let content know about this somehow.
+}
+
+void PeerConnectionImpl::DumpPacket_m(size_t level, dom::mozPacketDumpType type,
+                                      bool sending,
+                                      UniquePtr<uint8_t[]>& packet,
+                                      size_t size) {
+  if (IsClosed()) {
+    return;
+  }
+
+  // TODO: Is this efficient? Should we try grabbing our JS ctx from somewhere
+  // else?
+  AutoJSAPI jsapi;
+  if (!jsapi.Init(mWindow)) {
+    return;
+  }
+
+  JS::Rooted<JSObject*> jsobj(
+      jsapi.cx(),
+      JS::NewArrayBufferWithContents(jsapi.cx(), size, packet.release()));
+
+  RootedSpiderMonkeyInterface<ArrayBuffer> arrayBuffer(jsapi.cx());
+  if (!arrayBuffer.Init(jsobj)) {
+    return;
+  }
+
+  JSErrorResult jrv;
+  mPCObserver->OnPacket(level, type, sending, arrayBuffer, jrv);
+}
+
+bool PeerConnectionImpl::HostnameInPref(const char* aPref,
+                                        const nsCString& aHostName) {
+  auto HostInDomain = [](const nsCString& aHost, const nsCString& aPattern) {
+    int32_t patternOffset = 0;
+    int32_t hostOffset = 0;
+
+    // Act on '*.' wildcard in the left-most position in a domain pattern.
+    if (StringBeginsWith(aPattern, nsCString("*."))) {
+      patternOffset = 2;
+
+      // Ignore the lowest level sub-domain for the hostname.
+      hostOffset = aHost.FindChar('.') + 1;
+
+      if (hostOffset <= 1) {
+        // Reject a match between a wildcard and a TLD or '.foo' form.
+        return false;
+      }
+    }
+
+    nsDependentCString hostRoot(aHost, hostOffset);
+    return hostRoot.EqualsIgnoreCase(aPattern.BeginReading() + patternOffset);
+  };
+
+  nsCString domainList;
+  nsresult nr = Preferences::GetCString(aPref, domainList);
+
+  if (NS_FAILED(nr)) {
+    return false;
+  }
+
+  domainList.StripWhitespace();
+
+  if (domainList.IsEmpty() || aHostName.IsEmpty()) {
+    return false;
+  }
+
+  // Get UTF8 to ASCII domain name normalization service
+  nsresult rv;
+  nsCOMPtr<nsIIDNService> idnService =
+      do_GetService("@mozilla.org/network/idn-service;1", &rv);
+  if (NS_WARN_IF(NS_FAILED(rv))) {
+    return false;
+  }
+
+  // Test each domain name in the comma separated list
+  // after converting from UTF8 to ASCII. Each domain
+  // must match exactly or have a single leading '*.' wildcard.
+  for (const nsACString& each : domainList.Split(',')) {
+    nsCString domainPattern;
+    rv = idnService->ConvertUTF8toACE(each, domainPattern);
+    if (NS_SUCCEEDED(rv)) {
+      if (HostInDomain(aHostName, domainPattern)) {
+        return true;
+      }
+    } else {
+      NS_WARNING("Failed to convert UTF-8 host to ASCII");
+    }
+  }
+
+  return false;
+}
+
+nsresult PeerConnectionImpl::EnablePacketDump(unsigned long level,
+                                              dom::mozPacketDumpType type,
+                                              bool sending) {
+  return GetPacketDumper()->EnablePacketDump(level, type, sending);
+}
+
+nsresult PeerConnectionImpl::DisablePacketDump(unsigned long level,
+                                               dom::mozPacketDumpType type,
+                                               bool sending) {
+  return GetPacketDumper()->DisablePacketDump(level, type, sending);
+}
+
+void PeerConnectionImpl::StampTimecard(const char* aEvent) {
+  MOZ_ASSERT(NS_IsMainThread());
+  STAMP_TIMECARD(mTimeCard, aEvent);
+}
+
+void PeerConnectionImpl::SendWarningToConsole(const nsCString& aWarning) {
+  nsAutoString msg = NS_ConvertASCIItoUTF16(aWarning);
+  nsContentUtils::ReportToConsoleByWindowID(msg, nsIScriptError::warningFlag,
+                                            "WebRTC"_ns, mWindow->WindowID());
+}
+
+void PeerConnectionImpl::GetDefaultVideoCodecs(
+    std::vector<UniquePtr<JsepCodecDescription>>& aSupportedCodecs,
+    bool aUseRtx) {
+  // Supported video codecs.
+  // Note: order here implies priority for building offers!
+  aSupportedCodecs.emplace_back(
+      JsepVideoCodecDescription::CreateDefaultVP8(aUseRtx));
+  aSupportedCodecs.emplace_back(
+      JsepVideoCodecDescription::CreateDefaultVP9(aUseRtx));
+  aSupportedCodecs.emplace_back(
+      JsepVideoCodecDescription::CreateDefaultH264_1(aUseRtx));
+  aSupportedCodecs.emplace_back(
+      JsepVideoCodecDescription::CreateDefaultH264_0(aUseRtx));
+  aSupportedCodecs.emplace_back(
+      JsepVideoCodecDescription::CreateDefaultUlpFec());
+  aSupportedCodecs.emplace_back(
+      JsepApplicationCodecDescription::CreateDefault());
+  aSupportedCodecs.emplace_back(JsepVideoCodecDescription::CreateDefaultRed());
+}
+
+void PeerConnectionImpl::GetDefaultAudioCodecs(
+    std::vector<UniquePtr<JsepCodecDescription>>& aSupportedCodecs) {
+  aSupportedCodecs.emplace_back(JsepAudioCodecDescription::CreateDefaultOpus());
+  aSupportedCodecs.emplace_back(JsepAudioCodecDescription::CreateDefaultG722());
+  aSupportedCodecs.emplace_back(JsepAudioCodecDescription::CreateDefaultPCMU());
+  aSupportedCodecs.emplace_back(JsepAudioCodecDescription::CreateDefaultPCMA());
+  aSupportedCodecs.emplace_back(
+      JsepAudioCodecDescription::CreateDefaultTelephoneEvent());
+}
+
+void PeerConnectionImpl::GetDefaultRtpExtensions(
+    std::vector<RtpExtensionHeader>& aRtpExtensions) {
+  RtpExtensionHeader audioLevel = {JsepMediaType::kAudio,
+                                   SdpDirectionAttribute::Direction::kSendrecv,
+                                   webrtc::RtpExtension::kAudioLevelUri};
+  aRtpExtensions.push_back(audioLevel);
+
+  RtpExtensionHeader csrcAudioLevels = {
+      JsepMediaType::kAudio, SdpDirectionAttribute::Direction::kRecvonly,
+      webrtc::RtpExtension::kCsrcAudioLevelsUri};
+  aRtpExtensions.push_back(csrcAudioLevels);
+
+  RtpExtensionHeader mid = {JsepMediaType::kAudioVideo,
+                            SdpDirectionAttribute::Direction::kSendrecv,
+                            webrtc::RtpExtension::kMidUri};
+  aRtpExtensions.push_back(mid);
+
+  RtpExtensionHeader absSendTime = {JsepMediaType::kVideo,
+                                    SdpDirectionAttribute::Direction::kSendrecv,
+                                    webrtc::RtpExtension::kAbsSendTimeUri};
+  aRtpExtensions.push_back(absSendTime);
+
+  RtpExtensionHeader timestampOffset = {
+      JsepMediaType::kVideo, SdpDirectionAttribute::Direction::kSendrecv,
+      webrtc::RtpExtension::kTimestampOffsetUri};
+  aRtpExtensions.push_back(timestampOffset);
+
+  RtpExtensionHeader playoutDelay = {
+      JsepMediaType::kVideo, SdpDirectionAttribute::Direction::kRecvonly,
+      webrtc::RtpExtension::kPlayoutDelayUri};
+  aRtpExtensions.push_back(playoutDelay);
+
+  RtpExtensionHeader transportSequenceNumber = {
+      JsepMediaType::kVideo, SdpDirectionAttribute::Direction::kSendrecv,
+      webrtc::RtpExtension::kTransportSequenceNumberUri};
+  aRtpExtensions.push_back(transportSequenceNumber);
+}
+
+void PeerConnectionImpl::GetCapabilities(
+    const nsAString& aKind, dom::Nullable<dom::RTCRtpCapabilities>& aResult,
+    sdp::Direction aDirection) {
+  std::vector<UniquePtr<JsepCodecDescription>> codecs;
+  std::vector<RtpExtensionHeader> headers;
+  auto mediaType = JsepMediaType::kNone;
+
+  if (aKind.EqualsASCII("video")) {
+    GetDefaultVideoCodecs(codecs, true);
+    mediaType = JsepMediaType::kVideo;
+  } else if (aKind.EqualsASCII("audio")) {
+    GetDefaultAudioCodecs(codecs);
+    mediaType = JsepMediaType::kAudio;
+  } else {
+    return;
+  }
+
+  GetDefaultRtpExtensions(headers);
+
+  // Use the codecs for kind to fill out the RTCRtpCodecCapability
+  for (const auto& codec : codecs) {
+    // To avoid misleading information on codec capabilities skip those
+    // not signaled for audio/video (webrtc-datachannel)
+    // and any disabled by default (ulpfec and red).
+    if (codec->mName == "webrtc-datachannel" || codec->mName == "ulpfec" ||
+        codec->mName == "red") {
+      continue;
+    }
+
+    dom::RTCRtpCodecCapability capability;
+    capability.mMimeType = aKind + NS_ConvertASCIItoUTF16("/" + codec->mName);
+    capability.mClockRate = codec->mClock;
+
+    if (codec->mChannels) {
+      capability.mChannels.Construct(codec->mChannels);
+    }
+
+    UniquePtr<SdpFmtpAttributeList::Parameters> params;
+    codec->ApplyConfigToFmtp(params);
+
+    if (params != nullptr) {
+      std::ostringstream paramsString;
+      params->Serialize(paramsString);
+      nsTString<char16_t> fmtp;
+      fmtp.AssignASCII(paramsString.str());
+      capability.mSdpFmtpLine.Construct(fmtp);
+    }
+
+    if (!aResult.SetValue().mCodecs.AppendElement(capability, fallible)) {
+      mozalloc_handle_oom(0);
+    }
+  }
+
+  // We need to manually add rtx for video.
+  if (mediaType == JsepMediaType::kVideo) {
+    dom::RTCRtpCodecCapability capability;
+    capability.mMimeType = aKind + NS_ConvertASCIItoUTF16("/rtx");
+    capability.mClockRate = 90000;
+    if (!aResult.SetValue().mCodecs.AppendElement(capability, fallible)) {
+      mozalloc_handle_oom(0);
+    }
+  }
+
+  // Add headers that match the direction and media type requested.
+  for (const auto& header : headers) {
+    if ((header.direction & aDirection) && (header.mMediaType & mediaType)) {
+      dom::RTCRtpHeaderExtensionCapability rtpHeader;
+      rtpHeader.mUri.AssignASCII(header.extensionname);
+      if (!aResult.SetValue().mHeaderExtensions.AppendElement(rtpHeader,
+                                                              fallible)) {
+        mozalloc_handle_oom(0);
+      }
+    }
+  }
+}
+
+void PeerConnectionImpl::SetupPreferredCodecs(
+    std::vector<UniquePtr<JsepCodecDescription>>& aPreferredCodecs) {
+  bool useRtx =
+      Preferences::GetBool("media.peerconnection.video.use_rtx", false);
+
+  GetDefaultVideoCodecs(aPreferredCodecs, useRtx);
+  GetDefaultAudioCodecs(aPreferredCodecs);
+
+  // With red update the redundant encodings list
+  for (auto& videoCodec : aPreferredCodecs) {
+    if (videoCodec->mName == "red") {
+      JsepVideoCodecDescription& red =
+          static_cast<JsepVideoCodecDescription&>(*videoCodec);
+      red.UpdateRedundantEncodings(aPreferredCodecs);
+    }
+  }
+}
+
+void PeerConnectionImpl::SetupPreferredRtpExtensions(
+    std::vector<RtpExtensionHeader>& aPreferredheaders) {
+  GetDefaultRtpExtensions(aPreferredheaders);
+
+  if (!Preferences::GetBool("media.navigator.video.use_transport_cc", false)) {
+    aPreferredheaders.erase(
+        std::remove_if(
+            aPreferredheaders.begin(), aPreferredheaders.end(),
+            [&](const RtpExtensionHeader& header) {
+              return header.extensionname ==
+                     webrtc::RtpExtension::kTransportSequenceNumberUri;
+            }),
+        aPreferredheaders.end());
+  }
+}
+
+nsresult PeerConnectionImpl::CalculateFingerprint(
+    const std::string& algorithm, std::vector<uint8_t>* fingerprint) const {
+  DtlsDigest digest(algorithm);
+
+  MOZ_ASSERT(fingerprint);
+  const UniqueCERTCertificate& cert = mCertificate->Certificate();
+  nsresult rv = DtlsIdentity::ComputeFingerprint(cert, &digest);
+  if (NS_FAILED(rv)) {
+    CSFLogError(LOGTAG, "Unable to calculate certificate fingerprint, rv=%u",
+                static_cast<unsigned>(rv));
+    return rv;
+  }
+  *fingerprint = digest.value_;
+  return NS_OK;
+}
+
+NS_IMETHODIMP
+PeerConnectionImpl::GetFingerprint(char** fingerprint) {
+  MOZ_ASSERT(fingerprint);
+  MOZ_ASSERT(mCertificate);
+  std::vector<uint8_t> fp;
+  nsresult rv = CalculateFingerprint(DtlsIdentity::DEFAULT_HASH_ALGORITHM, &fp);
+  NS_ENSURE_SUCCESS(rv, rv);
+  std::ostringstream os;
+  os << DtlsIdentity::DEFAULT_HASH_ALGORITHM << ' '
+     << SdpFingerprintAttributeList::FormatFingerprint(fp);
+  std::string fpStr = os.str();
+
+  char* tmp = new char[fpStr.size() + 1];
+  std::copy(fpStr.begin(), fpStr.end(), tmp);
+  tmp[fpStr.size()] = '\0';
+
+  *fingerprint = tmp;
+  return NS_OK;
+}
+
+void PeerConnectionImpl::GetCurrentLocalDescription(nsAString& aSDP) const {
+  aSDP = NS_ConvertASCIItoUTF16(mCurrentLocalDescription.c_str());
+}
+
+void PeerConnectionImpl::GetPendingLocalDescription(nsAString& aSDP) const {
+  aSDP = NS_ConvertASCIItoUTF16(mPendingLocalDescription.c_str());
+}
+
+void PeerConnectionImpl::GetCurrentRemoteDescription(nsAString& aSDP) const {
+  aSDP = NS_ConvertASCIItoUTF16(mCurrentRemoteDescription.c_str());
+}
+
+void PeerConnectionImpl::GetPendingRemoteDescription(nsAString& aSDP) const {
+  aSDP = NS_ConvertASCIItoUTF16(mPendingRemoteDescription.c_str());
+}
+
+dom::Nullable<bool> PeerConnectionImpl::GetCurrentOfferer() const {
+  dom::Nullable<bool> result;
+  if (mCurrentOfferer.isSome()) {
+    result.SetValue(*mCurrentOfferer);
+  }
+  return result;
+}
+
+dom::Nullable<bool> PeerConnectionImpl::GetPendingOfferer() const {
+  dom::Nullable<bool> result;
+  if (mPendingOfferer.isSome()) {
+    result.SetValue(*mPendingOfferer);
+  }
+  return result;
+}
+
+NS_IMETHODIMP
+PeerConnectionImpl::SignalingState(RTCSignalingState* aState) {
+  PC_AUTO_ENTER_API_CALL_NO_CHECK();
+  MOZ_ASSERT(aState);
+
+  *aState = mSignalingState;
+  return NS_OK;
+}
+
+NS_IMETHODIMP
+PeerConnectionImpl::IceConnectionState(RTCIceConnectionState* aState) {
+  PC_AUTO_ENTER_API_CALL_NO_CHECK();
+  MOZ_ASSERT(aState);
+
+  *aState = mIceConnectionState;
+  return NS_OK;
+}
+
+NS_IMETHODIMP
+PeerConnectionImpl::IceGatheringState(RTCIceGatheringState* aState) {
+  PC_AUTO_ENTER_API_CALL_NO_CHECK();
+  MOZ_ASSERT(aState);
+
+  *aState = mIceGatheringState;
+  return NS_OK;
+}
+
+NS_IMETHODIMP
+PeerConnectionImpl::ConnectionState(RTCPeerConnectionState* aState) {
+  PC_AUTO_ENTER_API_CALL_NO_CHECK();
+  MOZ_ASSERT(aState);
+
+  *aState = mConnectionState;
+  return NS_OK;
+}
+
+nsresult PeerConnectionImpl::CheckApiState(bool assert_ice_ready) const {
+  PC_AUTO_ENTER_API_CALL_NO_CHECK();
+  MOZ_ASSERT(mTrickle || !assert_ice_ready ||
+             (mIceGatheringState == RTCIceGatheringState::Complete));
+
+  if (IsClosed()) {
+    CSFLogError(LOGTAG, "%s: called API while closed", __FUNCTION__);
+    return NS_ERROR_FAILURE;
+  }
+  return NS_OK;
+}
+
+void PeerConnectionImpl::StoreFinalStats(
+    UniquePtr<RTCStatsReportInternal>&& report) {
+  using namespace Telemetry;
+
+  report->mClosed = true;
+
+  for (const auto& inboundRtpStats : report->mInboundRtpStreamStats) {
+    bool isVideo = (inboundRtpStats.mId.Value().Find(u"video") != -1);
+    if (!isVideo) {
+      continue;
+    }
+    if (inboundRtpStats.mDiscardedPackets.WasPassed() &&
+        report->mCallDurationMs.WasPassed()) {
+      double mins = report->mCallDurationMs.Value() / (1000 * 60);
+      if (mins > 0) {
+        Accumulate(
+            WEBRTC_VIDEO_DECODER_DISCARDED_PACKETS_PER_CALL_PPM,
+            uint32_t(double(inboundRtpStats.mDiscardedPackets.Value()) / mins));
+      }
+    }
+  }
+
+  // Finally, store the stats
+  mFinalStats = std::move(report);
+}
+
+NS_IMETHODIMP
+PeerConnectionImpl::Close() {
+  CSFLogDebug(LOGTAG, "%s: for %s", __FUNCTION__, mHandle.c_str());
+  PC_AUTO_ENTER_API_CALL_NO_CHECK();
+
+  if (IsClosed()) {
+    return NS_OK;
+  }
+
+  STAMP_TIMECARD(mTimeCard, "Close");
+
+  // When ICE completes, we record some telemetry. We do this at the end of the
+  // call because we want to make sure we've waited for all trickle ICE
+  // candidates to come in; this can happen well after we've transitioned to
+  // connected. As a bonus, this allows us to detect race conditions where a
+  // stats dispatch happens right as the PC closes.
+  RecordEndOfCallTelemetry();
+
+  CSFLogInfo(LOGTAG,
+             "%s: Closing PeerConnectionImpl %s; "
+             "ending call",
+             __FUNCTION__, mHandle.c_str());
+  if (mJsepSession) {
+    mJsepSession->Close();
+  }
+  if (mDataConnection) {
+    CSFLogInfo(LOGTAG, "%s: Destroying DataChannelConnection %p for %s",
+               __FUNCTION__, (void*)mDataConnection.get(), mHandle.c_str());
+    mDataConnection->Destroy();
+    mDataConnection =
+        nullptr;  // it may not go away until the runnables are dead
+  }
+
+  if (mStunAddrsRequest) {
+    for (const auto& hostname : mRegisteredMDNSHostnames) {
+      mStunAddrsRequest->SendUnregisterMDNSHostname(
+          nsCString(hostname.c_str()));
+    }
+    mRegisteredMDNSHostnames.clear();
+    mStunAddrsRequest->Cancel();
+    mStunAddrsRequest = nullptr;
+  }
+
+  for (auto& transceiver : mTransceivers) {
+    transceiver->Close();
+  }
+
+  mTransportIdToRTCDtlsTransport.clear();
+
+  mQueuedIceCtxOperations.clear();
+
+  mOperations.Clear();
+
+  // Uncount this connection as active on the inner window upon close.
+  if (mWindow && mActiveOnWindow) {
+    mWindow->RemovePeerConnection();
+    mActiveOnWindow = false;
+  }
+
+  mSignalingState = RTCSignalingState::Closed;
+  mConnectionState = RTCPeerConnectionState::Closed;
+
+  if (!mTransportHandler) {
+    // We were never initialized, apparently.
+    return NS_OK;
+  }
+
+  // Clear any resources held by libwebrtc through our Call instance.
+  RefPtr<GenericPromise> callDestroyPromise;
+  if (mCall) {
+    // Make sure the compiler does not get confused and try to acquire a
+    // reference to this thread _after_ we null out mCall.
+    auto callThread = mCall->mCallThread;
+    callDestroyPromise =
+        InvokeAsync(callThread, __func__, [call = std::move(mCall)]() {
+          call->Destroy();
+          return GenericPromise::CreateAndResolve(
+              true, "PCImpl->WebRtcCallWrapper::Destroy");
+        });
+  } else {
+    callDestroyPromise = GenericPromise::CreateAndResolve(true, __func__);
+  }
+
+  mFinalStatsQuery =
+      GetStats(nullptr, true)
+          ->Then(
+              GetMainThreadSerialEventTarget(), __func__,
+              [this, self = RefPtr<PeerConnectionImpl>(this)](
+                  UniquePtr<dom::RTCStatsReportInternal>&& aReport) mutable {
+                StoreFinalStats(std::move(aReport));
+                return GenericNonExclusivePromise::CreateAndResolve(true,
+                                                                    __func__);
+              },
+              [](nsresult aError) {
+                return GenericNonExclusivePromise::CreateAndResolve(true,
+                                                                    __func__);
+              });
+
+  // 1. Allow final stats query to complete.
+  // 2. Tear down call, if necessary. We do this before we shut down the
+  //    transport handler, so RTCP BYE can be sent.
+  // 3. Unhook from the signal handler (sigslot) for transport stuff. This must
+  //    be done before we tear down the transport handler.
+  // 4. Tear down the transport handler, and deregister from PeerConnectionCtx.
+  //    When we deregister from PeerConnectionCtx, our final stats (if any)
+  //    will be stored.
+  MOZ_RELEASE_ASSERT(mSTSThread);
+  mFinalStatsQuery
+      ->Then(GetMainThreadSerialEventTarget(), __func__,
+             [callDestroyPromise]() mutable { return callDestroyPromise; })
+      ->Then(
+          mSTSThread, __func__,
+          [signalHandler = std::move(mSignalHandler)]() mutable {
+            CSFLogDebug(
+                LOGTAG,
+                "Destroying PeerConnectionImpl::SignalHandler on STS thread");
+            return GenericPromise::CreateAndResolve(
+                true, "PeerConnectionImpl::~SignalHandler");
+          })
+      ->Then(
+          GetMainThreadSerialEventTarget(), __func__,
+          [this, self = RefPtr<PeerConnectionImpl>(this)]() mutable {
+            CSFLogDebug(LOGTAG, "PCImpl->mTransportHandler::RemoveTransports");
+            mTransportHandler->RemoveTransportsExcept(std::set<std::string>());
+            if (mPrivateWindow) {
+              mTransportHandler->ExitPrivateMode();
+            }
+            mTransportHandler = nullptr;
+            if (PeerConnectionCtx::isActive()) {
+              // If we're shutting down xpcom, this Instance will be unset
+              // before calling Close() on all remaining PCs, to avoid
+              // reentrancy.
+              PeerConnectionCtx::GetInstance()->RemovePeerConnection(mHandle);
+            }
+          });
+
+  return NS_OK;
+}
+
+void PeerConnectionImpl::BreakCycles() {
+  for (auto& transceiver : mTransceivers) {
+    transceiver->BreakCycles();
+  }
+  mTransceivers.Clear();
+}
+
+bool PeerConnectionImpl::HasPendingSetParameters() const {
+  for (const auto& transceiver : mTransceivers) {
+    if (transceiver->Sender()->HasPendingSetParameters()) {
+      return true;
+    }
+  }
+  return false;
+}
+
+void PeerConnectionImpl::InvalidateLastReturnedParameters() {
+  for (const auto& transceiver : mTransceivers) {
+    transceiver->Sender()->InvalidateLastReturnedParameters();
+  }
+}
+
+nsresult PeerConnectionImpl::SetConfiguration(
+    const RTCConfiguration& aConfiguration) {
+  nsresult rv = mTransportHandler->SetIceConfig(
+      aConfiguration.mIceServers, aConfiguration.mIceTransportPolicy);
+  if (NS_WARN_IF(NS_FAILED(rv))) {
+    return rv;
+  }
+
+  JsepBundlePolicy bundlePolicy;
+  switch (aConfiguration.mBundlePolicy) {
+    case dom::RTCBundlePolicy::Balanced:
+      bundlePolicy = kBundleBalanced;
+      break;
+    case dom::RTCBundlePolicy::Max_compat:
+      bundlePolicy = kBundleMaxCompat;
+      break;
+    case dom::RTCBundlePolicy::Max_bundle:
+      bundlePolicy = kBundleMaxBundle;
+      break;
+    default:
+      MOZ_CRASH();
+  }
+
+  // Ignore errors, since those ought to be handled earlier.
+  Unused << mJsepSession->SetBundlePolicy(bundlePolicy);
+
+  if (!aConfiguration.mPeerIdentity.IsEmpty()) {
+    mPeerIdentity = new PeerIdentity(aConfiguration.mPeerIdentity);
+    mRequestedPrivacy = Some(PrincipalPrivacy::Private);
+  }
+
+  auto proxyConfig = GetProxyConfig();
+  if (proxyConfig) {
+    // Note that this could check if PrivacyRequested() is set on the PC and
+    // remove "webrtc" from the ALPN list.  But that would only work if the PC
+    // was constructed with a peerIdentity constraint, not when isolated
+    // streams are added.  If we ever need to signal to the proxy that the
+    // media is isolated, then we would need to restructure this code.
+    mTransportHandler->SetProxyConfig(std::move(*proxyConfig));
+  }
+
+  // Store the configuration for about:webrtc
+  StoreConfigurationForAboutWebrtc(aConfiguration);
+
+  return NS_OK;
+}
+
+RTCSctpTransport* PeerConnectionImpl::GetSctp() const {
+  return mSctpTransport.get();
+}
+
+void PeerConnectionImpl::RestartIce() {
+  RestartIceNoRenegotiationNeeded();
+  // Update the negotiation-needed flag for connection.
+  UpdateNegotiationNeeded();
+}
+
+// webrtc-pc does not specify any situations where this is done, but the JSEP
+// spec does, in some situations due to setConfiguration.
+void PeerConnectionImpl::RestartIceNoRenegotiationNeeded() {
+  // Empty connection.[[LocalIceCredentialsToReplace]], and populate it with
+  // all ICE credentials (ice-ufrag and ice-pwd as defined in section 15.4 of
+  // [RFC5245]) found in connection.[[CurrentLocalDescription]], as well as all
+  // ICE credentials found in connection.[[PendingLocalDescription]].
+  mLocalIceCredentialsToReplace = mJsepSession->GetLocalIceCredentials();
+}
+
+bool PeerConnectionImpl::PluginCrash(uint32_t aPluginID,
+                                     const nsAString& aPluginName) {
+  // fire an event to the DOM window if this is "ours"
+  if (!AnyCodecHasPluginID(aPluginID)) {
+    return false;
+  }
+
+  CSFLogError(LOGTAG, "%s: Our plugin %llu crashed", __FUNCTION__,
+              static_cast<unsigned long long>(aPluginID));
+
+  RefPtr<Document> doc = mWindow->GetExtantDoc();
+  if (!doc) {
+    NS_WARNING("Couldn't get document for PluginCrashed event!");
+    return true;
+  }
+
+  PluginCrashedEventInit init;
+  init.mPluginID = aPluginID;
+  init.mPluginName = aPluginName;
+  init.mSubmittedCrashReport = false;
+  init.mGmpPlugin = true;
+  init.mBubbles = true;
+  init.mCancelable = true;
+
+  RefPtr<PluginCrashedEvent> event =
+      PluginCrashedEvent::Constructor(doc, u"PluginCrashed"_ns, init);
+
+  event->SetTrusted(true);
+  event->WidgetEventPtr()->mFlags.mOnlyChromeDispatch = true;
+
+  nsCOMPtr<nsPIDOMWindowInner> window = mWindow;
+  EventDispatcher::DispatchDOMEvent(window, nullptr, event, nullptr, nullptr);
+
+  return true;
+}
+
+void PeerConnectionImpl::RecordEndOfCallTelemetry() {
+  if (!mCallTelemStarted) {
+    return;
+  }
+  MOZ_RELEASE_ASSERT(!mCallTelemEnded, "Don't end telemetry twice");
+  MOZ_RELEASE_ASSERT(mJsepSession,
+                     "Call telemetry only starts after jsep session start");
+  MOZ_RELEASE_ASSERT(mJsepSession->GetNegotiations() > 0,
+                     "Call telemetry only starts after first connection");
+
+  // Bitmask used for WEBRTC/LOOP_CALL_TYPE telemetry reporting
+  static const uint32_t kAudioTypeMask = 1;
+  static const uint32_t kVideoTypeMask = 2;
+  static const uint32_t kDataChannelTypeMask = 4;
+
+  // Report end-of-call Telemetry
+  Telemetry::Accumulate(Telemetry::WEBRTC_RENEGOTIATIONS,
+                        mJsepSession->GetNegotiations() - 1);
+  Telemetry::Accumulate(Telemetry::WEBRTC_MAX_VIDEO_SEND_TRACK,
+                        mMaxSending[SdpMediaSection::MediaType::kVideo]);
+  Telemetry::Accumulate(Telemetry::WEBRTC_MAX_VIDEO_RECEIVE_TRACK,
+                        mMaxReceiving[SdpMediaSection::MediaType::kVideo]);
+  Telemetry::Accumulate(Telemetry::WEBRTC_MAX_AUDIO_SEND_TRACK,
+                        mMaxSending[SdpMediaSection::MediaType::kAudio]);
+  Telemetry::Accumulate(Telemetry::WEBRTC_MAX_AUDIO_RECEIVE_TRACK,
+                        mMaxReceiving[SdpMediaSection::MediaType::kAudio]);
+  // DataChannels appear in both Sending and Receiving
+  Telemetry::Accumulate(Telemetry::WEBRTC_DATACHANNEL_NEGOTIATED,
+                        mMaxSending[SdpMediaSection::MediaType::kApplication]);
+  // Enumerated/bitmask: 1 = Audio, 2 = Video, 4 = DataChannel
+  // A/V = 3, A/V/D = 7, etc
+  uint32_t type = 0;
+  if (mMaxSending[SdpMediaSection::MediaType::kAudio] ||
+      mMaxReceiving[SdpMediaSection::MediaType::kAudio]) {
+    type = kAudioTypeMask;
+  }
+  if (mMaxSending[SdpMediaSection::MediaType::kVideo] ||
+      mMaxReceiving[SdpMediaSection::MediaType::kVideo]) {
+    type |= kVideoTypeMask;
+  }
+  if (mMaxSending[SdpMediaSection::MediaType::kApplication]) {
+    type |= kDataChannelTypeMask;
+  }
+  Telemetry::Accumulate(Telemetry::WEBRTC_CALL_TYPE, type);
+
+  MOZ_RELEASE_ASSERT(mWindow);
+  auto found = sCallDurationTimers.find(mWindow->WindowID());
+  if (found != sCallDurationTimers.end()) {
+    found->second.UnregisterConnection((type & kAudioTypeMask) ||
+                                       (type & kVideoTypeMask));
+    if (found->second.IsStopped()) {
+      sCallDurationTimers.erase(found);
+    }
+  }
+  mCallTelemEnded = true;
+}
+
+DOMMediaStream* PeerConnectionImpl::GetReceiveStream(
+    const std::string& aId) const {
+  nsString wanted = NS_ConvertASCIItoUTF16(aId.c_str());
+  for (auto& stream : mReceiveStreams) {
+    nsString id;
+    stream->GetId(id);
+    if (id == wanted) {
+      return stream;
+    }
+  }
+  return nullptr;
+}
+
+DOMMediaStream* PeerConnectionImpl::CreateReceiveStream(
+    const std::string& aId) {
+  mReceiveStreams.AppendElement(new DOMMediaStream(mWindow));
+  mReceiveStreams.LastElement()->AssignId(NS_ConvertASCIItoUTF16(aId.c_str()));
+  return mReceiveStreams.LastElement();
+}
+
+already_AddRefed<dom::Promise> PeerConnectionImpl::OnSetDescriptionSuccess(
+    dom::RTCSdpType aSdpType, bool aRemote, ErrorResult& aError) {
+  CSFLogDebug(LOGTAG, __FUNCTION__);
+
+  RefPtr<dom::Promise> p = MakePromise(aError);
+  if (aError.Failed()) {
+    return nullptr;
+  }
+
+  DoSetDescriptionSuccessPostProcessing(aSdpType, aRemote, p);
+
+  return p.forget();
+}
+
+void PeerConnectionImpl::DoSetDescriptionSuccessPostProcessing(
+    dom::RTCSdpType aSdpType, bool aRemote, const RefPtr<dom::Promise>& aP) {
+  // Spec says we queue a task for all the stuff that ends up back in JS
+  GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction(
+      __func__,
+      [this, self = RefPtr<PeerConnectionImpl>(this), aSdpType, aRemote, aP] {
+        if (IsClosed()) {
+          // Yes, we do not settle the promise here. Yes, this is what the spec
+          // wants.
+          return;
+        }
+
+        MOZ_ASSERT(mUncommittedJsepSession);
+
+        // sRD/sLD needs to be redone in certain circumstances
+        bool needsRedo = HasPendingSetParameters();
+        if (!needsRedo && aRemote && (aSdpType == dom::RTCSdpType::Offer)) {
+          for (auto& transceiver : mTransceivers) {
+            if (!mUncommittedJsepSession->GetTransceiver(
+                    transceiver->GetJsepTransceiverId())) {
+              needsRedo = true;
+              break;
+            }
+          }
+        }
+
+        if (needsRedo) {
+          // Spec says to abort, and re-do the sRD!
+          // This happens either when there is a SetParameters call in
+          // flight (that will race against the [[SendEncodings]]
+          // modification caused by sRD(offer)), or when addTrack has been
+          // called while sRD(offer) was in progress.
+          mUncommittedJsepSession.reset(mJsepSession->Clone());
+          JsepSession::Result result;
+          if (aRemote) {
+            mUncommittedJsepSession->SetRemoteDescription(
+                ToJsepSdpType(aSdpType), mRemoteRequestedSDP);
+          } else {
+            mUncommittedJsepSession->SetLocalDescription(
+                ToJsepSdpType(aSdpType), mLocalRequestedSDP);
+          }
+          if (result.mError.isSome()) {
+            // wat
+            nsCString error(
+                "When redoing sRD/sLD because it raced against "
+                "addTrack or setParameters, we encountered a failure that "
+                "did not happen "
+                "the first time. This should never happen. The error was: ");
+            error += mUncommittedJsepSession->GetLastError().c_str();
+            aP->MaybeRejectWithOperationError(error);
+            MOZ_ASSERT(false);
+          } else {
+            DoSetDescriptionSuccessPostProcessing(aSdpType, aRemote, aP);
+          }
+          return;
+        }
+
+        for (auto& transceiver : mTransceivers) {
+          if (!mUncommittedJsepSession->GetTransceiver(
+                  transceiver->GetJsepTransceiverId())) {
+            // sLD, or sRD(answer), just make sure the new transceiver is
+            // added, no need to re-do anything.
+            mUncommittedJsepSession->AddTransceiver(
+                transceiver->GetJsepTransceiver());
+          }
+        }
+
+        mJsepSession = std::move(mUncommittedJsepSession);
+
+        auto newSignalingState = GetSignalingState();
+        SyncFromJsep();
+        if (aRemote || aSdpType == dom::RTCSdpType::Pranswer ||
+            aSdpType == dom::RTCSdpType::Answer) {
+          InvalidateLastReturnedParameters();
+        }
+
+        // Section 4.4.1.5 Set the RTCSessionDescription:
+        if (aSdpType == dom::RTCSdpType::Rollback) {
+          // - step 4.5.10, type is rollback
+          RollbackRTCDtlsTransports();
+        } else if (!(aRemote && aSdpType == dom::RTCSdpType::Offer)) {
+          // - step 4.5.9 type is not rollback
+          // - step 4.5.9.1 when remote is false
+          // - step 4.5.9.2.13 when remote is true, type answer or pranswer
+          // More simply: not rollback, and not for remote offers.
+          bool markAsStable = aSdpType == dom::RTCSdpType::Offer &&
+                              mSignalingState == RTCSignalingState::Stable;
+          UpdateRTCDtlsTransports(markAsStable);
+        }
+
+        // Did we just apply a local description?
+        if (!aRemote) {
+          // We'd like to handle this in PeerConnectionImpl::UpdateNetworkState.
+          // Unfortunately, if the WiFi switch happens quickly, we never see
+          // that state change.  We need to detect the ice restart here and
+          // reset the PeerConnectionImpl's stun addresses so they are
+          // regathered when PeerConnectionImpl::GatherIfReady is called.
+          if (mJsepSession->IsIceRestarting()) {
+            ResetStunAddrsForIceRestart();
+          }
+          EnsureTransports(*mJsepSession);
+        }
+
+        if (mJsepSession->GetState() == kJsepStateStable) {
+          if (aSdpType != dom::RTCSdpType::Rollback) {
+            // We need this initted for UpdateTransports
+            InitializeDataChannel();
+          }
+
+          // If we're rolling back a local offer, we might need to remove some
+          // transports, and stomp some MediaPipeline setup, but nothing further
+          // needs to be done.
+          UpdateTransports(*mJsepSession, mForceIceTcp);
+          if (NS_FAILED(UpdateMediaPipelines())) {
+            CSFLogError(LOGTAG, "Error Updating MediaPipelines");
+            NS_ASSERTION(
+                false,
+                "Error Updating MediaPipelines in OnSetDescriptionSuccess()");
+            aP->MaybeRejectWithOperationError("Error Updating MediaPipelines");
+          }
+
+          if (aSdpType != dom::RTCSdpType::Rollback) {
+            StartIceChecks(*mJsepSession);
+          }
+
+          // Telemetry: record info on the current state of
+          // streams/renegotiations/etc Note: this code gets run on rollbacks as
+          // well!
+
+          // Update the max channels used with each direction for each type
+          uint16_t receiving[SdpMediaSection::kMediaTypes];
+          uint16_t sending[SdpMediaSection::kMediaTypes];
+          mJsepSession->CountTracksAndDatachannels(receiving, sending);
+          for (size_t i = 0; i < SdpMediaSection::kMediaTypes; i++) {
+            if (mMaxReceiving[i] < receiving[i]) {
+              mMaxReceiving[i] = receiving[i];
+            }
+            if (mMaxSending[i] < sending[i]) {
+              mMaxSending[i] = sending[i];
+            }
+          }
+        }
+
+        mPendingRemoteDescription =
+            mJsepSession->GetRemoteDescription(kJsepDescriptionPending);
+        mCurrentRemoteDescription =
+            mJsepSession->GetRemoteDescription(kJsepDescriptionCurrent);
+        mPendingLocalDescription =
+            mJsepSession->GetLocalDescription(kJsepDescriptionPending);
+        mCurrentLocalDescription =
+            mJsepSession->GetLocalDescription(kJsepDescriptionCurrent);
+        mPendingOfferer = mJsepSession->IsPendingOfferer();
+        mCurrentOfferer = mJsepSession->IsCurrentOfferer();
+
+        if (aSdpType == dom::RTCSdpType::Answer) {
+          std::set<std::pair<std::string, std::string>> iceCredentials =
+              mJsepSession->GetLocalIceCredentials();
+          std::vector<std::pair<std::string, std::string>>
+              iceCredentialsNotReplaced;
+          std::set_intersection(mLocalIceCredentialsToReplace.begin(),
+                                mLocalIceCredentialsToReplace.end(),
+                                iceCredentials.begin(), iceCredentials.end(),
+                                std::back_inserter(iceCredentialsNotReplaced));
+
+          if (iceCredentialsNotReplaced.empty()) {
+            mLocalIceCredentialsToReplace.clear();
+          }
+        }
+
+        if (newSignalingState == RTCSignalingState::Stable) {
+          mNegotiationNeeded = false;
+          UpdateNegotiationNeeded();
+        }
+
+        // Spec does not actually tell us to do this, but that is probably a
+        // spec bug.
+        UpdateConnectionState();
+
+        JSErrorResult jrv;
+        if (newSignalingState != mSignalingState) {
+          mSignalingState = newSignalingState;
+          mPCObserver->OnStateChange(PCObserverStateType::SignalingState, jrv);
+        }
+
+        if (aRemote) {
+          dom::RTCRtpReceiver::StreamAssociationChanges changes;
+          for (const auto& transceiver : mTransceivers) {
+            transceiver->Receiver()->UpdateStreams(&changes);
+          }
+
+          for (const auto& receiver : changes.mReceiversToMute) {
+            // This sets the muted state for the recv track and all its clones.
+            receiver->SetTrackMuteFromRemoteSdp();
+          }
+
+          for (const auto& association : changes.mStreamAssociationsRemoved) {
+            RefPtr<DOMMediaStream> stream =
+                GetReceiveStream(association.mStreamId);
+            if (stream && stream->HasTrack(*association.mTrack)) {
+              stream->RemoveTrackInternal(association.mTrack);
+            }
+          }
+
+          // TODO(Bug 1241291): For legacy event, remove eventually
+          std::vector<RefPtr<DOMMediaStream>> newStreams;
+
+          for (const auto& association : changes.mStreamAssociationsAdded) {
+            RefPtr<DOMMediaStream> stream =
+                GetReceiveStream(association.mStreamId);
+            if (!stream) {
+              stream = CreateReceiveStream(association.mStreamId);
+              newStreams.push_back(stream);
+            }
+
+            if (!stream->HasTrack(*association.mTrack)) {
+              stream->AddTrackInternal(association.mTrack);
+            }
+          }
+
+          // Make sure to wait until after we've calculated track changes before
+          // doing this.
+          for (size_t i = 0; i < mTransceivers.Length();) {
+            auto& transceiver = mTransceivers[i];
+            if (transceiver->ShouldRemove()) {
+              mTransceivers[i]->Close();
+              mTransceivers[i]->SetRemovedFromPc();
+              mTransceivers.RemoveElementAt(i);
+            } else {
+              ++i;
+            }
+          }
+
+          for (const auto& trackEvent : changes.mTrackEvents) {
+            dom::Sequence<OwningNonNull<DOMMediaStream>> streams;
+            for (const auto& id : trackEvent.mStreamIds) {
+              RefPtr<DOMMediaStream> stream = GetReceiveStream(id);
+              if (!stream) {
+                MOZ_ASSERT(false);
+                continue;
+              }
+              if (!streams.AppendElement(*stream, fallible)) {
+                // XXX(Bug 1632090) Instead of extending the array 1-by-1 (which
+                // might involve multiple reallocations) and potentially
+                // crashing here, SetCapacity could be called outside the loop
+                // once.
+                mozalloc_handle_oom(0);
+              }
+            }
+            mPCObserver->FireTrackEvent(*trackEvent.mReceiver, streams, jrv);
+          }
+
+          // TODO(Bug 1241291): Legacy event, remove eventually
+          for (const auto& stream : newStreams) {
+            mPCObserver->FireStreamEvent(*stream, jrv);
+          }
+        }
+        aP->MaybeResolveWithUndefined();
+      }));
+}
+
+void PeerConnectionImpl::OnSetDescriptionError() {
+  mUncommittedJsepSession = nullptr;
+}
+
+RTCSignalingState PeerConnectionImpl::GetSignalingState() const {
+  switch (mJsepSession->GetState()) {
+    case kJsepStateStable:
+      return RTCSignalingState::Stable;
+      break;
+    case kJsepStateHaveLocalOffer:
+      return RTCSignalingState::Have_local_offer;
+      break;
+    case kJsepStateHaveRemoteOffer:
+      return RTCSignalingState::Have_remote_offer;
+      break;
+    case kJsepStateHaveLocalPranswer:
+      return RTCSignalingState::Have_local_pranswer;
+      break;
+    case kJsepStateHaveRemotePranswer:
+      return RTCSignalingState::Have_remote_pranswer;
+      break;
+    case kJsepStateClosed:
+      return RTCSignalingState::Closed;
+      break;
+  }
+  MOZ_CRASH("Invalid JSEP state");
+}
+
+bool PeerConnectionImpl::IsClosed() const {
+  return mSignalingState == RTCSignalingState::Closed;
+}
+
+PeerConnectionWrapper::PeerConnectionWrapper(const std::string& handle)
+    : impl_(nullptr) {
+  if (PeerConnectionCtx::isActive()) {
+    impl_ = PeerConnectionCtx::GetInstance()->GetPeerConnection(handle);
+  }
+}
+
+const RefPtr<MediaTransportHandler> PeerConnectionImpl::GetTransportHandler()
+    const {
+  return mTransportHandler;
+}
+
+const std::string& PeerConnectionImpl::GetHandle() { return mHandle; }
+
+const std::string& PeerConnectionImpl::GetName() {
+  PC_AUTO_ENTER_API_CALL_NO_CHECK();
+  return mName;
+}
+
+void PeerConnectionImpl::CandidateReady(const std::string& candidate,
+                                        const std::string& transportId,
+                                        const std::string& ufrag) {
+  STAMP_TIMECARD(mTimeCard, "Ice Candidate gathered");
+  PC_AUTO_ENTER_API_CALL_VOID_RETURN(false);
+
+  if (mForceIceTcp && std::string::npos != candidate.find(" UDP ")) {
+    CSFLogWarn(LOGTAG, "Blocking local UDP candidate: %s", candidate.c_str());
+    STAMP_TIMECARD(mTimeCard, "UDP Ice Candidate blocked");
+    return;
+  }
+
+  // One of the very few places we still use level; required by the JSEP API
+  uint16_t level = 0;
+  std::string mid;
+  bool skipped = false;
+
+  if (mUncommittedJsepSession) {
+    // An sLD or sRD is in progress, and while that is the case, we need to add
+    // the candidate to both the current JSEP engine, and the uncommitted JSEP
+    // engine. We ignore errors because the spec says to only take into account
+    // the current/pending local descriptions when determining whether to
+    // surface the candidate to content, which does not take into account any
+    // in-progress sRD/sLD.
+    Unused << mUncommittedJsepSession->AddLocalIceCandidate(
+        candidate, transportId, ufrag, &level, &mid, &skipped);
+  }
+
+  nsresult res = mJsepSession->AddLocalIceCandidate(
+      candidate, transportId, ufrag, &level, &mid, &skipped);
+
+  if (NS_FAILED(res)) {
+    std::string errorString = mJsepSession->GetLastError();
+
+    STAMP_TIMECARD(mTimeCard, "Local Ice Candidate invalid");
+    CSFLogError(LOGTAG,
+                "Failed to incorporate local candidate into SDP:"
+                " res = %u, candidate = %s, transport-id = %s,"
+                " error = %s",
+                static_cast<unsigned>(res), candidate.c_str(),
+                transportId.c_str(), errorString.c_str());
+    return;
+  }
+
+  if (skipped) {
+    STAMP_TIMECARD(mTimeCard, "Local Ice Candidate skipped");
+    CSFLogInfo(LOGTAG,
+               "Skipped adding local candidate %s (transport-id %s) "
+               "to SDP, this typically happens because the m-section "
+               "is bundled, which means it doesn't make sense for it "
+               "to have its own transport-related attributes.",
+               candidate.c_str(), transportId.c_str());
+    return;
+  }
+
+  mPendingLocalDescription =
+      mJsepSession->GetLocalDescription(kJsepDescriptionPending);
+  mCurrentLocalDescription =
+      mJsepSession->GetLocalDescription(kJsepDescriptionCurrent);
+  CSFLogInfo(LOGTAG, "Passing local candidate to content: %s",
+             candidate.c_str());
+  SendLocalIceCandidateToContent(level, mid, candidate, ufrag);
+}
+
+void PeerConnectionImpl::SendLocalIceCandidateToContent(
+    uint16_t level, const std::string& mid, const std::string& candidate,
+    const std::string& ufrag) {
+  STAMP_TIMECARD(mTimeCard, "Send Ice Candidate to content");
+  JSErrorResult rv;
+  mPCObserver->OnIceCandidate(level, ObString(mid.c_str()),
+                              ObString(candidate.c_str()),
+                              ObString(ufrag.c_str()), rv);
+}
+
+void PeerConnectionImpl::IceConnectionStateChange(
+    dom::RTCIceConnectionState domState) {
+  PC_AUTO_ENTER_API_CALL_VOID_RETURN(false);
+
+  CSFLogDebug(LOGTAG, "%s: %d -> %d", __FUNCTION__,
+              static_cast<int>(mIceConnectionState),
+              static_cast<int>(domState));
+
+  if (domState == mIceConnectionState) {
+    // no work to be done since the states are the same.
+    // this can happen during ICE rollback situations.
+    return;
+  }
+
+  mIceConnectionState = domState;
+
+  // Would be nice if we had a means of converting one of these dom enums
+  // to a string that wasn't almost as much text as this switch statement...
+  switch (mIceConnectionState) {
+    case RTCIceConnectionState::New:
+      STAMP_TIMECARD(mTimeCard, "Ice state: new");
+      break;
+    case RTCIceConnectionState::Checking:
+      // For telemetry
+      mIceStartTime = TimeStamp::Now();
+      STAMP_TIMECARD(mTimeCard, "Ice state: checking");
+      break;
+    case RTCIceConnectionState::Connected:
+      STAMP_TIMECARD(mTimeCard, "Ice state: connected");
+      StartCallTelem();
+      break;
+    case RTCIceConnectionState::Completed:
+      STAMP_TIMECARD(mTimeCard, "Ice state: completed");
+      break;
+    case RTCIceConnectionState::Failed:
+      STAMP_TIMECARD(mTimeCard, "Ice state: failed");
+      break;
+    case RTCIceConnectionState::Disconnected:
+      STAMP_TIMECARD(mTimeCard, "Ice state: disconnected");
+      break;
+    case RTCIceConnectionState::Closed:
+      STAMP_TIMECARD(mTimeCard, "Ice state: closed");
+      break;
+    default:
+      MOZ_ASSERT_UNREACHABLE("Unexpected mIceConnectionState!");
+  }
+
+  WrappableJSErrorResult rv;
+  mPCObserver->OnStateChange(PCObserverStateType::IceConnectionState, rv);
+  UpdateConnectionState();
+}
+
+void PeerConnectionImpl::OnCandidateFound(const std::string& aTransportId,
+                                          const CandidateInfo& aCandidateInfo) {
+  if (mStunAddrsRequest && !aCandidateInfo.mMDNSAddress.empty()) {
+    MOZ_ASSERT(!aCandidateInfo.mActualAddress.empty());
+
+    if (mCanRegisterMDNSHostnamesDirectly) {
+      auto itor = mRegisteredMDNSHostnames.find(aCandidateInfo.mMDNSAddress);
+
+      // We'll see the address twice if we're generating both UDP and TCP
+      // candidates.
+      if (itor == mRegisteredMDNSHostnames.end()) {
+        mRegisteredMDNSHostnames.insert(aCandidateInfo.mMDNSAddress);
+        mStunAddrsRequest->SendRegisterMDNSHostname(
+            nsCString(aCandidateInfo.mMDNSAddress.c_str()),
+            nsCString(aCandidateInfo.mActualAddress.c_str()));
+      }
+    } else {
+      mMDNSHostnamesToRegister.emplace(aCandidateInfo.mMDNSAddress,
+                                       aCandidateInfo.mActualAddress);
+    }
+  }
+
+  if (!aCandidateInfo.mDefaultHostRtp.empty()) {
+    UpdateDefaultCandidate(aCandidateInfo.mDefaultHostRtp,
+                           aCandidateInfo.mDefaultPortRtp,
+                           aCandidateInfo.mDefaultHostRtcp,
+                           aCandidateInfo.mDefaultPortRtcp, aTransportId);
+  }
+  CandidateReady(aCandidateInfo.mCandidate, aTransportId,
+                 aCandidateInfo.mUfrag);
+}
+
+void PeerConnectionImpl::IceGatheringStateChange(
+    dom::RTCIceGatheringState state) {
+  PC_AUTO_ENTER_API_CALL_VOID_RETURN(false);
+
+  CSFLogDebug(LOGTAG, "%s %d", __FUNCTION__, static_cast<int>(state));
+  if (mIceGatheringState == state) {
+    return;
+  }
+
+  mIceGatheringState = state;
+
+  // Would be nice if we had a means of converting one of these dom enums
+  // to a string that wasn't almost as much text as this switch statement...
+  switch (mIceGatheringState) {
+    case RTCIceGatheringState::New:
+      STAMP_TIMECARD(mTimeCard, "Ice gathering state: new");
+      break;
+    case RTCIceGatheringState::Gathering:
+      STAMP_TIMECARD(mTimeCard, "Ice gathering state: gathering");
+      break;
+    case RTCIceGatheringState::Complete:
+      STAMP_TIMECARD(mTimeCard, "Ice gathering state: complete");
+      break;
+    default:
+      MOZ_ASSERT_UNREACHABLE("Unexpected mIceGatheringState!");
+  }
+
+  JSErrorResult rv;
+  mPCObserver->OnStateChange(PCObserverStateType::IceGatheringState, rv);
+}
+
+void PeerConnectionImpl::UpdateDefaultCandidate(
+    const std::string& defaultAddr, uint16_t defaultPort,
+    const std::string& defaultRtcpAddr, uint16_t defaultRtcpPort,
+    const std::string& transportId) {
+  CSFLogDebug(LOGTAG, "%s", __FUNCTION__);
+  mJsepSession->UpdateDefaultCandidate(
+      defaultAddr, defaultPort, defaultRtcpAddr, defaultRtcpPort, transportId);
+  if (mUncommittedJsepSession) {
+    mUncommittedJsepSession->UpdateDefaultCandidate(
+        defaultAddr, defaultPort, defaultRtcpAddr, defaultRtcpPort,
+        transportId);
+  }
+}
+
+static UniquePtr<dom::RTCStatsCollection> GetDataChannelStats_s(
+    const RefPtr<DataChannelConnection>& aDataConnection,
+    const DOMHighResTimeStamp aTimestamp) {
+  UniquePtr<dom::RTCStatsCollection> report(new dom::RTCStatsCollection);
+  if (aDataConnection) {
+    aDataConnection->AppendStatsToReport(report, aTimestamp);
+  }
+  return report;
+}
+
+RefPtr<dom::RTCStatsPromise> PeerConnectionImpl::GetDataChannelStats(
+    const RefPtr<DataChannelConnection>& aDataChannelConnection,
+    const DOMHighResTimeStamp aTimestamp) {
+  // Gather stats from DataChannels
+  return InvokeAsync(
+      GetMainThreadSerialEventTarget(), __func__,
+      [aDataChannelConnection, aTimestamp]() {
+        return dom::RTCStatsPromise::CreateAndResolve(
+            GetDataChannelStats_s(aDataChannelConnection, aTimestamp),
+            __func__);
+      });
+}
+
+void PeerConnectionImpl::CollectConduitTelemetryData() {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  nsTArray<RefPtr<VideoSessionConduit>> conduits;
+  for (const auto& transceiver : mTransceivers) {
+    if (RefPtr<MediaSessionConduit> conduit = transceiver->GetConduit()) {
+      conduit->AsVideoSessionConduit().apply(
+          [&](const auto& aVideo) { conduits.AppendElement(aVideo); });
+    }
+  }
+
+  if (!conduits.IsEmpty() && mCall) {
+    mCall->mCallThread->Dispatch(
+        NS_NewRunnableFunction(__func__, [conduits = std::move(conduits)] {
+          for (const auto& conduit : conduits) {
+            conduit->CollectTelemetryData();
+          }
+        }));
+  }
+}
+
+nsTArray<dom::RTCCodecStats> PeerConnectionImpl::GetCodecStats(
+    DOMHighResTimeStamp aNow) {
+  MOZ_ASSERT(NS_IsMainThread());
+  nsTArray<dom::RTCCodecStats> result;
+
+  struct CodecComparator {
+    bool operator()(const JsepCodecDescription* aA,
+                    const JsepCodecDescription* aB) const {
+      return aA->StatsId() < aB->StatsId();
+    }
+  };
+
+  // transportId -> codec; per direction (whether the codecType
+  // shall be "encode", "decode" or absent (if a codec exists in both maps for a
+  // transport)). These do the bookkeeping to ensure codec stats get coalesced
+  // to transport level.
+  std::map<std::string, std::set<JsepCodecDescription*, CodecComparator>>
+      sendCodecMap;
+  std::map<std::string, std::set<JsepCodecDescription*, CodecComparator>>
+      recvCodecMap;
+
+  // Find all JsepCodecDescription instances we want to turn into codec stats.
+  for (const auto& transceiver : mTransceivers) {
+    // TODO: Grab these from the JSEP transceivers instead
+    auto sendCodecs = transceiver->GetNegotiatedSendCodecs();
+    auto recvCodecs = transceiver->GetNegotiatedRecvCodecs();
+
+    const std::string transportId = transceiver->GetTransportId();
+    // This ensures both codec maps have the same size.
+    auto& sendMap = sendCodecMap[transportId];
+    auto& recvMap = recvCodecMap[transportId];
+
+    sendCodecs.apply([&](const auto& aCodecs) {
+      for (const auto& codec : aCodecs) {
+        sendMap.insert(codec.get());
+      }
+    });
+    recvCodecs.apply([&](const auto& aCodecs) {
+      for (const auto& codec : aCodecs) {
+        recvMap.insert(codec.get());
+      }
+    });
+  }
+
+  auto createCodecStat = [&](const JsepCodecDescription* aCodec,
+                             const nsString& aTransportId,
+                             Maybe<RTCCodecType> aCodecType) {
+    uint16_t pt;
+    {
+      DebugOnly<bool> rv = aCodec->GetPtAsInt(&pt);
+      MOZ_ASSERT(rv);
+    }
+    nsString mimeType;
+    mimeType.AppendPrintf(
+        "%s/%s", aCodec->Type() == SdpMediaSection::kVideo ? "video" : "audio",
+        aCodec->mName.c_str());
+    nsString id = aTransportId;
+    id.Append(u"_");
+    id.Append(aCodec->StatsId());
+
+    dom::RTCCodecStats codec;
+    codec.mId.Construct(std::move(id));
+    codec.mTimestamp.Construct(aNow);
+    codec.mType.Construct(RTCStatsType::Codec);
+    codec.mPayloadType = pt;
+    if (aCodecType) {
+      codec.mCodecType.Construct(*aCodecType);
+    }
+    codec.mTransportId = aTransportId;
+    codec.mMimeType = std::move(mimeType);
+    codec.mClockRate.Construct(aCodec->mClock);
+    if (aCodec->Type() == SdpMediaSection::MediaType::kAudio) {
+      codec.mChannels.Construct(aCodec->mChannels);
+    }
+    if (aCodec->mSdpFmtpLine) {
+      codec.mSdpFmtpLine.Construct(
+          NS_ConvertUTF8toUTF16(aCodec->mSdpFmtpLine->c_str()));
+    }
+
+    result.AppendElement(std::move(codec));
+  };
+
+  // Create codec stats for the gathered codec descriptions, sorted primarily
+  // by transportId, secondarily by payload type (from StatsId()).
+  for (const auto& [transportId, sendCodecs] : sendCodecMap) {
+    const auto& recvCodecs = recvCodecMap[transportId];
+    const nsString tid = NS_ConvertASCIItoUTF16(transportId);
+    AutoTArray<JsepCodecDescription*, 16> bidirectionalCodecs;
+    AutoTArray<JsepCodecDescription*, 16> unidirectionalCodecs;
+    std::set_intersection(sendCodecs.cbegin(), sendCodecs.cend(),
+                          recvCodecs.cbegin(), recvCodecs.cend(),
+                          MakeBackInserter(bidirectionalCodecs),
+                          CodecComparator());
+    std::set_symmetric_difference(sendCodecs.cbegin(), sendCodecs.cend(),
+                                  recvCodecs.cbegin(), recvCodecs.cend(),
+                                  MakeBackInserter(unidirectionalCodecs),
+                                  CodecComparator());
+    for (const auto* codec : bidirectionalCodecs) {
+      createCodecStat(codec, tid, Nothing());
+    }
+    for (const auto* codec : unidirectionalCodecs) {
+      createCodecStat(
+          codec, tid,
+          Some(codec->mDirection == sdp::kSend ? RTCCodecType::Encode
+                                               : RTCCodecType::Decode));
+    }
+  }
+
+  return result;
+}
+
+RefPtr<dom::RTCStatsReportPromise> PeerConnectionImpl::GetStats(
+    dom::MediaStreamTrack* aSelector, bool aInternalStats) {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  if (mFinalStatsQuery) {
+    // This case should be _extremely_ rare; this will basically only happen
+    // when WebrtcGlobalInformation tries to get our stats while we are tearing
+    // down.
+    return mFinalStatsQuery->Then(
+        GetMainThreadSerialEventTarget(), __func__,
+        [this, self = RefPtr<PeerConnectionImpl>(this)]() {
+          UniquePtr<dom::RTCStatsReportInternal> finalStats =
+              MakeUnique<dom::RTCStatsReportInternal>();
+          // Might not be set if this encountered some error.
+          if (mFinalStats) {
+            *finalStats = *mFinalStats;
+          }
+          return RTCStatsReportPromise::CreateAndResolve(std::move(finalStats),
+                                                         __func__);
+        });
+  }
+
+  nsTArray<RefPtr<dom::RTCStatsPromise>> promises;
+  DOMHighResTimeStamp now = mTimestampMaker.GetNow().ToDom();
+
+  nsTArray<dom::RTCCodecStats> codecStats = GetCodecStats(now);
+  std::set<std::string> transportIds;
+
+  if (!aSelector) {
+    // There might not be any senders/receivers if we're DataChannel only, so we
+    // don't handle the null selector case in the loop below.
+    transportIds.insert("");
+  }
+
+  nsTArray<
+      std::tuple<RTCRtpTransceiver*, RefPtr<RTCStatsPromise::AllPromiseType>>>
+      transceiverStatsPromises;
+  for (const auto& transceiver : mTransceivers) {
+    const bool sendSelected = transceiver->Sender()->HasTrack(aSelector);
+    const bool recvSelected = transceiver->Receiver()->HasTrack(aSelector);
+    if (!sendSelected && !recvSelected) {
+      continue;
+    }
+
+    if (aSelector) {
+      transportIds.insert(transceiver->GetTransportId());
+    }
+
+    nsTArray<RefPtr<RTCStatsPromise>> rtpStreamPromises;
+    // Get all rtp stream stats for the given selector. Then filter away any
+    // codec stat not related to the selector, and assign codec ids to the
+    // stream stats.
+    // Skips the ICE stats; we do our own queries based on |transportIds| to
+    // avoid duplicates
+    if (sendSelected) {
+      rtpStreamPromises.AppendElements(
+          transceiver->Sender()->GetStatsInternal(true));
+    }
+    if (recvSelected) {
+      rtpStreamPromises.AppendElements(
+          transceiver->Receiver()->GetStatsInternal(true));
+    }
+    transceiverStatsPromises.AppendElement(
+        std::make_tuple(transceiver.get(),
+                        RTCStatsPromise::All(GetMainThreadSerialEventTarget(),
+                                             rtpStreamPromises)));
+  }
+
+  promises.AppendElement(RTCRtpTransceiver::ApplyCodecStats(
+      std::move(codecStats), std::move(transceiverStatsPromises)));
+
+  for (const auto& transportId : transportIds) {
+    promises.AppendElement(mTransportHandler->GetIceStats(transportId, now));
+  }
+
+  promises.AppendElement(GetDataChannelStats(mDataConnection, now));
+
+  auto pcStatsCollection = MakeUnique<dom::RTCStatsCollection>();
+  RTCPeerConnectionStats pcStats;
+  pcStats.mTimestamp.Construct(now);
+  pcStats.mType.Construct(RTCStatsType::Peer_connection);
+  pcStats.mId.Construct(NS_ConvertUTF8toUTF16(mHandle.c_str()));
+  pcStats.mDataChannelsOpened.Construct(mDataChannelsOpened);
+  pcStats.mDataChannelsClosed.Construct(mDataChannelsClosed);
+  if (!pcStatsCollection->mPeerConnectionStats.AppendElement(std::move(pcStats),
+                                                             fallible)) {
+    mozalloc_handle_oom(0);
+  }
+  promises.AppendElement(RTCStatsPromise::CreateAndResolve(
+      std::move(pcStatsCollection), __func__));
+
+  // This is what we're going to return; all the stuff in |promises| will be
+  // accumulated here.
+  UniquePtr<dom::RTCStatsReportInternal> report(
+      new dom::RTCStatsReportInternal);
+  report->mPcid = NS_ConvertASCIItoUTF16(mName.c_str());
+  if (mWindow && mWindow->GetBrowsingContext()) {
+    report->mBrowserId = mWindow->GetBrowsingContext()->BrowserId();
+  }
+  report->mConfiguration.Construct(mJsConfiguration);
+  // TODO(bug 1589416): We need to do better here.
+  if (!mIceStartTime.IsNull()) {
+    report->mCallDurationMs.Construct(
+        (TimeStamp::Now() - mIceStartTime).ToMilliseconds());
+  }
+  report->mIceRestarts = mIceRestartCount;
+  report->mIceRollbacks = mIceRollbackCount;
+  report->mClosed = false;
+  report->mTimestamp = now;
+
+  if (aInternalStats && mJsepSession) {
+    for (const auto& candidate : mRawTrickledCandidates) {
+      if (!report->mRawRemoteCandidates.AppendElement(
+              NS_ConvertASCIItoUTF16(candidate.c_str()), fallible)) {
+        // XXX(Bug 1632090) Instead of extending the array 1-by-1 (which might
+        // involve multiple reallocations) and potentially crashing here,
+        // SetCapacity could be called outside the loop once.
+        mozalloc_handle_oom(0);
+      }
+    }
+
+    if (mJsepSession) {
+      // TODO we probably should report Current and Pending SDPs here
+      // separately. Plus the raw SDP we got from JS (mLocalRequestedSDP).
+      // And if it's the offer or answer would also be nice.
+      std::string localDescription =
+          mJsepSession->GetLocalDescription(kJsepDescriptionPendingOrCurrent);
+      std::string remoteDescription =
+          mJsepSession->GetRemoteDescription(kJsepDescriptionPendingOrCurrent);
+      if (!report->mSdpHistory.AppendElements(mSdpHistory, fallible)) {
+        mozalloc_handle_oom(0);
+      }
+      if (mJsepSession->IsPendingOfferer().isSome()) {
+        report->mOfferer.Construct(*mJsepSession->IsPendingOfferer());
+      } else if (mJsepSession->IsCurrentOfferer().isSome()) {
+        report->mOfferer.Construct(*mJsepSession->IsCurrentOfferer());
+      } else {
+        // Silly.
+        report->mOfferer.Construct(false);
+      }
+    }
+  }
+
+  return dom::RTCStatsPromise::All(GetMainThreadSerialEventTarget(), promises)
+      ->Then(
+          GetMainThreadSerialEventTarget(), __func__,
+          [report = std::move(report), idGen = mIdGenerator](
+              nsTArray<UniquePtr<dom::RTCStatsCollection>> aStats) mutable {
+            idGen->RewriteIds(std::move(aStats), report.get());
+            return dom::RTCStatsReportPromise::CreateAndResolve(
+                std::move(report), __func__);
+          },
+          [](nsresult rv) {
+            return dom::RTCStatsReportPromise::CreateAndReject(rv, __func__);
+          });
+}
+
+void PeerConnectionImpl::RecordIceRestartStatistics(JsepSdpType type) {
+  switch (type) {
+    case mozilla::kJsepSdpOffer:
+    case mozilla::kJsepSdpPranswer:
+      break;
+    case mozilla::kJsepSdpAnswer:
+      ++mIceRestartCount;
+      break;
+    case mozilla::kJsepSdpRollback:
+      ++mIceRollbackCount;
+      break;
+  }
+}
+
+void PeerConnectionImpl::StoreConfigurationForAboutWebrtc(
+    const dom::RTCConfiguration& aConfig) {
+  // This will only be called once, when the PeerConnection is initially
+  // configured, at least until setConfiguration is implemented
+  // see https://bugzilla.mozilla.org/show_bug.cgi?id=1253706
+  // @TODO bug 1739451 call this from setConfiguration
+  mJsConfiguration.mIceServers.Clear();
+  for (const auto& server : aConfig.mIceServers) {
+    RTCIceServerInternal internal;
+    internal.mCredentialProvided = server.mCredential.WasPassed();
+    internal.mUserNameProvided = server.mUsername.WasPassed();
+    if (server.mUrl.WasPassed()) {
+      if (!internal.mUrls.AppendElement(server.mUrl.Value(), fallible)) {
+        mozalloc_handle_oom(0);
+      }
+    }
+    if (server.mUrls.WasPassed()) {
+      for (const auto& url : server.mUrls.Value().GetAsStringSequence()) {
+        if (!internal.mUrls.AppendElement(url, fallible)) {
+          mozalloc_handle_oom(0);
+        }
+      }
+    }
+    if (!mJsConfiguration.mIceServers.AppendElement(internal, fallible)) {
+      mozalloc_handle_oom(0);
+    }
+  }
+  mJsConfiguration.mSdpSemantics.Reset();
+  if (aConfig.mSdpSemantics.WasPassed()) {
+    mJsConfiguration.mSdpSemantics.Construct(aConfig.mSdpSemantics.Value());
+  }
+
+  mJsConfiguration.mIceTransportPolicy.Reset();
+  mJsConfiguration.mIceTransportPolicy.Construct(aConfig.mIceTransportPolicy);
+  mJsConfiguration.mBundlePolicy.Reset();
+  mJsConfiguration.mBundlePolicy.Construct(aConfig.mBundlePolicy);
+  mJsConfiguration.mPeerIdentityProvided = !aConfig.mPeerIdentity.IsEmpty();
+  mJsConfiguration.mCertificatesProvided = !aConfig.mCertificates.Length();
+}
+
+dom::Sequence<dom::RTCSdpParsingErrorInternal>
+PeerConnectionImpl::GetLastSdpParsingErrors() const {
+  const auto& sdpErrors = mJsepSession->GetLastSdpParsingErrors();
+  dom::Sequence<dom::RTCSdpParsingErrorInternal> domErrors;
+  if (!domErrors.SetCapacity(domErrors.Length(), fallible)) {
+    mozalloc_handle_oom(0);
+  }
+  for (const auto& error : sdpErrors) {
+    mozilla::dom::RTCSdpParsingErrorInternal internal;
+    internal.mLineNumber = error.first;
+    if (!AppendASCIItoUTF16(MakeStringSpan(error.second.c_str()),
+                            internal.mError, fallible)) {
+      mozalloc_handle_oom(0);
+    }
+    if (!domErrors.AppendElement(std::move(internal), fallible)) {
+      mozalloc_handle_oom(0);
+    }
+  }
+  return domErrors;
+}
+
+// Telemetry for when calls start
+void PeerConnectionImpl::StartCallTelem() {
+  if (mCallTelemStarted) {
+    return;
+  }
+  MOZ_RELEASE_ASSERT(mWindow);
+  uint64_t windowId = mWindow->WindowID();
+  auto found = sCallDurationTimers.find(windowId);
+  if (found == sCallDurationTimers.end()) {
+    found =
+        sCallDurationTimers.emplace(windowId, PeerConnectionAutoTimer()).first;
+  }
+  found->second.RegisterConnection();
+  mCallTelemStarted = true;
+
+  // Increment session call counter
+  // If we want to track Loop calls independently here, we need two
+  // histograms.
+  //
+  // NOTE: As of bug 1654248 landing we are no longer counting renegotiations
+  // as separate calls. Expect numbers to drop compared to
+  // WEBRTC_CALL_COUNT_2.
+  Telemetry::Accumulate(Telemetry::WEBRTC_CALL_COUNT_3, 1);
+}
+
+void PeerConnectionImpl::StunAddrsHandler::OnMDNSQueryComplete(
+    const nsCString& hostname, const Maybe<nsCString>& address) {
+  MOZ_ASSERT(NS_IsMainThread());
+  PeerConnectionWrapper pcw(mPcHandle);
+  if (!pcw.impl()) {
+    return;
+  }
+  auto itor = pcw.impl()->mQueriedMDNSHostnames.find(hostname.BeginReading());
+  if (itor != pcw.impl()->mQueriedMDNSHostnames.end()) {
+    if (address) {
+      for (auto& cand : itor->second) {
+        // Replace obfuscated address with actual address
+        std::string obfuscatedAddr = cand.mTokenizedCandidate[4];
+        cand.mTokenizedCandidate[4] = address->BeginReading();
+        std::ostringstream o;
+        for (size_t i = 0; i < cand.mTokenizedCandidate.size(); ++i) {
+          o << cand.mTokenizedCandidate[i];
+          if (i + 1 != cand.mTokenizedCandidate.size()) {
+            o << " ";
+          }
+        }
+        std::string mungedCandidate = o.str();
+        pcw.impl()->StampTimecard("Done looking up mDNS name");
+        pcw.impl()->mTransportHandler->AddIceCandidate(
+            cand.mTransportId, mungedCandidate, cand.mUfrag, obfuscatedAddr);
+      }
+    } else {
+      pcw.impl()->StampTimecard("Failed looking up mDNS name");
+    }
+    pcw.impl()->mQueriedMDNSHostnames.erase(itor);
+  }
+}
+
+void PeerConnectionImpl::StunAddrsHandler::OnStunAddrsAvailable(
+    const mozilla::net::NrIceStunAddrArray& addrs) {
+  CSFLogInfo(LOGTAG, "%s: receiving (%d) stun addrs", __FUNCTION__,
+             (int)addrs.Length());
+  PeerConnectionWrapper pcw(mPcHandle);
+  if (!pcw.impl()) {
+    return;
+  }
+  pcw.impl()->mStunAddrs = addrs.Clone();
+  pcw.impl()->mLocalAddrsRequestState = STUN_ADDR_REQUEST_COMPLETE;
+  pcw.impl()->FlushIceCtxOperationQueueIfReady();
+  // If parent process returns 0 STUN addresses, change ICE connection
+  // state to failed.
+  if (!pcw.impl()->mStunAddrs.Length()) {
+    pcw.impl()->IceConnectionStateChange(dom::RTCIceConnectionState::Failed);
+  }
+}
+
+void PeerConnectionImpl::InitLocalAddrs() {
+  if (mLocalAddrsRequestState == STUN_ADDR_REQUEST_PENDING) {
+    return;
+  }
+  if (mStunAddrsRequest) {
+    mLocalAddrsRequestState = STUN_ADDR_REQUEST_PENDING;
+    mStunAddrsRequest->SendGetStunAddrs();
+  } else {
+    mLocalAddrsRequestState = STUN_ADDR_REQUEST_COMPLETE;
+  }
+}
+
+bool PeerConnectionImpl::ShouldForceProxy() const {
+  if (Preferences::GetBool("media.peerconnection.ice.proxy_only", false)) {
+    return true;
+  }
+
+  bool isPBM = false;
+  // This complicated null check is being extra conservative to avoid
+  // introducing crashes. It may not be needed.
+  if (mWindow && mWindow->GetExtantDoc() &&
+      mWindow->GetExtantDoc()->GetPrincipal() &&
+      mWindow->GetExtantDoc()
+              ->GetPrincipal()
+              ->OriginAttributesRef()
+              .mPrivateBrowsingId > 0) {
+    isPBM = true;
+  }
+
+  if (isPBM && Preferences::GetBool(
+                   "media.peerconnection.ice.proxy_only_if_pbmode", false)) {
+    return true;
+  }
+
+  if (!Preferences::GetBool(
+          "media.peerconnection.ice.proxy_only_if_behind_proxy", false)) {
+    return false;
+  }
+
+  // Ok, we're supposed to be proxy_only, but only if a proxy is configured.
+  // Let's just see if the document was loaded via a proxy.
+
+  nsCOMPtr<nsIHttpChannelInternal> httpChannelInternal = GetChannel();
+  if (!httpChannelInternal) {
+    return false;
+  }
+
+  bool proxyUsed = false;
+  Unused << httpChannelInternal->GetIsProxyUsed(&proxyUsed);
+  return proxyUsed;
+}
+
+void PeerConnectionImpl::EnsureTransports(const JsepSession& aSession) {
+  mJsepSession->ForEachTransceiver([this,
+                                    self = RefPtr<PeerConnectionImpl>(this)](
+                                       const JsepTransceiver& transceiver) {
+    if (transceiver.HasOwnTransport()) {
+      mTransportHandler->EnsureProvisionalTransport(
+          transceiver.mTransport.mTransportId,
+          transceiver.mTransport.mLocalUfrag, transceiver.mTransport.mLocalPwd,
+          transceiver.mTransport.mComponents);
+    }
+  });
+
+  GatherIfReady();
+}
+
+void PeerConnectionImpl::UpdateRTCDtlsTransports(bool aMarkAsStable) {
+  mJsepSession->ForEachTransceiver(
+      [this, self = RefPtr<PeerConnectionImpl>(this)](
+          const JsepTransceiver& jsepTransceiver) {
+        std::string transportId = jsepTransceiver.mTransport.mTransportId;
+        if (transportId.empty()) {
+          return;
+        }
+        if (!mTransportIdToRTCDtlsTransport.count(transportId)) {
+          mTransportIdToRTCDtlsTransport.emplace(
+              transportId, new RTCDtlsTransport(GetParentObject()));
+        }
+      });
+
+  for (auto& transceiver : mTransceivers) {
+    std::string transportId = transceiver->GetTransportId();
+    if (transportId.empty()) {
+      continue;
+    }
+    if (mTransportIdToRTCDtlsTransport.count(transportId)) {
+      transceiver->SetDtlsTransport(mTransportIdToRTCDtlsTransport[transportId],
+                                    aMarkAsStable);
+    }
+  }
+
+  // Spec says we only update the RTCSctpTransport when negotiation completes
+}
+
+void PeerConnectionImpl::RollbackRTCDtlsTransports() {
+  for (auto& transceiver : mTransceivers) {
+    transceiver->RollbackToStableDtlsTransport();
+  }
+}
+
+void PeerConnectionImpl::RemoveRTCDtlsTransportsExcept(
+    const std::set<std::string>& aTransportIds) {
+  for (auto iter = mTransportIdToRTCDtlsTransport.begin();
+       iter != mTransportIdToRTCDtlsTransport.end();) {
+    if (!aTransportIds.count(iter->first)) {
+      iter = mTransportIdToRTCDtlsTransport.erase(iter);
+    } else {
+      ++iter;
+    }
+  }
+}
+
+nsresult PeerConnectionImpl::UpdateTransports(const JsepSession& aSession,
+                                              const bool forceIceTcp) {
+  std::set<std::string> finalTransports;
+  Maybe<std::string> sctpTransport;
+  mJsepSession->ForEachTransceiver(
+      [&, this, self = RefPtr<PeerConnectionImpl>(this)](
+          const JsepTransceiver& transceiver) {
+        if (transceiver.GetMediaType() == SdpMediaSection::kApplication &&
+            transceiver.HasTransport()) {
+          sctpTransport = Some(transceiver.mTransport.mTransportId);
+        }
+
+        if (transceiver.HasOwnTransport()) {
+          finalTransports.insert(transceiver.mTransport.mTransportId);
+          UpdateTransport(transceiver, forceIceTcp);
+        }
+      });
+
+  // clean up the unused RTCDtlsTransports
+  RemoveRTCDtlsTransportsExcept(finalTransports);
+
+  mTransportHandler->RemoveTransportsExcept(finalTransports);
+
+  for (const auto& transceiverImpl : mTransceivers) {
+    transceiverImpl->UpdateTransport();
+  }
+
+  if (sctpTransport.isSome()) {
+    auto it = mTransportIdToRTCDtlsTransport.find(*sctpTransport);
+    if (it == mTransportIdToRTCDtlsTransport.end()) {
+      // What?
+      MOZ_ASSERT(false);
+      return NS_ERROR_FAILURE;
+    }
+    if (!mDataConnection) {
+      // What?
+      MOZ_ASSERT(false);
+      return NS_ERROR_FAILURE;
+    }
+    RefPtr<RTCDtlsTransport> dtlsTransport = it->second;
+    // Why on earth does the spec use a floating point for this?
+    double maxMessageSize =
+        static_cast<double>(mDataConnection->GetMaxMessageSize());
+    Nullable<uint16_t> maxChannels;
+
+    if (!mSctpTransport) {
+      mSctpTransport = new RTCSctpTransport(GetParentObject(), *dtlsTransport,
+                                            maxMessageSize, maxChannels);
+    } else {
+      mSctpTransport->SetTransport(*dtlsTransport);
+      mSctpTransport->SetMaxMessageSize(maxMessageSize);
+      mSctpTransport->SetMaxChannels(maxChannels);
+    }
+  } else {
+    mSctpTransport = nullptr;
+  }
+
+  return NS_OK;
+}
+
+void PeerConnectionImpl::UpdateTransport(const JsepTransceiver& aTransceiver,
+                                         bool aForceIceTcp) {
+  std::string ufrag;
+  std::string pwd;
+  std::vector<std::string> candidates;
+  size_t components = 0;
+
+  const JsepTransport& transport = aTransceiver.mTransport;
+  unsigned level = aTransceiver.GetLevel();
+
+  CSFLogDebug(LOGTAG, "ACTIVATING TRANSPORT! - PC %s: level=%u components=%u",
+              mHandle.c_str(), (unsigned)level,
+              (unsigned)transport.mComponents);
+
+  ufrag = transport.mIce->GetUfrag();
+  pwd = transport.mIce->GetPassword();
+  candidates = transport.mIce->GetCandidates();
+  components = transport.mComponents;
+  if (aForceIceTcp) {
+    candidates.erase(
+        std::remove_if(candidates.begin(), candidates.end(),
+                       [](const std::string& s) {
+                         return s.find(" UDP ") != std::string::npos ||
+                                s.find(" udp ") != std::string::npos;
+                       }),
+        candidates.end());
+  }
+
+  nsTArray<uint8_t> keyDer;
+  nsTArray<uint8_t> certDer;
+  nsresult rv = Identity()->Serialize(&keyDer, &certDer);
+  if (NS_FAILED(rv)) {
+    CSFLogError(LOGTAG, "%s: Failed to serialize DTLS identity: %d",
+                __FUNCTION__, (int)rv);
+    return;
+  }
+
+  DtlsDigestList digests;
+  for (const auto& fingerprint :
+       transport.mDtls->GetFingerprints().mFingerprints) {
+    std::ostringstream ss;
+    ss << fingerprint.hashFunc;
+    digests.emplace_back(ss.str(), fingerprint.fingerprint);
+  }
+
+  mTransportHandler->ActivateTransport(
+      transport.mTransportId, transport.mLocalUfrag, transport.mLocalPwd,
+      components, ufrag, pwd, keyDer, certDer, Identity()->auth_type(),
+      transport.mDtls->GetRole() == JsepDtlsTransport::kJsepDtlsClient, digests,
+      PrivacyRequested());
+
+  for (auto& candidate : candidates) {
+    AddIceCandidate("candidate:" + candidate, transport.mTransportId, ufrag);
+  }
+}
+
+nsresult PeerConnectionImpl::UpdateMediaPipelines() {
+  for (RefPtr<RTCRtpTransceiver>& transceiver : mTransceivers) {
+    transceiver->ResetSync();
+  }
+
+  for (RefPtr<RTCRtpTransceiver>& transceiver : mTransceivers) {
+    if (!transceiver->IsVideo()) {
+      nsresult rv = transceiver->SyncWithMatchingVideoConduits(mTransceivers);
+      if (NS_FAILED(rv)) {
+        return rv;
+      }
+    }
+
+    transceiver->UpdatePrincipalPrivacy(PrivacyRequested()
+                                            ? PrincipalPrivacy::Private
+                                            : PrincipalPrivacy::NonPrivate);
+
+    nsresult rv = transceiver->UpdateConduit();
+    if (NS_FAILED(rv)) {
+      return rv;
+    }
+  }
+
+  return NS_OK;
+}
+
+void PeerConnectionImpl::StartIceChecks(const JsepSession& aSession) {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  if (!mCanRegisterMDNSHostnamesDirectly) {
+    for (auto& pair : mMDNSHostnamesToRegister) {
+      mRegisteredMDNSHostnames.insert(pair.first);
+      mStunAddrsRequest->SendRegisterMDNSHostname(
+          nsCString(pair.first.c_str()), nsCString(pair.second.c_str()));
+    }
+    mMDNSHostnamesToRegister.clear();
+    mCanRegisterMDNSHostnamesDirectly = true;
+  }
+
+  std::vector<std::string> attributes;
+  if (aSession.RemoteIsIceLite()) {
+    attributes.push_back("ice-lite");
+  }
+
+  if (!aSession.GetIceOptions().empty()) {
+    attributes.push_back("ice-options:");
+    for (const auto& option : aSession.GetIceOptions()) {
+      attributes.back() += option + ' ';
+    }
+  }
+
+  nsCOMPtr<nsIRunnable> runnable(
+      WrapRunnable(mTransportHandler, &MediaTransportHandler::StartIceChecks,
+                   aSession.IsIceControlling(), attributes));
+
+  PerformOrEnqueueIceCtxOperation(runnable);
+}
+
+bool PeerConnectionImpl::GetPrefDefaultAddressOnly() const {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  uint64_t winId = mWindow->WindowID();
+
+  bool default_address_only = Preferences::GetBool(
+      "media.peerconnection.ice.default_address_only", false);
+  default_address_only |=
+      !MediaManager::Get()->IsActivelyCapturingOrHasAPermission(winId);
+  return default_address_only;
+}
+
+bool PeerConnectionImpl::GetPrefObfuscateHostAddresses() const {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  uint64_t winId = mWindow->WindowID();
+
+  bool obfuscate_host_addresses = Preferences::GetBool(
+      "media.peerconnection.ice.obfuscate_host_addresses", false);
+  obfuscate_host_addresses &=
+      !MediaManager::Get()->IsActivelyCapturingOrHasAPermission(winId);
+  obfuscate_host_addresses &= !PeerConnectionImpl::HostnameInPref(
+      "media.peerconnection.ice.obfuscate_host_addresses.blocklist", mHostname);
+  obfuscate_host_addresses &= XRE_IsContentProcess();
+
+  return obfuscate_host_addresses;
+}
+
+PeerConnectionImpl::SignalHandler::SignalHandler(PeerConnectionImpl* aPc,
+                                                 MediaTransportHandler* aSource)
+    : mHandle(aPc->GetHandle()),
+      mSource(aSource),
+      mSTSThread(aPc->GetSTSThread()) {
+  ConnectSignals();
+}
+
+PeerConnectionImpl::SignalHandler::~SignalHandler() {
+  ASSERT_ON_THREAD(mSTSThread);
+}
+
+void PeerConnectionImpl::SignalHandler::ConnectSignals() {
+  mSource->SignalGatheringStateChange.connect(
+      this, &PeerConnectionImpl::SignalHandler::IceGatheringStateChange_s);
+  mSource->SignalConnectionStateChange.connect(
+      this, &PeerConnectionImpl::SignalHandler::IceConnectionStateChange_s);
+  mSource->SignalCandidate.connect(
+      this, &PeerConnectionImpl::SignalHandler::OnCandidateFound_s);
+  mSource->SignalAlpnNegotiated.connect(
+      this, &PeerConnectionImpl::SignalHandler::AlpnNegotiated_s);
+  mSource->SignalStateChange.connect(
+      this, &PeerConnectionImpl::SignalHandler::ConnectionStateChange_s);
+  mSource->SignalRtcpStateChange.connect(
+      this, &PeerConnectionImpl::SignalHandler::ConnectionStateChange_s);
+}
+
+void PeerConnectionImpl::AddIceCandidate(const std::string& aCandidate,
+                                         const std::string& aTransportId,
+                                         const std::string& aUfrag) {
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_ASSERT(!aTransportId.empty());
+
+  bool obfuscate_host_addresses = Preferences::GetBool(
+      "media.peerconnection.ice.obfuscate_host_addresses", false);
+
+  if (obfuscate_host_addresses && !RelayOnly()) {
+    std::vector<std::string> tokens;
+    TokenizeCandidate(aCandidate, tokens);
+
+    if (tokens.size() > 4) {
+      std::string addr = tokens[4];
+
+      // Check for address ending with .local
+      size_t nPeriods = std::count(addr.begin(), addr.end(), '.');
+      size_t dotLocalLength = 6;  // length of ".local"
+
+      if (nPeriods == 1 &&
+          addr.rfind(".local") + dotLocalLength == addr.length()) {
+        if (mStunAddrsRequest) {
+          PendingIceCandidate cand;
+          cand.mTokenizedCandidate = std::move(tokens);
+          cand.mTransportId = aTransportId;
+          cand.mUfrag = aUfrag;
+          mQueriedMDNSHostnames[addr].push_back(cand);
+
+          GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction(
+              "PeerConnectionImpl::SendQueryMDNSHostname",
+              [self = RefPtr<PeerConnectionImpl>(this), addr]() mutable {
+                if (self->mStunAddrsRequest) {
+                  self->StampTimecard("Look up mDNS name");
+                  self->mStunAddrsRequest->SendQueryMDNSHostname(
+                      nsCString(nsAutoCString(addr.c_str())));
+                }
+                NS_ReleaseOnMainThread(
+                    "PeerConnectionImpl::SendQueryMDNSHostname", self.forget());
+              }));
+        }
+        // TODO: Bug 1535690, we don't want to tell the ICE context that remote
+        // trickle is done if we are waiting to resolve a mDNS candidate.
+        return;
+      }
+    }
+  }
+
+  mTransportHandler->AddIceCandidate(aTransportId, aCandidate, aUfrag, "");
+}
+
+void PeerConnectionImpl::UpdateNetworkState(bool online) {
+  if (mTransportHandler) {
+    mTransportHandler->UpdateNetworkState(online);
+  }
+}
+
+void PeerConnectionImpl::FlushIceCtxOperationQueueIfReady() {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  if (IsIceCtxReady()) {
+    for (auto& queuedIceCtxOperation : mQueuedIceCtxOperations) {
+      queuedIceCtxOperation->Run();
+    }
+    mQueuedIceCtxOperations.clear();
+  }
+}
+
+void PeerConnectionImpl::PerformOrEnqueueIceCtxOperation(
+    nsIRunnable* runnable) {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  if (IsIceCtxReady()) {
+    runnable->Run();
+  } else {
+    mQueuedIceCtxOperations.push_back(runnable);
+  }
+}
+
+void PeerConnectionImpl::GatherIfReady() {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  // Init local addrs here so that if we re-gather after an ICE restart
+  // resulting from changing WiFi networks, we get new local addrs.
+  // Otherwise, we would reuse the addrs from the original WiFi network
+  // and the ICE restart will fail.
+  if (!mStunAddrs.Length()) {
+    InitLocalAddrs();
+  }
+
+  // If we had previously queued gathering or ICE start, unqueue them
+  mQueuedIceCtxOperations.clear();
+  nsCOMPtr<nsIRunnable> runnable(WrapRunnable(
+      RefPtr<PeerConnectionImpl>(this), &PeerConnectionImpl::EnsureIceGathering,
+      GetPrefDefaultAddressOnly(), GetPrefObfuscateHostAddresses()));
+
+  PerformOrEnqueueIceCtxOperation(runnable);
+}
+
+already_AddRefed<nsIHttpChannelInternal> PeerConnectionImpl::GetChannel()
+    const {
+  Document* doc = mWindow->GetExtantDoc();
+  if (NS_WARN_IF(!doc)) {
+    NS_WARNING("Unable to get document from window");
+    return nullptr;
+  }
+
+  if (!doc->GetDocumentURI()->SchemeIs("file")) {
+    nsIChannel* channel = doc->GetChannel();
+    if (!channel) {
+      NS_WARNING("Unable to get channel from document");
+      return nullptr;
+    }
+
+    nsCOMPtr<nsIHttpChannelInternal> httpChannelInternal =
+        do_QueryInterface(channel);
+    if (NS_WARN_IF(!httpChannelInternal)) {
+      CSFLogInfo(LOGTAG, "%s: Document does not have an HTTP channel",
+                 __FUNCTION__);
+      return nullptr;
+    }
+    return httpChannelInternal.forget();
+  }
+  return nullptr;
+}
+
+nsresult PeerConnectionImpl::SetTargetForDefaultLocalAddressLookup() {
+  nsCOMPtr<nsIHttpChannelInternal> httpChannelInternal = GetChannel();
+  if (!httpChannelInternal) {
+    return NS_OK;
+  }
+
+  nsCString remoteIp;
+  nsresult rv = httpChannelInternal->GetRemoteAddress(remoteIp);
+  if (NS_FAILED(rv) || remoteIp.IsEmpty()) {
+    CSFLogError(LOGTAG, "%s: Failed to get remote IP address: %d", __FUNCTION__,
+                (int)rv);
+    return rv;
+  }
+
+  int32_t remotePort;
+  rv = httpChannelInternal->GetRemotePort(&remotePort);
+  if (NS_FAILED(rv)) {
+    CSFLogError(LOGTAG, "%s: Failed to get remote port number: %d",
+                __FUNCTION__, (int)rv);
+    return rv;
+  }
+
+  mTransportHandler->SetTargetForDefaultLocalAddressLookup(remoteIp.get(),
+                                                           remotePort);
+
+  return NS_OK;
+}
+
+void PeerConnectionImpl::EnsureIceGathering(bool aDefaultRouteOnly,
+                                            bool aObfuscateHostAddresses) {
+  if (!mTargetForDefaultLocalAddressLookupIsSet) {
+    nsresult rv = SetTargetForDefaultLocalAddressLookup();
+    if (NS_FAILED(rv)) {
+      NS_WARNING("Unable to set target for default local address lookup");
+    }
+    mTargetForDefaultLocalAddressLookupIsSet = true;
+  }
+
+  // Make sure we don't call StartIceGathering if we're in e10s mode
+  // and we received no STUN addresses from the parent process.  In the
+  // absence of previously provided STUN addresses, StartIceGathering will
+  // attempt to gather them (as in non-e10s mode), and this will cause a
+  // sandboxing exception in e10s mode.
+  if (!mStunAddrs.Length() && XRE_IsContentProcess()) {
+    CSFLogInfo(LOGTAG, "%s: No STUN addresses returned from parent process",
+               __FUNCTION__);
+    return;
+  }
+
+  mTransportHandler->StartIceGathering(aDefaultRouteOnly,
+                                       aObfuscateHostAddresses, mStunAddrs);
+}
+
+already_AddRefed<dom::RTCRtpTransceiver> PeerConnectionImpl::CreateTransceiver(
+    const std::string& aId, bool aIsVideo, const RTCRtpTransceiverInit& aInit,
+    dom::MediaStreamTrack* aSendTrack, bool aAddTrackMagic, ErrorResult& aRv) {
+  PeerConnectionCtx* ctx = PeerConnectionCtx::GetInstance();
+  if (!mCall) {
+    mCall = WebrtcCallWrapper::Create(
+        GetTimestampMaker(),
+        media::ShutdownBlockingTicket::Create(
+            u"WebrtcCallWrapper shutdown blocker"_ns,
+            NS_LITERAL_STRING_FROM_CSTRING(__FILE__), __LINE__),
+        ctx->GetSharedWebrtcState());
+  }
+
+  if (aAddTrackMagic) {
+    mJsepSession->ApplyToTransceiver(aId, [](JsepTransceiver& aTransceiver) {
+      aTransceiver.SetAddTrackMagic();
+    });
+  }
+
+  RefPtr<RTCRtpTransceiver> transceiver = new RTCRtpTransceiver(
+      mWindow, PrivacyRequested(), this, mTransportHandler, mJsepSession.get(),
+      aId, aIsVideo, mSTSThread.get(), aSendTrack, mCall.get(), mIdGenerator);
+
+  transceiver->Init(aInit, aRv);
+  if (aRv.Failed()) {
+    return nullptr;
+  }
+
+  if (aSendTrack) {
+    // implement checking for peerIdentity (where failure == black/silence)
+    Document* doc = mWindow->GetExtantDoc();
+    if (doc) {
+      transceiver->Sender()->GetPipeline()->UpdateSinkIdentity(
+          doc->NodePrincipal(), GetPeerIdentity());
+    } else {
+      MOZ_CRASH();
+      aRv = NS_ERROR_FAILURE;
+      return nullptr;  // Don't remove this till we know it's safe.
+    }
+  }
+
+  return transceiver.forget();
+}
+
+std::string PeerConnectionImpl::GetTransportIdMatchingSendTrack(
+    const dom::MediaStreamTrack& aTrack) const {
+  for (const RefPtr<RTCRtpTransceiver>& transceiver : mTransceivers) {
+    if (transceiver->Sender()->HasTrack(&aTrack)) {
+      return transceiver->GetTransportId();
+    }
+  }
+  return std::string();
+}
+
+void PeerConnectionImpl::SignalHandler::IceGatheringStateChange_s(
+    dom::RTCIceGatheringState aState) {
+  ASSERT_ON_THREAD(mSTSThread);
+
+  GetMainThreadSerialEventTarget()->Dispatch(
+      NS_NewRunnableFunction(__func__,
+                             [handle = mHandle, aState] {
+                               PeerConnectionWrapper wrapper(handle);
+                               if (wrapper.impl()) {
+                                 wrapper.impl()->IceGatheringStateChange(
+                                     aState);
+                               }
+                             }),
+      NS_DISPATCH_NORMAL);
+}
+
+void PeerConnectionImpl::SignalHandler::IceConnectionStateChange_s(
+    dom::RTCIceConnectionState aState) {
+  ASSERT_ON_THREAD(mSTSThread);
+
+  GetMainThreadSerialEventTarget()->Dispatch(
+      NS_NewRunnableFunction(__func__,
+                             [handle = mHandle, aState] {
+                               PeerConnectionWrapper wrapper(handle);
+                               if (wrapper.impl()) {
+                                 wrapper.impl()->IceConnectionStateChange(
+                                     aState);
+                               }
+                             }),
+      NS_DISPATCH_NORMAL);
+}
+
+void PeerConnectionImpl::SignalHandler::OnCandidateFound_s(
+    const std::string& aTransportId, const CandidateInfo& aCandidateInfo) {
+  ASSERT_ON_THREAD(mSTSThread);
+  CSFLogDebug(LOGTAG, "%s: %s", __FUNCTION__, aTransportId.c_str());
+
+  MOZ_ASSERT(!aCandidateInfo.mUfrag.empty());
+
+  GetMainThreadSerialEventTarget()->Dispatch(
+      NS_NewRunnableFunction(__func__,
+                             [handle = mHandle, aTransportId, aCandidateInfo] {
+                               PeerConnectionWrapper wrapper(handle);
+                               if (wrapper.impl()) {
+                                 wrapper.impl()->OnCandidateFound(
+                                     aTransportId, aCandidateInfo);
+                               }
+                             }),
+      NS_DISPATCH_NORMAL);
+}
+
+void PeerConnectionImpl::SignalHandler::AlpnNegotiated_s(
+    const std::string& aAlpn, bool aPrivacyRequested) {
+  MOZ_DIAGNOSTIC_ASSERT((aAlpn == "c-webrtc") == aPrivacyRequested);
+  GetMainThreadSerialEventTarget()->Dispatch(
+      NS_NewRunnableFunction(__func__,
+                             [handle = mHandle, aPrivacyRequested] {
+                               PeerConnectionWrapper wrapper(handle);
+                               if (wrapper.impl()) {
+                                 wrapper.impl()->OnAlpnNegotiated(
+                                     aPrivacyRequested);
+                               }
+                             }),
+      NS_DISPATCH_NORMAL);
+}
+
+void PeerConnectionImpl::SignalHandler::ConnectionStateChange_s(
+    const std::string& aTransportId, TransportLayer::State aState) {
+  GetMainThreadSerialEventTarget()->Dispatch(
+      NS_NewRunnableFunction(__func__,
+                             [handle = mHandle, aTransportId, aState] {
+                               PeerConnectionWrapper wrapper(handle);
+                               if (wrapper.impl()) {
+                                 wrapper.impl()->OnDtlsStateChange(aTransportId,
+                                                                   aState);
+                               }
+                             }),
+      NS_DISPATCH_NORMAL);
+}
+
+/**
+ * Tells you if any local track is isolated to a specific peer identity.
+ * Obviously, we want all the tracks to be isolated equally so that they can
+ * all be sent or not.  We check once when we are setting a local description
+ * and that determines if we flip the "privacy requested" bit on.  Once the bit
+ * is on, all media originating from this peer connection is isolated.
+ *
+ * @returns true if any track has a peerIdentity set on it
+ */
+bool PeerConnectionImpl::AnyLocalTrackHasPeerIdentity() const {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  for (const RefPtr<RTCRtpTransceiver>& transceiver : mTransceivers) {
+    if (transceiver->Sender()->GetTrack() &&
+        transceiver->Sender()->GetTrack()->GetPeerIdentity()) {
+      return true;
+    }
+  }
+  return false;
+}
+
+bool PeerConnectionImpl::AnyCodecHasPluginID(uint64_t aPluginID) {
+  for (RefPtr<RTCRtpTransceiver>& transceiver : mTransceivers) {
+    if (transceiver->ConduitHasPluginID(aPluginID)) {
+      return true;
+    }
+  }
+  return false;
+}
+
+std::unique_ptr<NrSocketProxyConfig> PeerConnectionImpl::GetProxyConfig()
+    const {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  if (!mForceProxy &&
+      Preferences::GetBool("media.peerconnection.disable_http_proxy", false)) {
+    return nullptr;
+  }
+
+  nsCString alpn = "webrtc,c-webrtc"_ns;
+  auto* browserChild = BrowserChild::GetFrom(mWindow);
+  if (!browserChild) {
+    // Android doesn't have browser child apparently...
+    return nullptr;
+  }
+
+  Document* doc = mWindow->GetExtantDoc();
+  if (NS_WARN_IF(!doc)) {
+    NS_WARNING("Unable to get document from window");
+    return nullptr;
+  }
+
+  TabId id = browserChild->GetTabId();
+  nsCOMPtr<nsILoadInfo> loadInfo =
+      new net::LoadInfo(doc->NodePrincipal(), doc->NodePrincipal(), doc, 0,
+                        nsIContentPolicy::TYPE_INVALID);
+
+  Maybe<net::LoadInfoArgs> loadInfoArgs;
+  MOZ_ALWAYS_SUCCEEDS(
+      mozilla::ipc::LoadInfoToLoadInfoArgs(loadInfo, &loadInfoArgs));
+  return std::unique_ptr<NrSocketProxyConfig>(new NrSocketProxyConfig(
+      net::WebrtcProxyConfig(id, alpn, *loadInfoArgs, mForceProxy)));
+}
+
+std::map<uint64_t, PeerConnectionAutoTimer>
+    PeerConnectionImpl::sCallDurationTimers;
+}  // namespace mozilla
diff --git a/dom/media/webrtc/jsapi/PeerConnectionImpl.h b/dom/media/webrtc/jsapi/PeerConnectionImpl.h
new file mode 100644
index 0000000000..66af1aa0e9
--- /dev/null
+++ b/dom/media/webrtc/jsapi/PeerConnectionImpl.h
@@ -0,0 +1,969 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _PEER_CONNECTION_IMPL_H_
+#define _PEER_CONNECTION_IMPL_H_
+
+#include <string>
+#include <vector>
+#include <map>
+#include <cmath>
+
+#include "prlock.h"
+#include "mozilla/RefPtr.h"
+#include "nsComponentManagerUtils.h"
+#include "nsPIDOMWindow.h"
+#include "nsIUUIDGenerator.h"
+#include "nsIThread.h"
+#include "mozilla/Mutex.h"
+#include "mozilla/Attributes.h"
+
+// Work around nasty macro in webrtc/voice_engine/voice_engine_defines.h
+#ifdef GetLastError
+#  undef GetLastError
+#endif
+
+#include "jsep/JsepSession.h"
+#include "jsep/JsepSessionImpl.h"
+#include "sdp/SdpMediaSection.h"
+
+#include "mozilla/ErrorResult.h"
+#include "jsapi/PacketDumper.h"
+#include "mozilla/dom/RTCPeerConnectionBinding.h"  // mozPacketDumpType, maybe move?
+#include "mozilla/dom/PeerConnectionImplBinding.h"  // ChainedOperation
+#include "mozilla/dom/RTCRtpCapabilitiesBinding.h"
+#include "mozilla/dom/RTCRtpTransceiverBinding.h"
+#include "mozilla/dom/RTCConfigurationBinding.h"
+#include "PrincipalChangeObserver.h"
+#include "mozilla/dom/PromiseNativeHandler.h"
+
+#include "mozilla/TimeStamp.h"
+#include "mozilla/net/DataChannel.h"
+#include "VideoUtils.h"
+#include "VideoSegment.h"
+#include "mozilla/dom/RTCStatsReportBinding.h"
+#include "mozilla/PeerIdentity.h"
+#include "RTCStatsIdGenerator.h"
+#include "RTCStatsReport.h"
+
+#include "mozilla/net/StunAddrsRequestChild.h"
+#include "MediaTransportHandler.h"
+#include "nsIHttpChannelInternal.h"
+#include "RTCDtlsTransport.h"
+#include "RTCRtpTransceiver.h"
+
+namespace test {
+#ifdef USE_FAKE_PCOBSERVER
+class AFakePCObserver;
+#endif
+}  // namespace test
+
+class nsDOMDataChannel;
+class nsIPrincipal;
+
+namespace mozilla {
+struct CandidateInfo;
+class DataChannel;
+class DtlsIdentity;
+class MediaPipeline;
+class MediaPipelineReceive;
+class MediaPipelineTransmit;
+enum class PrincipalPrivacy : uint8_t;
+class SharedWebrtcState;
+
+namespace dom {
+class RTCCertificate;
+struct RTCConfiguration;
+struct RTCRtpSourceEntry;
+struct RTCIceServer;
+struct RTCOfferOptions;
+struct RTCRtpParameters;
+class RTCRtpSender;
+class MediaStreamTrack;
+
+#ifdef USE_FAKE_PCOBSERVER
+typedef test::AFakePCObserver PeerConnectionObserver;
+typedef const char* PCObserverString;
+#else
+class PeerConnectionObserver;
+typedef NS_ConvertUTF8toUTF16 PCObserverString;
+#endif
+}  // namespace dom
+}  // namespace mozilla
+
+#if defined(__cplusplus) && __cplusplus >= 201103L
+typedef struct Timecard Timecard;
+#else
+#  include "common/time_profiling/timecard.h"
+#endif
+
+// To preserve blame, convert nsresult to ErrorResult with wrappers. These
+// macros help declare wrappers w/function being wrapped when there are no
+// differences.
+
+#define NS_IMETHODIMP_TO_ERRORRESULT(func, rv, ...) \
+  NS_IMETHODIMP func(__VA_ARGS__);                  \
+  void func(__VA_ARGS__, rv)
+
+#define NS_IMETHODIMP_TO_ERRORRESULT_RETREF(resulttype, func, rv, ...) \
+  NS_IMETHODIMP func(__VA_ARGS__, resulttype** result);                \
+  already_AddRefed<resulttype> func(__VA_ARGS__, rv)
+
+namespace mozilla {
+
+using mozilla::DtlsIdentity;
+using mozilla::ErrorResult;
+using mozilla::PeerIdentity;
+using mozilla::dom::PeerConnectionObserver;
+using mozilla::dom::RTCConfiguration;
+using mozilla::dom::RTCIceServer;
+using mozilla::dom::RTCOfferOptions;
+
+class PeerConnectionWrapper;
+class RemoteSourceStreamInfo;
+
+// Uuid Generator
+class PCUuidGenerator : public mozilla::JsepUuidGenerator {
+ public:
+  virtual bool Generate(std::string* idp) override;
+  virtual mozilla::JsepUuidGenerator* Clone() const override {
+    return new PCUuidGenerator(*this);
+  }
+
+ private:
+  nsCOMPtr<nsIUUIDGenerator> mGenerator;
+};
+
+// This is a variation of Telemetry::AutoTimer that keeps a reference
+// count and records the elapsed time when the count falls to zero. The
+// elapsed time is recorded in seconds.
+struct PeerConnectionAutoTimer {
+  PeerConnectionAutoTimer()
+      : mRefCnt(0), mStart(TimeStamp::Now()), mUsedAV(false){};
+  void RegisterConnection();
+  void UnregisterConnection(bool aContainedAV);
+  bool IsStopped();
+
+ private:
+  int64_t mRefCnt;
+  TimeStamp mStart;
+  bool mUsedAV;
+};
+
+// Enter an API call and check that the state is OK,
+// the PC isn't closed, etc.
+#define PC_AUTO_ENTER_API_CALL(assert_ice_ready)             \
+  do {                                                       \
+    /* do/while prevents res from conflicting with locals */ \
+    nsresult res = CheckApiState(assert_ice_ready);          \
+    if (NS_FAILED(res)) return res;                          \
+  } while (0)
+#define PC_AUTO_ENTER_API_CALL_VOID_RETURN(assert_ice_ready) \
+  do {                                                       \
+    /* do/while prevents res from conflicting with locals */ \
+    nsresult res = CheckApiState(assert_ice_ready);          \
+    if (NS_FAILED(res)) return;                              \
+  } while (0)
+#define PC_AUTO_ENTER_API_CALL_NO_CHECK() CheckThread()
+
+class PeerConnectionImpl final
+    : public nsISupports,
+      public nsWrapperCache,
+      public mozilla::DataChannelConnection::DataConnectionListener {
+  struct Internal;  // Avoid exposing c includes to bindings
+
+ public:
+  explicit PeerConnectionImpl(
+      const mozilla::dom::GlobalObject* aGlobal = nullptr);
+
+  NS_DECL_CYCLE_COLLECTING_ISUPPORTS
+  NS_DECL_CYCLE_COLLECTION_WRAPPERCACHE_CLASS(PeerConnectionImpl)
+
+  struct RtpExtensionHeader {
+    JsepMediaType mMediaType;
+    SdpDirectionAttribute::Direction direction;
+    std::string extensionname;
+  };
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+  nsPIDOMWindowInner* GetParentObject() const;
+
+  static already_AddRefed<PeerConnectionImpl> Constructor(
+      const mozilla::dom::GlobalObject& aGlobal);
+
+  // DataConnection observers
+  void NotifyDataChannel(already_AddRefed<mozilla::DataChannel> aChannel)
+      // PeerConnectionImpl only inherits from mozilla::DataChannelConnection
+      // inside libxul.
+      override;
+
+  void NotifyDataChannelOpen(DataChannel*) override;
+
+  void NotifyDataChannelClosed(DataChannel*) override;
+
+  void NotifySctpConnected() override;
+
+  void NotifySctpClosed() override;
+
+  const RefPtr<MediaTransportHandler> GetTransportHandler() const;
+
+  // Handle system to allow weak references to be passed through C code
+  virtual const std::string& GetHandle();
+
+  // Name suitable for exposing to content
+  virtual const std::string& GetName();
+
+  // ICE events
+  void IceConnectionStateChange(dom::RTCIceConnectionState state);
+  void IceGatheringStateChange(dom::RTCIceGatheringState state);
+  void OnCandidateFound(const std::string& aTransportId,
+                        const CandidateInfo& aCandidateInfo);
+  void UpdateDefaultCandidate(const std::string& defaultAddr,
+                              uint16_t defaultPort,
+                              const std::string& defaultRtcpAddr,
+                              uint16_t defaultRtcpPort,
+                              const std::string& transportId);
+
+  static void ListenThread(void* aData);
+  static void ConnectThread(void* aData);
+
+  // Get the STS thread
+  nsISerialEventTarget* GetSTSThread() {
+    PC_AUTO_ENTER_API_CALL_NO_CHECK();
+    return mSTSThread;
+  }
+
+  nsresult Initialize(PeerConnectionObserver& aObserver,
+                      nsGlobalWindowInner* aWindow);
+
+  // Initialize PeerConnection from an RTCConfiguration object (JS entrypoint)
+  void Initialize(PeerConnectionObserver& aObserver,
+                  nsGlobalWindowInner& aWindow, ErrorResult& rv);
+
+  void SetCertificate(mozilla::dom::RTCCertificate& aCertificate);
+  const RefPtr<mozilla::dom::RTCCertificate>& Certificate() const;
+  // This is a hack to support external linkage.
+  RefPtr<DtlsIdentity> Identity() const;
+
+  NS_IMETHODIMP_TO_ERRORRESULT(CreateOffer, ErrorResult& rv,
+                               const RTCOfferOptions& aOptions) {
+    rv = CreateOffer(aOptions);
+  }
+
+  NS_IMETHODIMP CreateAnswer();
+  void CreateAnswer(ErrorResult& rv) { rv = CreateAnswer(); }
+
+  NS_IMETHODIMP CreateOffer(const mozilla::JsepOfferOptions& aConstraints);
+
+  NS_IMETHODIMP SetLocalDescription(int32_t aAction, const char* aSDP);
+
+  void SetLocalDescription(int32_t aAction, const nsAString& aSDP,
+                           ErrorResult& rv) {
+    rv = SetLocalDescription(aAction, NS_ConvertUTF16toUTF8(aSDP).get());
+  }
+
+  NS_IMETHODIMP SetRemoteDescription(int32_t aAction, const char* aSDP);
+
+  void SetRemoteDescription(int32_t aAction, const nsAString& aSDP,
+                            ErrorResult& rv) {
+    rv = SetRemoteDescription(aAction, NS_ConvertUTF16toUTF8(aSDP).get());
+  }
+
+  already_AddRefed<dom::Promise> GetStats(dom::MediaStreamTrack* aSelector);
+
+  void GetRemoteStreams(nsTArray<RefPtr<DOMMediaStream>>& aStreamsOut) const;
+
+  NS_IMETHODIMP AddIceCandidate(const char* aCandidate, const char* aMid,
+                                const char* aUfrag,
+                                const dom::Nullable<unsigned short>& aLevel);
+
+  void AddIceCandidate(const nsAString& aCandidate, const nsAString& aMid,
+                       const nsAString& aUfrag,
+                       const dom::Nullable<unsigned short>& aLevel,
+                       ErrorResult& rv) {
+    rv = AddIceCandidate(NS_ConvertUTF16toUTF8(aCandidate).get(),
+                         NS_ConvertUTF16toUTF8(aMid).get(),
+                         NS_ConvertUTF16toUTF8(aUfrag).get(), aLevel);
+  }
+
+  void UpdateNetworkState(bool online);
+
+  NS_IMETHODIMP CloseStreams();
+
+  void CloseStreams(ErrorResult& rv) { rv = CloseStreams(); }
+
+  already_AddRefed<dom::RTCRtpTransceiver> AddTransceiver(
+      const dom::RTCRtpTransceiverInit& aInit, const nsAString& aKind,
+      dom::MediaStreamTrack* aSendTrack, bool aAddTrackMagic, ErrorResult& aRv);
+
+  bool CheckNegotiationNeeded();
+  bool CreatedSender(const dom::RTCRtpSender& aSender) const;
+
+  // test-only
+  NS_IMETHODIMP_TO_ERRORRESULT(EnablePacketDump, ErrorResult& rv,
+                               unsigned long level, dom::mozPacketDumpType type,
+                               bool sending) {
+    rv = EnablePacketDump(level, type, sending);
+  }
+
+  // test-only
+  NS_IMETHODIMP_TO_ERRORRESULT(DisablePacketDump, ErrorResult& rv,
+                               unsigned long level, dom::mozPacketDumpType type,
+                               bool sending) {
+    rv = DisablePacketDump(level, type, sending);
+  }
+
+  void GetPeerIdentity(nsAString& peerIdentity) {
+    if (mPeerIdentity) {
+      peerIdentity = mPeerIdentity->ToString();
+      return;
+    }
+
+    peerIdentity.SetIsVoid(true);
+  }
+
+  const PeerIdentity* GetPeerIdentity() const { return mPeerIdentity; }
+  NS_IMETHODIMP_TO_ERRORRESULT(SetPeerIdentity, ErrorResult& rv,
+                               const nsAString& peerIdentity) {
+    rv = SetPeerIdentity(peerIdentity);
+  }
+
+  const std::string& GetIdAsAscii() const { return mName; }
+
+  void GetId(nsAString& id) { id = NS_ConvertASCIItoUTF16(mName.c_str()); }
+
+  void SetId(const nsAString& id) { mName = NS_ConvertUTF16toUTF8(id).get(); }
+
+  // this method checks to see if we've made a promise to protect media.
+  bool PrivacyRequested() const {
+    return mRequestedPrivacy.valueOr(PrincipalPrivacy::NonPrivate) ==
+           PrincipalPrivacy::Private;
+  }
+
+  NS_IMETHODIMP GetFingerprint(char** fingerprint);
+  void GetFingerprint(nsAString& fingerprint) {
+    char* tmp;
+    nsresult rv = GetFingerprint(&tmp);
+    NS_ENSURE_SUCCESS_VOID(rv);
+    fingerprint.AssignASCII(tmp);
+    delete[] tmp;
+  }
+
+  void GetCurrentLocalDescription(nsAString& aSDP) const;
+  void GetPendingLocalDescription(nsAString& aSDP) const;
+
+  void GetCurrentRemoteDescription(nsAString& aSDP) const;
+  void GetPendingRemoteDescription(nsAString& aSDP) const;
+
+  dom::Nullable<bool> GetCurrentOfferer() const;
+  dom::Nullable<bool> GetPendingOfferer() const;
+
+  NS_IMETHODIMP SignalingState(mozilla::dom::RTCSignalingState* aState);
+
+  mozilla::dom::RTCSignalingState SignalingState() {
+    mozilla::dom::RTCSignalingState state;
+    SignalingState(&state);
+    return state;
+  }
+
+  NS_IMETHODIMP IceConnectionState(mozilla::dom::RTCIceConnectionState* aState);
+
+  mozilla::dom::RTCIceConnectionState IceConnectionState() {
+    mozilla::dom::RTCIceConnectionState state;
+    IceConnectionState(&state);
+    return state;
+  }
+
+  NS_IMETHODIMP IceGatheringState(mozilla::dom::RTCIceGatheringState* aState);
+
+  mozilla::dom::RTCIceGatheringState IceGatheringState() {
+    return mIceGatheringState;
+  }
+
+  NS_IMETHODIMP ConnectionState(mozilla::dom::RTCPeerConnectionState* aState);
+
+  mozilla::dom::RTCPeerConnectionState ConnectionState() {
+    mozilla::dom::RTCPeerConnectionState state;
+    ConnectionState(&state);
+    return state;
+  }
+
+  NS_IMETHODIMP Close();
+
+  void Close(ErrorResult& rv) { rv = Close(); }
+
+  // TODO: Convert this to MOZ_CAN_RUN_SCRIPT (bug 1415230)
+  MOZ_CAN_RUN_SCRIPT_BOUNDARY bool PluginCrash(uint32_t aPluginID,
+                                               const nsAString& aPluginName);
+
+  NS_IMETHODIMP_TO_ERRORRESULT(SetConfiguration, ErrorResult& rv,
+                               const RTCConfiguration& aConfiguration) {
+    rv = SetConfiguration(aConfiguration);
+  }
+
+  dom::RTCSctpTransport* GetSctp() const;
+
+  void RestartIce();
+  void RestartIceNoRenegotiationNeeded();
+
+  void RecordEndOfCallTelemetry();
+
+  nsresult InitializeDataChannel();
+
+  NS_IMETHODIMP_TO_ERRORRESULT_RETREF(nsDOMDataChannel, CreateDataChannel,
+                                      ErrorResult& rv, const nsAString& aLabel,
+                                      const nsAString& aProtocol,
+                                      uint16_t aType, bool outOfOrderAllowed,
+                                      uint16_t aMaxTime, uint16_t aMaxNum,
+                                      bool aExternalNegotiated,
+                                      uint16_t aStream);
+
+  // Base class for chained operations. Necessary right now because some
+  // operations come from JS (in the form of dom::ChainedOperation), and others
+  // come from c++ (dom::ChainedOperation is very unwieldy and arcane to build
+  // in c++). Once we stop using JSImpl, we should be able to simplify this.
+  class Operation : public dom::PromiseNativeHandler {
+   public:
+    NS_DECL_CYCLE_COLLECTING_ISUPPORTS
+    NS_DECL_CYCLE_COLLECTION_CLASS(Operation)
+    Operation(PeerConnectionImpl* aPc, ErrorResult& aError);
+    MOZ_CAN_RUN_SCRIPT
+    void Call(ErrorResult& aError);
+    dom::Promise* GetPromise() { return mPromise; }
+    MOZ_CAN_RUN_SCRIPT
+    void ResolvedCallback(JSContext* aCx, JS::Handle<JS::Value> aValue,
+                          ErrorResult& aRv) override;
+
+    MOZ_CAN_RUN_SCRIPT
+    void RejectedCallback(JSContext* aCx, JS::Handle<JS::Value> aValue,
+                          ErrorResult& aRv) override;
+
+   protected:
+    MOZ_CAN_RUN_SCRIPT
+    virtual RefPtr<dom::Promise> CallImpl(ErrorResult& aError) = 0;
+    virtual ~Operation();
+    // This is the promise p from https://w3c.github.io/webrtc-pc/#dfn-chain
+    // This will be a content promise, since we return this to the caller of
+    // Chain.
+    RefPtr<dom::Promise> mPromise;
+    RefPtr<PeerConnectionImpl> mPc;
+  };
+
+  class JSOperation final : public Operation {
+   public:
+    JSOperation(PeerConnectionImpl* aPc, dom::ChainedOperation& aOp,
+                ErrorResult& aError);
+    NS_DECL_ISUPPORTS_INHERITED
+    NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(JSOperation, Operation)
+
+   private:
+    MOZ_CAN_RUN_SCRIPT
+    RefPtr<dom::Promise> CallImpl(ErrorResult& aError) override;
+    ~JSOperation() = default;
+    RefPtr<dom::ChainedOperation> mOperation;
+  };
+
+  MOZ_CAN_RUN_SCRIPT
+  already_AddRefed<dom::Promise> Chain(dom::ChainedOperation& aOperation,
+                                       ErrorResult& aError);
+  MOZ_CAN_RUN_SCRIPT
+  already_AddRefed<dom::Promise> Chain(const RefPtr<Operation>& aOperation,
+                                       ErrorResult& aError);
+  already_AddRefed<dom::Promise> MakePromise(ErrorResult& aError) const;
+
+  void UpdateNegotiationNeeded();
+
+  void GetTransceivers(
+      nsTArray<RefPtr<dom::RTCRtpTransceiver>>& aTransceiversOut) {
+    aTransceiversOut = mTransceivers.Clone();
+  }
+
+  // Gets the RTC Signaling State of the JSEP session
+  dom::RTCSignalingState GetSignalingState() const;
+
+  already_AddRefed<dom::Promise> OnSetDescriptionSuccess(
+      dom::RTCSdpType aSdpType, bool aRemote, ErrorResult& aError);
+
+  void OnSetDescriptionError();
+
+  bool IsClosed() const;
+
+  // called when DTLS connects; we only need this once
+  nsresult OnAlpnNegotiated(bool aPrivacyRequested);
+
+  void OnDtlsStateChange(const std::string& aTransportId,
+                         TransportLayer::State aState);
+  void UpdateConnectionState();
+  dom::RTCPeerConnectionState GetNewConnectionState() const;
+
+  // initialize telemetry for when calls start
+  void StartCallTelem();
+
+  // Gets all codec stats for all transports, coalesced to transport level.
+  nsTArray<dom::RTCCodecStats> GetCodecStats(DOMHighResTimeStamp aNow);
+
+  RefPtr<dom::RTCStatsReportPromise> GetStats(dom::MediaStreamTrack* aSelector,
+                                              bool aInternalStats);
+
+  void CollectConduitTelemetryData();
+
+  void OnMediaError(const std::string& aError);
+
+  void DumpPacket_m(size_t level, dom::mozPacketDumpType type, bool sending,
+                    UniquePtr<uint8_t[]>& packet, size_t size);
+
+  const dom::RTCStatsTimestampMaker& GetTimestampMaker() const {
+    return mTimestampMaker;
+  }
+
+  // Utility function, given a string pref and an URI, returns whether or not
+  // the URI occurs in the pref. Wildcards are supported (e.g. *.example.com)
+  // and multiple hostnames can be present, separated by commas.
+  static bool HostnameInPref(const char* aPrefList, const nsCString& aHostName);
+
+  void StampTimecard(const char* aEvent);
+
+  bool RelayOnly() const {
+    return mJsConfiguration.mIceTransportPolicy.WasPassed() &&
+           mJsConfiguration.mIceTransportPolicy.Value() ==
+               dom::RTCIceTransportPolicy::Relay;
+  }
+
+  RefPtr<PacketDumper> GetPacketDumper() {
+    if (!mPacketDumper) {
+      mPacketDumper = new PacketDumper(mHandle);
+    }
+
+    return mPacketDumper;
+  }
+
+  nsString GenerateUUID() const {
+    std::string result;
+    if (!mUuidGen->Generate(&result)) {
+      MOZ_CRASH();
+    }
+    return NS_ConvertUTF8toUTF16(result.c_str());
+  }
+
+  bool ShouldAllowOldSetParameters() const { return mAllowOldSetParameters; }
+
+  nsCString GetHostname() const { return mHostname; }
+  nsCString GetEffectiveTLDPlus1() const { return mEffectiveTLDPlus1; }
+
+  void SendWarningToConsole(const nsCString& aWarning);
+
+  const UniquePtr<dom::RTCStatsReportInternal>& GetFinalStats() const {
+    return mFinalStats;
+  }
+
+  void DisableLongTermStats() { mDisableLongTermStats = true; }
+
+  bool LongTermStatsIsDisabled() const { return mDisableLongTermStats; }
+
+  static void GetDefaultVideoCodecs(
+      std::vector<UniquePtr<JsepCodecDescription>>& aSupportedCodecs,
+      bool aUseRtx);
+
+  static void GetDefaultAudioCodecs(
+      std::vector<UniquePtr<JsepCodecDescription>>& aSupportedCodecs);
+
+  static void GetDefaultRtpExtensions(
+      std::vector<RtpExtensionHeader>& aRtpExtensions);
+
+  static void GetCapabilities(const nsAString& aKind,
+                              dom::Nullable<dom::RTCRtpCapabilities>& aResult,
+                              sdp::Direction aDirection);
+  static void SetupPreferredCodecs(
+      std::vector<UniquePtr<JsepCodecDescription>>& aPreferredCodecs);
+
+  static void SetupPreferredRtpExtensions(
+      std::vector<RtpExtensionHeader>& aPreferredheaders);
+
+ private:
+  virtual ~PeerConnectionImpl();
+  PeerConnectionImpl(const PeerConnectionImpl& rhs);
+  PeerConnectionImpl& operator=(PeerConnectionImpl);
+
+  RefPtr<dom::RTCStatsPromise> GetDataChannelStats(
+      const RefPtr<DataChannelConnection>& aDataChannelConnection,
+      const DOMHighResTimeStamp aTimestamp);
+  nsresult CalculateFingerprint(const std::string& algorithm,
+                                std::vector<uint8_t>* fingerprint) const;
+  nsresult ConfigureJsepSessionCodecs();
+
+  NS_IMETHODIMP EnsureDataConnection(uint16_t aLocalPort, uint16_t aNumstreams,
+                                     uint32_t aMaxMessageSize, bool aMMSSet);
+
+  nsresult CheckApiState(bool assert_ice_ready) const;
+  void StoreFinalStats(UniquePtr<dom::RTCStatsReportInternal>&& report);
+  void CheckThread() const { MOZ_ASSERT(NS_IsMainThread(), "Wrong thread"); }
+
+  // test-only: called from AddRIDExtension and AddRIDFilter
+  // for simulcast mochitests.
+  RefPtr<MediaPipeline> GetMediaPipelineForTrack(
+      dom::MediaStreamTrack& aRecvTrack);
+
+  void CandidateReady(const std::string& candidate,
+                      const std::string& transportId, const std::string& ufrag);
+  void SendLocalIceCandidateToContent(uint16_t level, const std::string& mid,
+                                      const std::string& candidate,
+                                      const std::string& ufrag);
+
+  nsresult GetDatachannelParameters(uint32_t* channels, uint16_t* localport,
+                                    uint16_t* remoteport,
+                                    uint32_t* maxmessagesize, bool* mmsset,
+                                    std::string* transportId,
+                                    bool* client) const;
+
+  nsresult AddRtpTransceiverToJsepSession(JsepTransceiver& transceiver);
+
+  void RecordIceRestartStatistics(JsepSdpType type);
+
+  void StoreConfigurationForAboutWebrtc(const RTCConfiguration& aConfig);
+
+  dom::Sequence<dom::RTCSdpParsingErrorInternal> GetLastSdpParsingErrors()
+      const;
+
+  MOZ_CAN_RUN_SCRIPT
+  void RunNextOperation(ErrorResult& aError);
+
+  void SyncToJsep();
+  void SyncFromJsep();
+
+  void DoSetDescriptionSuccessPostProcessing(dom::RTCSdpType aSdpType,
+                                             bool aRemote,
+                                             const RefPtr<dom::Promise>& aP);
+
+  // Timecard used to measure processing time. This should be the first class
+  // attribute so that we accurately measure the time required to instantiate
+  // any other attributes of this class.
+  Timecard* mTimeCard;
+
+  // Configuration used to initialize the PeerConnection
+  dom::RTCConfigurationInternal mJsConfiguration;
+
+  mozilla::dom::RTCSignalingState mSignalingState;
+
+  // ICE State
+  mozilla::dom::RTCIceConnectionState mIceConnectionState;
+  mozilla::dom::RTCIceGatheringState mIceGatheringState;
+
+  mozilla::dom::RTCPeerConnectionState mConnectionState;
+
+  RefPtr<PeerConnectionObserver> mPCObserver;
+
+  nsCOMPtr<nsPIDOMWindowInner> mWindow;
+
+  // The SDP sent in from JS
+  std::string mLocalRequestedSDP;
+  std::string mRemoteRequestedSDP;
+  // Only accessed from main
+  mozilla::dom::Sequence<mozilla::dom::RTCSdpHistoryEntryInternal> mSdpHistory;
+  std::string mPendingLocalDescription;
+  std::string mPendingRemoteDescription;
+  std::string mCurrentLocalDescription;
+  std::string mCurrentRemoteDescription;
+  Maybe<bool> mPendingOfferer;
+  Maybe<bool> mCurrentOfferer;
+
+  // DTLS fingerprint
+  std::string mFingerprint;
+  std::string mRemoteFingerprint;
+
+  // identity-related fields
+  // The entity on the other end of the peer-to-peer connection;
+  // void if they are not yet identified, and no identity setting has been set
+  RefPtr<PeerIdentity> mPeerIdentity;
+  // The certificate we are using.
+  RefPtr<mozilla::dom::RTCCertificate> mCertificate;
+  // Whether an app should be prevented from accessing media produced by the PC
+  // If this is true, then media will not be sent until mPeerIdentity matches
+  // local streams PeerIdentity; and remote streams are protected from content
+  //
+  // This can be false if mPeerIdentity is set, in the case where identity is
+  // provided, but the media is not protected from the app on either side
+  Maybe<PrincipalPrivacy> mRequestedPrivacy;
+
+  // A handle to refer to this PC with
+  std::string mHandle;
+
+  // A name for this PC that we are willing to expose to content.
+  std::string mName;
+  nsCString mHostname;
+  nsCString mEffectiveTLDPlus1;
+
+  // The target to run stuff on
+  nsCOMPtr<nsISerialEventTarget> mSTSThread;
+
+  // DataConnection that's used to get all the DataChannels
+  RefPtr<mozilla::DataChannelConnection> mDataConnection;
+  unsigned int mDataChannelsOpened = 0;
+  unsigned int mDataChannelsClosed = 0;
+
+  bool mForceIceTcp;
+  RefPtr<MediaTransportHandler> mTransportHandler;
+
+  // The JSEP negotiation session.
+  mozilla::UniquePtr<PCUuidGenerator> mUuidGen;
+  mozilla::UniquePtr<mozilla::JsepSession> mJsepSession;
+  // There are lots of error cases where we want to abandon an sRD/sLD _after_
+  // it has already been applied to the JSEP engine, and revert back to the
+  // previous state. We also want to ensure that the various modifications
+  // to the JSEP engine are not exposed to JS until the sRD/sLD completes,
+  // which is why we have a new "uncommitted" JSEP engine.
+  mozilla::UniquePtr<mozilla::JsepSession> mUncommittedJsepSession;
+  unsigned long mIceRestartCount;
+  unsigned long mIceRollbackCount;
+
+  // The following are used for Telemetry:
+  bool mCallTelemStarted = false;
+  bool mCallTelemEnded = false;
+
+  // We _could_ make mFinalStatsQuery be an RTCStatsReportPromise, but that
+  // would require RTCStatsReportPromise to no longer be exclusive, which is
+  // a bit of a hassle, and not very performant.
+  RefPtr<GenericNonExclusivePromise> mFinalStatsQuery;
+  UniquePtr<dom::RTCStatsReportInternal> mFinalStats;
+  bool mDisableLongTermStats = false;
+
+  // Start time of ICE.
+  mozilla::TimeStamp mIceStartTime;
+  // Hold PeerConnectionAutoTimer instances for each window.
+  static std::map<uint64_t, PeerConnectionAutoTimer> sCallDurationTimers;
+
+  bool mHaveConfiguredCodecs;
+
+  bool mTrickle;
+
+  bool mPrivateWindow;
+
+  // Whether this PeerConnection is being counted as active by mWindow
+  bool mActiveOnWindow;
+
+  // storage for Telemetry data
+  uint16_t mMaxReceiving[SdpMediaSection::kMediaTypes];
+  uint16_t mMaxSending[SdpMediaSection::kMediaTypes];
+
+  // used to store the raw trickle candidate string for display
+  // on the about:webrtc raw candidates table.
+  std::vector<std::string> mRawTrickledCandidates;
+
+  dom::RTCStatsTimestampMaker mTimestampMaker;
+
+  RefPtr<RTCStatsIdGenerator> mIdGenerator;
+  // Ordinarily, I would use a std::map here, but this used to be a JS Map
+  // which iterates in insertion order, and I want to avoid changing this.
+  nsTArray<RefPtr<DOMMediaStream>> mReceiveStreams;
+
+  DOMMediaStream* GetReceiveStream(const std::string& aId) const;
+  DOMMediaStream* CreateReceiveStream(const std::string& aId);
+
+  void InitLocalAddrs();  // for stun local address IPC request
+  bool ShouldForceProxy() const;
+  std::unique_ptr<NrSocketProxyConfig> GetProxyConfig() const;
+
+  class StunAddrsHandler : public net::StunAddrsListener {
+   public:
+    explicit StunAddrsHandler(PeerConnectionImpl* aPc)
+        : mPcHandle(aPc->GetHandle()) {}
+
+    void OnMDNSQueryComplete(const nsCString& hostname,
+                             const Maybe<nsCString>& address) override;
+
+    void OnStunAddrsAvailable(
+        const mozilla::net::NrIceStunAddrArray& addrs) override;
+
+   private:
+    // This class is not cycle-collected, so we must avoid grabbing a strong
+    // reference.
+    const std::string mPcHandle;
+    virtual ~StunAddrsHandler() {}
+  };
+
+  // Manage ICE transports.
+  void UpdateTransport(const JsepTransceiver& aTransceiver, bool aForceIceTcp);
+
+  void GatherIfReady();
+  void FlushIceCtxOperationQueueIfReady();
+  void PerformOrEnqueueIceCtxOperation(nsIRunnable* runnable);
+  nsresult SetTargetForDefaultLocalAddressLookup();
+  void EnsureIceGathering(bool aDefaultRouteOnly, bool aObfuscateHostAddresses);
+
+  bool GetPrefDefaultAddressOnly() const;
+  bool GetPrefObfuscateHostAddresses() const;
+
+  bool IsIceCtxReady() const {
+    return mLocalAddrsRequestState == STUN_ADDR_REQUEST_COMPLETE;
+  }
+
+  // Ensure ICE transports exist that we might need when offer/answer concludes
+  void EnsureTransports(const JsepSession& aSession);
+
+  void UpdateRTCDtlsTransports(bool aMarkAsStable);
+  void RollbackRTCDtlsTransports();
+  void RemoveRTCDtlsTransportsExcept(
+      const std::set<std::string>& aTransportIds);
+
+  // Activate ICE transports at the conclusion of offer/answer,
+  // or when rollback occurs.
+  nsresult UpdateTransports(const JsepSession& aSession,
+                            const bool forceIceTcp);
+
+  void ResetStunAddrsForIceRestart() { mStunAddrs.Clear(); }
+
+  // Start ICE checks.
+  void StartIceChecks(const JsepSession& session);
+
+  // Process a trickle ICE candidate.
+  void AddIceCandidate(const std::string& candidate,
+                       const std::string& aTransportId,
+                       const std::string& aUFrag);
+
+  // Handle complete media pipelines.
+  // This updates codec parameters, starts/stops send/receive, and other
+  // stuff that doesn't necessarily require negotiation. This can be called at
+  // any time, not just when an offer/answer exchange completes.
+  nsresult UpdateMediaPipelines();
+
+  already_AddRefed<dom::RTCRtpTransceiver> CreateTransceiver(
+      const std::string& aId, bool aIsVideo,
+      const dom::RTCRtpTransceiverInit& aInit,
+      dom::MediaStreamTrack* aSendTrack, bool aAddTrackMagic, ErrorResult& aRv);
+
+  std::string GetTransportIdMatchingSendTrack(
+      const dom::MediaStreamTrack& aTrack) const;
+
+  // this determines if any track is peerIdentity constrained
+  bool AnyLocalTrackHasPeerIdentity() const;
+
+  bool AnyCodecHasPluginID(uint64_t aPluginID);
+
+  already_AddRefed<nsIHttpChannelInternal> GetChannel() const;
+
+  void BreakCycles();
+
+  bool HasPendingSetParameters() const;
+  void InvalidateLastReturnedParameters();
+
+  RefPtr<WebrtcCallWrapper> mCall;
+
+  // See Bug 1642419, this can be removed when all sites are working with RTX.
+  bool mRtxIsAllowed = true;
+
+  nsTArray<RefPtr<Operation>> mOperations;
+  bool mChainingOperation = false;
+  bool mUpdateNegotiationNeededFlagOnEmptyChain = false;
+  bool mNegotiationNeeded = false;
+  std::set<std::pair<std::string, std::string>> mLocalIceCredentialsToReplace;
+
+  nsTArray<RefPtr<dom::RTCRtpTransceiver>> mTransceivers;
+  std::map<std::string, RefPtr<dom::RTCDtlsTransport>>
+      mTransportIdToRTCDtlsTransport;
+  RefPtr<dom::RTCSctpTransport> mSctpTransport;
+
+  // Used whenever we need to dispatch a runnable to STS to tweak something
+  // on our ICE ctx, but are not ready to do so at the moment (eg; we are
+  // waiting to get a callback with our http proxy config before we start
+  // gathering or start checking)
+  std::vector<nsCOMPtr<nsIRunnable>> mQueuedIceCtxOperations;
+
+  // Set if prefs dictate that we should force the use of a web proxy.
+  bool mForceProxy = false;
+
+  // Used to cancel incoming stun addrs response
+  RefPtr<net::StunAddrsRequestChild> mStunAddrsRequest;
+
+  enum StunAddrRequestState {
+    STUN_ADDR_REQUEST_NONE,
+    STUN_ADDR_REQUEST_PENDING,
+    STUN_ADDR_REQUEST_COMPLETE
+  };
+  // Used to track the state of the stun addr IPC request
+  StunAddrRequestState mLocalAddrsRequestState = STUN_ADDR_REQUEST_NONE;
+
+  // Used to store the result of the stun addr IPC request
+  nsTArray<NrIceStunAddr> mStunAddrs;
+
+  // Used to ensure the target for default local address lookup is only set
+  // once.
+  bool mTargetForDefaultLocalAddressLookupIsSet = false;
+
+  // Keep track of local hostnames to register. Registration is deferred
+  // until StartIceChecks has run. Accessed on main thread only.
+  std::map<std::string, std::string> mMDNSHostnamesToRegister;
+  bool mCanRegisterMDNSHostnamesDirectly = false;
+
+  // Used to store the mDNS hostnames that we have registered
+  std::set<std::string> mRegisteredMDNSHostnames;
+
+  // web-compat stopgap
+  bool mAllowOldSetParameters = false;
+
+  // Used to store the mDNS hostnames that we have queried
+  struct PendingIceCandidate {
+    std::vector<std::string> mTokenizedCandidate;
+    std::string mTransportId;
+    std::string mUfrag;
+  };
+  std::map<std::string, std::list<PendingIceCandidate>> mQueriedMDNSHostnames;
+
+  // Connecting PCImpl to sigslot is not safe, because sigslot takes strong
+  // references without any reference counting, and JS holds refcounted strong
+  // references to PCImpl (meaning JS can cause PCImpl to be destroyed).  This
+  // is not ref-counted (since sigslot holds onto non-refcounted strong refs)
+  // Must be destroyed on STS. Holds a weak reference to PCImpl.
+  class SignalHandler : public sigslot::has_slots<> {
+   public:
+    SignalHandler(PeerConnectionImpl* aPc, MediaTransportHandler* aSource);
+    virtual ~SignalHandler();
+
+    void ConnectSignals();
+
+    // ICE events
+    void IceGatheringStateChange_s(dom::RTCIceGatheringState aState);
+    void IceConnectionStateChange_s(dom::RTCIceConnectionState aState);
+    void OnCandidateFound_s(const std::string& aTransportId,
+                            const CandidateInfo& aCandidateInfo);
+    void AlpnNegotiated_s(const std::string& aAlpn, bool aPrivacyRequested);
+    void ConnectionStateChange_s(const std::string& aTransportId,
+                                 TransportLayer::State aState);
+
+   private:
+    const std::string mHandle;
+    RefPtr<MediaTransportHandler> mSource;
+    RefPtr<nsISerialEventTarget> mSTSThread;
+  };
+
+  mozilla::UniquePtr<SignalHandler> mSignalHandler;
+
+  // Make absolutely sure our refcount does not go to 0 before Close() is called
+  // This is because Close does a stats query, which needs the
+  // PeerConnectionImpl to stick around until the query is done.
+  RefPtr<PeerConnectionImpl> mKungFuDeathGrip;
+  RefPtr<PacketDumper> mPacketDumper;
+
+ public:
+  // these are temporary until the DataChannel Listen/Connect API is removed
+  unsigned short listenPort;
+  unsigned short connectPort;
+  char* connectStr;  // XXX ownership/free
+};
+
+// This is what is returned when you acquire on a handle
+class PeerConnectionWrapper {
+ public:
+  explicit PeerConnectionWrapper(const std::string& handle);
+
+  PeerConnectionImpl* impl() { return impl_; }
+
+ private:
+  RefPtr<PeerConnectionImpl> impl_;
+};
+
+}  // namespace mozilla
+
+#undef NS_IMETHODIMP_TO_ERRORRESULT
+#undef NS_IMETHODIMP_TO_ERRORRESULT_RETREF
+#endif  // _PEER_CONNECTION_IMPL_H_
diff --git a/dom/media/webrtc/jsapi/RTCDTMFSender.cpp b/dom/media/webrtc/jsapi/RTCDTMFSender.cpp
new file mode 100644
index 0000000000..30355aca26
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCDTMFSender.cpp
@@ -0,0 +1,159 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "RTCDTMFSender.h"
+#include "libwebrtcglue/MediaConduitInterface.h"
+#include "transport/logging.h"
+#include "RTCRtpTransceiver.h"
+#include "nsITimer.h"
+#include "mozilla/dom/RTCDTMFSenderBinding.h"
+#include "mozilla/dom/RTCDTMFToneChangeEvent.h"
+#include <algorithm>
+#include <bitset>
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(RTCDTMFSender, DOMEventTargetHelper,
+                                   mTransceiver, mSendTimer)
+
+NS_IMPL_ADDREF_INHERITED(RTCDTMFSender, DOMEventTargetHelper)
+NS_IMPL_RELEASE_INHERITED(RTCDTMFSender, DOMEventTargetHelper)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(RTCDTMFSender)
+  NS_WRAPPERCACHE_INTERFACE_MAP_ENTRY
+  NS_INTERFACE_MAP_ENTRY(nsITimerCallback)
+  NS_INTERFACE_MAP_ENTRY(nsINamed)
+NS_INTERFACE_MAP_END_INHERITING(DOMEventTargetHelper)
+
+LazyLogModule gDtmfLog("RTCDTMFSender");
+
+RTCDTMFSender::RTCDTMFSender(nsPIDOMWindowInner* aWindow,
+                             RTCRtpTransceiver* aTransceiver)
+    : DOMEventTargetHelper(aWindow), mTransceiver(aTransceiver) {}
+
+JSObject* RTCDTMFSender::WrapObject(JSContext* aCx,
+                                    JS::Handle<JSObject*> aGivenProto) {
+  return RTCDTMFSender_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+static int GetDTMFToneCode(uint16_t c) {
+  const char* DTMF_TONECODES = "0123456789*#ABCD";
+
+  if (c == ',') {
+    // , is a special character indicating a 2 second delay
+    return -1;
+  }
+
+  const char* i = strchr(DTMF_TONECODES, c);
+  MOZ_ASSERT(i);
+  return static_cast<int>(i - DTMF_TONECODES);
+}
+
+static std::bitset<256> GetCharacterBitset(const std::string& aCharsInSet) {
+  std::bitset<256> result;
+  for (auto c : aCharsInSet) {
+    result[c] = true;
+  }
+  return result;
+}
+
+static bool IsUnrecognizedChar(const char c) {
+  static const std::bitset<256> recognized =
+      GetCharacterBitset("0123456789ABCD#*,");
+  return !recognized[c];
+}
+
+void RTCDTMFSender::SetPayloadType(int32_t aPayloadType,
+                                   int32_t aPayloadFrequency) {
+  MOZ_ASSERT(NS_IsMainThread());
+  mPayloadType = Some(aPayloadType);
+  mPayloadFrequency = Some(aPayloadFrequency);
+}
+
+void RTCDTMFSender::InsertDTMF(const nsAString& aTones, uint32_t aDuration,
+                               uint32_t aInterToneGap, ErrorResult& aRv) {
+  if (!mTransceiver->CanSendDTMF()) {
+    aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
+    return;
+  }
+
+  std::string utf8Tones = NS_ConvertUTF16toUTF8(aTones).get();
+
+  std::transform(utf8Tones.begin(), utf8Tones.end(), utf8Tones.begin(),
+                 [](const char c) { return std::toupper(c); });
+
+  if (std::any_of(utf8Tones.begin(), utf8Tones.end(), IsUnrecognizedChar)) {
+    aRv.Throw(NS_ERROR_DOM_INVALID_CHARACTER_ERR);
+    return;
+  }
+
+  CopyUTF8toUTF16(utf8Tones, mToneBuffer);
+  mDuration = std::clamp(aDuration, 40U, 6000U);
+  mInterToneGap = std::clamp(aInterToneGap, 30U, 6000U);
+
+  if (mToneBuffer.Length()) {
+    StartPlayout(0);
+  }
+}
+
+void RTCDTMFSender::StopPlayout() {
+  if (mSendTimer) {
+    mSendTimer->Cancel();
+    mSendTimer = nullptr;
+  }
+}
+
+void RTCDTMFSender::StartPlayout(uint32_t aDelay) {
+  if (!mSendTimer) {
+    mSendTimer = NS_NewTimer();
+    mSendTimer->InitWithCallback(this, aDelay, nsITimer::TYPE_ONE_SHOT);
+  }
+}
+
+nsresult RTCDTMFSender::Notify(nsITimer* timer) {
+  MOZ_ASSERT(NS_IsMainThread());
+  StopPlayout();
+
+  if (!mTransceiver->IsSending()) {
+    return NS_OK;
+  }
+
+  RTCDTMFToneChangeEventInit init;
+  if (!mToneBuffer.IsEmpty()) {
+    uint16_t toneChar = mToneBuffer.CharAt(0);
+    int tone = GetDTMFToneCode(toneChar);
+
+    init.mTone.Assign(toneChar);
+
+    mToneBuffer.Cut(0, 1);
+
+    if (tone == -1) {
+      StartPlayout(2000);
+    } else {
+      // Reset delay if necessary
+      StartPlayout(mDuration + mInterToneGap);
+      mDtmfEvent.Notify(DtmfEvent(mPayloadType.ref(), mPayloadFrequency.ref(),
+                                  tone, mDuration));
+    }
+  }
+
+  RefPtr<RTCDTMFToneChangeEvent> event =
+      RTCDTMFToneChangeEvent::Constructor(this, u"tonechange"_ns, init);
+  DispatchTrustedEvent(event);
+
+  return NS_OK;
+}
+
+nsresult RTCDTMFSender::GetName(nsACString& aName) {
+  aName.AssignLiteral("RTCDTMFSender");
+  return NS_OK;
+}
+
+void RTCDTMFSender::GetToneBuffer(nsAString& aOutToneBuffer) {
+  aOutToneBuffer = mToneBuffer;
+}
+
+}  // namespace mozilla::dom
+
+#undef LOGTAG
diff --git a/dom/media/webrtc/jsapi/RTCDTMFSender.h b/dom/media/webrtc/jsapi/RTCDTMFSender.h
new file mode 100644
index 0000000000..14be7eb6ee
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCDTMFSender.h
@@ -0,0 +1,78 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _RTCDTMFSender_h_
+#define _RTCDTMFSender_h_
+
+#include "MediaEventSource.h"
+#include "mozilla/DOMEventTargetHelper.h"
+#include "mozilla/RefPtr.h"
+#include "js/RootingAPI.h"
+#include "nsITimer.h"
+
+class nsPIDOMWindowInner;
+class nsITimer;
+
+namespace mozilla {
+class AudioSessionConduit;
+
+struct DtmfEvent {
+  DtmfEvent(int aPayloadType, int aPayloadFrequency, int aEventCode,
+            int aLengthMs)
+      : mPayloadType(aPayloadType),
+        mPayloadFrequency(aPayloadFrequency),
+        mEventCode(aEventCode),
+        mLengthMs(aLengthMs) {}
+  const int mPayloadType;
+  const int mPayloadFrequency;
+  const int mEventCode;
+  const int mLengthMs;
+};
+
+namespace dom {
+class RTCRtpTransceiver;
+
+class RTCDTMFSender : public DOMEventTargetHelper,
+                      public nsITimerCallback,
+                      public nsINamed {
+ public:
+  RTCDTMFSender(nsPIDOMWindowInner* aWindow, RTCRtpTransceiver* aTransceiver);
+
+  // nsISupports
+  NS_DECL_NSITIMERCALLBACK
+  NS_DECL_NSINAMED
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(RTCDTMFSender, DOMEventTargetHelper)
+
+  // webidl
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+  void SetPayloadType(int32_t aPayloadType, int32_t aPayloadFrequency);
+  void InsertDTMF(const nsAString& aTones, uint32_t aDuration,
+                  uint32_t aInterToneGap, ErrorResult& aRv);
+  void GetToneBuffer(nsAString& aOutToneBuffer);
+  IMPL_EVENT_HANDLER(tonechange)
+
+  void StopPlayout();
+
+  MediaEventSource<DtmfEvent>& OnDtmfEvent() { return mDtmfEvent; }
+
+ private:
+  virtual ~RTCDTMFSender() = default;
+
+  void StartPlayout(uint32_t aDelay);
+
+  RefPtr<RTCRtpTransceiver> mTransceiver;
+  MediaEventProducer<DtmfEvent> mDtmfEvent;
+  Maybe<int32_t> mPayloadType;
+  Maybe<int32_t> mPayloadFrequency;
+  nsString mToneBuffer;
+  uint32_t mDuration = 0;
+  uint32_t mInterToneGap = 0;
+  nsCOMPtr<nsITimer> mSendTimer;
+};
+
+}  // namespace dom
+}  // namespace mozilla
+#endif  // _RTCDTMFSender_h_
diff --git a/dom/media/webrtc/jsapi/RTCDtlsTransport.cpp b/dom/media/webrtc/jsapi/RTCDtlsTransport.cpp
new file mode 100644
index 0000000000..442e9c7a17
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCDtlsTransport.cpp
@@ -0,0 +1,69 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "RTCDtlsTransport.h"
+#include "mozilla/dom/Event.h"
+#include "mozilla/dom/EventBinding.h"
+#include "mozilla/dom/RTCDtlsTransportBinding.h"
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(RTCDtlsTransport, DOMEventTargetHelper)
+
+NS_IMPL_ADDREF_INHERITED(RTCDtlsTransport, DOMEventTargetHelper)
+NS_IMPL_RELEASE_INHERITED(RTCDtlsTransport, DOMEventTargetHelper)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(RTCDtlsTransport)
+  NS_WRAPPERCACHE_INTERFACE_MAP_ENTRY
+NS_INTERFACE_MAP_END_INHERITING(DOMEventTargetHelper)
+
+RTCDtlsTransport::RTCDtlsTransport(nsPIDOMWindowInner* aWindow)
+    : DOMEventTargetHelper(aWindow), mState(RTCDtlsTransportState::New) {}
+
+JSObject* RTCDtlsTransport::WrapObject(JSContext* aCx,
+                                       JS::Handle<JSObject*> aGivenProto) {
+  return RTCDtlsTransport_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+void RTCDtlsTransport::UpdateState(TransportLayer::State aState) {
+  if (mState == RTCDtlsTransportState::Closed) {
+    return;
+  }
+
+  RTCDtlsTransportState newState = mState;
+  switch (aState) {
+    case TransportLayer::TS_NONE:
+      break;
+    case TransportLayer::TS_INIT:
+      break;
+    case TransportLayer::TS_CONNECTING:
+      newState = RTCDtlsTransportState::Connecting;
+      break;
+    case TransportLayer::TS_OPEN:
+      newState = RTCDtlsTransportState::Connected;
+      break;
+    case TransportLayer::TS_CLOSED:
+      newState = RTCDtlsTransportState::Closed;
+      break;
+    case TransportLayer::TS_ERROR:
+      newState = RTCDtlsTransportState::Failed;
+      break;
+  }
+
+  if (newState == mState) {
+    return;
+  }
+
+  mState = newState;
+
+  EventInit init;
+  init.mBubbles = false;
+  init.mCancelable = false;
+
+  RefPtr<Event> event = Event::Constructor(this, u"statechange"_ns, init);
+
+  DispatchTrustedEvent(event);
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webrtc/jsapi/RTCDtlsTransport.h b/dom/media/webrtc/jsapi/RTCDtlsTransport.h
new file mode 100644
index 0000000000..74a4b5f618
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCDtlsTransport.h
@@ -0,0 +1,43 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _RTCDtlsTransport_h_
+#define _RTCDtlsTransport_h_
+
+#include "mozilla/DOMEventTargetHelper.h"
+#include "mozilla/RefPtr.h"
+#include "js/RootingAPI.h"
+#include "transport/transportlayer.h"
+
+class nsPIDOMWindowInner;
+
+namespace mozilla::dom {
+
+enum class RTCDtlsTransportState : uint8_t;
+
+class RTCDtlsTransport : public DOMEventTargetHelper {
+ public:
+  explicit RTCDtlsTransport(nsPIDOMWindowInner* aWindow);
+
+  // nsISupports
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(RTCDtlsTransport,
+                                           DOMEventTargetHelper)
+
+  // webidl
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+  IMPL_EVENT_HANDLER(statechange)
+  RTCDtlsTransportState State() const { return mState; }
+
+  void UpdateState(TransportLayer::State aState);
+
+ private:
+  virtual ~RTCDtlsTransport() = default;
+
+  RTCDtlsTransportState mState;
+};
+
+}  // namespace mozilla::dom
+#endif  // _RTCDtlsTransport_h_
diff --git a/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp b/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp
new file mode 100644
index 0000000000..136aa5142f
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp
@@ -0,0 +1,942 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "RTCRtpReceiver.h"
+#include "PeerConnectionImpl.h"
+#include "mozilla/dom/RTCRtpCapabilitiesBinding.h"
+#include "transport/logging.h"
+#include "mozilla/dom/MediaStreamTrack.h"
+#include "mozilla/dom/Promise.h"
+#include "nsPIDOMWindow.h"
+#include "PrincipalHandle.h"
+#include "nsIPrincipal.h"
+#include "mozilla/dom/Document.h"
+#include "mozilla/NullPrincipal.h"
+#include "MediaTrackGraph.h"
+#include "RemoteTrackSource.h"
+#include "libwebrtcglue/RtpRtcpConfig.h"
+#include "nsString.h"
+#include "mozilla/dom/AudioStreamTrack.h"
+#include "mozilla/dom/VideoStreamTrack.h"
+#include "MediaTransportHandler.h"
+#include "jsep/JsepTransceiver.h"
+#include "mozilla/dom/RTCRtpReceiverBinding.h"
+#include "mozilla/dom/RTCRtpSourcesBinding.h"
+#include "RTCStatsReport.h"
+#include "mozilla/Preferences.h"
+#include "PeerConnectionCtx.h"
+#include "RTCRtpTransceiver.h"
+#include "libwebrtcglue/AudioConduit.h"
+#include "call/call.h"
+
+namespace mozilla::dom {
+
+LazyLogModule gReceiverLog("RTCRtpReceiver");
+
+NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE_CLASS(RTCRtpReceiver)
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(RTCRtpReceiver)
+  // We do not do anything here, we wait for BreakCycles to be called
+  NS_IMPL_CYCLE_COLLECTION_UNLINK_PRESERVED_WRAPPER
+NS_IMPL_CYCLE_COLLECTION_UNLINK_END
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN(RTCRtpReceiver)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mWindow, mPc, mTransceiver, mTrack,
+                                    mTrackSource)
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
+
+NS_IMPL_CYCLE_COLLECTING_ADDREF(RTCRtpReceiver)
+NS_IMPL_CYCLE_COLLECTING_RELEASE(RTCRtpReceiver)
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(RTCRtpReceiver)
+  NS_WRAPPERCACHE_INTERFACE_MAP_ENTRY
+  NS_INTERFACE_MAP_ENTRY(nsISupports)
+NS_INTERFACE_MAP_END
+
+static PrincipalHandle GetPrincipalHandle(nsPIDOMWindowInner* aWindow,
+                                          PrincipalPrivacy aPrivacy) {
+  // Set the principal used for creating the tracks. This makes the track
+  // data (audio/video samples) accessible to the receiving page. We're
+  // only certain that privacy hasn't been requested if we're connected.
+  nsCOMPtr<nsIScriptObjectPrincipal> winPrincipal = do_QueryInterface(aWindow);
+  RefPtr<nsIPrincipal> principal = winPrincipal->GetPrincipal();
+  if (NS_WARN_IF(!principal)) {
+    principal = NullPrincipal::CreateWithoutOriginAttributes();
+  } else if (aPrivacy == PrincipalPrivacy::Private) {
+    principal = NullPrincipal::CreateWithInheritedAttributes(principal);
+  }
+  return MakePrincipalHandle(principal);
+}
+
+#define INIT_CANONICAL(name, val)         \
+  name(AbstractThread::MainThread(), val, \
+       "RTCRtpReceiver::" #name " (Canonical)")
+
+RTCRtpReceiver::RTCRtpReceiver(
+    nsPIDOMWindowInner* aWindow, PrincipalPrivacy aPrivacy,
+    PeerConnectionImpl* aPc, MediaTransportHandler* aTransportHandler,
+    AbstractThread* aCallThread, nsISerialEventTarget* aStsThread,
+    MediaSessionConduit* aConduit, RTCRtpTransceiver* aTransceiver,
+    const TrackingId& aTrackingId)
+    : mWatchManager(this, AbstractThread::MainThread()),
+      mWindow(aWindow),
+      mPc(aPc),
+      mCallThread(aCallThread),
+      mStsThread(aStsThread),
+      mTransportHandler(aTransportHandler),
+      mTransceiver(aTransceiver),
+      INIT_CANONICAL(mSsrc, 0),
+      INIT_CANONICAL(mVideoRtxSsrc, 0),
+      INIT_CANONICAL(mLocalRtpExtensions, RtpExtList()),
+      INIT_CANONICAL(mAudioCodecs, std::vector<AudioCodecConfig>()),
+      INIT_CANONICAL(mVideoCodecs, std::vector<VideoCodecConfig>()),
+      INIT_CANONICAL(mVideoRtpRtcpConfig, Nothing()),
+      INIT_CANONICAL(mReceiving, false) {
+  PrincipalHandle principalHandle = GetPrincipalHandle(aWindow, aPrivacy);
+  const bool isAudio = aConduit->type() == MediaSessionConduit::AUDIO;
+
+  MediaTrackGraph* graph = MediaTrackGraph::GetInstance(
+      isAudio ? MediaTrackGraph::AUDIO_THREAD_DRIVER
+              : MediaTrackGraph::SYSTEM_THREAD_DRIVER,
+      aWindow, MediaTrackGraph::REQUEST_DEFAULT_SAMPLE_RATE,
+      MediaTrackGraph::DEFAULT_OUTPUT_DEVICE);
+
+  if (isAudio) {
+    auto* source = graph->CreateSourceTrack(MediaSegment::AUDIO);
+    mTrackSource = MakeAndAddRef<RemoteTrackSource>(
+        source, this, principalHandle, u"remote audio"_ns, aTrackingId);
+    mTrack = MakeAndAddRef<AudioStreamTrack>(aWindow, source, mTrackSource);
+    mPipeline = MakeAndAddRef<MediaPipelineReceiveAudio>(
+        mPc->GetHandle(), aTransportHandler, aCallThread, mStsThread.get(),
+        *aConduit->AsAudioSessionConduit(), mTrackSource->Stream(), aTrackingId,
+        principalHandle, aPrivacy);
+  } else {
+    auto* source = graph->CreateSourceTrack(MediaSegment::VIDEO);
+    mTrackSource = MakeAndAddRef<RemoteTrackSource>(
+        source, this, principalHandle, u"remote video"_ns, aTrackingId);
+    mTrack = MakeAndAddRef<VideoStreamTrack>(aWindow, source, mTrackSource);
+    mPipeline = MakeAndAddRef<MediaPipelineReceiveVideo>(
+        mPc->GetHandle(), aTransportHandler, aCallThread, mStsThread.get(),
+        *aConduit->AsVideoSessionConduit(), mTrackSource->Stream(), aTrackingId,
+        principalHandle, aPrivacy);
+  }
+
+  mPipeline->InitControl(this);
+
+  // Spec says remote tracks start out muted.
+  mTrackSource->SetMuted(true);
+
+  // Until Bug 1232234 is fixed, we'll get extra RTCP BYES during renegotiation,
+  // so we'll disable muting on RTCP BYE and timeout for now.
+  if (Preferences::GetBool("media.peerconnection.mute_on_bye_or_timeout",
+                           false)) {
+    mRtcpByeListener = aConduit->RtcpByeEvent().Connect(
+        GetMainThreadSerialEventTarget(), this, &RTCRtpReceiver::OnRtcpBye);
+    mRtcpTimeoutListener = aConduit->RtcpTimeoutEvent().Connect(
+        GetMainThreadSerialEventTarget(), this, &RTCRtpReceiver::OnRtcpTimeout);
+  }
+
+  mWatchManager.Watch(mReceiveTrackMute,
+                      &RTCRtpReceiver::UpdateReceiveTrackMute);
+}
+
+#undef INIT_CANONICAL
+
+RTCRtpReceiver::~RTCRtpReceiver() { MOZ_ASSERT(!mPipeline); }
+
+JSObject* RTCRtpReceiver::WrapObject(JSContext* aCx,
+                                     JS::Handle<JSObject*> aGivenProto) {
+  return RTCRtpReceiver_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+RTCDtlsTransport* RTCRtpReceiver::GetTransport() const {
+  if (!mTransceiver) {
+    return nullptr;
+  }
+  return mTransceiver->GetDtlsTransport();
+}
+
+void RTCRtpReceiver::GetCapabilities(
+    const GlobalObject&, const nsAString& aKind,
+    Nullable<dom::RTCRtpCapabilities>& aResult) {
+  PeerConnectionImpl::GetCapabilities(aKind, aResult, sdp::Direction::kRecv);
+}
+
+already_AddRefed<Promise> RTCRtpReceiver::GetStats(ErrorResult& aError) {
+  nsCOMPtr<nsIGlobalObject> global = do_QueryInterface(mWindow);
+  RefPtr<Promise> promise = Promise::Create(global, aError);
+  if (NS_WARN_IF(aError.Failed())) {
+    return nullptr;
+  }
+
+  if (NS_WARN_IF(!mTransceiver)) {
+    // TODO(bug 1056433): When we stop nulling this out when the PC is closed
+    // (or when the transceiver is stopped), we can remove this code. We
+    // resolve instead of reject in order to make this eventual change in
+    // behavior a little smaller.
+    promise->MaybeResolve(new RTCStatsReport(mWindow));
+    return promise.forget();
+  }
+
+  mTransceiver->ChainToDomPromiseWithCodecStats(GetStatsInternal(), promise);
+  return promise.forget();
+}
+
+nsTArray<RefPtr<RTCStatsPromise>> RTCRtpReceiver::GetStatsInternal(
+    bool aSkipIceStats) {
+  MOZ_ASSERT(NS_IsMainThread());
+  nsTArray<RefPtr<RTCStatsPromise>> promises(3);
+
+  if (!mPipeline) {
+    return promises;
+  }
+
+  if (!mHaveStartedReceiving) {
+    return promises;
+  }
+
+  nsString recvTrackId;
+  MOZ_ASSERT(mTrack);
+  if (mTrack) {
+    mTrack->GetId(recvTrackId);
+  }
+
+  {
+    // Add bandwidth estimation stats
+    promises.AppendElement(InvokeAsync(
+        mCallThread, __func__,
+        [conduit = mPipeline->mConduit, recvTrackId]() mutable {
+          auto report = MakeUnique<dom::RTCStatsCollection>();
+          const Maybe<webrtc::Call::Stats> stats = conduit->GetCallStats();
+          stats.apply([&](const auto& aStats) {
+            dom::RTCBandwidthEstimationInternal bw;
+            bw.mTrackIdentifier = recvTrackId;
+            bw.mSendBandwidthBps.Construct(aStats.send_bandwidth_bps / 8);
+            bw.mMaxPaddingBps.Construct(aStats.max_padding_bitrate_bps / 8);
+            bw.mReceiveBandwidthBps.Construct(aStats.recv_bandwidth_bps / 8);
+            bw.mPacerDelayMs.Construct(aStats.pacer_delay_ms);
+            if (aStats.rtt_ms >= 0) {
+              bw.mRttMs.Construct(aStats.rtt_ms);
+            }
+            if (!report->mBandwidthEstimations.AppendElement(std::move(bw),
+                                                             fallible)) {
+              mozalloc_handle_oom(0);
+            }
+          });
+          return RTCStatsPromise::CreateAndResolve(std::move(report), __func__);
+        }));
+  }
+
+  promises.AppendElement(
+      InvokeAsync(
+          mCallThread, __func__,
+          [pipeline = mPipeline, recvTrackId] {
+            auto report = MakeUnique<dom::RTCStatsCollection>();
+            auto asAudio = pipeline->mConduit->AsAudioSessionConduit();
+            auto asVideo = pipeline->mConduit->AsVideoSessionConduit();
+
+            nsString kind = asVideo.isNothing() ? u"audio"_ns : u"video"_ns;
+            nsString idstr = kind + u"_"_ns;
+            idstr.AppendInt(static_cast<uint32_t>(pipeline->Level()));
+
+            Maybe<uint32_t> ssrc = pipeline->mConduit->GetRemoteSSRC();
+
+            // Add frame history
+            asVideo.apply([&](const auto& conduit) {
+              if (conduit->AddFrameHistory(&report->mVideoFrameHistories)) {
+                auto& history = report->mVideoFrameHistories.LastElement();
+                history.mTrackIdentifier = recvTrackId;
+              }
+            });
+
+            // TODO(@@NG):ssrcs handle Conduits having multiple stats at the
+            // same level.
+            // This is pending spec work.
+            // Gather pipeline stats.
+            nsString localId = u"inbound_rtp_"_ns + idstr;
+            nsString remoteId;
+
+            auto constructCommonRemoteOutboundRtpStats =
+                [&](RTCRemoteOutboundRtpStreamStats& aRemote,
+                    const DOMHighResTimeStamp& aTimestamp) {
+                  remoteId = u"inbound_rtcp_"_ns + idstr;
+                  aRemote.mTimestamp.Construct(aTimestamp);
+                  aRemote.mId.Construct(remoteId);
+                  aRemote.mType.Construct(RTCStatsType::Remote_outbound_rtp);
+                  ssrc.apply([&](uint32_t aSsrc) { aRemote.mSsrc = aSsrc; });
+                  aRemote.mKind = kind;
+                  aRemote.mMediaType.Construct(
+                      kind);  // mediaType is the old name for kind.
+                  aRemote.mLocalId.Construct(localId);
+                };
+
+            auto constructCommonInboundRtpStats =
+                [&](RTCInboundRtpStreamStats& aLocal) {
+                  aLocal.mTrackIdentifier = recvTrackId;
+                  aLocal.mTimestamp.Construct(
+                      pipeline->GetTimestampMaker().GetNow().ToDom());
+                  aLocal.mId.Construct(localId);
+                  aLocal.mType.Construct(RTCStatsType::Inbound_rtp);
+                  ssrc.apply([&](uint32_t aSsrc) { aLocal.mSsrc = aSsrc; });
+                  aLocal.mKind = kind;
+                  aLocal.mMediaType.Construct(
+                      kind);  // mediaType is the old name for kind.
+                  if (remoteId.Length()) {
+                    aLocal.mRemoteId.Construct(remoteId);
+                  }
+                };
+
+            asAudio.apply([&](auto& aConduit) {
+              Maybe<webrtc::AudioReceiveStreamInterface::Stats> audioStats =
+                  aConduit->GetReceiverStats();
+              if (audioStats.isNothing()) {
+                return;
+              }
+
+              if (!audioStats->last_packet_received_timestamp_ms) {
+                // By spec: "The lifetime of all RTP monitored objects starts
+                // when the RTP stream is first used: When the first RTP packet
+                // is sent or received on the SSRC it represents"
+                return;
+              }
+
+              // First, fill in remote stat with rtcp sender data, if present.
+              if (audioStats->last_sender_report_timestamp_ms) {
+                RTCRemoteOutboundRtpStreamStats remote;
+                constructCommonRemoteOutboundRtpStats(
+                    remote,
+                    RTCStatsTimestamp::FromNtp(
+                        aConduit->GetTimestampMaker(),
+                        webrtc::Timestamp::Millis(
+                            *audioStats->last_sender_report_timestamp_ms) +
+                            webrtc::TimeDelta::Seconds(webrtc::kNtpJan1970))
+                        .ToDom());
+                remote.mPacketsSent.Construct(
+                    audioStats->sender_reports_packets_sent);
+                remote.mBytesSent.Construct(
+                    audioStats->sender_reports_bytes_sent);
+                remote.mRemoteTimestamp.Construct(
+                    *audioStats->last_sender_report_remote_timestamp_ms);
+                if (!report->mRemoteOutboundRtpStreamStats.AppendElement(
+                        std::move(remote), fallible)) {
+                  mozalloc_handle_oom(0);
+                }
+              }
+
+              // Then, fill in local side (with cross-link to remote only if
+              // present)
+              RTCInboundRtpStreamStats local;
+              constructCommonInboundRtpStats(local);
+              local.mJitter.Construct(audioStats->jitter_ms / 1000.0);
+              local.mPacketsLost.Construct(audioStats->packets_lost);
+              local.mPacketsReceived.Construct(audioStats->packets_rcvd);
+              local.mPacketsDiscarded.Construct(audioStats->packets_discarded);
+              local.mBytesReceived.Construct(audioStats->payload_bytes_rcvd);
+              // Always missing from libwebrtc stats
+              // if (audioStats->estimated_playout_ntp_timestamp_ms) {
+              //   local.mEstimatedPlayoutTimestamp.Construct(
+              //       RTCStatsTimestamp::FromNtp(
+              //           aConduit->GetTimestampMaker(),
+              //           webrtc::Timestamp::Millis(
+              //               *audioStats->estimated_playout_ntp_timestamp_ms))
+              //           .ToDom());
+              // }
+              local.mJitterBufferDelay.Construct(
+                  audioStats->jitter_buffer_delay_seconds);
+              local.mJitterBufferEmittedCount.Construct(
+                  audioStats->jitter_buffer_emitted_count);
+              local.mTotalSamplesReceived.Construct(
+                  audioStats->total_samples_received);
+              local.mConcealedSamples.Construct(audioStats->concealed_samples);
+              local.mSilentConcealedSamples.Construct(
+                  audioStats->silent_concealed_samples);
+              if (audioStats->last_packet_received_timestamp_ms) {
+                local.mLastPacketReceivedTimestamp.Construct(
+                    RTCStatsTimestamp::FromNtp(
+                        aConduit->GetTimestampMaker(),
+                        webrtc::Timestamp::Millis(
+                            *audioStats->last_packet_received_timestamp_ms) +
+                            webrtc::TimeDelta::Seconds(webrtc::kNtpJan1970))
+                        .ToDom());
+              }
+              local.mHeaderBytesReceived.Construct(
+                  audioStats->header_and_padding_bytes_rcvd);
+              local.mFecPacketsReceived.Construct(
+                  audioStats->fec_packets_received);
+              local.mFecPacketsDiscarded.Construct(
+                  audioStats->fec_packets_discarded);
+              local.mConcealmentEvents.Construct(
+                  audioStats->concealment_events);
+
+              local.mInsertedSamplesForDeceleration.Construct(
+                  audioStats->inserted_samples_for_deceleration);
+              local.mRemovedSamplesForAcceleration.Construct(
+                  audioStats->removed_samples_for_acceleration);
+              if (audioStats->audio_level >= 0 &&
+                  audioStats->audio_level <= 32767) {
+                local.mAudioLevel.Construct(audioStats->audio_level / 32767.0);
+              }
+              local.mTotalAudioEnergy.Construct(
+                  audioStats->total_output_energy);
+              local.mTotalSamplesDuration.Construct(
+                  audioStats->total_output_duration);
+
+              if (!report->mInboundRtpStreamStats.AppendElement(
+                      std::move(local), fallible)) {
+                mozalloc_handle_oom(0);
+              }
+            });
+
+            asVideo.apply([&](auto& aConduit) {
+              Maybe<webrtc::VideoReceiveStreamInterface::Stats> videoStats =
+                  aConduit->GetReceiverStats();
+              if (videoStats.isNothing()) {
+                return;
+              }
+
+              if (!videoStats->rtp_stats.last_packet_received_timestamp_ms) {
+                // By spec: "The lifetime of all RTP monitored objects starts
+                // when the RTP stream is first used: When the first RTP packet
+                // is sent or received on the SSRC it represents"
+                return;
+              }
+
+              // First, fill in remote stat with rtcp sender data, if present.
+              if (videoStats->rtcp_sender_ntp_timestamp_ms) {
+                RTCRemoteOutboundRtpStreamStats remote;
+                constructCommonRemoteOutboundRtpStats(
+                    remote, RTCStatsTimestamp::FromNtp(
+                                aConduit->GetTimestampMaker(),
+                                webrtc::Timestamp::Millis(
+                                    videoStats->rtcp_sender_ntp_timestamp_ms))
+                                .ToDom());
+                remote.mPacketsSent.Construct(
+                    videoStats->rtcp_sender_packets_sent);
+                remote.mBytesSent.Construct(
+                    videoStats->rtcp_sender_octets_sent);
+                remote.mRemoteTimestamp.Construct(
+                    (webrtc::TimeDelta::Millis(
+                         videoStats->rtcp_sender_remote_ntp_timestamp_ms) -
+                     webrtc::TimeDelta::Seconds(webrtc::kNtpJan1970))
+                        .ms());
+                if (!report->mRemoteOutboundRtpStreamStats.AppendElement(
+                        std::move(remote), fallible)) {
+                  mozalloc_handle_oom(0);
+                }
+              }
+
+              // Then, fill in local side (with cross-link to remote only if
+              // present)
+              RTCInboundRtpStreamStats local;
+              constructCommonInboundRtpStats(local);
+              local.mJitter.Construct(
+                  static_cast<double>(videoStats->rtp_stats.jitter) /
+                  webrtc::kVideoPayloadTypeFrequency);
+              local.mPacketsLost.Construct(videoStats->rtp_stats.packets_lost);
+              local.mPacketsReceived.Construct(
+                  videoStats->rtp_stats.packet_counter.packets);
+              local.mPacketsDiscarded.Construct(videoStats->packets_discarded);
+              local.mDiscardedPackets.Construct(videoStats->packets_discarded);
+              local.mBytesReceived.Construct(
+                  videoStats->rtp_stats.packet_counter.payload_bytes);
+
+              // Fill in packet type statistics
+              local.mNackCount.Construct(
+                  videoStats->rtcp_packet_type_counts.nack_packets);
+              local.mFirCount.Construct(
+                  videoStats->rtcp_packet_type_counts.fir_packets);
+              local.mPliCount.Construct(
+                  videoStats->rtcp_packet_type_counts.pli_packets);
+
+              // Lastly, fill in video decoder stats
+              local.mFramesDecoded.Construct(videoStats->frames_decoded);
+
+              local.mFramesPerSecond.Construct(videoStats->decode_frame_rate);
+              local.mFrameWidth.Construct(videoStats->width);
+              local.mFrameHeight.Construct(videoStats->height);
+              // XXX: key_frames + delta_frames may undercount frames because
+              // they were dropped in FrameBuffer::InsertFrame. (bug 1766553)
+              local.mFramesReceived.Construct(
+                  videoStats->frame_counts.key_frames +
+                  videoStats->frame_counts.delta_frames);
+              local.mJitterBufferDelay.Construct(
+                  videoStats->jitter_buffer_delay_seconds);
+              local.mJitterBufferEmittedCount.Construct(
+                  videoStats->jitter_buffer_emitted_count);
+
+              if (videoStats->qp_sum) {
+                local.mQpSum.Construct(videoStats->qp_sum.value());
+              }
+              local.mTotalDecodeTime.Construct(
+                  double(videoStats->total_decode_time.ms()) / 1000);
+              local.mTotalInterFrameDelay.Construct(
+                  videoStats->total_inter_frame_delay);
+              local.mTotalSquaredInterFrameDelay.Construct(
+                  videoStats->total_squared_inter_frame_delay);
+              if (videoStats->rtp_stats.last_packet_received_timestamp_ms) {
+                local.mLastPacketReceivedTimestamp.Construct(
+                    RTCStatsTimestamp::FromNtp(
+                        aConduit->GetTimestampMaker(),
+                        webrtc::Timestamp::Millis(
+                            *videoStats->rtp_stats
+                                 .last_packet_received_timestamp_ms) +
+                            webrtc::TimeDelta::Seconds(webrtc::kNtpJan1970))
+                        .ToDom());
+              }
+              local.mHeaderBytesReceived.Construct(
+                  videoStats->rtp_stats.packet_counter.header_bytes +
+                  videoStats->rtp_stats.packet_counter.padding_bytes);
+              local.mTotalProcessingDelay.Construct(
+                  videoStats->total_processing_delay.seconds<double>());
+              /*
+               * Potential new stats that are now available upstream
+                   .if (videoStats->estimated_playout_ntp_timestamp_ms) {
+                local.mEstimatedPlayoutTimestamp.Construct(
+                    RTCStatsTimestamp::FromNtp(
+                        aConduit->GetTimestampMaker(),
+                        webrtc::Timestamp::Millis(
+                            *videoStats->estimated_playout_ntp_timestamp_ms))
+                        .ToDom());
+              }
+               */
+              // Not including frames dropped in the rendering pipe, which
+              // is not of webrtc's concern anyway?!
+              local.mFramesDropped.Construct(videoStats->frames_dropped);
+              if (!report->mInboundRtpStreamStats.AppendElement(
+                      std::move(local), fallible)) {
+                mozalloc_handle_oom(0);
+              }
+            });
+            return RTCStatsPromise::CreateAndResolve(std::move(report),
+                                                     __func__);
+          })
+          ->Then(
+              mStsThread, __func__,
+              [pipeline = mPipeline](UniquePtr<RTCStatsCollection> aReport) {
+                // Fill in Contributing Source statistics
+                if (!aReport->mInboundRtpStreamStats.IsEmpty() &&
+                    aReport->mInboundRtpStreamStats[0].mId.WasPassed()) {
+                  pipeline->GetContributingSourceStats(
+                      aReport->mInboundRtpStreamStats[0].mId.Value(),
+                      aReport->mRtpContributingSourceStats);
+                }
+                return RTCStatsPromise::CreateAndResolve(std::move(aReport),
+                                                         __func__);
+              },
+              [] {
+                MOZ_CRASH("Unexpected reject");
+                return RTCStatsPromise::CreateAndReject(NS_ERROR_UNEXPECTED,
+                                                        __func__);
+              }));
+
+  if (!aSkipIceStats && GetJsepTransceiver().mTransport.mComponents) {
+    promises.AppendElement(mTransportHandler->GetIceStats(
+        GetJsepTransceiver().mTransport.mTransportId,
+        mPipeline->GetTimestampMaker().GetNow().ToDom()));
+  }
+
+  return promises;
+}
+
+void RTCRtpReceiver::SetJitterBufferTarget(
+    const Nullable<DOMHighResTimeStamp>& aTargetMs, ErrorResult& aError) {
+  // Spec says jitter buffer target cannot be negative or larger than 4000
+  // milliseconds and to throw RangeError if it is. If an invalid value is
+  // received we return early to preserve the current JitterBufferTarget
+  // internal slot and jitter buffer values.
+  if (mPipeline && mPipeline->mConduit) {
+    if (!aTargetMs.IsNull() &&
+        (aTargetMs.Value() < 0.0 || aTargetMs.Value() > 4000.0)) {
+      aError.ThrowRangeError<MSG_VALUE_OUT_OF_RANGE>("jitterBufferTarget");
+      return;
+    }
+
+    mJitterBufferTarget.reset();
+
+    if (!aTargetMs.IsNull()) {
+      mJitterBufferTarget = Some(aTargetMs.Value());
+    }
+    // If aJitterBufferTarget is null then we are resetting the jitter buffer so
+    // pass the default target of 0.0.
+    mPipeline->mConduit->SetJitterBufferTarget(
+        mJitterBufferTarget.valueOr(0.0));
+  }
+}
+
+void RTCRtpReceiver::GetContributingSources(
+    nsTArray<RTCRtpContributingSource>& aSources) {
+  // Duplicate code...
+  if (mPipeline && mPipeline->mConduit) {
+    nsTArray<dom::RTCRtpSourceEntry> sources;
+    mPipeline->mConduit->GetRtpSources(sources);
+    sources.RemoveElementsBy([](const dom::RTCRtpSourceEntry& aEntry) {
+      return aEntry.mSourceType != dom::RTCRtpSourceEntryType::Contributing;
+    });
+    aSources.ReplaceElementsAt(0, aSources.Length(), sources.Elements(),
+                               sources.Length());
+  }
+}
+
+void RTCRtpReceiver::GetSynchronizationSources(
+    nsTArray<dom::RTCRtpSynchronizationSource>& aSources) {
+  // Duplicate code...
+  if (mPipeline && mPipeline->mConduit) {
+    nsTArray<dom::RTCRtpSourceEntry> sources;
+    mPipeline->mConduit->GetRtpSources(sources);
+    sources.RemoveElementsBy([](const dom::RTCRtpSourceEntry& aEntry) {
+      return aEntry.mSourceType != dom::RTCRtpSourceEntryType::Synchronization;
+    });
+    aSources.ReplaceElementsAt(0, aSources.Length(), sources.Elements(),
+                               sources.Length());
+  }
+}
+
+nsPIDOMWindowInner* RTCRtpReceiver::GetParentObject() const { return mWindow; }
+
+void RTCRtpReceiver::Shutdown() {
+  MOZ_ASSERT(NS_IsMainThread());
+  mWatchManager.Shutdown();
+  if (mPipeline) {
+    mPipeline->Shutdown();
+    mPipeline = nullptr;
+  }
+  if (mTrackSource) {
+    mTrackSource->Destroy();
+  }
+  mCallThread = nullptr;
+  mRtcpByeListener.DisconnectIfExists();
+  mRtcpTimeoutListener.DisconnectIfExists();
+  mUnmuteListener.DisconnectIfExists();
+}
+
+void RTCRtpReceiver::BreakCycles() {
+  mWindow = nullptr;
+  mPc = nullptr;
+  mTrack = nullptr;
+  mTrackSource = nullptr;
+}
+
+void RTCRtpReceiver::UpdateTransport() {
+  MOZ_ASSERT(NS_IsMainThread());
+  if (!mHaveSetupTransport) {
+    mPipeline->SetLevel(GetJsepTransceiver().GetLevel());
+    mHaveSetupTransport = true;
+  }
+
+  UniquePtr<MediaPipelineFilter> filter;
+
+  auto const& details = GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails();
+  if (GetJsepTransceiver().HasBundleLevel() && details) {
+    std::vector<webrtc::RtpExtension> extmaps;
+    details->ForEachRTPHeaderExtension(
+        [&extmaps](const SdpExtmapAttributeList::Extmap& extmap) {
+          extmaps.emplace_back(extmap.extensionname, extmap.entry);
+        });
+    filter = MakeUnique<MediaPipelineFilter>(extmaps);
+
+    // Add remote SSRCs so we can distinguish which RTP packets actually
+    // belong to this pipeline (also RTCP sender reports).
+    for (uint32_t ssrc : GetJsepTransceiver().mRecvTrack.GetSsrcs()) {
+      filter->AddRemoteSSRC(ssrc);
+    }
+    for (uint32_t ssrc : GetJsepTransceiver().mRecvTrack.GetRtxSsrcs()) {
+      filter->AddRemoteSSRC(ssrc);
+    }
+    auto mid = Maybe<std::string>();
+    if (GetMid() != "") {
+      mid = Some(GetMid());
+    }
+    filter->SetRemoteMediaStreamId(mid);
+
+    // Add unique payload types as a last-ditch fallback
+    auto uniquePts = GetJsepTransceiver()
+                         .mRecvTrack.GetNegotiatedDetails()
+                         ->GetUniquePayloadTypes();
+    for (unsigned char& uniquePt : uniquePts) {
+      filter->AddUniquePT(uniquePt);
+    }
+  }
+
+  mPipeline->UpdateTransport_m(GetJsepTransceiver().mTransport.mTransportId,
+                               std::move(filter));
+}
+
+void RTCRtpReceiver::UpdateConduit() {
+  if (mPipeline->mConduit->type() == MediaSessionConduit::VIDEO) {
+    UpdateVideoConduit();
+  } else {
+    UpdateAudioConduit();
+  }
+
+  if ((mReceiving = mTransceiver->IsReceiving())) {
+    mHaveStartedReceiving = true;
+  }
+}
+
+void RTCRtpReceiver::UpdateVideoConduit() {
+  RefPtr<VideoSessionConduit> conduit =
+      *mPipeline->mConduit->AsVideoSessionConduit();
+
+  // NOTE(pkerr) - this is new behavior. Needed because the
+  // CreateVideoReceiveStream method of the Call API will assert (in debug)
+  // and fail if a value is not provided for the remote_ssrc that will be used
+  // by the far-end sender.
+  if (!GetJsepTransceiver().mRecvTrack.GetSsrcs().empty()) {
+    MOZ_LOG(gReceiverLog, LogLevel::Debug,
+            ("%s[%s]: %s Setting remote SSRC %u", mPc->GetHandle().c_str(),
+             GetMid().c_str(), __FUNCTION__,
+             GetJsepTransceiver().mRecvTrack.GetSsrcs().front()));
+    uint32_t rtxSsrc =
+        GetJsepTransceiver().mRecvTrack.GetRtxSsrcs().empty()
+            ? 0
+            : GetJsepTransceiver().mRecvTrack.GetRtxSsrcs().front();
+    mSsrc = GetJsepTransceiver().mRecvTrack.GetSsrcs().front();
+    mVideoRtxSsrc = rtxSsrc;
+
+    // TODO (bug 1423041) once we pay attention to receiving MID's in RTP
+    // packets (see bug 1405495) we could make this depending on the presence of
+    // MID in the RTP packets instead of relying on the signaling.
+    // In any case, do not disable SSRC changes if no SSRCs were negotiated
+    if (GetJsepTransceiver().HasBundleLevel() &&
+        (!GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails() ||
+         !GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails()->GetExt(
+             webrtc::RtpExtension::kMidUri))) {
+      mCallThread->Dispatch(
+          NewRunnableMethod("VideoSessionConduit::DisableSsrcChanges", conduit,
+                            &VideoSessionConduit::DisableSsrcChanges));
+    }
+  }
+
+  if (GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails() &&
+      GetJsepTransceiver().mRecvTrack.GetActive()) {
+    const auto& details(
+        *GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails());
+
+    {
+      std::vector<webrtc::RtpExtension> extmaps;
+      // @@NG read extmap from track
+      details.ForEachRTPHeaderExtension(
+          [&extmaps](const SdpExtmapAttributeList::Extmap& extmap) {
+            extmaps.emplace_back(extmap.extensionname, extmap.entry);
+          });
+      mLocalRtpExtensions = extmaps;
+    }
+
+    std::vector<VideoCodecConfig> configs;
+    RTCRtpTransceiver::NegotiatedDetailsToVideoCodecConfigs(details, &configs);
+    if (configs.empty()) {
+      // TODO: Are we supposed to plumb this error back to JS? This does not
+      // seem like a failure to set an answer, it just means that codec
+      // negotiation failed. For now, we're just doing the same thing we do
+      // if negotiation as a whole failed.
+      MOZ_LOG(gReceiverLog, LogLevel::Error,
+              ("%s[%s]: %s  No video codecs were negotiated (recv).",
+               mPc->GetHandle().c_str(), GetMid().c_str(), __FUNCTION__));
+      return;
+    }
+
+    mVideoCodecs = configs;
+    mVideoRtpRtcpConfig = Some(details.GetRtpRtcpConfig());
+  }
+}
+
+void RTCRtpReceiver::UpdateAudioConduit() {
+  RefPtr<AudioSessionConduit> conduit =
+      *mPipeline->mConduit->AsAudioSessionConduit();
+
+  if (!GetJsepTransceiver().mRecvTrack.GetSsrcs().empty()) {
+    MOZ_LOG(gReceiverLog, LogLevel::Debug,
+            ("%s[%s]: %s Setting remote SSRC %u", mPc->GetHandle().c_str(),
+             GetMid().c_str(), __FUNCTION__,
+             GetJsepTransceiver().mRecvTrack.GetSsrcs().front()));
+    mSsrc = GetJsepTransceiver().mRecvTrack.GetSsrcs().front();
+
+    // TODO (bug 1423041) once we pay attention to receiving MID's in RTP
+    // packets (see bug 1405495) we could make this depending on the presence of
+    // MID in the RTP packets instead of relying on the signaling.
+    // In any case, do not disable SSRC changes if no SSRCs were negotiated
+    if (GetJsepTransceiver().HasBundleLevel() &&
+        (!GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails() ||
+         !GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails()->GetExt(
+             webrtc::RtpExtension::kMidUri))) {
+      mCallThread->Dispatch(
+          NewRunnableMethod("AudioSessionConduit::DisableSsrcChanges", conduit,
+                            &AudioSessionConduit::DisableSsrcChanges));
+    }
+  }
+
+  if (GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails() &&
+      GetJsepTransceiver().mRecvTrack.GetActive()) {
+    const auto& details(
+        *GetJsepTransceiver().mRecvTrack.GetNegotiatedDetails());
+    std::vector<AudioCodecConfig> configs;
+    RTCRtpTransceiver::NegotiatedDetailsToAudioCodecConfigs(details, &configs);
+    if (configs.empty()) {
+      // TODO: Are we supposed to plumb this error back to JS? This does not
+      // seem like a failure to set an answer, it just means that codec
+      // negotiation failed. For now, we're just doing the same thing we do
+      // if negotiation as a whole failed.
+      MOZ_LOG(gReceiverLog, LogLevel::Error,
+              ("%s[%s]: %s No audio codecs were negotiated (recv)",
+               mPc->GetHandle().c_str(), GetMid().c_str(), __FUNCTION__));
+      return;
+    }
+
+    // Ensure conduit knows about extensions prior to creating streams
+    {
+      std::vector<webrtc::RtpExtension> extmaps;
+      // @@NG read extmap from track
+      details.ForEachRTPHeaderExtension(
+          [&extmaps](const SdpExtmapAttributeList::Extmap& extmap) {
+            extmaps.emplace_back(extmap.extensionname, extmap.entry);
+          });
+      mLocalRtpExtensions = extmaps;
+    }
+
+    mAudioCodecs = configs;
+  }
+}
+
+void RTCRtpReceiver::Stop() {
+  MOZ_ASSERT(mTransceiver->Stopped());
+  mReceiving = false;
+}
+
+bool RTCRtpReceiver::HasTrack(const dom::MediaStreamTrack* aTrack) const {
+  return !aTrack || (mTrack == aTrack);
+}
+
+void RTCRtpReceiver::SyncFromJsep(const JsepTransceiver& aJsepTransceiver) {
+  if (!mPipeline) {
+    return;
+  }
+
+  // Spec says we set [[Receptive]] to true on sLD(sendrecv/recvonly), and to
+  // false on sRD(recvonly/inactive), sLD(sendonly/inactive), or when stop()
+  // is called.
+  bool wasReceptive = mReceptive;
+  mReceptive = aJsepTransceiver.mRecvTrack.GetReceptive();
+  if (!wasReceptive && mReceptive) {
+    mUnmuteListener = mPipeline->mConduit->RtpPacketEvent().Connect(
+        GetMainThreadSerialEventTarget(), this, &RTCRtpReceiver::OnRtpPacket);
+  } else if (wasReceptive && !mReceptive) {
+    mUnmuteListener.DisconnectIfExists();
+  }
+}
+
+void RTCRtpReceiver::SyncToJsep(JsepTransceiver& aJsepTransceiver) const {}
+
+void RTCRtpReceiver::UpdateStreams(StreamAssociationChanges* aChanges) {
+  // We don't sort and use set_difference, because we need to report the
+  // added/removed streams in the order that they appear in the SDP.
+  std::set<std::string> newIds(
+      GetJsepTransceiver().mRecvTrack.GetStreamIds().begin(),
+      GetJsepTransceiver().mRecvTrack.GetStreamIds().end());
+  MOZ_ASSERT(GetJsepTransceiver().mRecvTrack.GetRemoteSetSendBit() ||
+             newIds.empty());
+  bool needsTrackEvent = false;
+  for (const auto& id : mStreamIds) {
+    if (!newIds.count(id)) {
+      aChanges->mStreamAssociationsRemoved.push_back({mTrack, id});
+    }
+  }
+
+  std::set<std::string> oldIds(mStreamIds.begin(), mStreamIds.end());
+  for (const auto& id : GetJsepTransceiver().mRecvTrack.GetStreamIds()) {
+    if (!oldIds.count(id)) {
+      needsTrackEvent = true;
+      aChanges->mStreamAssociationsAdded.push_back({mTrack, id});
+    }
+  }
+
+  mStreamIds = GetJsepTransceiver().mRecvTrack.GetStreamIds();
+
+  if (mRemoteSetSendBit !=
+      GetJsepTransceiver().mRecvTrack.GetRemoteSetSendBit()) {
+    mRemoteSetSendBit = GetJsepTransceiver().mRecvTrack.GetRemoteSetSendBit();
+    if (mRemoteSetSendBit) {
+      needsTrackEvent = true;
+    } else {
+      aChanges->mReceiversToMute.push_back(this);
+    }
+  }
+
+  if (needsTrackEvent) {
+    aChanges->mTrackEvents.push_back({this, mStreamIds});
+  }
+}
+
+void RTCRtpReceiver::UpdatePrincipalPrivacy(PrincipalPrivacy aPrivacy) {
+  if (!mPipeline) {
+    return;
+  }
+
+  if (aPrivacy != PrincipalPrivacy::Private) {
+    return;
+  }
+
+  mPipeline->SetPrivatePrincipal(GetPrincipalHandle(mWindow, aPrivacy));
+}
+
+// test-only: adds fake CSRCs and audio data
+void RTCRtpReceiver::MozInsertAudioLevelForContributingSource(
+    const uint32_t aSource, const DOMHighResTimeStamp aTimestamp,
+    const uint32_t aRtpTimestamp, const bool aHasLevel, const uint8_t aLevel) {
+  if (!mPipeline || mPipeline->IsVideo() || !mPipeline->mConduit) {
+    return;
+  }
+  mPipeline->mConduit->InsertAudioLevelForContributingSource(
+      aSource, aTimestamp, aRtpTimestamp, aHasLevel, aLevel);
+}
+
+void RTCRtpReceiver::OnRtcpBye() { mReceiveTrackMute = true; }
+
+void RTCRtpReceiver::OnRtcpTimeout() { mReceiveTrackMute = true; }
+
+void RTCRtpReceiver::SetTrackMuteFromRemoteSdp() {
+  MOZ_ASSERT(!mReceptive,
+             "PeerConnectionImpl should have blocked unmute events prior to "
+             "firing mute");
+  mReceiveTrackMute = true;
+  // Set the mute state (and fire the mute event) synchronously. Unmute is
+  // handled asynchronously after receiving RTP packets.
+  UpdateReceiveTrackMute();
+  MOZ_ASSERT(mTrack->Muted(), "Muted state was indeed set synchronously");
+}
+
+void RTCRtpReceiver::OnRtpPacket() {
+  MOZ_ASSERT(mReceptive, "We should not be registered unless this is set!");
+  // We should be registered since we're currently getting a callback.
+  mUnmuteListener.Disconnect();
+  if (mReceptive) {
+    mReceiveTrackMute = false;
+  }
+}
+
+void RTCRtpReceiver::UpdateReceiveTrackMute() {
+  if (!mTrack) {
+    return;
+  }
+  if (!mTrackSource) {
+    return;
+  }
+  // This sets the muted state for mTrack and all its clones.
+  // Idempotent -- only reacts to changes.
+  mTrackSource->SetMuted(mReceiveTrackMute);
+}
+
+std::string RTCRtpReceiver::GetMid() const {
+  return mTransceiver->GetMidAscii();
+}
+
+JsepTransceiver& RTCRtpReceiver::GetJsepTransceiver() {
+  MOZ_ASSERT(mTransceiver);
+  return mTransceiver->GetJsepTransceiver();
+}
+
+const JsepTransceiver& RTCRtpReceiver::GetJsepTransceiver() const {
+  MOZ_ASSERT(mTransceiver);
+  return mTransceiver->GetJsepTransceiver();
+}
+
+}  // namespace mozilla::dom
+
+#undef LOGTAG
diff --git a/dom/media/webrtc/jsapi/RTCRtpReceiver.h b/dom/media/webrtc/jsapi/RTCRtpReceiver.h
new file mode 100644
index 0000000000..2c050bceb1
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCRtpReceiver.h
@@ -0,0 +1,198 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _RTCRtpReceiver_h_
+#define _RTCRtpReceiver_h_
+
+#include "nsISupports.h"
+#include "nsWrapperCache.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/StateMirroring.h"
+#include "mozilla/Maybe.h"
+#include "js/RootingAPI.h"
+#include "libwebrtcglue/RtpRtcpConfig.h"
+#include "nsTArray.h"
+#include "mozilla/dom/RTCRtpCapabilitiesBinding.h"
+#include "mozilla/dom/RTCStatsReportBinding.h"
+#include "PerformanceRecorder.h"
+#include "RTCStatsReport.h"
+#include "transportbridge/MediaPipeline.h"
+#include <vector>
+
+class nsPIDOMWindowInner;
+
+namespace mozilla {
+class MediaSessionConduit;
+class MediaTransportHandler;
+class JsepTransceiver;
+class PeerConnectionImpl;
+enum class PrincipalPrivacy : uint8_t;
+class RemoteTrackSource;
+
+namespace dom {
+class MediaStreamTrack;
+class Promise;
+class RTCDtlsTransport;
+struct RTCRtpCapabilities;
+struct RTCRtpContributingSource;
+struct RTCRtpSynchronizationSource;
+class RTCRtpTransceiver;
+
+class RTCRtpReceiver : public nsISupports,
+                       public nsWrapperCache,
+                       public MediaPipelineReceiveControlInterface {
+ public:
+  RTCRtpReceiver(nsPIDOMWindowInner* aWindow, PrincipalPrivacy aPrivacy,
+                 PeerConnectionImpl* aPc,
+                 MediaTransportHandler* aTransportHandler,
+                 AbstractThread* aCallThread, nsISerialEventTarget* aStsThread,
+                 MediaSessionConduit* aConduit, RTCRtpTransceiver* aTransceiver,
+                 const TrackingId& aTrackingId);
+
+  // nsISupports
+  NS_DECL_CYCLE_COLLECTING_ISUPPORTS
+  NS_DECL_CYCLE_COLLECTION_WRAPPERCACHE_CLASS(RTCRtpReceiver)
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  // webidl
+  MediaStreamTrack* Track() const { return mTrack; }
+  RTCDtlsTransport* GetTransport() const;
+  static void GetCapabilities(const GlobalObject&, const nsAString& aKind,
+                              Nullable<dom::RTCRtpCapabilities>& aResult);
+  already_AddRefed<Promise> GetStats(ErrorResult& aError);
+  void GetContributingSources(
+      nsTArray<dom::RTCRtpContributingSource>& aSources);
+  void GetSynchronizationSources(
+      nsTArray<dom::RTCRtpSynchronizationSource>& aSources);
+  // test-only: insert fake CSRCs and audio levels for testing
+  void MozInsertAudioLevelForContributingSource(
+      const uint32_t aSource, const DOMHighResTimeStamp aTimestamp,
+      const uint32_t aRtpTimestamp, const bool aHasLevel, const uint8_t aLevel);
+
+  nsPIDOMWindowInner* GetParentObject() const;
+  nsTArray<RefPtr<RTCStatsPromise>> GetStatsInternal(
+      bool aSkipIceStats = false);
+  Nullable<DOMHighResTimeStamp> GetJitterBufferTarget(
+      ErrorResult& aError) const {
+    return mJitterBufferTarget.isSome() ? Nullable(mJitterBufferTarget.value())
+                                        : Nullable<DOMHighResTimeStamp>();
+  }
+  void SetJitterBufferTarget(const Nullable<DOMHighResTimeStamp>& aTargetMs,
+                             ErrorResult& aError);
+
+  void Shutdown();
+  void BreakCycles();
+  // Terminal state, reached through stopping RTCRtpTransceiver.
+  void Stop();
+  bool HasTrack(const dom::MediaStreamTrack* aTrack) const;
+  void SyncToJsep(JsepTransceiver& aJsepTransceiver) const;
+  void SyncFromJsep(const JsepTransceiver& aJsepTransceiver);
+  const std::vector<std::string>& GetStreamIds() const { return mStreamIds; }
+
+  struct StreamAssociation {
+    RefPtr<MediaStreamTrack> mTrack;
+    std::string mStreamId;
+  };
+
+  struct TrackEventInfo {
+    RefPtr<RTCRtpReceiver> mReceiver;
+    std::vector<std::string> mStreamIds;
+  };
+
+  struct StreamAssociationChanges {
+    std::vector<RefPtr<RTCRtpReceiver>> mReceiversToMute;
+    std::vector<StreamAssociation> mStreamAssociationsRemoved;
+    std::vector<StreamAssociation> mStreamAssociationsAdded;
+    std::vector<TrackEventInfo> mTrackEvents;
+  };
+
+  // This is called when we set an answer (ie; when the transport is finalized).
+  void UpdateTransport();
+  void UpdateConduit();
+
+  // This is called when we set a remote description; may be an offer or answer.
+  void UpdateStreams(StreamAssociationChanges* aChanges);
+
+  // Called when the privacy-needed state changes on the fly, as a result of
+  // ALPN negotiation.
+  void UpdatePrincipalPrivacy(PrincipalPrivacy aPrivacy);
+
+  void OnRtcpBye();
+  void OnRtcpTimeout();
+
+  void SetTrackMuteFromRemoteSdp();
+  void OnRtpPacket();
+  void UpdateUnmuteBlockingState();
+  void UpdateReceiveTrackMute();
+
+  AbstractCanonical<Ssrc>* CanonicalSsrc() { return &mSsrc; }
+  AbstractCanonical<Ssrc>* CanonicalVideoRtxSsrc() { return &mVideoRtxSsrc; }
+  AbstractCanonical<RtpExtList>* CanonicalLocalRtpExtensions() {
+    return &mLocalRtpExtensions;
+  }
+
+  AbstractCanonical<std::vector<AudioCodecConfig>>* CanonicalAudioCodecs() {
+    return &mAudioCodecs;
+  }
+
+  AbstractCanonical<std::vector<VideoCodecConfig>>* CanonicalVideoCodecs() {
+    return &mVideoCodecs;
+  }
+  AbstractCanonical<Maybe<RtpRtcpConfig>>* CanonicalVideoRtpRtcpConfig() {
+    return &mVideoRtpRtcpConfig;
+  }
+  AbstractCanonical<bool>* CanonicalReceiving() override { return &mReceiving; }
+
+ private:
+  virtual ~RTCRtpReceiver();
+
+  void UpdateVideoConduit();
+  void UpdateAudioConduit();
+
+  std::string GetMid() const;
+  JsepTransceiver& GetJsepTransceiver();
+  const JsepTransceiver& GetJsepTransceiver() const;
+
+  WatchManager<RTCRtpReceiver> mWatchManager;
+  nsCOMPtr<nsPIDOMWindowInner> mWindow;
+  RefPtr<PeerConnectionImpl> mPc;
+  bool mHaveStartedReceiving = false;
+  bool mHaveSetupTransport = false;
+  RefPtr<AbstractThread> mCallThread;
+  nsCOMPtr<nsISerialEventTarget> mStsThread;
+  RefPtr<dom::MediaStreamTrack> mTrack;
+  RefPtr<RemoteTrackSource> mTrackSource;
+  RefPtr<MediaPipelineReceive> mPipeline;
+  RefPtr<MediaTransportHandler> mTransportHandler;
+  RefPtr<RTCRtpTransceiver> mTransceiver;
+  // This is [[AssociatedRemoteMediaStreams]], basically. We do not keep the
+  // streams themselves here, because that would require this object to know
+  // where the stream list for the whole RTCPeerConnection lives..
+  std::vector<std::string> mStreamIds;
+  bool mRemoteSetSendBit = false;
+  Watchable<bool> mReceiveTrackMute{true, "RTCRtpReceiver::mReceiveTrackMute"};
+  // This corresponds to the [[Receptive]] slot on RTCRtpTransceiver.
+  // Its only purpose is suppressing unmute events if true.
+  bool mReceptive = false;
+  // This is the [[JitterBufferTarget]] internal slot.
+  Maybe<DOMHighResTimeStamp> mJitterBufferTarget;
+
+  MediaEventListener mRtcpByeListener;
+  MediaEventListener mRtcpTimeoutListener;
+  MediaEventListener mUnmuteListener;
+
+  Canonical<Ssrc> mSsrc;
+  Canonical<Ssrc> mVideoRtxSsrc;
+  Canonical<RtpExtList> mLocalRtpExtensions;
+  Canonical<std::vector<AudioCodecConfig>> mAudioCodecs;
+  Canonical<std::vector<VideoCodecConfig>> mVideoCodecs;
+  Canonical<Maybe<RtpRtcpConfig>> mVideoRtpRtcpConfig;
+  Canonical<bool> mReceiving;
+};
+
+}  // namespace dom
+}  // namespace mozilla
+#endif  // _RTCRtpReceiver_h_
diff --git a/dom/media/webrtc/jsapi/RTCRtpSender.cpp b/dom/media/webrtc/jsapi/RTCRtpSender.cpp
new file mode 100644
index 0000000000..568a83e8d1
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCRtpSender.cpp
@@ -0,0 +1,1654 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "RTCRtpSender.h"
+#include "transport/logging.h"
+#include "mozilla/dom/MediaStreamTrack.h"
+#include "mozilla/dom/Promise.h"
+#include "mozilla/glean/GleanMetrics.h"
+#include "nsPIDOMWindow.h"
+#include "nsString.h"
+#include "mozilla/dom/VideoStreamTrack.h"
+#include "jsep/JsepTransceiver.h"
+#include "mozilla/dom/RTCRtpSenderBinding.h"
+#include "RTCStatsReport.h"
+#include "mozilla/Preferences.h"
+#include "RTCRtpTransceiver.h"
+#include "PeerConnectionImpl.h"
+#include "libwebrtcglue/AudioConduit.h"
+#include <vector>
+#include "call/call.h"
+
+namespace mozilla::dom {
+
+LazyLogModule gSenderLog("RTCRtpSender");
+
+NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE_CLASS(RTCRtpSender)
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(RTCRtpSender)
+  // We do not do anything here, we wait for BreakCycles to be called
+  NS_IMPL_CYCLE_COLLECTION_UNLINK_PRESERVED_WRAPPER
+NS_IMPL_CYCLE_COLLECTION_UNLINK_END
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN(RTCRtpSender)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mWindow, mPc, mSenderTrack, mTransceiver,
+                                    mStreams, mDtmf)
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
+
+NS_IMPL_CYCLE_COLLECTING_ADDREF(RTCRtpSender)
+NS_IMPL_CYCLE_COLLECTING_RELEASE(RTCRtpSender)
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(RTCRtpSender)
+  NS_WRAPPERCACHE_INTERFACE_MAP_ENTRY
+  NS_INTERFACE_MAP_ENTRY(nsISupports)
+NS_INTERFACE_MAP_END
+
+#define INIT_CANONICAL(name, val) \
+  name(AbstractThread::MainThread(), val, "RTCRtpSender::" #name " (Canonical)")
+
+RTCRtpSender::RTCRtpSender(nsPIDOMWindowInner* aWindow, PeerConnectionImpl* aPc,
+                           MediaTransportHandler* aTransportHandler,
+                           AbstractThread* aCallThread,
+                           nsISerialEventTarget* aStsThread,
+                           MediaSessionConduit* aConduit,
+                           dom::MediaStreamTrack* aTrack,
+                           const Sequence<RTCRtpEncodingParameters>& aEncodings,
+                           RTCRtpTransceiver* aTransceiver)
+    : mWatchManager(this, AbstractThread::MainThread()),
+      mWindow(aWindow),
+      mPc(aPc),
+      mSenderTrack(aTrack),
+      mTransportHandler(aTransportHandler),
+      mTransceiver(aTransceiver),
+      INIT_CANONICAL(mSsrcs, Ssrcs()),
+      INIT_CANONICAL(mVideoRtxSsrcs, Ssrcs()),
+      INIT_CANONICAL(mLocalRtpExtensions, RtpExtList()),
+      INIT_CANONICAL(mAudioCodec, Nothing()),
+      INIT_CANONICAL(mVideoCodec, Nothing()),
+      INIT_CANONICAL(mVideoRtpRtcpConfig, Nothing()),
+      INIT_CANONICAL(mVideoCodecMode, webrtc::VideoCodecMode::kRealtimeVideo),
+      INIT_CANONICAL(mCname, std::string()),
+      INIT_CANONICAL(mTransmitting, false) {
+  mPipeline = new MediaPipelineTransmit(
+      mPc->GetHandle(), aTransportHandler, aCallThread, aStsThread,
+      aConduit->type() == MediaSessionConduit::VIDEO, aConduit);
+  mPipeline->InitControl(this);
+
+  if (aConduit->type() == MediaSessionConduit::AUDIO) {
+    mDtmf = new RTCDTMFSender(aWindow, mTransceiver);
+  }
+  mPipeline->SetTrack(mSenderTrack);
+
+  mozilla::glean::rtcrtpsender::count.Add(1);
+
+  if (mPc->ShouldAllowOldSetParameters()) {
+    mAllowOldSetParameters = true;
+    mozilla::glean::rtcrtpsender::count_setparameters_compat.Add(1);
+  }
+
+  if (aEncodings.Length()) {
+    // This sender was created by addTransceiver with sendEncodings.
+    mParameters.mEncodings = aEncodings;
+    mSimulcastEnvelopeSet = true;
+    mozilla::glean::rtcrtpsender::used_sendencodings.AddToNumerator(1);
+  } else {
+    // This sender was created by addTrack, sRD(offer), or addTransceiver
+    // without sendEncodings.
+    RTCRtpEncodingParameters defaultEncoding;
+    defaultEncoding.mActive = true;
+    if (aConduit->type() == MediaSessionConduit::VIDEO) {
+      defaultEncoding.mScaleResolutionDownBy.Construct(1.0f);
+    }
+    Unused << mParameters.mEncodings.AppendElement(defaultEncoding, fallible);
+    UpdateRestorableEncodings(mParameters.mEncodings);
+    MaybeGetJsepRids();
+  }
+
+  if (mDtmf) {
+    mWatchManager.Watch(mTransmitting, &RTCRtpSender::UpdateDtmfSender);
+  }
+}
+
+#undef INIT_CANONICAL
+
+RTCRtpSender::~RTCRtpSender() = default;
+
+JSObject* RTCRtpSender::WrapObject(JSContext* aCx,
+                                   JS::Handle<JSObject*> aGivenProto) {
+  return RTCRtpSender_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+RTCDtlsTransport* RTCRtpSender::GetTransport() const {
+  if (!mTransceiver) {
+    return nullptr;
+  }
+  return mTransceiver->GetDtlsTransport();
+}
+
+RTCDTMFSender* RTCRtpSender::GetDtmf() const { return mDtmf; }
+
+already_AddRefed<Promise> RTCRtpSender::GetStats(ErrorResult& aError) {
+  RefPtr<Promise> promise = MakePromise(aError);
+  if (aError.Failed()) {
+    return nullptr;
+  }
+  if (NS_WARN_IF(!mPipeline)) {
+    // TODO(bug 1056433): When we stop nulling this out when the PC is closed
+    // (or when the transceiver is stopped), we can remove this code. We
+    // resolve instead of reject in order to make this eventual change in
+    // behavior a little smaller.
+    promise->MaybeResolve(new RTCStatsReport(mWindow));
+    return promise.forget();
+  }
+
+  if (!mSenderTrack) {
+    promise->MaybeResolve(new RTCStatsReport(mWindow));
+    return promise.forget();
+  }
+
+  mTransceiver->ChainToDomPromiseWithCodecStats(GetStatsInternal(), promise);
+  return promise.forget();
+}
+
+nsTArray<RefPtr<dom::RTCStatsPromise>> RTCRtpSender::GetStatsInternal(
+    bool aSkipIceStats) {
+  MOZ_ASSERT(NS_IsMainThread());
+  nsTArray<RefPtr<RTCStatsPromise>> promises(2);
+  if (!mSenderTrack || !mPipeline) {
+    return promises;
+  }
+
+  nsAutoString trackName;
+  if (auto track = mPipeline->GetTrack()) {
+    track->GetId(trackName);
+  }
+
+  {
+    // Add bandwidth estimation stats
+    promises.AppendElement(InvokeAsync(
+        mPipeline->mCallThread, __func__,
+        [conduit = mPipeline->mConduit, trackName]() mutable {
+          auto report = MakeUnique<dom::RTCStatsCollection>();
+          Maybe<webrtc::Call::Stats> stats = conduit->GetCallStats();
+          stats.apply([&](const auto aStats) {
+            dom::RTCBandwidthEstimationInternal bw;
+            bw.mTrackIdentifier = trackName;
+            bw.mSendBandwidthBps.Construct(aStats.send_bandwidth_bps / 8);
+            bw.mMaxPaddingBps.Construct(aStats.max_padding_bitrate_bps / 8);
+            bw.mReceiveBandwidthBps.Construct(aStats.recv_bandwidth_bps / 8);
+            bw.mPacerDelayMs.Construct(aStats.pacer_delay_ms);
+            if (aStats.rtt_ms >= 0) {
+              bw.mRttMs.Construct(aStats.rtt_ms);
+            }
+            if (!report->mBandwidthEstimations.AppendElement(std::move(bw),
+                                                             fallible)) {
+              mozalloc_handle_oom(0);
+            }
+          });
+          return RTCStatsPromise::CreateAndResolve(std::move(report), __func__);
+        }));
+  }
+
+  promises.AppendElement(InvokeAsync(
+      mPipeline->mCallThread, __func__, [pipeline = mPipeline, trackName] {
+        auto report = MakeUnique<dom::RTCStatsCollection>();
+        auto asAudio = pipeline->mConduit->AsAudioSessionConduit();
+        auto asVideo = pipeline->mConduit->AsVideoSessionConduit();
+
+        nsString kind = asVideo.isNothing() ? u"audio"_ns : u"video"_ns;
+        nsString idstr = kind + u"_"_ns;
+        idstr.AppendInt(static_cast<uint32_t>(pipeline->Level()));
+
+        for (uint32_t ssrc : pipeline->mConduit->GetLocalSSRCs()) {
+          nsString localId = u"outbound_rtp_"_ns + idstr + u"_"_ns;
+          localId.AppendInt(ssrc);
+          nsString remoteId;
+          Maybe<uint16_t> base_seq =
+              pipeline->mConduit->RtpSendBaseSeqFor(ssrc);
+
+          auto constructCommonRemoteInboundRtpStats =
+              [&](RTCRemoteInboundRtpStreamStats& aRemote,
+                  const webrtc::ReportBlockData& aRtcpData) {
+                remoteId = u"outbound_rtcp_"_ns + idstr + u"_"_ns;
+                remoteId.AppendInt(ssrc);
+                aRemote.mTimestamp.Construct(
+                    RTCStatsTimestamp::FromNtp(
+                        pipeline->GetTimestampMaker(),
+                        webrtc::Timestamp::Micros(
+                            aRtcpData.report_block_timestamp_utc_us()) +
+                            webrtc::TimeDelta::Seconds(webrtc::kNtpJan1970))
+                        .ToDom());
+                aRemote.mId.Construct(remoteId);
+                aRemote.mType.Construct(RTCStatsType::Remote_inbound_rtp);
+                aRemote.mSsrc = ssrc;
+                aRemote.mKind = kind;
+                aRemote.mMediaType.Construct(
+                    kind);  // mediaType is the old name for kind.
+                aRemote.mLocalId.Construct(localId);
+                if (base_seq) {
+                  if (aRtcpData.report_block()
+                          .extended_highest_sequence_number < *base_seq) {
+                    aRemote.mPacketsReceived.Construct(0);
+                  } else {
+                    aRemote.mPacketsReceived.Construct(
+                        aRtcpData.report_block()
+                            .extended_highest_sequence_number -
+                        aRtcpData.report_block().packets_lost - *base_seq + 1);
+                  }
+                }
+              };
+
+          auto constructCommonOutboundRtpStats =
+              [&](RTCOutboundRtpStreamStats& aLocal) {
+                aLocal.mSsrc = ssrc;
+                aLocal.mTimestamp.Construct(
+                    pipeline->GetTimestampMaker().GetNow().ToDom());
+                aLocal.mId.Construct(localId);
+                aLocal.mType.Construct(RTCStatsType::Outbound_rtp);
+                aLocal.mKind = kind;
+                aLocal.mMediaType.Construct(
+                    kind);  // mediaType is the old name for kind.
+                if (remoteId.Length()) {
+                  aLocal.mRemoteId.Construct(remoteId);
+                }
+              };
+
+          asAudio.apply([&](auto& aConduit) {
+            Maybe<webrtc::AudioSendStream::Stats> audioStats =
+                aConduit->GetSenderStats();
+            if (audioStats.isNothing()) {
+              return;
+            }
+
+            if (audioStats->packets_sent == 0) {
+              // By spec: "The lifetime of all RTP monitored objects starts
+              // when the RTP stream is first used: When the first RTP packet
+              // is sent or received on the SSRC it represents"
+              return;
+            }
+
+            // First, fill in remote stat with rtcp receiver data, if present.
+            // ReceiverReports have less information than SenderReports, so fill
+            // in what we can.
+            Maybe<webrtc::ReportBlockData> reportBlockData;
+            {
+              if (const auto remoteSsrc = aConduit->GetRemoteSSRC();
+                  remoteSsrc) {
+                for (auto& data : audioStats->report_block_datas) {
+                  if (data.report_block().source_ssrc == ssrc &&
+                      data.report_block().sender_ssrc == *remoteSsrc) {
+                    reportBlockData.emplace(data);
+                    break;
+                  }
+                }
+              }
+            }
+            reportBlockData.apply([&](auto& aReportBlockData) {
+              RTCRemoteInboundRtpStreamStats remote;
+              constructCommonRemoteInboundRtpStats(remote, aReportBlockData);
+              if (audioStats->jitter_ms >= 0) {
+                remote.mJitter.Construct(audioStats->jitter_ms / 1000.0);
+              }
+              if (audioStats->packets_lost >= 0) {
+                remote.mPacketsLost.Construct(audioStats->packets_lost);
+              }
+              if (audioStats->rtt_ms >= 0) {
+                remote.mRoundTripTime.Construct(
+                    static_cast<double>(audioStats->rtt_ms) / 1000.0);
+              }
+              remote.mFractionLost.Construct(audioStats->fraction_lost);
+              remote.mTotalRoundTripTime.Construct(
+                  double(aReportBlockData.sum_rtt_ms()) / 1000);
+              remote.mRoundTripTimeMeasurements.Construct(
+                  aReportBlockData.num_rtts());
+              if (!report->mRemoteInboundRtpStreamStats.AppendElement(
+                      std::move(remote), fallible)) {
+                mozalloc_handle_oom(0);
+              }
+            });
+
+            // Then, fill in local side (with cross-link to remote only if
+            // present)
+            RTCOutboundRtpStreamStats local;
+            constructCommonOutboundRtpStats(local);
+            local.mPacketsSent.Construct(audioStats->packets_sent);
+            local.mBytesSent.Construct(audioStats->payload_bytes_sent);
+            local.mNackCount.Construct(
+                audioStats->rtcp_packet_type_counts.nack_packets);
+            local.mHeaderBytesSent.Construct(
+                audioStats->header_and_padding_bytes_sent);
+            local.mRetransmittedPacketsSent.Construct(
+                audioStats->retransmitted_packets_sent);
+            local.mRetransmittedBytesSent.Construct(
+                audioStats->retransmitted_bytes_sent);
+            /*
+             * Potential new stats that are now available upstream.
+             * Note: when we last tried exposing this we were getting
+             * targetBitrate for audio was ending up as 0. We did not
+             * investigate why.
+            local.mTargetBitrate.Construct(audioStats->target_bitrate_bps);
+             */
+            if (!report->mOutboundRtpStreamStats.AppendElement(std::move(local),
+                                                               fallible)) {
+              mozalloc_handle_oom(0);
+            }
+          });
+
+          asVideo.apply([&](auto& aConduit) {
+            Maybe<webrtc::VideoSendStream::Stats> videoStats =
+                aConduit->GetSenderStats();
+            if (videoStats.isNothing()) {
+              return;
+            }
+
+            Maybe<webrtc::VideoSendStream::StreamStats> streamStats;
+            auto kv = videoStats->substreams.find(ssrc);
+            if (kv != videoStats->substreams.end()) {
+              streamStats = Some(kv->second);
+            }
+
+            if (!streamStats) {
+              // By spec: "The lifetime of all RTP monitored objects starts
+              // when the RTP stream is first used: When the first RTP packet
+              // is sent or received on the SSRC it represents"
+              return;
+            }
+
+            aConduit->GetAssociatedLocalRtxSSRC(ssrc).apply(
+                [&](const auto rtxSsrc) {
+                  auto kv = videoStats->substreams.find(rtxSsrc);
+                  if (kv != videoStats->substreams.end()) {
+                    streamStats->rtp_stats.Add(kv->second.rtp_stats);
+                  }
+                });
+
+            if (streamStats->rtp_stats.first_packet_time_ms == -1) {
+              return;
+            }
+
+            // First, fill in remote stat with rtcp receiver data, if present.
+            // ReceiverReports have less information than SenderReports, so fill
+            // in what we can.
+            if (streamStats->report_block_data) {
+              const webrtc::ReportBlockData& rtcpReportData =
+                  *streamStats->report_block_data;
+              RTCRemoteInboundRtpStreamStats remote;
+              remote.mJitter.Construct(
+                  static_cast<double>(rtcpReportData.report_block().jitter) /
+                  webrtc::kVideoPayloadTypeFrequency);
+              remote.mPacketsLost.Construct(
+                  rtcpReportData.report_block().packets_lost);
+              if (rtcpReportData.has_rtt()) {
+                remote.mRoundTripTime.Construct(
+                    static_cast<double>(rtcpReportData.last_rtt_ms()) / 1000.0);
+              }
+              constructCommonRemoteInboundRtpStats(remote, rtcpReportData);
+              remote.mTotalRoundTripTime.Construct(
+                  streamStats->report_block_data->sum_rtt_ms() / 1000.0);
+              remote.mFractionLost.Construct(
+                  static_cast<float>(
+                      rtcpReportData.report_block().fraction_lost) /
+                  (1 << 8));
+              remote.mRoundTripTimeMeasurements.Construct(
+                  streamStats->report_block_data->num_rtts());
+              if (!report->mRemoteInboundRtpStreamStats.AppendElement(
+                      std::move(remote), fallible)) {
+                mozalloc_handle_oom(0);
+              }
+            }
+
+            // Then, fill in local side (with cross-link to remote only if
+            // present)
+            RTCOutboundRtpStreamStats local;
+            constructCommonOutboundRtpStats(local);
+            local.mPacketsSent.Construct(
+                streamStats->rtp_stats.transmitted.packets);
+            local.mBytesSent.Construct(
+                streamStats->rtp_stats.transmitted.payload_bytes);
+            local.mNackCount.Construct(
+                streamStats->rtcp_packet_type_counts.nack_packets);
+            local.mFirCount.Construct(
+                streamStats->rtcp_packet_type_counts.fir_packets);
+            local.mPliCount.Construct(
+                streamStats->rtcp_packet_type_counts.pli_packets);
+            local.mFramesEncoded.Construct(streamStats->frames_encoded);
+            if (streamStats->qp_sum) {
+              local.mQpSum.Construct(*streamStats->qp_sum);
+            }
+            local.mHeaderBytesSent.Construct(
+                streamStats->rtp_stats.transmitted.header_bytes +
+                streamStats->rtp_stats.transmitted.padding_bytes);
+            local.mRetransmittedPacketsSent.Construct(
+                streamStats->rtp_stats.retransmitted.packets);
+            local.mRetransmittedBytesSent.Construct(
+                streamStats->rtp_stats.retransmitted.payload_bytes);
+            local.mTotalEncodedBytesTarget.Construct(
+                videoStats->total_encoded_bytes_target);
+            local.mFrameWidth.Construct(streamStats->width);
+            local.mFrameHeight.Construct(streamStats->height);
+            local.mFramesSent.Construct(streamStats->frames_encoded);
+            local.mHugeFramesSent.Construct(streamStats->huge_frames_sent);
+            local.mTotalEncodeTime.Construct(
+                double(streamStats->total_encode_time_ms) / 1000.);
+            /*
+             * Potential new stats that are now available upstream.
+            local.mTargetBitrate.Construct(videoStats->target_media_bitrate_bps);
+             */
+            if (!report->mOutboundRtpStreamStats.AppendElement(std::move(local),
+                                                               fallible)) {
+              mozalloc_handle_oom(0);
+            }
+          });
+        }
+
+        auto constructCommonMediaSourceStats =
+            [&](RTCMediaSourceStats& aStats) {
+              nsString id = u"mediasource_"_ns + idstr + trackName;
+              aStats.mTimestamp.Construct(
+                  pipeline->GetTimestampMaker().GetNow().ToDom());
+              aStats.mId.Construct(id);
+              aStats.mType.Construct(RTCStatsType::Media_source);
+              aStats.mTrackIdentifier = trackName;
+              aStats.mKind = kind;
+            };
+
+        // TODO(bug 1804678): Use RTCAudioSourceStats/RTCVideoSourceStats
+        RTCMediaSourceStats mediaSourceStats;
+        constructCommonMediaSourceStats(mediaSourceStats);
+        if (!report->mMediaSourceStats.AppendElement(
+                std::move(mediaSourceStats), fallible)) {
+          mozalloc_handle_oom(0);
+        }
+
+        return RTCStatsPromise::CreateAndResolve(std::move(report), __func__);
+      }));
+
+  if (!aSkipIceStats && GetJsepTransceiver().mTransport.mComponents) {
+    promises.AppendElement(mTransportHandler->GetIceStats(
+        GetJsepTransceiver().mTransport.mTransportId,
+        mPipeline->GetTimestampMaker().GetNow().ToDom()));
+  }
+
+  return promises;
+}
+
+void RTCRtpSender::GetCapabilities(const GlobalObject&, const nsAString& aKind,
+                                   Nullable<dom::RTCRtpCapabilities>& aResult) {
+  PeerConnectionImpl::GetCapabilities(aKind, aResult, sdp::Direction::kSend);
+}
+
+void RTCRtpSender::WarnAboutBadSetParameters(const nsCString& aError) {
+  nsCString warning(
+      "WARNING! Invalid setParameters call detected! The good news? Firefox "
+      "supports sendEncodings in addTransceiver now, so we ask that you switch "
+      "over to using the parameters code you use for other browsers. Thank you "
+      "for your patience and support. The specific error was: ");
+  warning += aError;
+  mPc->SendWarningToConsole(warning);
+}
+
+nsCString RTCRtpSender::GetEffectiveTLDPlus1() const {
+  return mPc->GetEffectiveTLDPlus1();
+}
+
+already_AddRefed<Promise> RTCRtpSender::SetParameters(
+    const dom::RTCRtpSendParameters& aParameters, ErrorResult& aError) {
+  dom::RTCRtpSendParameters paramsCopy(aParameters);
+  // When the setParameters method is called, the user agent MUST run the
+  // following steps:
+  // Let parameters be the method's first argument.
+  // Let sender be the RTCRtpSender object on which setParameters is invoked.
+  // Let transceiver be the RTCRtpTransceiver object associated with sender
+  // (i.e.sender is transceiver.[[Sender]]).
+
+  RefPtr<dom::Promise> p = MakePromise(aError);
+  if (aError.Failed()) {
+    return nullptr;
+  }
+
+  if (mPc->IsClosed()) {
+    p->MaybeRejectWithInvalidStateError("Peer connection is closed");
+    return p.forget();
+  }
+
+  // If transceiver.[[Stopped]] is true, return a promise rejected with a newly
+  // created InvalidStateError.
+  if (mTransceiver->Stopped()) {
+    p->MaybeRejectWithInvalidStateError("This sender's transceiver is stopped");
+    return p.forget();
+  }
+
+  // If sender.[[LastReturnedParameters]] is null, return a promise rejected
+  // with a newly created InvalidStateError.
+  if (!mLastReturnedParameters.isSome()) {
+    nsCString error(
+        "Cannot call setParameters without first calling getParameters");
+    if (mAllowOldSetParameters) {
+      if (!mHaveWarnedBecauseNoGetParameters) {
+        mHaveWarnedBecauseNoGetParameters = true;
+        mozilla::glean::rtcrtpsender_setparameters::warn_no_getparameters
+            .AddToNumerator(1);
+#ifdef EARLY_BETA_OR_EARLIER
+        mozilla::glean::rtcrtpsender_setparameters::blame_no_getparameters
+            .Get(GetEffectiveTLDPlus1())
+            .Add(1);
+#endif
+      }
+      WarnAboutBadSetParameters(error);
+    } else {
+      if (!mHaveFailedBecauseNoGetParameters) {
+        mHaveFailedBecauseNoGetParameters = true;
+        mozilla::glean::rtcrtpsender_setparameters::fail_no_getparameters
+            .AddToNumerator(1);
+      }
+      p->MaybeRejectWithInvalidStateError(error);
+      return p.forget();
+    }
+  }
+
+  // According to the spec, our consistency checking is based on
+  // [[LastReturnedParameters]], but if we're letting
+  // [[LastReturnedParameters]]==null slide, we still want to do
+  // consistency checking on _something_ so we can warn implementers if they
+  // are messing that up also. Just find something, _anything_, to do that
+  // checking with.
+  // TODO(bug 1803388): Remove this stuff once it is no longer needed.
+  // TODO(bug 1803389): Remove the glean errors once they are no longer needed.
+  Maybe<RTCRtpSendParameters> oldParams;
+  if (mAllowOldSetParameters) {
+    if (mLastReturnedParameters.isSome()) {
+      oldParams = mLastReturnedParameters;
+    } else if (mPendingParameters.isSome()) {
+      oldParams = mPendingParameters;
+    } else {
+      oldParams = Some(mParameters);
+    }
+    MOZ_ASSERT(oldParams.isSome());
+  } else {
+    oldParams = mLastReturnedParameters;
+  }
+  MOZ_ASSERT(oldParams.isSome());
+
+  // Validate parameters by running the following steps:
+  // Let encodings be parameters.encodings.
+  // Let codecs be parameters.codecs.
+  // Let N be the number of RTCRtpEncodingParameters stored in
+  // sender.[[SendEncodings]].
+  // If any of the following conditions are met,
+  // return a promise rejected with a newly created InvalidModificationError:
+
+  bool pendingRidChangeFromCompatMode = false;
+  // encodings.length is different from N.
+  if (paramsCopy.mEncodings.Length() != oldParams->mEncodings.Length()) {
+    nsCString error("Cannot change the number of encodings with setParameters");
+    if (!mAllowOldSetParameters) {
+      if (!mHaveFailedBecauseEncodingCountChange) {
+        mHaveFailedBecauseEncodingCountChange = true;
+        mozilla::glean::rtcrtpsender_setparameters::fail_length_changed
+            .AddToNumerator(1);
+      }
+      p->MaybeRejectWithInvalidModificationError(error);
+      return p.forget();
+    }
+    // Make sure we don't use the old rids in SyncToJsep while we wait for the
+    // queued task below to update mParameters.
+    pendingRidChangeFromCompatMode = true;
+    mSimulcastEnvelopeSet = true;
+    if (!mHaveWarnedBecauseEncodingCountChange) {
+      mHaveWarnedBecauseEncodingCountChange = true;
+      mozilla::glean::rtcrtpsender_setparameters::warn_length_changed
+          .AddToNumerator(1);
+#ifdef EARLY_BETA_OR_EARLIER
+      mozilla::glean::rtcrtpsender_setparameters::blame_length_changed
+          .Get(GetEffectiveTLDPlus1())
+          .Add(1);
+#endif
+    }
+    WarnAboutBadSetParameters(error);
+  } else {
+    // encodings has been re-ordered.
+    for (size_t i = 0; i < paramsCopy.mEncodings.Length(); ++i) {
+      const auto& oldEncoding = oldParams->mEncodings[i];
+      const auto& newEncoding = paramsCopy.mEncodings[i];
+      if (oldEncoding.mRid != newEncoding.mRid) {
+        nsCString error("Cannot change rid, or reorder encodings");
+        if (!mHaveFailedBecauseRidChange) {
+          mHaveFailedBecauseRidChange = true;
+          mozilla::glean::rtcrtpsender_setparameters::fail_rid_changed
+              .AddToNumerator(1);
+        }
+        p->MaybeRejectWithInvalidModificationError(error);
+        return p.forget();
+      }
+    }
+  }
+
+  // TODO(bug 1803388): Handle this in webidl, once we stop allowing the old
+  // setParameters style.
+  if (!paramsCopy.mTransactionId.WasPassed()) {
+    nsCString error("transactionId is not set!");
+    if (!mAllowOldSetParameters) {
+      if (!mHaveFailedBecauseNoTransactionId) {
+        mHaveFailedBecauseNoTransactionId = true;
+        mozilla::glean::rtcrtpsender_setparameters::fail_no_transactionid
+            .AddToNumerator(1);
+      }
+      p->MaybeRejectWithTypeError(error);
+      return p.forget();
+    }
+    if (!mHaveWarnedBecauseNoTransactionId) {
+      mHaveWarnedBecauseNoTransactionId = true;
+      mozilla::glean::rtcrtpsender_setparameters::warn_no_transactionid
+          .AddToNumerator(1);
+#ifdef EARLY_BETA_OR_EARLIER
+      mozilla::glean::rtcrtpsender_setparameters::blame_no_transactionid
+          .Get(GetEffectiveTLDPlus1())
+          .Add(1);
+#endif
+    }
+    WarnAboutBadSetParameters(error);
+  } else if (oldParams->mTransactionId != paramsCopy.mTransactionId) {
+    // Any parameter in parameters is marked as a Read-only parameter (such as
+    // RID) and has a value that is different from the corresponding parameter
+    // value in sender.[[LastReturnedParameters]]. Note that this also applies
+    // to transactionId.
+    nsCString error(
+        "Cannot change transaction id: call getParameters, modify the result, "
+        "and then call setParameters");
+    if (!mAllowOldSetParameters) {
+      if (!mHaveFailedBecauseStaleTransactionId) {
+        mHaveFailedBecauseStaleTransactionId = true;
+        mozilla::glean::rtcrtpsender_setparameters::fail_stale_transactionid
+            .AddToNumerator(1);
+      }
+      p->MaybeRejectWithInvalidModificationError(error);
+      return p.forget();
+    }
+    if (!mHaveWarnedBecauseStaleTransactionId) {
+      mHaveWarnedBecauseStaleTransactionId = true;
+      mozilla::glean::rtcrtpsender_setparameters::warn_stale_transactionid
+          .AddToNumerator(1);
+#ifdef EARLY_BETA_OR_EARLIER
+      mozilla::glean::rtcrtpsender_setparameters::blame_stale_transactionid
+          .Get(GetEffectiveTLDPlus1())
+          .Add(1);
+#endif
+    }
+    WarnAboutBadSetParameters(error);
+  }
+
+  // This could conceivably happen if we are allowing the old setParameters
+  // behavior.
+  if (!paramsCopy.mEncodings.Length()) {
+    nsCString error("Cannot set an empty encodings array");
+    if (!mAllowOldSetParameters) {
+      if (!mHaveFailedBecauseNoEncodings) {
+        mHaveFailedBecauseNoEncodings = true;
+        mozilla::glean::rtcrtpsender_setparameters::fail_no_encodings
+            .AddToNumerator(1);
+      }
+
+      p->MaybeRejectWithInvalidModificationError(error);
+      return p.forget();
+    }
+    // TODO: Add some warning telemetry here
+    WarnAboutBadSetParameters(error);
+    // Just don't do this; it's stupid.
+    paramsCopy.mEncodings = oldParams->mEncodings;
+  }
+
+  // TODO: Verify remaining read-only parameters
+  // headerExtensions (bug 1765851)
+  // rtcp (bug 1765852)
+  // codecs (bug 1534687)
+
+  // CheckAndRectifyEncodings handles the following steps:
+  // If transceiver kind is "audio", remove the scaleResolutionDownBy member
+  // from all encodings that contain one.
+  //
+  // If transceiver kind is "video", and any encoding in encodings contains a
+  // scaleResolutionDownBy member whose value is less than 1.0, return a
+  // promise rejected with a newly created RangeError.
+  //
+  // Verify that each encoding in encodings has a maxFramerate member whose
+  // value is greater than or equal to 0.0. If one of the maxFramerate values
+  // does not meet this requirement, return a promise rejected with a newly
+  // created RangeError.
+  ErrorResult rv;
+  CheckAndRectifyEncodings(paramsCopy.mEncodings, mTransceiver->IsVideo(), rv);
+  if (rv.Failed()) {
+    if (!mHaveFailedBecauseOtherError) {
+      mHaveFailedBecauseOtherError = true;
+      mozilla::glean::rtcrtpsender_setparameters::fail_other.AddToNumerator(1);
+    }
+    p->MaybeReject(std::move(rv));
+    return p.forget();
+  }
+
+  // If transceiver kind is "video", then for each encoding in encodings that
+  // doesn't contain a scaleResolutionDownBy member, add a
+  // scaleResolutionDownBy member with the value 1.0.
+  if (mTransceiver->IsVideo()) {
+    for (auto& encoding : paramsCopy.mEncodings) {
+      if (!encoding.mScaleResolutionDownBy.WasPassed()) {
+        encoding.mScaleResolutionDownBy.Construct(1.0);
+      }
+    }
+  }
+
+  // Let p be a new promise. (see above)
+
+  // In parallel, configure the media stack to use parameters to transmit
+  // sender.[[SenderTrack]].
+  // Right now this is infallible. That may change someday.
+
+  // We need to put this in a member variable, since MaybeUpdateConduit needs it
+  // This also allows PeerConnectionImpl to detect when there is a pending
+  // setParameters, which has implcations for the handling of
+  // setRemoteDescription.
+  mPendingRidChangeFromCompatMode = pendingRidChangeFromCompatMode;
+  mPendingParameters = Some(paramsCopy);
+  uint32_t serialNumber = ++mNumSetParametersCalls;
+  MaybeUpdateConduit();
+
+  // If the media stack is successfully configured with parameters,
+  // queue a task to run the following steps:
+  GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction(
+      __func__,
+      [this, self = RefPtr<RTCRtpSender>(this), p, paramsCopy, serialNumber] {
+        // Set sender.[[LastReturnedParameters]] to null.
+        mLastReturnedParameters = Nothing();
+        // Set sender.[[SendEncodings]] to parameters.encodings.
+        mParameters = paramsCopy;
+        UpdateRestorableEncodings(mParameters.mEncodings);
+        // Only clear mPendingParameters if it matches; there could have been
+        // back-to-back calls to setParameters, and we only want to clear this
+        // if no subsequent setParameters is pending.
+        if (serialNumber == mNumSetParametersCalls) {
+          mPendingParameters = Nothing();
+          // Ok, nothing has called SyncToJsep while this async task was
+          // pending. No need for special handling anymore.
+          mPendingRidChangeFromCompatMode = false;
+        }
+        MOZ_ASSERT(mParameters.mEncodings.Length());
+        // Resolve p with undefined.
+        p->MaybeResolveWithUndefined();
+      }));
+
+  // Return p.
+  return p.forget();
+}
+
+// static
+void RTCRtpSender::CheckAndRectifyEncodings(
+    Sequence<RTCRtpEncodingParameters>& aEncodings, bool aVideo,
+    ErrorResult& aRv) {
+  // If any encoding contains a rid member whose value does not conform to the
+  // grammar requirements specified in Section 10 of [RFC8851], throw a
+  // TypeError.
+  for (const auto& encoding : aEncodings) {
+    if (encoding.mRid.WasPassed()) {
+      std::string utf8Rid = NS_ConvertUTF16toUTF8(encoding.mRid.Value()).get();
+      std::string error;
+      if (!SdpRidAttributeList::CheckRidValidity(utf8Rid, &error)) {
+        aRv.ThrowTypeError(nsCString(error));
+        return;
+      }
+      if (utf8Rid.size() > SdpRidAttributeList::kMaxRidLength) {
+        std::ostringstream ss;
+        ss << "Rid can be at most " << SdpRidAttributeList::kMaxRidLength
+           << " characters long (due to internal limitations)";
+        aRv.ThrowTypeError(nsCString(ss.str()));
+        return;
+      }
+    }
+  }
+
+  if (aEncodings.Length() > 1) {
+    // If some but not all encodings contain a rid member, throw a TypeError.
+    // rid must be set if there is more than one encoding
+    // NOTE: Since rid is read-only, and the number of encodings cannot grow,
+    // this should never happen in setParameters.
+    for (const auto& encoding : aEncodings) {
+      if (!encoding.mRid.WasPassed()) {
+        aRv.ThrowTypeError("Missing rid");
+        return;
+      }
+    }
+
+    // If any encoding contains a rid member whose value is the same as that of
+    // a rid contained in another encoding in sendEncodings, throw a TypeError.
+    // NOTE: Since rid is read-only, and the number of encodings cannot grow,
+    // this should never happen in setParameters.
+    std::set<nsString> uniqueRids;
+    for (const auto& encoding : aEncodings) {
+      if (uniqueRids.count(encoding.mRid.Value())) {
+        aRv.ThrowTypeError("Duplicate rid");
+        return;
+      }
+      uniqueRids.insert(encoding.mRid.Value());
+    }
+  }
+  // TODO: ptime/adaptivePtime validation (bug 1733647)
+
+  // If kind is "audio", remove the scaleResolutionDownBy member from all
+  // encodings that contain one.
+  if (!aVideo) {
+    for (auto& encoding : aEncodings) {
+      if (encoding.mScaleResolutionDownBy.WasPassed()) {
+        encoding.mScaleResolutionDownBy.Reset();
+      }
+      if (encoding.mMaxFramerate.WasPassed()) {
+        encoding.mMaxFramerate.Reset();
+      }
+    }
+  }
+
+  // If any encoding contains a scaleResolutionDownBy member whose value is
+  // less than 1.0, throw a RangeError.
+  for (const auto& encoding : aEncodings) {
+    if (encoding.mScaleResolutionDownBy.WasPassed()) {
+      if (encoding.mScaleResolutionDownBy.Value() < 1.0f) {
+        aRv.ThrowRangeError("scaleResolutionDownBy must be >= 1.0");
+        return;
+      }
+    }
+  }
+
+  // Verify that the value of each maxFramerate member in sendEncodings that is
+  // defined is greater than 0.0. If one of the maxFramerate values does not
+  // meet this requirement, throw a RangeError.
+  for (const auto& encoding : aEncodings) {
+    if (encoding.mMaxFramerate.WasPassed()) {
+      if (encoding.mMaxFramerate.Value() < 0.0f) {
+        aRv.ThrowRangeError("maxFramerate must be non-negative");
+        return;
+      }
+    }
+  }
+}
+
+void RTCRtpSender::GetParameters(RTCRtpSendParameters& aParameters) {
+  MOZ_ASSERT(mParameters.mEncodings.Length());
+  // If sender.[[LastReturnedParameters]] is not null, return
+  // sender.[[LastReturnedParameters]], and abort these steps.
+  if (mLastReturnedParameters.isSome()) {
+    aParameters = *mLastReturnedParameters;
+    return;
+  }
+
+  // Let result be a new RTCRtpSendParameters dictionary constructed as follows:
+
+  // transactionId is set to a new unique identifier
+  aParameters.mTransactionId.Construct(mPc->GenerateUUID());
+
+  // encodings is set to the value of the [[SendEncodings]] internal slot.
+  aParameters.mEncodings = mParameters.mEncodings;
+
+  // The headerExtensions sequence is populated based on the header extensions
+  // that have been negotiated for sending
+  // TODO(bug 1765851): We do not support this yet
+  // aParameters.mHeaderExtensions.Construct();
+
+  // codecs is set to the value of the [[SendCodecs]] internal slot
+  // TODO(bug 1534687): We do not support this yet
+
+  // rtcp.cname is set to the CNAME of the associated RTCPeerConnection.
+  // rtcp.reducedSize is set to true if reduced-size RTCP has been negotiated
+  // for sending, and false otherwise.
+  // TODO(bug 1765852): We do not support this yet
+  aParameters.mRtcp.Construct();
+  aParameters.mRtcp.Value().mCname.Construct();
+  aParameters.mRtcp.Value().mReducedSize.Construct(false);
+  aParameters.mHeaderExtensions.Construct();
+  aParameters.mCodecs.Construct();
+
+  // Set sender.[[LastReturnedParameters]] to result.
+  mLastReturnedParameters = Some(aParameters);
+
+  // Queue a task that sets sender.[[LastReturnedParameters]] to null.
+  GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction(
+      __func__, [this, self = RefPtr<RTCRtpSender>(this)] {
+        mLastReturnedParameters = Nothing();
+      }));
+}
+
+bool operator==(const RTCRtpEncodingParameters& a1,
+                const RTCRtpEncodingParameters& a2) {
+  // webidl does not generate types that are equality comparable
+  return a1.mActive == a2.mActive && a1.mFec == a2.mFec &&
+         a1.mMaxBitrate == a2.mMaxBitrate &&
+         a1.mMaxFramerate == a2.mMaxFramerate && a1.mPriority == a2.mPriority &&
+         a1.mRid == a2.mRid && a1.mRtx == a2.mRtx &&
+         a1.mScaleResolutionDownBy == a2.mScaleResolutionDownBy &&
+         a1.mSsrc == a2.mSsrc;
+}
+
+// static
+void RTCRtpSender::ApplyJsEncodingToConduitEncoding(
+    const RTCRtpEncodingParameters& aJsEncoding,
+    VideoCodecConfig::Encoding* aConduitEncoding) {
+  aConduitEncoding->active = aJsEncoding.mActive;
+  if (aJsEncoding.mMaxBitrate.WasPassed()) {
+    aConduitEncoding->constraints.maxBr = aJsEncoding.mMaxBitrate.Value();
+  }
+  if (aJsEncoding.mMaxFramerate.WasPassed()) {
+    aConduitEncoding->constraints.maxFps =
+        Some(aJsEncoding.mMaxFramerate.Value());
+  }
+  if (aJsEncoding.mScaleResolutionDownBy.WasPassed()) {
+    // Optional does not have a valueOr, despite being based on Maybe
+    // :(
+    aConduitEncoding->constraints.scaleDownBy =
+        aJsEncoding.mScaleResolutionDownBy.Value();
+  } else {
+    aConduitEncoding->constraints.scaleDownBy = 1.0f;
+  }
+}
+
+void RTCRtpSender::UpdateRestorableEncodings(
+    const Sequence<RTCRtpEncodingParameters>& aEncodings) {
+  MOZ_ASSERT(aEncodings.Length());
+
+  if (GetJsepTransceiver().mSendTrack.GetNegotiatedDetails()) {
+    // Once initial negotiation completes, we are no longer allowed to restore
+    // the unicast encoding.
+    mUnicastEncoding.reset();
+  } else if (mParameters.mEncodings.Length() == 1 &&
+             !mParameters.mEncodings[0].mRid.WasPassed()) {
+    // If we have not completed the initial negotiation, and we currently are
+    // ridless unicast, we need to save our unicast encoding in case a
+    // rollback occurs.
+    mUnicastEncoding = Some(mParameters.mEncodings[0]);
+  }
+}
+
+Sequence<RTCRtpEncodingParameters> RTCRtpSender::ToSendEncodings(
+    const std::vector<std::string>& aRids) const {
+  MOZ_ASSERT(!aRids.empty());
+
+  Sequence<RTCRtpEncodingParameters> result;
+  // If sendEncodings is given as input to this algorithm, and is non-empty,
+  // set the [[SendEncodings]] slot to sendEncodings.
+  for (const auto& rid : aRids) {
+    MOZ_ASSERT(!rid.empty());
+    RTCRtpEncodingParameters encoding;
+    encoding.mActive = true;
+    encoding.mRid.Construct(NS_ConvertUTF8toUTF16(rid.c_str()));
+    Unused << result.AppendElement(encoding, fallible);
+  }
+
+  // If sendEncodings is non-empty, set each encoding's scaleResolutionDownBy
+  // to 2^(length of sendEncodings - encoding index - 1).
+  if (mTransceiver->IsVideo()) {
+    double scale = 1.0f;
+    for (auto it = result.rbegin(); it != result.rend(); ++it) {
+      it->mScaleResolutionDownBy.Construct(scale);
+      scale *= 2;
+    }
+  }
+
+  return result;
+}
+
+void RTCRtpSender::MaybeGetJsepRids() {
+  MOZ_ASSERT(!mSimulcastEnvelopeSet);
+  MOZ_ASSERT(mParameters.mEncodings.Length());
+
+  auto jsepRids = GetJsepTransceiver().mSendTrack.GetRids();
+  if (!jsepRids.empty()) {
+    UpdateRestorableEncodings(mParameters.mEncodings);
+    if (jsepRids.size() != 1 || !jsepRids[0].empty()) {
+      // JSEP is using at least one rid. Stomp our single ridless encoding
+      mParameters.mEncodings = ToSendEncodings(jsepRids);
+    }
+    mSimulcastEnvelopeSet = true;
+    mSimulcastEnvelopeSetByJSEP = true;
+  }
+}
+
+Sequence<RTCRtpEncodingParameters> RTCRtpSender::GetMatchingEncodings(
+    const std::vector<std::string>& aRids) const {
+  Sequence<RTCRtpEncodingParameters> result;
+
+  if (!aRids.empty() && !aRids[0].empty()) {
+    // Simulcast, or unicast with rid
+    for (const auto& encoding : mParameters.mEncodings) {
+      for (const auto& rid : aRids) {
+        auto utf16Rid = NS_ConvertUTF8toUTF16(rid.c_str());
+        if (!encoding.mRid.WasPassed() || (utf16Rid == encoding.mRid.Value())) {
+          auto encodingCopy(encoding);
+          if (!encodingCopy.mRid.WasPassed()) {
+            encodingCopy.mRid.Construct(NS_ConvertUTF8toUTF16(rid.c_str()));
+          }
+          Unused << result.AppendElement(encodingCopy, fallible);
+          break;
+        }
+      }
+    }
+  }
+
+  // If we're allowing the old setParameters behavior, we _might_ be able to
+  // get into this situation even if there were rids above. Be extra careful.
+  // Under normal circumstances, this just handles the ridless case.
+  if (!result.Length()) {
+    // Unicast with no specified rid. Restore mUnicastEncoding, if
+    // it exists, otherwise pick the first encoding.
+    if (mUnicastEncoding.isSome()) {
+      Unused << result.AppendElement(*mUnicastEncoding, fallible);
+    } else {
+      Unused << result.AppendElement(mParameters.mEncodings[0], fallible);
+    }
+  }
+
+  return result;
+}
+
+void RTCRtpSender::SetStreams(
+    const Sequence<OwningNonNull<DOMMediaStream>>& aStreams, ErrorResult& aRv) {
+  if (mPc->IsClosed()) {
+    aRv.ThrowInvalidStateError(
+        "Cannot call setStreams if the peer connection is closed");
+    return;
+  }
+
+  SetStreamsImpl(aStreams);
+  mPc->UpdateNegotiationNeeded();
+}
+
+void RTCRtpSender::SetStreamsImpl(
+    const Sequence<OwningNonNull<DOMMediaStream>>& aStreams) {
+  mStreams.Clear();
+  std::set<nsString> ids;
+  for (const auto& stream : aStreams) {
+    nsString id;
+    stream->GetId(id);
+    if (!ids.count(id)) {
+      ids.insert(id);
+      mStreams.AppendElement(stream);
+    }
+  }
+}
+
+void RTCRtpSender::GetStreams(nsTArray<RefPtr<DOMMediaStream>>& aStreams) {
+  aStreams = mStreams.Clone();
+}
+
+class ReplaceTrackOperation final : public PeerConnectionImpl::Operation {
+ public:
+  ReplaceTrackOperation(PeerConnectionImpl* aPc,
+                        const RefPtr<RTCRtpTransceiver>& aTransceiver,
+                        const RefPtr<MediaStreamTrack>& aTrack,
+                        ErrorResult& aError);
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(ReplaceTrackOperation,
+                                           PeerConnectionImpl::Operation)
+
+ private:
+  MOZ_CAN_RUN_SCRIPT
+  RefPtr<dom::Promise> CallImpl(ErrorResult& aError) override;
+  ~ReplaceTrackOperation() = default;
+  RefPtr<RTCRtpTransceiver> mTransceiver;
+  RefPtr<MediaStreamTrack> mNewTrack;
+};
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(ReplaceTrackOperation,
+                                   PeerConnectionImpl::Operation, mTransceiver,
+                                   mNewTrack)
+
+NS_IMPL_ADDREF_INHERITED(ReplaceTrackOperation, PeerConnectionImpl::Operation)
+NS_IMPL_RELEASE_INHERITED(ReplaceTrackOperation, PeerConnectionImpl::Operation)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(ReplaceTrackOperation)
+NS_INTERFACE_MAP_END_INHERITING(PeerConnectionImpl::Operation)
+
+ReplaceTrackOperation::ReplaceTrackOperation(
+    PeerConnectionImpl* aPc, const RefPtr<RTCRtpTransceiver>& aTransceiver,
+    const RefPtr<MediaStreamTrack>& aTrack, ErrorResult& aError)
+    : PeerConnectionImpl::Operation(aPc, aError),
+      mTransceiver(aTransceiver),
+      mNewTrack(aTrack) {}
+
+RefPtr<dom::Promise> ReplaceTrackOperation::CallImpl(ErrorResult& aError) {
+  RefPtr<RTCRtpSender> sender = mTransceiver->Sender();
+  // If transceiver.[[Stopped]] is true, return a promise rejected with a newly
+  // created InvalidStateError.
+  if (mTransceiver->Stopped()) {
+    RefPtr<dom::Promise> error = sender->MakePromise(aError);
+    if (aError.Failed()) {
+      return nullptr;
+    }
+    MOZ_LOG(gSenderLog, LogLevel::Debug,
+            ("%s Cannot call replaceTrack when transceiver is stopped",
+             __FUNCTION__));
+    error->MaybeRejectWithInvalidStateError(
+        "Cannot call replaceTrack when transceiver is stopped");
+    return error;
+  }
+
+  // Let p be a new promise.
+  RefPtr<dom::Promise> p = sender->MakePromise(aError);
+  if (aError.Failed()) {
+    return nullptr;
+  }
+
+  if (!sender->SeamlessTrackSwitch(mNewTrack)) {
+    MOZ_LOG(gSenderLog, LogLevel::Info,
+            ("%s Could not seamlessly replace track", __FUNCTION__));
+    p->MaybeRejectWithInvalidModificationError(
+        "Could not seamlessly replace track");
+    return p;
+  }
+
+  // Queue a task that runs the following steps:
+  GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction(
+      __func__, [p, sender, track = mNewTrack]() MOZ_CAN_RUN_SCRIPT_BOUNDARY {
+        // If connection.[[IsClosed]] is true, abort these steps.
+        // Set sender.[[SenderTrack]] to withTrack.
+        if (sender->SetSenderTrackWithClosedCheck(track)) {
+          // Resolve p with undefined.
+          p->MaybeResolveWithUndefined();
+        }
+      }));
+
+  // Return p.
+  return p;
+}
+
+already_AddRefed<dom::Promise> RTCRtpSender::ReplaceTrack(
+    dom::MediaStreamTrack* aWithTrack, ErrorResult& aError) {
+  // If withTrack is non-null and withTrack.kind differs from the transceiver
+  // kind of transceiver, return a promise rejected with a newly created
+  // TypeError.
+  if (aWithTrack) {
+    nsString newKind;
+    aWithTrack->GetKind(newKind);
+    nsString oldKind;
+    mTransceiver->GetKind(oldKind);
+    if (newKind != oldKind) {
+      RefPtr<dom::Promise> error = MakePromise(aError);
+      if (aError.Failed()) {
+        return nullptr;
+      }
+      error->MaybeRejectWithTypeError(
+          "Cannot replaceTrack with a different kind!");
+      return error.forget();
+    }
+  }
+
+  MOZ_LOG(gSenderLog, LogLevel::Debug,
+          ("%s[%s]: %s (%p to %p)", mPc->GetHandle().c_str(), GetMid().c_str(),
+           __FUNCTION__, mSenderTrack.get(), aWithTrack));
+
+  // Return the result of chaining the following steps to connection's
+  // operations chain:
+  RefPtr<PeerConnectionImpl::Operation> op =
+      new ReplaceTrackOperation(mPc, mTransceiver, aWithTrack, aError);
+  if (aError.Failed()) {
+    return nullptr;
+  }
+  // Static analysis forces us to use a temporary.
+  auto pc = mPc;
+  return pc->Chain(op, aError);
+}
+
+nsPIDOMWindowInner* RTCRtpSender::GetParentObject() const { return mWindow; }
+
+already_AddRefed<dom::Promise> RTCRtpSender::MakePromise(
+    ErrorResult& aError) const {
+  return mPc->MakePromise(aError);
+}
+
+bool RTCRtpSender::SeamlessTrackSwitch(
+    const RefPtr<MediaStreamTrack>& aWithTrack) {
+  // We do not actually update mSenderTrack here! Spec says that happens in a
+  // queued task after this is done (this happens in
+  // SetSenderTrackWithClosedCheck).
+
+  mPipeline->SetTrack(aWithTrack);
+
+  MaybeUpdateConduit();
+
+  // There may eventually be cases where a renegotiation is necessary to switch.
+  return true;
+}
+
+void RTCRtpSender::SetTrack(const RefPtr<MediaStreamTrack>& aTrack) {
+  // Used for RTCPeerConnection.removeTrack and RTCPeerConnection.addTrack
+  mSenderTrack = aTrack;
+  SeamlessTrackSwitch(aTrack);
+  if (aTrack) {
+    // RFC says (in the section on remote rollback):
+    // However, an RtpTransceiver MUST NOT be removed if a track was attached
+    // to the RtpTransceiver via the addTrack method.
+    mAddTrackCalled = true;
+  }
+}
+
+bool RTCRtpSender::SetSenderTrackWithClosedCheck(
+    const RefPtr<MediaStreamTrack>& aTrack) {
+  if (!mPc->IsClosed()) {
+    mSenderTrack = aTrack;
+    return true;
+  }
+
+  return false;
+}
+
+void RTCRtpSender::Shutdown() {
+  MOZ_ASSERT(NS_IsMainThread());
+  mWatchManager.Shutdown();
+  mPipeline->Shutdown();
+  mPipeline = nullptr;
+}
+
+void RTCRtpSender::BreakCycles() {
+  mWindow = nullptr;
+  mPc = nullptr;
+  mSenderTrack = nullptr;
+  mTransceiver = nullptr;
+  mStreams.Clear();
+  mDtmf = nullptr;
+}
+
+void RTCRtpSender::UpdateTransport() {
+  MOZ_ASSERT(NS_IsMainThread());
+  if (!mHaveSetupTransport) {
+    mPipeline->SetLevel(GetJsepTransceiver().GetLevel());
+    mHaveSetupTransport = true;
+  }
+
+  mPipeline->UpdateTransport_m(GetJsepTransceiver().mTransport.mTransportId,
+                               nullptr);
+}
+
+void RTCRtpSender::MaybeUpdateConduit() {
+  // NOTE(pkerr) - the Call API requires the both local_ssrc and remote_ssrc be
+  // set to a non-zero value or the CreateVideo...Stream call will fail.
+  if (NS_WARN_IF(GetJsepTransceiver().mSendTrack.GetSsrcs().empty())) {
+    MOZ_ASSERT(
+        false,
+        "No local ssrcs! This is a bug in the jsep engine, and should never "
+        "happen!");
+    return;
+  }
+
+  if (!mPipeline) {
+    return;
+  }
+
+  bool wasTransmitting = mTransmitting;
+
+  if (mPipeline->mConduit->type() == MediaSessionConduit::VIDEO) {
+    Maybe<VideoConfig> newConfig = GetNewVideoConfig();
+    if (newConfig.isSome()) {
+      ApplyVideoConfig(*newConfig);
+    }
+  } else {
+    Maybe<AudioConfig> newConfig = GetNewAudioConfig();
+    if (newConfig.isSome()) {
+      ApplyAudioConfig(*newConfig);
+    }
+  }
+
+  if (!mSenderTrack && !wasTransmitting && mTransmitting) {
+    MOZ_LOG(gSenderLog, LogLevel::Debug,
+            ("%s[%s]: %s Starting transmit conduit without send track!",
+             mPc->GetHandle().c_str(), GetMid().c_str(), __FUNCTION__));
+  }
+}
+
+void RTCRtpSender::SyncFromJsep(const JsepTransceiver& aJsepTransceiver) {
+  if (!mSimulcastEnvelopeSet) {
+    // JSEP is establishing the simulcast envelope for the first time, right now
+    // This is the addTrack (or addTransceiver without sendEncodings) case.
+    MaybeGetJsepRids();
+  } else if (!aJsepTransceiver.mSendTrack.GetNegotiatedDetails() ||
+             !aJsepTransceiver.mSendTrack.IsInHaveRemote()) {
+    // Spec says that we do not update our encodings until we're in stable,
+    // _unless_ this is the first negotiation.
+    std::vector<std::string> rids = aJsepTransceiver.mSendTrack.GetRids();
+    if (mSimulcastEnvelopeSetByJSEP && rids.empty()) {
+      // JSEP previously set the simulcast envelope, but now it has no opinion
+      // regarding unicast/simulcast. This can only happen on rollback of the
+      // initial remote offer.
+      mParameters.mEncodings = GetMatchingEncodings(rids);
+      MOZ_ASSERT(mParameters.mEncodings.Length());
+      mSimulcastEnvelopeSetByJSEP = false;
+      mSimulcastEnvelopeSet = false;
+    } else if (!rids.empty()) {
+      // JSEP has an opinion on the simulcast envelope, which trumps anything
+      // we have already.
+      mParameters.mEncodings = GetMatchingEncodings(rids);
+      MOZ_ASSERT(mParameters.mEncodings.Length());
+    }
+  }
+
+  MaybeUpdateConduit();
+}
+
+void RTCRtpSender::SyncToJsep(JsepTransceiver& aJsepTransceiver) const {
+  std::vector<std::string> streamIds;
+  for (const auto& stream : mStreams) {
+    nsString wideStreamId;
+    stream->GetId(wideStreamId);
+    std::string streamId = NS_ConvertUTF16toUTF8(wideStreamId).get();
+    MOZ_ASSERT(!streamId.empty());
+    streamIds.push_back(streamId);
+  }
+
+  aJsepTransceiver.mSendTrack.UpdateStreamIds(streamIds);
+
+  if (mSimulcastEnvelopeSet) {
+    std::vector<std::string> rids;
+    Maybe<RTCRtpSendParameters> parameters;
+    if (mPendingRidChangeFromCompatMode) {
+      // *sigh* If we have just let a setParameters change our rids, but we have
+      // not yet updated mParameters because the queued task hasn't run yet,
+      // we want to set the _new_ rids on the JsepTrack. So, we are forced to
+      // grab them from mPendingParameters.
+      parameters = mPendingParameters;
+    } else {
+      parameters = Some(mParameters);
+    }
+    for (const auto& encoding : parameters->mEncodings) {
+      if (encoding.mRid.WasPassed()) {
+        rids.push_back(NS_ConvertUTF16toUTF8(encoding.mRid.Value()).get());
+      } else {
+        rids.push_back("");
+      }
+    }
+
+    aJsepTransceiver.mSendTrack.SetRids(rids);
+  }
+
+  if (mTransceiver->IsVideo()) {
+    aJsepTransceiver.mSendTrack.SetMaxEncodings(webrtc::kMaxSimulcastStreams);
+  } else {
+    aJsepTransceiver.mSendTrack.SetMaxEncodings(1);
+  }
+
+  if (mAddTrackCalled) {
+    aJsepTransceiver.SetOnlyExistsBecauseOfSetRemote(false);
+  }
+}
+
+Maybe<RTCRtpSender::VideoConfig> RTCRtpSender::GetNewVideoConfig() {
+  // It is possible for SDP to signal that there is a send track, but there not
+  // actually be a send track, according to the specification; all that needs to
+  // happen is for the transceiver to be configured to send...
+  if (!GetJsepTransceiver().mSendTrack.GetNegotiatedDetails()) {
+    return Nothing();
+  }
+
+  VideoConfig oldConfig;
+  oldConfig.mSsrcs = mSsrcs;
+  oldConfig.mLocalRtpExtensions = mLocalRtpExtensions;
+  oldConfig.mCname = mCname;
+  oldConfig.mTransmitting = mTransmitting;
+  oldConfig.mVideoRtxSsrcs = mVideoRtxSsrcs;
+  oldConfig.mVideoCodec = mVideoCodec;
+  oldConfig.mVideoRtpRtcpConfig = mVideoRtpRtcpConfig;
+  oldConfig.mVideoCodecMode = mVideoCodecMode;
+
+  VideoConfig newConfig(oldConfig);
+
+  UpdateBaseConfig(&newConfig);
+
+  newConfig.mVideoRtxSsrcs = GetJsepTransceiver().mSendTrack.GetRtxSsrcs();
+
+  const JsepTrackNegotiatedDetails details(
+      *GetJsepTransceiver().mSendTrack.GetNegotiatedDetails());
+
+  if (mSenderTrack) {
+    RefPtr<mozilla::dom::VideoStreamTrack> videotrack =
+        mSenderTrack->AsVideoStreamTrack();
+
+    if (!videotrack) {
+      MOZ_CRASH(
+          "In ConfigureVideoCodecMode, mSenderTrack is not video! This should "
+          "never happen!");
+    }
+
+    dom::MediaSourceEnum source = videotrack->GetSource().GetMediaSource();
+    switch (source) {
+      case dom::MediaSourceEnum::Browser:
+      case dom::MediaSourceEnum::Screen:
+      case dom::MediaSourceEnum::Window:
+      case dom::MediaSourceEnum::Application:
+        newConfig.mVideoCodecMode = webrtc::VideoCodecMode::kScreensharing;
+        break;
+
+      case dom::MediaSourceEnum::Camera:
+      case dom::MediaSourceEnum::Other:
+        // Other is used by canvas capture, which we treat as realtime video.
+        // This seems debatable, but we've been doing it this way for a long
+        // time, so this is likely fine.
+        newConfig.mVideoCodecMode = webrtc::VideoCodecMode::kRealtimeVideo;
+        break;
+
+      case dom::MediaSourceEnum::Microphone:
+      case dom::MediaSourceEnum::AudioCapture:
+      case dom::MediaSourceEnum::EndGuard_:
+        MOZ_ASSERT(false);
+        break;
+    }
+  }
+
+  std::vector<VideoCodecConfig> configs;
+  RTCRtpTransceiver::NegotiatedDetailsToVideoCodecConfigs(details, &configs);
+
+  if (configs.empty()) {
+    // TODO: Are we supposed to plumb this error back to JS? This does not
+    // seem like a failure to set an answer, it just means that codec
+    // negotiation failed. For now, we're just doing the same thing we do
+    // if negotiation as a whole failed.
+    MOZ_LOG(gSenderLog, LogLevel::Error,
+            ("%s[%s]: %s  No video codecs were negotiated (send).",
+             mPc->GetHandle().c_str(), GetMid().c_str(), __FUNCTION__));
+    return Nothing();
+  }
+
+  newConfig.mVideoCodec = Some(configs[0]);
+  // Spec says that we start using new parameters right away, _before_ we
+  // update the parameters that are visible to JS (ie; mParameters).
+  const RTCRtpSendParameters& parameters =
+      mPendingParameters.isSome() ? *mPendingParameters : mParameters;
+  for (VideoCodecConfig::Encoding& conduitEncoding :
+       newConfig.mVideoCodec->mEncodings) {
+    for (const RTCRtpEncodingParameters& jsEncoding : parameters.mEncodings) {
+      std::string rid;
+      if (jsEncoding.mRid.WasPassed()) {
+        rid = NS_ConvertUTF16toUTF8(jsEncoding.mRid.Value()).get();
+      }
+      if (conduitEncoding.rid == rid) {
+        ApplyJsEncodingToConduitEncoding(jsEncoding, &conduitEncoding);
+        break;
+      }
+    }
+  }
+
+  newConfig.mVideoRtpRtcpConfig = Some(details.GetRtpRtcpConfig());
+
+  if (newConfig == oldConfig) {
+    MOZ_LOG(gSenderLog, LogLevel::Debug,
+            ("%s[%s]: %s  No change in video config", mPc->GetHandle().c_str(),
+             GetMid().c_str(), __FUNCTION__));
+    return Nothing();
+  }
+
+  if (newConfig.mVideoCodec.isSome()) {
+    MOZ_ASSERT(newConfig.mSsrcs.size() ==
+               newConfig.mVideoCodec->mEncodings.size());
+  }
+  return Some(newConfig);
+}
+
+Maybe<RTCRtpSender::AudioConfig> RTCRtpSender::GetNewAudioConfig() {
+  AudioConfig oldConfig;
+  oldConfig.mSsrcs = mSsrcs;
+  oldConfig.mLocalRtpExtensions = mLocalRtpExtensions;
+  oldConfig.mCname = mCname;
+  oldConfig.mTransmitting = mTransmitting;
+  oldConfig.mAudioCodec = mAudioCodec;
+
+  AudioConfig newConfig(oldConfig);
+
+  UpdateBaseConfig(&newConfig);
+
+  if (GetJsepTransceiver().mSendTrack.GetNegotiatedDetails() &&
+      GetJsepTransceiver().mSendTrack.GetActive()) {
+    const auto& details(
+        *GetJsepTransceiver().mSendTrack.GetNegotiatedDetails());
+
+    std::vector<AudioCodecConfig> configs;
+    RTCRtpTransceiver::NegotiatedDetailsToAudioCodecConfigs(details, &configs);
+    if (configs.empty()) {
+      // TODO: Are we supposed to plumb this error back to JS? This does not
+      // seem like a failure to set an answer, it just means that codec
+      // negotiation failed. For now, we're just doing the same thing we do
+      // if negotiation as a whole failed.
+      MOZ_LOG(gSenderLog, LogLevel::Error,
+              ("%s[%s]: %s No audio codecs were negotiated (send)",
+               mPc->GetHandle().c_str(), GetMid().c_str(), __FUNCTION__));
+      return Nothing();
+    }
+
+    std::vector<AudioCodecConfig> dtmfConfigs;
+    std::copy_if(
+        configs.begin(), configs.end(), std::back_inserter(dtmfConfigs),
+        [](const auto& value) { return value.mName == "telephone-event"; });
+
+    const AudioCodecConfig& sendCodec = configs[0];
+
+    if (!dtmfConfigs.empty()) {
+      // There is at least one telephone-event codec.
+      // We primarily choose the codec whose frequency matches the send codec.
+      // Secondarily we choose the one with the lowest frequency.
+      auto dtmfIterator =
+          std::find_if(dtmfConfigs.begin(), dtmfConfigs.end(),
+                       [&sendCodec](const auto& dtmfCodec) {
+                         return dtmfCodec.mFreq == sendCodec.mFreq;
+                       });
+      if (dtmfIterator == dtmfConfigs.end()) {
+        dtmfIterator = std::min_element(
+            dtmfConfigs.begin(), dtmfConfigs.end(),
+            [](const auto& a, const auto& b) { return a.mFreq < b.mFreq; });
+      }
+      MOZ_ASSERT(dtmfIterator != dtmfConfigs.end());
+      newConfig.mDtmfPt = dtmfIterator->mType;
+      newConfig.mDtmfFreq = dtmfIterator->mFreq;
+    }
+
+    newConfig.mAudioCodec = Some(sendCodec);
+  }
+
+  if (newConfig == oldConfig) {
+    MOZ_LOG(gSenderLog, LogLevel::Debug,
+            ("%s[%s]: %s  No change in audio config", mPc->GetHandle().c_str(),
+             GetMid().c_str(), __FUNCTION__));
+    return Nothing();
+  }
+
+  return Some(newConfig);
+}
+
+void RTCRtpSender::UpdateBaseConfig(BaseConfig* aConfig) {
+  aConfig->mSsrcs = GetJsepTransceiver().mSendTrack.GetSsrcs();
+  aConfig->mCname = GetJsepTransceiver().mSendTrack.GetCNAME();
+
+  if (GetJsepTransceiver().mSendTrack.GetNegotiatedDetails() &&
+      GetJsepTransceiver().mSendTrack.GetActive()) {
+    const auto& details(
+        *GetJsepTransceiver().mSendTrack.GetNegotiatedDetails());
+    {
+      std::vector<webrtc::RtpExtension> extmaps;
+      // @@NG read extmap from track
+      details.ForEachRTPHeaderExtension(
+          [&extmaps](const SdpExtmapAttributeList::Extmap& extmap) {
+            extmaps.emplace_back(extmap.extensionname, extmap.entry);
+          });
+      aConfig->mLocalRtpExtensions = extmaps;
+    }
+  }
+  // RTCRtpTransceiver::IsSending is updated after negotiation completes, in a
+  // queued task (which we may be in right now). Don't use
+  // JsepTrack::GetActive, because that updates before the queued task, which
+  // is too early for some of the things we interact with here (eg;
+  // RTCDTMFSender).
+  aConfig->mTransmitting = mTransceiver->IsSending();
+}
+
+void RTCRtpSender::ApplyVideoConfig(const VideoConfig& aConfig) {
+  if (aConfig.mVideoCodec.isSome()) {
+    MOZ_ASSERT(aConfig.mSsrcs.size() == aConfig.mVideoCodec->mEncodings.size());
+  }
+
+  mSsrcs = aConfig.mSsrcs;
+  mCname = aConfig.mCname;
+  mLocalRtpExtensions = aConfig.mLocalRtpExtensions;
+
+  mVideoRtxSsrcs = aConfig.mVideoRtxSsrcs;
+  mVideoCodec = aConfig.mVideoCodec;
+  mVideoRtpRtcpConfig = aConfig.mVideoRtpRtcpConfig;
+  mVideoCodecMode = aConfig.mVideoCodecMode;
+
+  mTransmitting = aConfig.mTransmitting;
+}
+
+void RTCRtpSender::ApplyAudioConfig(const AudioConfig& aConfig) {
+  mTransmitting = false;
+
+  mSsrcs = aConfig.mSsrcs;
+  mCname = aConfig.mCname;
+  mLocalRtpExtensions = aConfig.mLocalRtpExtensions;
+
+  mAudioCodec = aConfig.mAudioCodec;
+
+  if (aConfig.mDtmfPt >= 0) {
+    mDtmf->SetPayloadType(aConfig.mDtmfPt, aConfig.mDtmfFreq);
+  }
+
+  mTransmitting = aConfig.mTransmitting;
+}
+
+void RTCRtpSender::Stop() {
+  MOZ_ASSERT(mTransceiver->Stopped());
+  mTransmitting = false;
+}
+
+bool RTCRtpSender::HasTrack(const dom::MediaStreamTrack* aTrack) const {
+  if (!mSenderTrack) {
+    return false;
+  }
+
+  if (!aTrack) {
+    return true;
+  }
+
+  return mSenderTrack.get() == aTrack;
+}
+
+RefPtr<MediaPipelineTransmit> RTCRtpSender::GetPipeline() const {
+  return mPipeline;
+}
+
+std::string RTCRtpSender::GetMid() const { return mTransceiver->GetMidAscii(); }
+
+JsepTransceiver& RTCRtpSender::GetJsepTransceiver() {
+  return mTransceiver->GetJsepTransceiver();
+}
+
+void RTCRtpSender::UpdateDtmfSender() {
+  if (!mDtmf) {
+    return;
+  }
+
+  if (mTransmitting) {
+    return;
+  }
+
+  mDtmf->StopPlayout();
+}
+
+}  // namespace mozilla::dom
+
+#undef LOGTAG
diff --git a/dom/media/webrtc/jsapi/RTCRtpSender.h b/dom/media/webrtc/jsapi/RTCRtpSender.h
new file mode 100644
index 0000000000..0c1282e0db
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCRtpSender.h
@@ -0,0 +1,260 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _RTCRtpSender_h_
+#define _RTCRtpSender_h_
+
+#include "nsISupports.h"
+#include "nsWrapperCache.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/StateMirroring.h"
+#include "mozilla/Maybe.h"
+#include "js/RootingAPI.h"
+#include "libwebrtcglue/RtpRtcpConfig.h"
+#include "nsTArray.h"
+#include "mozilla/dom/RTCStatsReportBinding.h"
+#include "mozilla/dom/RTCRtpCapabilitiesBinding.h"
+#include "mozilla/dom/RTCRtpParametersBinding.h"
+#include "RTCStatsReport.h"
+#include "jsep/JsepTrack.h"
+#include "transportbridge/MediaPipeline.h"
+
+class nsPIDOMWindowInner;
+
+namespace mozilla {
+class MediaSessionConduit;
+class MediaTransportHandler;
+class JsepTransceiver;
+class PeerConnectionImpl;
+class DOMMediaStream;
+
+namespace dom {
+class MediaStreamTrack;
+class Promise;
+class RTCDtlsTransport;
+class RTCDTMFSender;
+struct RTCRtpCapabilities;
+class RTCRtpTransceiver;
+
+class RTCRtpSender : public nsISupports,
+                     public nsWrapperCache,
+                     public MediaPipelineTransmitControlInterface {
+ public:
+  RTCRtpSender(nsPIDOMWindowInner* aWindow, PeerConnectionImpl* aPc,
+               MediaTransportHandler* aTransportHandler,
+               AbstractThread* aCallThread, nsISerialEventTarget* aStsThread,
+               MediaSessionConduit* aConduit, dom::MediaStreamTrack* aTrack,
+               const Sequence<RTCRtpEncodingParameters>& aEncodings,
+               RTCRtpTransceiver* aTransceiver);
+
+  // nsISupports
+  NS_DECL_CYCLE_COLLECTING_ISUPPORTS
+  NS_DECL_CYCLE_COLLECTION_WRAPPERCACHE_CLASS(RTCRtpSender)
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  // webidl
+  MediaStreamTrack* GetTrack() const { return mSenderTrack; }
+  RTCDtlsTransport* GetTransport() const;
+  RTCDTMFSender* GetDtmf() const;
+  MOZ_CAN_RUN_SCRIPT
+  already_AddRefed<Promise> ReplaceTrack(MediaStreamTrack* aWithTrack,
+                                         ErrorResult& aError);
+  already_AddRefed<Promise> GetStats(ErrorResult& aError);
+  static void GetCapabilities(const GlobalObject&, const nsAString& kind,
+                              Nullable<dom::RTCRtpCapabilities>& result);
+  already_AddRefed<Promise> SetParameters(
+      const dom::RTCRtpSendParameters& aParameters, ErrorResult& aError);
+  // Not a simple getter, so not const
+  // See https://w3c.github.io/webrtc-pc/#dom-rtcrtpsender-getparameters
+  void GetParameters(RTCRtpSendParameters& aParameters);
+
+  static void CheckAndRectifyEncodings(
+      Sequence<RTCRtpEncodingParameters>& aEncodings, bool aVideo,
+      ErrorResult& aRv);
+
+  nsPIDOMWindowInner* GetParentObject() const;
+  nsTArray<RefPtr<RTCStatsPromise>> GetStatsInternal(
+      bool aSkipIceStats = false);
+
+  void SetStreams(const Sequence<OwningNonNull<DOMMediaStream>>& aStreams,
+                  ErrorResult& aRv);
+  // ChromeOnly webidl
+  void GetStreams(nsTArray<RefPtr<DOMMediaStream>>& aStreams);
+  // ChromeOnly webidl
+  void SetStreamsImpl(const Sequence<OwningNonNull<DOMMediaStream>>& aStreams);
+  // ChromeOnly webidl
+  void SetTrack(const RefPtr<MediaStreamTrack>& aTrack);
+  void Shutdown();
+  void BreakCycles();
+  // Terminal state, reached through stopping RTCRtpTransceiver.
+  void Stop();
+  bool HasTrack(const dom::MediaStreamTrack* aTrack) const;
+  bool IsMyPc(const PeerConnectionImpl* aPc) const { return mPc.get() == aPc; }
+  RefPtr<MediaPipelineTransmit> GetPipeline() const;
+  already_AddRefed<dom::Promise> MakePromise(ErrorResult& aError) const;
+  bool SeamlessTrackSwitch(const RefPtr<MediaStreamTrack>& aWithTrack);
+  bool SetSenderTrackWithClosedCheck(const RefPtr<MediaStreamTrack>& aTrack);
+
+  // This is called when we set an answer (ie; when the transport is finalized).
+  void UpdateTransport();
+  void SyncToJsep(JsepTransceiver& aJsepTransceiver) const;
+  void SyncFromJsep(const JsepTransceiver& aJsepTransceiver);
+  void MaybeUpdateConduit();
+
+  AbstractCanonical<Ssrcs>* CanonicalSsrcs() { return &mSsrcs; }
+  AbstractCanonical<Ssrcs>* CanonicalVideoRtxSsrcs() { return &mVideoRtxSsrcs; }
+  AbstractCanonical<RtpExtList>* CanonicalLocalRtpExtensions() {
+    return &mLocalRtpExtensions;
+  }
+
+  AbstractCanonical<Maybe<AudioCodecConfig>>* CanonicalAudioCodec() {
+    return &mAudioCodec;
+  }
+
+  AbstractCanonical<Maybe<VideoCodecConfig>>* CanonicalVideoCodec() {
+    return &mVideoCodec;
+  }
+  AbstractCanonical<Maybe<RtpRtcpConfig>>* CanonicalVideoRtpRtcpConfig() {
+    return &mVideoRtpRtcpConfig;
+  }
+  AbstractCanonical<webrtc::VideoCodecMode>* CanonicalVideoCodecMode() {
+    return &mVideoCodecMode;
+  }
+  AbstractCanonical<std::string>* CanonicalCname() { return &mCname; }
+  AbstractCanonical<bool>* CanonicalTransmitting() override {
+    return &mTransmitting;
+  }
+
+  bool HasPendingSetParameters() const { return mPendingParameters.isSome(); }
+  void InvalidateLastReturnedParameters() {
+    mLastReturnedParameters = Nothing();
+  }
+
+ private:
+  virtual ~RTCRtpSender();
+
+  std::string GetMid() const;
+  JsepTransceiver& GetJsepTransceiver();
+  static void ApplyJsEncodingToConduitEncoding(
+      const RTCRtpEncodingParameters& aJsEncoding,
+      VideoCodecConfig::Encoding* aConduitEncoding);
+  void UpdateRestorableEncodings(
+      const Sequence<RTCRtpEncodingParameters>& aEncodings);
+  Sequence<RTCRtpEncodingParameters> GetMatchingEncodings(
+      const std::vector<std::string>& aRids) const;
+  Sequence<RTCRtpEncodingParameters> ToSendEncodings(
+      const std::vector<std::string>& aRids) const;
+  void MaybeGetJsepRids();
+  void UpdateDtmfSender();
+
+  void WarnAboutBadSetParameters(const nsCString& aError);
+  nsCString GetEffectiveTLDPlus1() const;
+
+  WatchManager<RTCRtpSender> mWatchManager;
+  nsCOMPtr<nsPIDOMWindowInner> mWindow;
+  RefPtr<PeerConnectionImpl> mPc;
+  RefPtr<dom::MediaStreamTrack> mSenderTrack;
+  bool mAddTrackCalled = false;
+  RTCRtpSendParameters mParameters;
+  Maybe<RTCRtpSendParameters> mPendingParameters;
+  uint32_t mNumSetParametersCalls = 0;
+  // When JSEP goes from simulcast to unicast without a rid, and we started out
+  // as unicast without a rid, we are supposed to restore that unicast encoding
+  // from before.
+  Maybe<RTCRtpEncodingParameters> mUnicastEncoding;
+  bool mSimulcastEnvelopeSet = false;
+  bool mSimulcastEnvelopeSetByJSEP = false;
+  bool mPendingRidChangeFromCompatMode = false;
+  Maybe<RTCRtpSendParameters> mLastReturnedParameters;
+  RefPtr<MediaPipelineTransmit> mPipeline;
+  RefPtr<MediaTransportHandler> mTransportHandler;
+  RefPtr<RTCRtpTransceiver> mTransceiver;
+  nsTArray<RefPtr<DOMMediaStream>> mStreams;
+  bool mHaveSetupTransport = false;
+  // TODO(bug 1803388): Remove this stuff once it is no longer needed.
+  bool mAllowOldSetParameters = false;
+
+  // TODO(bug 1803388): Remove the glean warnings once they are no longer needed
+  bool mHaveWarnedBecauseNoGetParameters = false;
+  bool mHaveWarnedBecauseEncodingCountChange = false;
+  bool mHaveWarnedBecauseNoTransactionId = false;
+  bool mHaveWarnedBecauseStaleTransactionId = false;
+  // TODO(bug 1803389): Remove the glean errors once they are no longer needed.
+  bool mHaveFailedBecauseNoGetParameters = false;
+  bool mHaveFailedBecauseEncodingCountChange = false;
+  bool mHaveFailedBecauseRidChange = false;
+  bool mHaveFailedBecauseNoTransactionId = false;
+  bool mHaveFailedBecauseStaleTransactionId = false;
+  bool mHaveFailedBecauseNoEncodings = false;
+  bool mHaveFailedBecauseOtherError = false;
+
+  RefPtr<dom::RTCDTMFSender> mDtmf;
+
+  class BaseConfig {
+   public:
+    // TODO(bug 1744116): Use = default here
+    bool operator==(const BaseConfig& aOther) const {
+      return mSsrcs == aOther.mSsrcs &&
+             mLocalRtpExtensions == aOther.mLocalRtpExtensions &&
+             mCname == aOther.mCname && mTransmitting == aOther.mTransmitting;
+    }
+    Ssrcs mSsrcs;
+    RtpExtList mLocalRtpExtensions;
+    std::string mCname;
+    bool mTransmitting = false;
+  };
+
+  class VideoConfig : public BaseConfig {
+   public:
+    // TODO(bug 1744116): Use = default here
+    bool operator==(const VideoConfig& aOther) const {
+      return BaseConfig::operator==(aOther) &&
+             mVideoRtxSsrcs == aOther.mVideoRtxSsrcs &&
+             mVideoCodec == aOther.mVideoCodec &&
+             mVideoRtpRtcpConfig == aOther.mVideoRtpRtcpConfig &&
+             mVideoCodecMode == aOther.mVideoCodecMode;
+    }
+    Ssrcs mVideoRtxSsrcs;
+    Maybe<VideoCodecConfig> mVideoCodec;
+    Maybe<RtpRtcpConfig> mVideoRtpRtcpConfig;
+    webrtc::VideoCodecMode mVideoCodecMode =
+        webrtc::VideoCodecMode::kRealtimeVideo;
+  };
+
+  class AudioConfig : public BaseConfig {
+   public:
+    // TODO(bug 1744116): Use = default here
+    bool operator==(const AudioConfig& aOther) const {
+      return BaseConfig::operator==(aOther) &&
+             mAudioCodec == aOther.mAudioCodec && mDtmfPt == aOther.mDtmfPt &&
+             mDtmfFreq == aOther.mDtmfFreq;
+    }
+    Maybe<AudioCodecConfig> mAudioCodec;
+    int32_t mDtmfPt = -1;
+    int32_t mDtmfFreq = 0;
+  };
+
+  Maybe<VideoConfig> GetNewVideoConfig();
+  Maybe<AudioConfig> GetNewAudioConfig();
+  void UpdateBaseConfig(BaseConfig* aConfig);
+  void ApplyVideoConfig(const VideoConfig& aConfig);
+  void ApplyAudioConfig(const AudioConfig& aConfig);
+
+  Canonical<Ssrcs> mSsrcs;
+  Canonical<Ssrcs> mVideoRtxSsrcs;
+  Canonical<RtpExtList> mLocalRtpExtensions;
+
+  Canonical<Maybe<AudioCodecConfig>> mAudioCodec;
+  Canonical<Maybe<VideoCodecConfig>> mVideoCodec;
+  Canonical<Maybe<RtpRtcpConfig>> mVideoRtpRtcpConfig;
+  Canonical<webrtc::VideoCodecMode> mVideoCodecMode;
+  Canonical<std::string> mCname;
+  Canonical<bool> mTransmitting;
+};
+
+}  // namespace dom
+}  // namespace mozilla
+#endif  // _RTCRtpSender_h_
diff --git a/dom/media/webrtc/jsapi/RTCRtpTransceiver.cpp b/dom/media/webrtc/jsapi/RTCRtpTransceiver.cpp
new file mode 100644
index 0000000000..bf4f74dd5d
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCRtpTransceiver.cpp
@@ -0,0 +1,1080 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "jsapi/RTCRtpTransceiver.h"
+#include "mozilla/UniquePtr.h"
+#include <algorithm>
+#include <string>
+#include <vector>
+#include "libwebrtcglue/AudioConduit.h"
+#include "libwebrtcglue/VideoConduit.h"
+#include "MediaTrackGraph.h"
+#include "transportbridge/MediaPipeline.h"
+#include "transportbridge/MediaPipelineFilter.h"
+#include "jsep/JsepTrack.h"
+#include "sdp/SdpHelper.h"
+#include "MediaTrackGraphImpl.h"
+#include "transport/logging.h"
+#include "MediaEngine.h"
+#include "nsIPrincipal.h"
+#include "MediaSegment.h"
+#include "RemoteTrackSource.h"
+#include "libwebrtcglue/RtpRtcpConfig.h"
+#include "MediaTransportHandler.h"
+#include "mozilla/dom/RTCRtpReceiverBinding.h"
+#include "mozilla/dom/RTCRtpSenderBinding.h"
+#include "mozilla/dom/RTCRtpTransceiverBinding.h"
+#include "mozilla/dom/Promise.h"
+#include "RTCDtlsTransport.h"
+#include "RTCRtpReceiver.h"
+#include "RTCRtpSender.h"
+#include "RTCDTMFSender.h"
+#include "systemservices/MediaUtils.h"
+#include "libwebrtcglue/WebrtcCallWrapper.h"
+#include "libwebrtcglue/WebrtcGmpVideoCodec.h"
+#include "utils/PerformanceRecorder.h"
+
+namespace mozilla {
+
+using namespace dom;
+
+namespace {
+struct ConduitControlState : public AudioConduitControlInterface,
+                             public VideoConduitControlInterface {
+  ConduitControlState(RTCRtpTransceiver* aTransceiver, RTCRtpSender* aSender,
+                      RTCRtpReceiver* aReceiver)
+      : mTransceiver(new nsMainThreadPtrHolder<RTCRtpTransceiver>(
+            "ConduitControlState::mTransceiver", aTransceiver, false)),
+        mSender(new nsMainThreadPtrHolder<dom::RTCRtpSender>(
+            "ConduitControlState::mSender", aSender, false)),
+        mReceiver(new nsMainThreadPtrHolder<dom::RTCRtpReceiver>(
+            "ConduitControlState::mReceiver", aReceiver, false)) {}
+
+  const nsMainThreadPtrHandle<RTCRtpTransceiver> mTransceiver;
+  const nsMainThreadPtrHandle<RTCRtpSender> mSender;
+  const nsMainThreadPtrHandle<RTCRtpReceiver> mReceiver;
+
+  // MediaConduitControlInterface
+  AbstractCanonical<bool>* CanonicalReceiving() override {
+    return mReceiver->CanonicalReceiving();
+  }
+  AbstractCanonical<bool>* CanonicalTransmitting() override {
+    return mSender->CanonicalTransmitting();
+  }
+  AbstractCanonical<Ssrcs>* CanonicalLocalSsrcs() override {
+    return mSender->CanonicalSsrcs();
+  }
+  AbstractCanonical<std::string>* CanonicalLocalCname() override {
+    return mSender->CanonicalCname();
+  }
+  AbstractCanonical<std::string>* CanonicalMid() override {
+    return mTransceiver->CanonicalMid();
+  }
+  AbstractCanonical<Ssrc>* CanonicalRemoteSsrc() override {
+    return mReceiver->CanonicalSsrc();
+  }
+  AbstractCanonical<std::string>* CanonicalSyncGroup() override {
+    return mTransceiver->CanonicalSyncGroup();
+  }
+  AbstractCanonical<RtpExtList>* CanonicalLocalRecvRtpExtensions() override {
+    return mReceiver->CanonicalLocalRtpExtensions();
+  }
+  AbstractCanonical<RtpExtList>* CanonicalLocalSendRtpExtensions() override {
+    return mSender->CanonicalLocalRtpExtensions();
+  }
+
+  // AudioConduitControlInterface
+  AbstractCanonical<Maybe<AudioCodecConfig>>* CanonicalAudioSendCodec()
+      override {
+    return mSender->CanonicalAudioCodec();
+  }
+  AbstractCanonical<std::vector<AudioCodecConfig>>* CanonicalAudioRecvCodecs()
+      override {
+    return mReceiver->CanonicalAudioCodecs();
+  }
+  MediaEventSource<DtmfEvent>& OnDtmfEvent() override {
+    return mSender->GetDtmf()->OnDtmfEvent();
+  }
+
+  // VideoConduitControlInterface
+  AbstractCanonical<Ssrcs>* CanonicalLocalVideoRtxSsrcs() override {
+    return mSender->CanonicalVideoRtxSsrcs();
+  }
+  AbstractCanonical<Ssrc>* CanonicalRemoteVideoRtxSsrc() override {
+    return mReceiver->CanonicalVideoRtxSsrc();
+  }
+  AbstractCanonical<Maybe<VideoCodecConfig>>* CanonicalVideoSendCodec()
+      override {
+    return mSender->CanonicalVideoCodec();
+  }
+  AbstractCanonical<Maybe<RtpRtcpConfig>>* CanonicalVideoSendRtpRtcpConfig()
+      override {
+    return mSender->CanonicalVideoRtpRtcpConfig();
+  }
+  AbstractCanonical<std::vector<VideoCodecConfig>>* CanonicalVideoRecvCodecs()
+      override {
+    return mReceiver->CanonicalVideoCodecs();
+  }
+  AbstractCanonical<Maybe<RtpRtcpConfig>>* CanonicalVideoRecvRtpRtcpConfig()
+      override {
+    return mReceiver->CanonicalVideoRtpRtcpConfig();
+  }
+  AbstractCanonical<webrtc::VideoCodecMode>* CanonicalVideoCodecMode()
+      override {
+    return mSender->CanonicalVideoCodecMode();
+  }
+};
+}  // namespace
+
+MOZ_MTLOG_MODULE("RTCRtpTransceiver")
+
+NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE_CLASS(RTCRtpTransceiver)
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(RTCRtpTransceiver)
+  if (tmp->mHandlingUnlink) {
+    tmp->BreakCycles();
+    tmp->mHandlingUnlink = false;
+  }
+  NS_IMPL_CYCLE_COLLECTION_UNLINK_PRESERVED_WRAPPER
+NS_IMPL_CYCLE_COLLECTION_UNLINK_END
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN(RTCRtpTransceiver)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mWindow, mPc, mSendTrack, mReceiver,
+                                    mSender, mDtlsTransport,
+                                    mLastStableDtlsTransport)
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
+
+NS_IMPL_CYCLE_COLLECTING_ADDREF(RTCRtpTransceiver)
+NS_IMPL_CYCLE_COLLECTING_RELEASE(RTCRtpTransceiver)
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(RTCRtpTransceiver)
+  NS_WRAPPERCACHE_INTERFACE_MAP_ENTRY
+  NS_INTERFACE_MAP_ENTRY(nsISupports)
+NS_INTERFACE_MAP_END
+
+#define INIT_CANONICAL(name, val)         \
+  name(AbstractThread::MainThread(), val, \
+       "RTCRtpTransceiver::" #name " (Canonical)")
+
+RTCRtpTransceiver::RTCRtpTransceiver(
+    nsPIDOMWindowInner* aWindow, bool aPrivacyNeeded, PeerConnectionImpl* aPc,
+    MediaTransportHandler* aTransportHandler, JsepSession* aJsepSession,
+    const std::string& aTransceiverId, bool aIsVideo,
+    nsISerialEventTarget* aStsThread, dom::MediaStreamTrack* aSendTrack,
+    WebrtcCallWrapper* aCallWrapper, RTCStatsIdGenerator* aIdGenerator)
+    : mWindow(aWindow),
+      mPc(aPc),
+      mTransportHandler(aTransportHandler),
+      mTransceiverId(aTransceiverId),
+      mJsepTransceiver(*aJsepSession->GetTransceiver(mTransceiverId)),
+      mStsThread(aStsThread),
+      mCallWrapper(aCallWrapper),
+      mSendTrack(aSendTrack),
+      mIdGenerator(aIdGenerator),
+      mPrincipalPrivacy(aPrivacyNeeded ? PrincipalPrivacy::Private
+                                       : PrincipalPrivacy::NonPrivate),
+      mIsVideo(aIsVideo),
+      INIT_CANONICAL(mMid, std::string()),
+      INIT_CANONICAL(mSyncGroup, std::string()) {}
+
+#undef INIT_CANONICAL
+
+RTCRtpTransceiver::~RTCRtpTransceiver() = default;
+
+SdpDirectionAttribute::Direction ToSdpDirection(
+    RTCRtpTransceiverDirection aDirection) {
+  switch (aDirection) {
+    case dom::RTCRtpTransceiverDirection::Sendrecv:
+      return SdpDirectionAttribute::Direction::kSendrecv;
+    case dom::RTCRtpTransceiverDirection::Sendonly:
+      return SdpDirectionAttribute::Direction::kSendonly;
+    case dom::RTCRtpTransceiverDirection::Recvonly:
+      return SdpDirectionAttribute::Direction::kRecvonly;
+    case dom::RTCRtpTransceiverDirection::Inactive:
+      return SdpDirectionAttribute::Direction::kInactive;
+    case dom::RTCRtpTransceiverDirection::EndGuard_:;
+  }
+  MOZ_CRASH("Invalid transceiver direction!");
+}
+
+static uint32_t sRemoteSourceId = 0;
+
+// TODO(bug 1401592): Once we implement the sendEncodings stuff, there will
+// need to be validation code in here.
+void RTCRtpTransceiver::Init(const RTCRtpTransceiverInit& aInit,
+                             ErrorResult& aRv) {
+  TrackingId trackingId(TrackingId::Source::RTCRtpReceiver, sRemoteSourceId++,
+                        TrackingId::TrackAcrossProcesses::Yes);
+  if (IsVideo()) {
+    InitVideo(trackingId);
+  } else {
+    InitAudio();
+  }
+
+  if (!IsValid()) {
+    aRv = NS_ERROR_UNEXPECTED;
+    return;
+  }
+
+  mReceiver = new RTCRtpReceiver(mWindow, mPrincipalPrivacy, mPc,
+                                 mTransportHandler, mCallWrapper->mCallThread,
+                                 mStsThread, mConduit, this, trackingId);
+
+  mSender = new RTCRtpSender(mWindow, mPc, mTransportHandler,
+                             mCallWrapper->mCallThread, mStsThread, mConduit,
+                             mSendTrack, aInit.mSendEncodings, this);
+
+  if (mConduit) {
+    InitConduitControl();
+  }
+
+  mSender->SetStreamsImpl(aInit.mStreams);
+  mDirection = aInit.mDirection;
+}
+
+void RTCRtpTransceiver::SetDtlsTransport(dom::RTCDtlsTransport* aDtlsTransport,
+                                         bool aStable) {
+  mDtlsTransport = aDtlsTransport;
+  if (aStable) {
+    mLastStableDtlsTransport = mDtlsTransport;
+  }
+}
+
+void RTCRtpTransceiver::RollbackToStableDtlsTransport() {
+  mDtlsTransport = mLastStableDtlsTransport;
+}
+
+void RTCRtpTransceiver::InitAudio() {
+  mConduit = AudioSessionConduit::Create(mCallWrapper, mStsThread);
+
+  if (!mConduit) {
+    MOZ_MTLOG(ML_ERROR, mPc->GetHandle()
+                            << "[" << mMid.Ref() << "]: " << __FUNCTION__
+                            << ": Failed to create AudioSessionConduit");
+    // TODO(bug 1422897): We need a way to record this when it happens in the
+    // wild.
+  }
+}
+
+void RTCRtpTransceiver::InitVideo(const TrackingId& aRecvTrackingId) {
+  VideoSessionConduit::Options options;
+  options.mVideoLatencyTestEnable =
+      Preferences::GetBool("media.video.test_latency", false);
+  options.mMinBitrate = std::max(
+      0,
+      Preferences::GetInt("media.peerconnection.video.min_bitrate", 0) * 1000);
+  options.mStartBitrate = std::max(
+      0, Preferences::GetInt("media.peerconnection.video.start_bitrate", 0) *
+             1000);
+  options.mPrefMaxBitrate = std::max(
+      0,
+      Preferences::GetInt("media.peerconnection.video.max_bitrate", 0) * 1000);
+  if (options.mMinBitrate != 0 &&
+      options.mMinBitrate < kViEMinCodecBitrate_bps) {
+    options.mMinBitrate = kViEMinCodecBitrate_bps;
+  }
+  if (options.mStartBitrate < options.mMinBitrate) {
+    options.mStartBitrate = options.mMinBitrate;
+  }
+  if (options.mPrefMaxBitrate &&
+      options.mStartBitrate > options.mPrefMaxBitrate) {
+    options.mStartBitrate = options.mPrefMaxBitrate;
+  }
+  // XXX We'd love if this was a live param for testing adaptation/etc
+  // in automation
+  options.mMinBitrateEstimate =
+      std::max(0, Preferences::GetInt(
+                      "media.peerconnection.video.min_bitrate_estimate", 0) *
+                      1000);
+  options.mSpatialLayers = std::max(
+      1, Preferences::GetInt("media.peerconnection.video.svc.spatial", 0));
+  options.mTemporalLayers = std::max(
+      1, Preferences::GetInt("media.peerconnection.video.svc.temporal", 0));
+  options.mDenoising =
+      Preferences::GetBool("media.peerconnection.video.denoising", false);
+  options.mLockScaling =
+      Preferences::GetBool("media.peerconnection.video.lock_scaling", false);
+
+  mConduit =
+      VideoSessionConduit::Create(mCallWrapper, mStsThread, std::move(options),
+                                  mPc->GetHandle(), aRecvTrackingId);
+
+  if (!mConduit) {
+    MOZ_MTLOG(ML_ERROR, mPc->GetHandle()
+                            << "[" << mMid.Ref() << "]: " << __FUNCTION__
+                            << ": Failed to create VideoSessionConduit");
+    // TODO(bug 1422897): We need a way to record this when it happens in the
+    // wild.
+  }
+}
+
+void RTCRtpTransceiver::InitConduitControl() {
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_ASSERT(mConduit);
+  ConduitControlState control(this, mSender, mReceiver);
+  mCallWrapper->mCallThread->Dispatch(NS_NewRunnableFunction(
+      __func__, [conduit = mConduit, control = std::move(control)]() mutable {
+        conduit->AsVideoSessionConduit().apply(
+            [&](VideoSessionConduit* aConduit) {
+              aConduit->InitControl(&control);
+            });
+        conduit->AsAudioSessionConduit().apply(
+            [&](AudioSessionConduit* aConduit) {
+              aConduit->InitControl(&control);
+            });
+      }));
+}
+
+void RTCRtpTransceiver::Close() {
+  // Called via PCImpl::Close -> PCImpl::CloseInt -> PCImpl::ShutdownMedia ->
+  // PCMedia::SelfDestruct.  Satisfies step 7 of
+  // https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-close
+  mShutdown = true;
+  if (mDtlsTransport) {
+    mDtlsTransport->UpdateState(TransportLayer::TS_CLOSED);
+  }
+  StopImpl();
+}
+
+void RTCRtpTransceiver::BreakCycles() {
+  mSender->BreakCycles();
+  mReceiver->BreakCycles();
+  mWindow = nullptr;
+  mSendTrack = nullptr;
+  mSender = nullptr;
+  mReceiver = nullptr;
+  mDtlsTransport = nullptr;
+  mLastStableDtlsTransport = nullptr;
+  mPc = nullptr;
+}
+
+// TODO: Only called from one place in PeerConnectionImpl, synchronously, when
+// the JSEP engine has successfully completed an offer/answer exchange. This is
+// a bit squirrely, since identity validation happens asynchronously in
+// PeerConnection.jsm. This probably needs to happen once all the "in parallel"
+// steps have succeeded, but before we queue the task for JS observable state
+// updates.
+nsresult RTCRtpTransceiver::UpdateTransport() {
+  if (!mHasTransport) {
+    return NS_OK;
+  }
+
+  mReceiver->UpdateTransport();
+  mSender->UpdateTransport();
+  return NS_OK;
+}
+
+nsresult RTCRtpTransceiver::UpdateConduit() {
+  if (mStopped) {
+    return NS_OK;
+  }
+
+  mReceiver->UpdateConduit();
+  mSender->MaybeUpdateConduit();
+
+  return NS_OK;
+}
+
+void RTCRtpTransceiver::UpdatePrincipalPrivacy(PrincipalPrivacy aPrivacy) {
+  if (mPrincipalPrivacy == aPrivacy) {
+    return;
+  }
+
+  mPrincipalPrivacy = aPrivacy;
+  mReceiver->UpdatePrincipalPrivacy(mPrincipalPrivacy);
+}
+
+void RTCRtpTransceiver::ResetSync() { mSyncGroup = std::string(); }
+
+// TODO: Only called from one place in PeerConnectionImpl, synchronously, when
+// the JSEP engine has successfully completed an offer/answer exchange. This is
+// a bit squirrely, since identity validation happens asynchronously in
+// PeerConnection.jsm. This probably needs to happen once all the "in parallel"
+// steps have succeeded, but before we queue the task for JS observable state
+// updates.
+nsresult RTCRtpTransceiver::SyncWithMatchingVideoConduits(
+    nsTArray<RefPtr<RTCRtpTransceiver>>& transceivers) {
+  if (mStopped) {
+    return NS_OK;
+  }
+
+  if (IsVideo()) {
+    MOZ_MTLOG(ML_ERROR, mPc->GetHandle()
+                            << "[" << mMid.Ref() << "]: " << __FUNCTION__
+                            << " called when transceiver is not "
+                               "video! This should never happen.");
+    MOZ_CRASH();
+    return NS_ERROR_UNEXPECTED;
+  }
+
+  std::set<std::string> myReceiveStreamIds;
+  myReceiveStreamIds.insert(mReceiver->GetStreamIds().begin(),
+                            mReceiver->GetStreamIds().end());
+
+  for (RefPtr<RTCRtpTransceiver>& transceiver : transceivers) {
+    if (!transceiver->IsValid()) {
+      continue;
+    }
+
+    if (!transceiver->IsVideo()) {
+      // |this| is an audio transceiver, so we skip other audio transceivers
+      continue;
+    }
+
+    // Maybe could make this more efficient by cacheing this set, but probably
+    // not worth it.
+    for (const std::string& streamId :
+         transceiver->Receiver()->GetStreamIds()) {
+      if (myReceiveStreamIds.count(streamId)) {
+        // Ok, we have one video, one non-video - cross the streams!
+        mSyncGroup = streamId;
+        transceiver->mSyncGroup = streamId;
+
+        MOZ_MTLOG(ML_DEBUG, mPc->GetHandle()
+                                << "[" << mMid.Ref() << "]: " << __FUNCTION__
+                                << " Syncing " << mConduit.get() << " to "
+                                << transceiver->mConduit.get());
+
+        // The sync code in call.cc only permits sync between audio stream and
+        // one video stream. They take the first match, so there's no point in
+        // continuing here. If we want to change the default, we should sort
+        // video streams here and only call SetSyncGroup on the chosen stream.
+        break;
+      }
+    }
+  }
+
+  return NS_OK;
+}
+
+bool RTCRtpTransceiver::ConduitHasPluginID(uint64_t aPluginID) {
+  return mConduit && mConduit->HasCodecPluginID(aPluginID);
+}
+
+void RTCRtpTransceiver::SyncFromJsep(const JsepSession& aSession) {
+  MOZ_MTLOG(ML_DEBUG, mPc->GetHandle()
+                          << "[" << mMid.Ref() << "]: " << __FUNCTION__
+                          << " Syncing from JSEP transceiver");
+  if (mShutdown) {
+    // Shutdown_m has already been called, probably due to pc.close(). Just
+    // nod and smile.
+    return;
+  }
+
+  mJsepTransceiver = *aSession.GetTransceiver(mTransceiverId);
+
+  // Transceivers can stop due to JSEP negotiation, so we need to check that
+  if (mJsepTransceiver.IsStopped()) {
+    StopImpl();
+  }
+
+  mReceiver->SyncFromJsep(mJsepTransceiver);
+  mSender->SyncFromJsep(mJsepTransceiver);
+
+  // mid from JSEP
+  if (mJsepTransceiver.IsAssociated()) {
+    mMid = mJsepTransceiver.GetMid();
+  } else {
+    mMid = std::string();
+  }
+
+  // currentDirection from JSEP, but not if "this transceiver has never been
+  // represented in an offer/answer exchange"
+  if (mJsepTransceiver.HasLevel() && mJsepTransceiver.IsNegotiated()) {
+    if (mJsepTransceiver.mRecvTrack.GetActive()) {
+      if (mJsepTransceiver.mSendTrack.GetActive()) {
+        mCurrentDirection.SetValue(dom::RTCRtpTransceiverDirection::Sendrecv);
+        mHasBeenUsedToSend = true;
+      } else {
+        mCurrentDirection.SetValue(dom::RTCRtpTransceiverDirection::Recvonly);
+      }
+    } else {
+      if (mJsepTransceiver.mSendTrack.GetActive()) {
+        mCurrentDirection.SetValue(dom::RTCRtpTransceiverDirection::Sendonly);
+        mHasBeenUsedToSend = true;
+      } else {
+        mCurrentDirection.SetValue(dom::RTCRtpTransceiverDirection::Inactive);
+      }
+    }
+  }
+
+  mShouldRemove = mJsepTransceiver.IsRemoved();
+  mHasTransport = mJsepTransceiver.HasLevel() && !mJsepTransceiver.IsStopped();
+}
+
+void RTCRtpTransceiver::SyncToJsep(JsepSession& aSession) const {
+  MOZ_MTLOG(ML_DEBUG, mPc->GetHandle()
+                          << "[" << mMid.Ref() << "]: " << __FUNCTION__
+                          << " Syncing to JSEP transceiver");
+
+  aSession.ApplyToTransceiver(
+      mTransceiverId, [this, self = RefPtr<const RTCRtpTransceiver>(this)](
+                          JsepTransceiver& aTransceiver) {
+        mReceiver->SyncToJsep(aTransceiver);
+        mSender->SyncToJsep(aTransceiver);
+        aTransceiver.mJsDirection = ToSdpDirection(mDirection);
+        if (mStopped) {
+          aTransceiver.Stop();
+        }
+      });
+}
+
+void RTCRtpTransceiver::GetKind(nsAString& aKind) const {
+  // The transceiver kind of an RTCRtpTransceiver is defined by the kind of the
+  // associated RTCRtpReceiver's MediaStreamTrack object.
+  MOZ_ASSERT(mReceiver && mReceiver->Track());
+  mReceiver->Track()->GetKind(aKind);
+}
+
+void RTCRtpTransceiver::GetMid(nsAString& aMid) const {
+  if (!mMid.Ref().empty()) {
+    aMid = NS_ConvertUTF8toUTF16(mMid.Ref());
+  } else {
+    aMid.SetIsVoid(true);
+  }
+}
+
+std::string RTCRtpTransceiver::GetMidAscii() const {
+  if (mMid.Ref().empty()) {
+    return std::string();
+  }
+
+  return mMid.Ref();
+}
+
+void RTCRtpTransceiver::SetDirection(RTCRtpTransceiverDirection aDirection,
+                                     ErrorResult& aRv) {
+  if (mStopped) {
+    aRv.ThrowInvalidStateError("Transceiver is stopped!");
+    return;
+  }
+
+  if (aDirection == mDirection) {
+    return;
+  }
+
+  SetDirectionInternal(aDirection);
+
+  mPc->UpdateNegotiationNeeded();
+}
+
+void RTCRtpTransceiver::SetDirectionInternal(
+    RTCRtpTransceiverDirection aDirection) {
+  // We do not update the direction on the JsepTransceiver until sync
+  mDirection = aDirection;
+}
+
+bool RTCRtpTransceiver::ShouldRemove() const { return mShouldRemove; }
+
+bool RTCRtpTransceiver::CanSendDTMF() const {
+  // Spec says: "If connection's RTCPeerConnectionState is not "connected"
+  // return false." We don't support that right now. This is supposed to be
+  // true once ICE is complete, and _all_ DTLS handshakes are also complete. We
+  // don't really have access to the state of _all_ of our DTLS states either.
+  // Our pipeline _does_ know whether SRTP/SRTCP is ready, which happens
+  // immediately after our transport finishes DTLS (unless there was an error),
+  // so this is pretty close.
+  // TODO (bug 1265827): Base this on RTCPeerConnectionState instead.
+  // TODO (bug 1623193): Tighten this up
+  if (!IsSending() || !mSender->GetTrack()) {
+    return false;
+  }
+
+  // Ok, it looks like the connection is up and sending. Did we negotiate
+  // telephone-event?
+  const JsepTrackNegotiatedDetails* details =
+      mJsepTransceiver.mSendTrack.GetNegotiatedDetails();
+  if (NS_WARN_IF(!details || !details->GetEncodingCount())) {
+    // What?
+    return false;
+  }
+
+  for (size_t i = 0; i < details->GetEncodingCount(); ++i) {
+    const auto& encoding = details->GetEncoding(i);
+    for (const auto& codec : encoding.GetCodecs()) {
+      if (codec->mName == "telephone-event") {
+        return true;
+      }
+    }
+  }
+
+  return false;
+}
+
+JSObject* RTCRtpTransceiver::WrapObject(JSContext* aCx,
+                                        JS::Handle<JSObject*> aGivenProto) {
+  return dom::RTCRtpTransceiver_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+nsPIDOMWindowInner* RTCRtpTransceiver::GetParentObject() const {
+  return mWindow;
+}
+
+static void JsepCodecDescToAudioCodecConfig(
+    const JsepAudioCodecDescription& aCodec, Maybe<AudioCodecConfig>* aConfig) {
+  uint16_t pt;
+
+  // TODO(bug 1761272): Getting the pt for a JsepAudioCodecDescription should be
+  // infallible.
+  if (NS_WARN_IF(!aCodec.GetPtAsInt(&pt))) {
+    MOZ_MTLOG(ML_ERROR, "Invalid payload type: " << aCodec.mDefaultPt);
+    MOZ_ASSERT(false);
+    return;
+  }
+
+  // libwebrtc crashes if we attempt to configure a mono recv codec
+  bool sendMono = aCodec.mForceMono && aCodec.mDirection == sdp::kSend;
+
+  *aConfig = Some(AudioCodecConfig(
+      pt, aCodec.mName, static_cast<int>(aCodec.mClock),
+      sendMono ? 1 : static_cast<int>(aCodec.mChannels), aCodec.mFECEnabled));
+  (*aConfig)->mMaxPlaybackRate = static_cast<int>(aCodec.mMaxPlaybackRate);
+  (*aConfig)->mDtmfEnabled = aCodec.mDtmfEnabled;
+  (*aConfig)->mDTXEnabled = aCodec.mDTXEnabled;
+  (*aConfig)->mMaxAverageBitrate = aCodec.mMaxAverageBitrate;
+  (*aConfig)->mFrameSizeMs = aCodec.mFrameSizeMs;
+  (*aConfig)->mMinFrameSizeMs = aCodec.mMinFrameSizeMs;
+  (*aConfig)->mMaxFrameSizeMs = aCodec.mMaxFrameSizeMs;
+  (*aConfig)->mCbrEnabled = aCodec.mCbrEnabled;
+}
+
+// TODO: This and the next function probably should move to JsepTransceiver
+Maybe<const std::vector<UniquePtr<JsepCodecDescription>>&>
+RTCRtpTransceiver::GetNegotiatedSendCodecs() const {
+  if (!mJsepTransceiver.mSendTrack.GetActive()) {
+    return Nothing();
+  }
+
+  const auto* details = mJsepTransceiver.mSendTrack.GetNegotiatedDetails();
+  if (!details) {
+    return Nothing();
+  }
+
+  if (details->GetEncodingCount() == 0) {
+    return Nothing();
+  }
+
+  return SomeRef(details->GetEncoding(0).GetCodecs());
+}
+
+Maybe<const std::vector<UniquePtr<JsepCodecDescription>>&>
+RTCRtpTransceiver::GetNegotiatedRecvCodecs() const {
+  if (!mJsepTransceiver.mRecvTrack.GetActive()) {
+    return Nothing();
+  }
+
+  const auto* details = mJsepTransceiver.mRecvTrack.GetNegotiatedDetails();
+  if (!details) {
+    return Nothing();
+  }
+
+  if (details->GetEncodingCount() == 0) {
+    return Nothing();
+  }
+
+  return SomeRef(details->GetEncoding(0).GetCodecs());
+}
+
+// TODO: Maybe move this someplace else?
+/*static*/
+void RTCRtpTransceiver::NegotiatedDetailsToAudioCodecConfigs(
+    const JsepTrackNegotiatedDetails& aDetails,
+    std::vector<AudioCodecConfig>* aConfigs) {
+  Maybe<AudioCodecConfig> telephoneEvent;
+
+  if (aDetails.GetEncodingCount()) {
+    for (const auto& codec : aDetails.GetEncoding(0).GetCodecs()) {
+      if (NS_WARN_IF(codec->Type() != SdpMediaSection::kAudio)) {
+        MOZ_ASSERT(false, "Codec is not audio! This is a JSEP bug.");
+        return;
+      }
+      Maybe<AudioCodecConfig> config;
+      const JsepAudioCodecDescription& audio =
+          static_cast<const JsepAudioCodecDescription&>(*codec);
+      JsepCodecDescToAudioCodecConfig(audio, &config);
+      if (config->mName == "telephone-event") {
+        telephoneEvent = std::move(config);
+      } else {
+        aConfigs->push_back(std::move(*config));
+      }
+    }
+  }
+
+  // Put telephone event at the back, because webrtc.org crashes if we don't
+  // If we need to do even more sorting, we should use std::sort.
+  if (telephoneEvent) {
+    aConfigs->push_back(std::move(*telephoneEvent));
+  }
+}
+
+auto RTCRtpTransceiver::GetActivePayloadTypes() const
+    -> RefPtr<ActivePayloadTypesPromise> {
+  if (!mConduit) {
+    return ActivePayloadTypesPromise::CreateAndResolve(PayloadTypes(),
+                                                       __func__);
+  }
+
+  if (!mCallWrapper) {
+    return ActivePayloadTypesPromise::CreateAndResolve(PayloadTypes(),
+                                                       __func__);
+  }
+
+  return InvokeAsync(mCallWrapper->mCallThread, __func__,
+                     [conduit = mConduit]() {
+                       PayloadTypes pts;
+                       pts.mSendPayloadType = conduit->ActiveSendPayloadType();
+                       pts.mRecvPayloadType = conduit->ActiveRecvPayloadType();
+                       return ActivePayloadTypesPromise::CreateAndResolve(
+                           std::move(pts), __func__);
+                     });
+}
+
+static void JsepCodecDescToVideoCodecConfig(
+    const JsepVideoCodecDescription& aCodec, Maybe<VideoCodecConfig>* aConfig) {
+  uint16_t pt;
+
+  // TODO(bug 1761272): Getting the pt for a JsepVideoCodecDescription should be
+  // infallible.
+  if (NS_WARN_IF(!aCodec.GetPtAsInt(&pt))) {
+    MOZ_MTLOG(ML_ERROR, "Invalid payload type: " << aCodec.mDefaultPt);
+    MOZ_ASSERT(false);
+    return;
+  }
+
+  UniquePtr<VideoCodecConfigH264> h264Config;
+
+  if (aCodec.mName == "H264") {
+    h264Config = MakeUnique<VideoCodecConfigH264>();
+    size_t spropSize = sizeof(h264Config->sprop_parameter_sets);
+    strncpy(h264Config->sprop_parameter_sets,
+            aCodec.mSpropParameterSets.c_str(), spropSize);
+    h264Config->sprop_parameter_sets[spropSize - 1] = '\0';
+    h264Config->packetization_mode =
+        static_cast<int>(aCodec.mPacketizationMode);
+    h264Config->profile_level_id = static_cast<int>(aCodec.mProfileLevelId);
+    h264Config->tias_bw = 0;  // TODO(bug 1403206)
+  }
+
+  *aConfig = Some(VideoCodecConfig(pt, aCodec.mName, aCodec.mConstraints,
+                                   h264Config.get()));
+
+  (*aConfig)->mAckFbTypes = aCodec.mAckFbTypes;
+  (*aConfig)->mNackFbTypes = aCodec.mNackFbTypes;
+  (*aConfig)->mCcmFbTypes = aCodec.mCcmFbTypes;
+  (*aConfig)->mRembFbSet = aCodec.RtcpFbRembIsSet();
+  (*aConfig)->mFECFbSet = aCodec.mFECEnabled;
+  (*aConfig)->mTransportCCFbSet = aCodec.RtcpFbTransportCCIsSet();
+  if (aCodec.mFECEnabled) {
+    uint16_t pt;
+    if (SdpHelper::GetPtAsInt(aCodec.mREDPayloadType, &pt)) {
+      (*aConfig)->mREDPayloadType = pt;
+    }
+    if (SdpHelper::GetPtAsInt(aCodec.mULPFECPayloadType, &pt)) {
+      (*aConfig)->mULPFECPayloadType = pt;
+    }
+  }
+  if (aCodec.mRtxEnabled) {
+    uint16_t pt;
+    if (SdpHelper::GetPtAsInt(aCodec.mRtxPayloadType, &pt)) {
+      (*aConfig)->mRTXPayloadType = pt;
+    }
+  }
+}
+
+// TODO: Maybe move this someplace else?
+/*static*/
+void RTCRtpTransceiver::NegotiatedDetailsToVideoCodecConfigs(
+    const JsepTrackNegotiatedDetails& aDetails,
+    std::vector<VideoCodecConfig>* aConfigs) {
+  if (aDetails.GetEncodingCount()) {
+    for (const auto& codec : aDetails.GetEncoding(0).GetCodecs()) {
+      if (NS_WARN_IF(codec->Type() != SdpMediaSection::kVideo)) {
+        MOZ_ASSERT(false, "Codec is not video! This is a JSEP bug.");
+        return;
+      }
+      Maybe<VideoCodecConfig> config;
+      const JsepVideoCodecDescription& video =
+          static_cast<const JsepVideoCodecDescription&>(*codec);
+
+      JsepCodecDescToVideoCodecConfig(video, &config);
+
+      config->mTias = aDetails.GetTias();
+
+      for (size_t i = 0; i < aDetails.GetEncodingCount(); ++i) {
+        const JsepTrackEncoding& jsepEncoding(aDetails.GetEncoding(i));
+        if (jsepEncoding.HasFormat(video.mDefaultPt)) {
+          VideoCodecConfig::Encoding encoding;
+          encoding.rid = jsepEncoding.mRid;
+          config->mEncodings.push_back(encoding);
+        }
+      }
+
+      aConfigs->push_back(std::move(*config));
+    }
+  }
+}
+
+void RTCRtpTransceiver::Stop(ErrorResult& aRv) {
+  if (mPc->IsClosed()) {
+    aRv.ThrowInvalidStateError("Peer connection is closed");
+    return;
+  }
+
+  StopImpl();
+  mPc->UpdateNegotiationNeeded();
+}
+
+void RTCRtpTransceiver::StopImpl() {
+  if (mStopped) {
+    return;
+  }
+
+  if (mCallWrapper) {
+    auto conduit = std::move(mConduit);
+    (conduit ? conduit->Shutdown()
+             : GenericPromise::CreateAndResolve(true, __func__))
+        ->Then(GetMainThreadSerialEventTarget(), __func__,
+               [sender = mSender, receiver = mReceiver]() mutable {
+                 // Shutdown pipelines when conduits are guaranteed shut down,
+                 // so that all packets sent from conduits can be delivered.
+                 sender->Shutdown();
+                 receiver->Shutdown();
+               });
+    mCallWrapper = nullptr;
+  }
+  mStopped = true;
+  mCurrentDirection.SetNull();
+
+  mSender->Stop();
+  mReceiver->Stop();
+
+  auto self = nsMainThreadPtrHandle<RTCRtpTransceiver>(
+      new nsMainThreadPtrHolder<RTCRtpTransceiver>(
+          "RTCRtpTransceiver::StopImpl::self", this, false));
+  mStsThread->Dispatch(NS_NewRunnableFunction(
+      __func__, [self] { self->mTransportHandler = nullptr; }));
+}
+
+bool RTCRtpTransceiver::IsVideo() const { return mIsVideo; }
+
+bool RTCRtpTransceiver::IsSending() const {
+  return mCurrentDirection == Nullable(RTCRtpTransceiverDirection::Sendonly) ||
+         mCurrentDirection == Nullable(RTCRtpTransceiverDirection::Sendrecv);
+}
+
+bool RTCRtpTransceiver::IsReceiving() const {
+  return mCurrentDirection == Nullable(RTCRtpTransceiverDirection::Recvonly) ||
+         mCurrentDirection == Nullable(RTCRtpTransceiverDirection::Sendrecv);
+}
+
+void RTCRtpTransceiver::ChainToDomPromiseWithCodecStats(
+    nsTArray<RefPtr<RTCStatsPromise>> aStats,
+    const RefPtr<dom::Promise>& aDomPromise) {
+  nsTArray<RTCCodecStats> codecStats =
+      mPc->GetCodecStats(mPc->GetTimestampMaker().GetNow().ToDom());
+
+  AutoTArray<
+      std::tuple<RTCRtpTransceiver*, RefPtr<RTCStatsPromise::AllPromiseType>>,
+      1>
+      statsPromises;
+  statsPromises.AppendElement(std::make_tuple(
+      this, RTCStatsPromise::All(GetMainThreadSerialEventTarget(), aStats)));
+
+  ApplyCodecStats(std::move(codecStats), std::move(statsPromises))
+      ->Then(
+          GetMainThreadSerialEventTarget(), __func__,
+          [aDomPromise, window = mWindow,
+           idGen = mIdGenerator](UniquePtr<RTCStatsCollection> aStats) mutable {
+            // Rewrite ids and merge stats collections into the final report.
+            AutoTArray<UniquePtr<RTCStatsCollection>, 1> stats;
+            stats.AppendElement(std::move(aStats));
+
+            RTCStatsCollection opaqueStats;
+            idGen->RewriteIds(std::move(stats), &opaqueStats);
+
+            RefPtr<RTCStatsReport> report(new RTCStatsReport(window));
+            report->Incorporate(opaqueStats);
+
+            aDomPromise->MaybeResolve(std::move(report));
+          },
+          [aDomPromise](nsresult aError) {
+            aDomPromise->MaybeReject(NS_ERROR_FAILURE);
+          });
+}
+
+RefPtr<RTCStatsPromise> RTCRtpTransceiver::ApplyCodecStats(
+    nsTArray<RTCCodecStats> aCodecStats,
+    nsTArray<
+        std::tuple<RTCRtpTransceiver*, RefPtr<RTCStatsPromise::AllPromiseType>>>
+        aTransceiverStatsPromises) {
+  MOZ_ASSERT(NS_IsMainThread());
+  // The process here is roughly:
+  // - Gather all inputs to the codec filtering process, including:
+  //   - Each transceiver's transportIds
+  //   - Each transceiver's active payload types (resolved)
+  //   - Each transceiver's resolved stats
+  //
+  //   Waiting (async) for multiple promises of different types is not supported
+  //   by the MozPromise API (bug 1752318), so we are a bit finicky here. We
+  //   create media::Refcountables of the types we want to resolve, and let
+  //   these be shared across Then-functions through RefPtrs.
+  //
+  // - For each active payload type in a transceiver:
+  //   - Register the codec stats for this payload type and transport if we
+  //     haven't already done so
+  //   - If it was a send payload type, assign the codec stats id for this
+  //     payload type and transport to the transceiver's outbound-rtp and
+  //     remote-inbound-rtp stats as codecId
+  //   - If it was a recv payload type, assign the codec stats id for this
+  //     payload type and transport to the transceiver's inbound-rtp and
+  //     remote-outbound-rtp stats as codecId
+  //
+  // - Flatten all transceiver stats collections into one, and set the
+  //   registered codec stats on it
+
+  // Wrap codec stats in a Refcountable<> to allow sharing across promise
+  // handlers.
+  auto codecStats = MakeRefPtr<media::Refcountable<nsTArray<RTCCodecStats>>>();
+  *codecStats = std::move(aCodecStats);
+
+  struct IdComparator {
+    bool operator()(const RTCCodecStats& aA, const RTCCodecStats& aB) const {
+      return aA.mId.Value() < aB.mId.Value();
+    }
+  };
+
+  // Stores distinct codec stats by id; to avoid dupes within a transport.
+  auto finalCodecStats =
+      MakeRefPtr<media::Refcountable<std::set<RTCCodecStats, IdComparator>>>();
+
+  // All the transceiver rtp stream stats in a single array. These stats will,
+  // when resolved, contain codecIds.
+  nsTArray<RefPtr<RTCStatsPromise>> promises(
+      aTransceiverStatsPromises.Length());
+
+  for (const auto& [transceiver, allPromise] : aTransceiverStatsPromises) {
+    // Per transceiver, gather up what we need to assign codecId to this
+    // transceiver's rtp stream stats. Register codec stats while we're at it.
+    auto payloadTypes =
+        MakeRefPtr<media::Refcountable<RTCRtpTransceiver::PayloadTypes>>();
+    promises.AppendElement(
+        transceiver->GetActivePayloadTypes()
+            ->Then(
+                GetMainThreadSerialEventTarget(), __func__,
+                [payloadTypes, allPromise = allPromise](
+                    RTCRtpTransceiver::PayloadTypes aPayloadTypes) {
+                  // Forward active payload types to the next Then-handler.
+                  *payloadTypes = std::move(aPayloadTypes);
+                  return allPromise;
+                },
+                [] {
+                  MOZ_CRASH("Unexpected reject");
+                  return RTCStatsPromise::AllPromiseType::CreateAndReject(
+                      NS_ERROR_UNEXPECTED, __func__);
+                })
+            ->Then(
+                GetMainThreadSerialEventTarget(), __func__,
+                [codecStats, finalCodecStats, payloadTypes,
+                 transportId =
+                     NS_ConvertASCIItoUTF16(transceiver->GetTransportId())](
+                    nsTArray<UniquePtr<RTCStatsCollection>>
+                        aTransceiverStats) mutable {
+                  // We have all the data we need to register codec stats and
+                  // assign codecIds for this transceiver's rtp stream stats.
+
+                  auto report = MakeUnique<RTCStatsCollection>();
+                  FlattenStats(std::move(aTransceiverStats), report.get());
+
+                  // Find the codec stats we are looking for, based on the
+                  // transportId and the active payload types.
+                  Maybe<RTCCodecStats&> sendCodec;
+                  Maybe<RTCCodecStats&> recvCodec;
+                  for (auto& codec : *codecStats) {
+                    if (payloadTypes->mSendPayloadType.isSome() ==
+                            sendCodec.isSome() &&
+                        payloadTypes->mRecvPayloadType.isSome() ==
+                            recvCodec.isSome()) {
+                      // We have found all the codec stats we were looking for.
+                      break;
+                    }
+                    if (codec.mTransportId != transportId) {
+                      continue;
+                    }
+                    if (payloadTypes->mSendPayloadType &&
+                        *payloadTypes->mSendPayloadType ==
+                            static_cast<int>(codec.mPayloadType) &&
+                        (!codec.mCodecType.WasPassed() ||
+                         codec.mCodecType.Value() == RTCCodecType::Encode)) {
+                      MOZ_ASSERT(!sendCodec,
+                                 "At most one send codec stat per transceiver");
+                      sendCodec = SomeRef(codec);
+                    }
+                    if (payloadTypes->mRecvPayloadType &&
+                        *payloadTypes->mRecvPayloadType ==
+                            static_cast<int>(codec.mPayloadType) &&
+                        (!codec.mCodecType.WasPassed() ||
+                         codec.mCodecType.Value() == RTCCodecType::Decode)) {
+                      MOZ_ASSERT(!recvCodec,
+                                 "At most one recv codec stat per transceiver");
+                      recvCodec = SomeRef(codec);
+                    }
+                  }
+
+                  // Register and assign codecIds for the found codec stats.
+                  if (sendCodec) {
+                    finalCodecStats->insert(*sendCodec);
+                    for (auto& stat : report->mOutboundRtpStreamStats) {
+                      stat.mCodecId.Construct(sendCodec->mId.Value());
+                    }
+                    for (auto& stat : report->mRemoteInboundRtpStreamStats) {
+                      stat.mCodecId.Construct(sendCodec->mId.Value());
+                    }
+                  }
+                  if (recvCodec) {
+                    finalCodecStats->insert(*recvCodec);
+                    for (auto& stat : report->mInboundRtpStreamStats) {
+                      stat.mCodecId.Construct(recvCodec->mId.Value());
+                    }
+                    for (auto& stat : report->mRemoteOutboundRtpStreamStats) {
+                      stat.mCodecId.Construct(recvCodec->mId.Value());
+                    }
+                  }
+
+                  return RTCStatsPromise::CreateAndResolve(std::move(report),
+                                                           __func__);
+                },
+                [] {
+                  MOZ_CRASH("Unexpected reject");
+                  return RTCStatsPromise::CreateAndReject(NS_ERROR_UNEXPECTED,
+                                                          __func__);
+                }));
+  }
+
+  return RTCStatsPromise::All(GetMainThreadSerialEventTarget(), promises)
+      ->Then(
+          GetMainThreadSerialEventTarget(), __func__,
+          [finalCodecStats = std::move(finalCodecStats)](
+              nsTArray<UniquePtr<RTCStatsCollection>> aStats) mutable {
+            auto finalStats = MakeUnique<RTCStatsCollection>();
+            FlattenStats(std::move(aStats), finalStats.get());
+            MOZ_ASSERT(finalStats->mCodecStats.IsEmpty());
+            if (!finalStats->mCodecStats.SetCapacity(finalCodecStats->size(),
+                                                     fallible)) {
+              mozalloc_handle_oom(0);
+            }
+            while (!finalCodecStats->empty()) {
+              auto node = finalCodecStats->extract(finalCodecStats->begin());
+              if (!finalStats->mCodecStats.AppendElement(
+                      std::move(node.value()), fallible)) {
+                mozalloc_handle_oom(0);
+              }
+            }
+            return RTCStatsPromise::CreateAndResolve(std::move(finalStats),
+                                                     __func__);
+          },
+          [] {
+            MOZ_CRASH("Unexpected reject");
+            return RTCStatsPromise::CreateAndReject(NS_ERROR_UNEXPECTED,
+                                                    __func__);
+          });
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/jsapi/RTCRtpTransceiver.h b/dom/media/webrtc/jsapi/RTCRtpTransceiver.h
new file mode 100644
index 0000000000..4830e465a0
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCRtpTransceiver.h
@@ -0,0 +1,243 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+#ifndef _TRANSCEIVERIMPL_H_
+#define _TRANSCEIVERIMPL_H_
+
+#include <string>
+#include "mozilla/StateMirroring.h"
+#include "mozilla/RefPtr.h"
+#include "nsCOMPtr.h"
+#include "nsISerialEventTarget.h"
+#include "nsTArray.h"
+#include "mozilla/dom/MediaStreamTrack.h"
+#include "ErrorList.h"
+#include "jsep/JsepSession.h"
+#include "transport/transportlayer.h"  // For TransportLayer::State
+#include "mozilla/dom/RTCRtpTransceiverBinding.h"
+#include "RTCStatsReport.h"
+
+class nsIPrincipal;
+
+namespace mozilla {
+class PeerIdentity;
+class MediaSessionConduit;
+class VideoSessionConduit;
+class AudioSessionConduit;
+struct AudioCodecConfig;
+class VideoCodecConfig;  // Why is this a class, but AudioCodecConfig a struct?
+class MediaPipelineTransmit;
+class MediaPipeline;
+class MediaPipelineFilter;
+class MediaTransportHandler;
+class RTCStatsIdGenerator;
+class WebrtcCallWrapper;
+class JsepTrackNegotiatedDetails;
+class PeerConnectionImpl;
+enum class PrincipalPrivacy : uint8_t;
+
+namespace dom {
+class RTCDtlsTransport;
+class RTCDTMFSender;
+class RTCRtpTransceiver;
+struct RTCRtpSourceEntry;
+class RTCRtpReceiver;
+class RTCRtpSender;
+
+/**
+ * This is what ties all the various pieces that make up a transceiver
+ * together. This includes:
+ * MediaStreamTrack for rendering and capture
+ * MediaTransportHandler for RTP transmission/reception
+ * Audio/VideoConduit for feeding RTP/RTCP into webrtc.org for decoding, and
+ * feeding audio/video frames into webrtc.org for encoding into RTP/RTCP.
+ */
+class RTCRtpTransceiver : public nsISupports, public nsWrapperCache {
+ public:
+  /**
+   * |aSendTrack| might or might not be set.
+   */
+  RTCRtpTransceiver(
+      nsPIDOMWindowInner* aWindow, bool aPrivacyNeeded, PeerConnectionImpl* aPc,
+      MediaTransportHandler* aTransportHandler, JsepSession* aJsepSession,
+      const std::string& aTransceiverId, bool aIsVideo,
+      nsISerialEventTarget* aStsThread, MediaStreamTrack* aSendTrack,
+      WebrtcCallWrapper* aCallWrapper, RTCStatsIdGenerator* aIdGenerator);
+
+  void Init(const RTCRtpTransceiverInit& aInit, ErrorResult& aRv);
+
+  bool IsValid() const { return !!mConduit; }
+
+  nsresult UpdateTransport();
+
+  nsresult UpdateConduit();
+
+  void UpdatePrincipalPrivacy(PrincipalPrivacy aPrivacy);
+
+  void ResetSync();
+
+  nsresult SyncWithMatchingVideoConduits(
+      nsTArray<RefPtr<RTCRtpTransceiver>>& transceivers);
+
+  void Close();
+
+  void BreakCycles();
+
+  bool ConduitHasPluginID(uint64_t aPluginID);
+
+  // for webidl
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+  nsPIDOMWindowInner* GetParentObject() const;
+  RTCRtpReceiver* Receiver() const { return mReceiver; }
+  RTCRtpSender* Sender() const { return mSender; }
+  RTCDtlsTransport* GetDtlsTransport() const { return mDtlsTransport; }
+  void GetKind(nsAString& aKind) const;
+  void GetMid(nsAString& aMid) const;
+  RTCRtpTransceiverDirection Direction() const { return mDirection; }
+  void SetDirection(RTCRtpTransceiverDirection aDirection, ErrorResult& aRv);
+  Nullable<RTCRtpTransceiverDirection> GetCurrentDirection() {
+    return mCurrentDirection;
+  }
+  void Stop(ErrorResult& aRv);
+  void SetDirectionInternal(RTCRtpTransceiverDirection aDirection);
+  bool HasBeenUsedToSend() const { return mHasBeenUsedToSend; }
+
+  bool CanSendDTMF() const;
+  bool Stopped() const { return mStopped; }
+  void SyncToJsep(JsepSession& aSession) const;
+  void SyncFromJsep(const JsepSession& aSession);
+  std::string GetMidAscii() const;
+
+  void SetDtlsTransport(RTCDtlsTransport* aDtlsTransport, bool aStable);
+  void RollbackToStableDtlsTransport();
+
+  std::string GetTransportId() const {
+    return mJsepTransceiver.mTransport.mTransportId;
+  }
+
+  JsepTransceiver& GetJsepTransceiver() { return mJsepTransceiver; }
+
+  bool IsVideo() const;
+
+  bool IsSending() const;
+
+  bool IsReceiving() const;
+
+  bool ShouldRemove() const;
+
+  Maybe<const std::vector<UniquePtr<JsepCodecDescription>>&>
+  GetNegotiatedSendCodecs() const;
+
+  Maybe<const std::vector<UniquePtr<JsepCodecDescription>>&>
+  GetNegotiatedRecvCodecs() const;
+
+  struct PayloadTypes {
+    Maybe<int> mSendPayloadType;
+    Maybe<int> mRecvPayloadType;
+  };
+  using ActivePayloadTypesPromise = MozPromise<PayloadTypes, nsresult, true>;
+  RefPtr<ActivePayloadTypesPromise> GetActivePayloadTypes() const;
+
+  MediaSessionConduit* GetConduit() const { return mConduit; }
+
+  const std::string& GetJsepTransceiverId() const { return mTransceiverId; }
+
+  void SetRemovedFromPc() { mHandlingUnlink = true; }
+
+  // nsISupports
+  NS_DECL_CYCLE_COLLECTING_ISUPPORTS
+  NS_DECL_CYCLE_COLLECTION_WRAPPERCACHE_CLASS(RTCRtpTransceiver)
+
+  static void NegotiatedDetailsToAudioCodecConfigs(
+      const JsepTrackNegotiatedDetails& aDetails,
+      std::vector<AudioCodecConfig>* aConfigs);
+
+  static void NegotiatedDetailsToVideoCodecConfigs(
+      const JsepTrackNegotiatedDetails& aDetails,
+      std::vector<VideoCodecConfig>* aConfigs);
+
+  /* Returns a promise that will contain the stats in aStats, along with the
+   * codec stats (which is a PC-wide thing) */
+  void ChainToDomPromiseWithCodecStats(nsTArray<RefPtr<RTCStatsPromise>> aStats,
+                                       const RefPtr<Promise>& aDomPromise);
+
+  /**
+   * Takes a set of codec stats (per-peerconnection) and a set of
+   * transceiver/transceiver-stats-promise tuples. Filters out all referenced
+   * codec stats based on the transceiver's transport and rtp stream stats.
+   * Finally returns the flattened stats containing the filtered codec stats and
+   * all given per-transceiver-stats.
+   */
+  static RefPtr<RTCStatsPromise> ApplyCodecStats(
+      nsTArray<RTCCodecStats> aCodecStats,
+      nsTArray<std::tuple<RTCRtpTransceiver*,
+                          RefPtr<RTCStatsPromise::AllPromiseType>>>
+          aTransceiverStatsPromises);
+
+  AbstractCanonical<std::string>* CanonicalMid() { return &mMid; }
+  AbstractCanonical<std::string>* CanonicalSyncGroup() { return &mSyncGroup; }
+
+ private:
+  virtual ~RTCRtpTransceiver();
+  void InitAudio();
+  void InitVideo(const TrackingId& aRecvTrackingId);
+  void InitConduitControl();
+  void StopImpl();
+
+  nsCOMPtr<nsPIDOMWindowInner> mWindow;
+  RefPtr<PeerConnectionImpl> mPc;
+  RefPtr<MediaTransportHandler> mTransportHandler;
+  const std::string mTransceiverId;
+  // Copy of latest from the JSEP engine.
+  JsepTransceiver mJsepTransceiver;
+  nsCOMPtr<nsISerialEventTarget> mStsThread;
+  // state for webrtc.org that is shared between all transceivers
+  RefPtr<WebrtcCallWrapper> mCallWrapper;
+  RefPtr<MediaStreamTrack> mSendTrack;
+  RefPtr<RTCStatsIdGenerator> mIdGenerator;
+  RefPtr<MediaSessionConduit> mConduit;
+  // The spec says both RTCRtpReceiver and RTCRtpSender have a slot for
+  // an RTCDtlsTransport.  They are always the same, so we'll store it
+  // here.
+  RefPtr<RTCDtlsTransport> mDtlsTransport;
+  // The spec says both RTCRtpReceiver and RTCRtpSender have a slot for
+  // a last stable state RTCDtlsTransport.  They are always the same, so
+  // we'll store it here.
+  RefPtr<RTCDtlsTransport> mLastStableDtlsTransport;
+  RefPtr<RTCRtpReceiver> mReceiver;
+  RefPtr<RTCRtpSender> mSender;
+  RTCRtpTransceiverDirection mDirection = RTCRtpTransceiverDirection::Sendrecv;
+  Nullable<RTCRtpTransceiverDirection> mCurrentDirection;
+  bool mStopped = false;
+  bool mShutdown = false;
+  bool mHasBeenUsedToSend = false;
+  PrincipalPrivacy mPrincipalPrivacy;
+  bool mShouldRemove = false;
+  bool mHasTransport = false;
+  bool mIsVideo;
+  // This is really nasty. Most of the time, PeerConnectionImpl needs to be in
+  // charge of unlinking each RTCRtpTransceiver, because it needs to perform
+  // stats queries on its way out, which requires all of the RTCRtpTransceivers
+  // (and their transitive dependencies) to stick around until those stats
+  // queries are finished. However, when an RTCRtpTransceiver is removed from
+  // the PeerConnectionImpl due to negotiation, the PeerConnectionImpl
+  // releases its reference, which means the PeerConnectionImpl cannot be in
+  // charge of the unlink anymore. We cannot do the unlink when this reference
+  // is released either, because RTCRtpTransceiver might have some work it needs
+  // to do first. Also, JS may be maintaining a reference to the
+  // RTCRtpTransceiver (or one of its dependencies), which means it must remain
+  // fully functional after it is removed (meaning it cannot release any of its
+  // dependencies, or vice versa).
+  bool mHandlingUnlink = false;
+  std::string mTransportId;
+
+  Canonical<std::string> mMid;
+  Canonical<std::string> mSyncGroup;
+};
+
+}  // namespace dom
+
+}  // namespace mozilla
+
+#endif  // _TRANSCEIVERIMPL_H_
diff --git a/dom/media/webrtc/jsapi/RTCSctpTransport.cpp b/dom/media/webrtc/jsapi/RTCSctpTransport.cpp
new file mode 100644
index 0000000000..63424968ae
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCSctpTransport.cpp
@@ -0,0 +1,53 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "RTCSctpTransport.h"
+#include "mozilla/dom/Event.h"
+#include "mozilla/dom/EventBinding.h"
+#include "mozilla/dom/RTCSctpTransportBinding.h"
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(RTCSctpTransport, DOMEventTargetHelper,
+                                   mDtlsTransport)
+
+NS_IMPL_ADDREF_INHERITED(RTCSctpTransport, DOMEventTargetHelper)
+NS_IMPL_RELEASE_INHERITED(RTCSctpTransport, DOMEventTargetHelper)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(RTCSctpTransport)
+  NS_WRAPPERCACHE_INTERFACE_MAP_ENTRY
+NS_INTERFACE_MAP_END_INHERITING(DOMEventTargetHelper)
+
+RTCSctpTransport::RTCSctpTransport(nsPIDOMWindowInner* aWindow,
+                                   RTCDtlsTransport& aDtlsTransport,
+                                   double aMaxMessageSize,
+                                   const Nullable<uint16_t>& aMaxChannels)
+    : DOMEventTargetHelper(aWindow),
+      mState(RTCSctpTransportState::Connecting),
+      mDtlsTransport(&aDtlsTransport),
+      mMaxMessageSize(aMaxMessageSize),
+      mMaxChannels(aMaxChannels) {}
+
+JSObject* RTCSctpTransport::WrapObject(JSContext* aCx,
+                                       JS::Handle<JSObject*> aGivenProto) {
+  return RTCSctpTransport_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+void RTCSctpTransport::UpdateState(RTCSctpTransportState aState) {
+  if (mState == RTCSctpTransportState::Closed || mState == aState) {
+    return;
+  }
+
+  mState = aState;
+
+  EventInit init;
+  init.mBubbles = false;
+  init.mCancelable = false;
+
+  RefPtr<Event> event = Event::Constructor(this, u"statechange"_ns, init);
+
+  DispatchTrustedEvent(event);
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webrtc/jsapi/RTCSctpTransport.h b/dom/media/webrtc/jsapi/RTCSctpTransport.h
new file mode 100644
index 0000000000..8e21db5e73
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCSctpTransport.h
@@ -0,0 +1,65 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _RTCSctpTransport_h_
+#define _RTCSctpTransport_h_
+
+#include "mozilla/DOMEventTargetHelper.h"
+#include "mozilla/RefPtr.h"
+#include "js/RootingAPI.h"
+#include "RTCDtlsTransport.h"
+
+class nsPIDOMWindowInner;
+
+namespace mozilla::dom {
+
+enum class RTCSctpTransportState : uint8_t;
+
+class RTCSctpTransport : public DOMEventTargetHelper {
+ public:
+  explicit RTCSctpTransport(nsPIDOMWindowInner* aWindow,
+                            RTCDtlsTransport& aDtlsTransport,
+                            double aMaxMessageSize,
+                            const Nullable<uint16_t>& aMaxChannels);
+
+  // nsISupports
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(RTCSctpTransport,
+                                           DOMEventTargetHelper)
+
+  // webidl
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+  IMPL_EVENT_HANDLER(statechange)
+
+  RTCDtlsTransport* Transport() const { return mDtlsTransport; }
+  RTCSctpTransportState State() const { return mState; }
+  double MaxMessageSize() const { return mMaxMessageSize; }
+  Nullable<uint16_t> GetMaxChannels() const { return mMaxChannels; }
+
+  void SetTransport(RTCDtlsTransport& aTransport) {
+    mDtlsTransport = &aTransport;
+  }
+
+  void SetMaxMessageSize(double aMaxMessageSize) {
+    mMaxMessageSize = aMaxMessageSize;
+  }
+
+  void SetMaxChannels(const Nullable<uint16_t>& aMaxChannels) {
+    mMaxChannels = aMaxChannels;
+  }
+
+  void UpdateState(RTCSctpTransportState aState);
+
+ private:
+  virtual ~RTCSctpTransport() = default;
+
+  RTCSctpTransportState mState;
+  RefPtr<RTCDtlsTransport> mDtlsTransport;
+  double mMaxMessageSize;
+  Nullable<uint16_t> mMaxChannels;
+};
+
+}  // namespace mozilla::dom
+#endif  // _RTCSctpTransport_h_
diff --git a/dom/media/webrtc/jsapi/RTCStatsIdGenerator.cpp b/dom/media/webrtc/jsapi/RTCStatsIdGenerator.cpp
new file mode 100644
index 0000000000..8b0462e223
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCStatsIdGenerator.cpp
@@ -0,0 +1,90 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+
+#include "RTCStatsIdGenerator.h"
+
+#include <iostream>
+
+#include "mozilla/RandomNum.h"
+#include "RTCStatsReport.h"
+#include "WebrtcGlobal.h"
+
+namespace mozilla {
+
+RTCStatsIdGenerator::RTCStatsIdGenerator()
+    : mSalt(RandomUint64().valueOr(0xa5a5a5a5)), mCounter(0) {}
+
+void RTCStatsIdGenerator::RewriteIds(
+    nsTArray<UniquePtr<dom::RTCStatsCollection>> aFromStats,
+    dom::RTCStatsCollection* aIntoReport) {
+  // Rewrite an Optional id
+  auto rewriteId = [&](dom::Optional<nsString>& id) {
+    if (id.WasPassed()) {
+      id.Value() = Id(id.Value());
+    }
+  };
+
+  auto rewriteIds = [&](auto& aList, auto... aParam) {
+    for (auto& stat : aList) {
+      (rewriteId(stat.*aParam), ...);
+    }
+  };
+
+  // Involves a lot of copying, since webidl dictionaries don't have
+  // move semantics. Oh well.
+
+  // Create a temporary to avoid double-rewriting any stats already in
+  // aIntoReport.
+  auto stats = MakeUnique<dom::RTCStatsCollection>();
+  dom::FlattenStats(std::move(aFromStats), stats.get());
+
+  using S = dom::RTCStats;
+  using ICPS = dom::RTCIceCandidatePairStats;
+  using RSS = dom::RTCRtpStreamStats;
+  using IRSS = dom::RTCInboundRtpStreamStats;
+  using ORSS = dom::RTCOutboundRtpStreamStats;
+  using RIRSS = dom::RTCRemoteInboundRtpStreamStats;
+  using RORSS = dom::RTCRemoteOutboundRtpStreamStats;
+
+  rewriteIds(stats->mIceCandidatePairStats, &S::mId, &ICPS::mLocalCandidateId,
+             &ICPS::mRemoteCandidateId);
+  rewriteIds(stats->mIceCandidateStats, &S::mId);
+  rewriteIds(stats->mInboundRtpStreamStats, &S::mId, &IRSS::mRemoteId,
+             &RSS::mCodecId);
+  rewriteIds(stats->mOutboundRtpStreamStats, &S::mId, &ORSS::mRemoteId,
+             &RSS::mCodecId);
+  rewriteIds(stats->mRemoteInboundRtpStreamStats, &S::mId, &RIRSS::mLocalId,
+             &RSS::mCodecId);
+  rewriteIds(stats->mRemoteOutboundRtpStreamStats, &S::mId, &RORSS::mLocalId,
+             &RSS::mCodecId);
+  rewriteIds(stats->mCodecStats, &S::mId);
+  rewriteIds(stats->mRtpContributingSourceStats, &S::mId);
+  rewriteIds(stats->mTrickledIceCandidateStats, &S::mId);
+  rewriteIds(stats->mDataChannelStats, &S::mId);
+
+  dom::MergeStats(std::move(stats), aIntoReport);
+}
+
+nsString RTCStatsIdGenerator::Id(const nsString& aKey) {
+  if (!aKey.Length()) {
+    MOZ_ASSERT(aKey.Length(), "Stats IDs should never be empty.");
+    return aKey;
+  }
+  if (mAllocated.find(aKey) == mAllocated.end()) {
+    mAllocated[aKey] = Generate();
+  }
+  return mAllocated[aKey];
+}
+
+nsString RTCStatsIdGenerator::Generate() {
+  auto random = RandomUint64().valueOr(0x1a22);
+  auto idNum = static_cast<uint32_t>(mSalt ^ ((mCounter++ << 16) | random));
+  nsString id;
+  id.AppendInt(idNum, 16);  // Append as hex
+  return id;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/jsapi/RTCStatsIdGenerator.h b/dom/media/webrtc/jsapi/RTCStatsIdGenerator.h
new file mode 100644
index 0000000000..1391c029ad
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCStatsIdGenerator.h
@@ -0,0 +1,42 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+
+#ifndef _RTCSTATSIDGENERATOR_H_
+#define _RTCSTATSIDGENERATOR_H_
+
+#include <map>
+
+#include "mozilla/Atomics.h"
+#include "mozilla/UniquePtr.h"
+#include "nsISupportsImpl.h"
+#include "nsString.h"
+#include "nsTArray.h"
+
+namespace mozilla {
+namespace dom {
+struct RTCStatsCollection;
+}  // namespace dom
+
+class RTCStatsIdGenerator {
+ public:
+  RTCStatsIdGenerator();
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(RTCStatsIdGenerator);
+
+  void RewriteIds(nsTArray<UniquePtr<dom::RTCStatsCollection>> aFromStats,
+                  dom::RTCStatsCollection* aIntoReport);
+
+ private:
+  virtual ~RTCStatsIdGenerator(){};
+  nsString Id(const nsString& aKey);
+  nsString Generate();
+
+  const uint64_t mSalt;
+  uint64_t mCounter;
+  std::map<nsString, nsString> mAllocated;
+};
+
+}  // namespace mozilla
+#endif
diff --git a/dom/media/webrtc/jsapi/RTCStatsReport.cpp b/dom/media/webrtc/jsapi/RTCStatsReport.cpp
new file mode 100644
index 0000000000..9f39eb3865
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCStatsReport.cpp
@@ -0,0 +1,213 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/. */
+
+#include "RTCStatsReport.h"
+#include "libwebrtcglue/SystemTime.h"
+#include "mozilla/dom/Performance.h"
+#include "nsRFPService.h"
+#include "WebrtcGlobal.h"
+
+namespace mozilla::dom {
+
+RTCStatsTimestampState::RTCStatsTimestampState()
+    : mRandomTimelineSeed(0),
+      mStartDomRealtime(WebrtcSystemTimeBase()),
+      mStartRealtime(
+          WebrtcSystemTime() -
+          webrtc::TimeDelta::Micros(
+              (TimeStamp::Now() - mStartDomRealtime).ToMicroseconds())),
+      mRTPCallerType(RTPCallerType::Normal),
+      mStartWallClockRaw(
+          PerformanceService::GetOrCreate()->TimeOrigin(mStartDomRealtime)) {}
+
+RTCStatsTimestampState::RTCStatsTimestampState(Performance& aPerformance)
+    : mRandomTimelineSeed(aPerformance.GetRandomTimelineSeed()),
+      mStartDomRealtime(aPerformance.CreationTimeStamp()),
+      mStartRealtime(
+          WebrtcSystemTime() -
+          webrtc::TimeDelta::Micros(
+              (TimeStamp::Now() - mStartDomRealtime).ToMicroseconds())),
+      mRTPCallerType(aPerformance.GetRTPCallerType()),
+      mStartWallClockRaw(
+          PerformanceService::GetOrCreate()->TimeOrigin(mStartDomRealtime)) {}
+
+TimeStamp RTCStatsTimestamp::ToMozTime() const { return mMozTime; }
+
+webrtc::Timestamp RTCStatsTimestamp::ToRealtime() const {
+  return ToDomRealtime() +
+         webrtc::TimeDelta::Micros(mState.mStartRealtime.us());
+}
+
+webrtc::Timestamp RTCStatsTimestamp::To1Jan1970() const {
+  return ToDomRealtime() + webrtc::TimeDelta::Millis(mState.mStartWallClockRaw);
+}
+
+webrtc::Timestamp RTCStatsTimestamp::ToNtp() const {
+  return To1Jan1970() + webrtc::TimeDelta::Seconds(webrtc::kNtpJan1970);
+}
+
+webrtc::Timestamp RTCStatsTimestamp::ToDomRealtime() const {
+  return webrtc::Timestamp::Micros(
+      (mMozTime - mState.mStartDomRealtime).ToMicroseconds());
+}
+
+DOMHighResTimeStamp RTCStatsTimestamp::ToDom() const {
+  // webrtc-pc says to use performance.timeOrigin + performance.now(), but
+  // keeping a Performance object around is difficult because it is
+  // main-thread-only. So, we perform the same calculation here. Note that this
+  // can be very different from the current wall-clock time because of changes
+  // to the wall clock, or monotonic clock drift over long periods of time.
+  // We are very careful to do exactly what Performance does, to avoid timestamp
+  // discrepancies.
+
+  DOMHighResTimeStamp realtime = ToDomRealtime().ms<double>();
+  // mRandomTimelineSeed is not set in the unit-tests.
+  if (mState.mRandomTimelineSeed) {
+    realtime = nsRFPService::ReduceTimePrecisionAsMSecs(
+        realtime, mState.mRandomTimelineSeed, mState.mRTPCallerType);
+  }
+
+  // Ugh. Performance::TimeOrigin is not constant, which means we need to
+  // emulate this weird behavior so our time stamps are consistent with JS
+  // timeOrigin. This is based on the code here:
+  // https://searchfox.org/mozilla-central/rev/
+  // 053826b10f838f77c27507e5efecc96e34718541/dom/performance/Performance.cpp#111-117
+  DOMHighResTimeStamp start = nsRFPService::ReduceTimePrecisionAsMSecs(
+      mState.mStartWallClockRaw, 0, mState.mRTPCallerType);
+
+  return start + realtime;
+}
+
+/* static */ RTCStatsTimestamp RTCStatsTimestamp::FromMozTime(
+    const RTCStatsTimestampMaker& aMaker, TimeStamp aMozTime) {
+  return RTCStatsTimestamp(aMaker.mState, aMozTime);
+}
+
+/* static */ RTCStatsTimestamp RTCStatsTimestamp::FromRealtime(
+    const RTCStatsTimestampMaker& aMaker, webrtc::Timestamp aRealtime) {
+  return FromDomRealtime(
+      aMaker,
+      aRealtime - webrtc::TimeDelta::Micros(aMaker.mState.mStartRealtime.us()));
+}
+
+/* static */ RTCStatsTimestamp RTCStatsTimestamp::From1Jan1970(
+    const RTCStatsTimestampMaker& aMaker, webrtc::Timestamp a1Jan1970) {
+  const auto& state = aMaker.mState;
+  return FromDomRealtime(
+      aMaker, a1Jan1970 - webrtc::TimeDelta::Millis(state.mStartWallClockRaw));
+}
+
+/* static */ RTCStatsTimestamp RTCStatsTimestamp::FromNtp(
+    const RTCStatsTimestampMaker& aMaker, webrtc::Timestamp aNtpTime) {
+  const auto& state = aMaker.mState;
+  const auto domRealtime = aNtpTime -
+                           webrtc::TimeDelta::Seconds(webrtc::kNtpJan1970) -
+                           webrtc::TimeDelta::Millis(state.mStartWallClockRaw);
+  // Ntp times exposed by libwebrtc to stats are always **rounded** to
+  // milliseconds. That means they can jump up to half a millisecond into the
+  // future. We compensate for that here so that things seem consistent to js.
+  return FromDomRealtime(aMaker, domRealtime - webrtc::TimeDelta::Micros(500));
+}
+
+/* static */ RTCStatsTimestamp RTCStatsTimestamp::FromDomRealtime(
+    const RTCStatsTimestampMaker& aMaker, webrtc::Timestamp aDomRealtime) {
+  return RTCStatsTimestamp(aMaker.mState, aMaker.mState.mStartDomRealtime +
+                                              TimeDuration::FromMicroseconds(
+                                                  aDomRealtime.us<double>()));
+}
+
+RTCStatsTimestamp::RTCStatsTimestamp(RTCStatsTimestampState aState,
+                                     TimeStamp aMozTime)
+    : mState(aState), mMozTime(aMozTime) {}
+
+RTCStatsTimestampMaker::RTCStatsTimestampMaker(RTCStatsTimestampState aState)
+    : mState(aState) {}
+
+/* static */
+RTCStatsTimestampMaker RTCStatsTimestampMaker::Create(
+    nsPIDOMWindowInner* aWindow /* = nullptr */) {
+  if (!aWindow) {
+    return RTCStatsTimestampMaker(RTCStatsTimestampState());
+  }
+  if (Performance* p = aWindow->GetPerformance()) {
+    return RTCStatsTimestampMaker(RTCStatsTimestampState(*p));
+  }
+  return RTCStatsTimestampMaker(RTCStatsTimestampState());
+}
+
+RTCStatsTimestamp RTCStatsTimestampMaker::GetNow() const {
+  return RTCStatsTimestamp::FromMozTime(*this, TimeStamp::Now());
+}
+
+NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE(RTCStatsReport, mParent)
+
+RTCStatsReport::RTCStatsReport(nsPIDOMWindowInner* aParent)
+    : mParent(aParent) {}
+
+/*static*/
+already_AddRefed<RTCStatsReport> RTCStatsReport::Constructor(
+    const GlobalObject& aGlobal) {
+  nsCOMPtr<nsPIDOMWindowInner> window(
+      do_QueryInterface(aGlobal.GetAsSupports()));
+  RefPtr<RTCStatsReport> report(new RTCStatsReport(window));
+  return report.forget();
+}
+
+JSObject* RTCStatsReport::WrapObject(JSContext* aCx,
+                                     JS::Handle<JSObject*> aGivenProto) {
+  return RTCStatsReport_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+void RTCStatsReport::Incorporate(RTCStatsCollection& aStats) {
+  ForAllPublicRTCStatsCollectionMembers(
+      aStats, [&](auto... aMember) { (SetRTCStats(aMember), ...); });
+}
+
+void RTCStatsReport::Set(const nsAString& aKey, JS::Handle<JSObject*> aValue,
+                         ErrorResult& aRv) {
+  RTCStatsReport_Binding::MaplikeHelpers::Set(this, aKey, aValue, aRv);
+}
+
+namespace {
+template <size_t I, typename... Ts>
+bool MoveInto(std::tuple<Ts...>& aFrom, std::tuple<Ts*...>& aInto) {
+  return std::get<I>(aInto)->AppendElements(std::move(std::get<I>(aFrom)),
+                                            fallible);
+}
+
+template <size_t... Is, typename... Ts>
+bool MoveInto(std::tuple<Ts...>&& aFrom, std::tuple<Ts*...>& aInto,
+              std::index_sequence<Is...>) {
+  return (... && MoveInto<Is>(aFrom, aInto));
+}
+
+template <typename... Ts>
+bool MoveInto(std::tuple<Ts...>&& aFrom, std::tuple<Ts*...>& aInto) {
+  return MoveInto(std::move(aFrom), aInto, std::index_sequence_for<Ts...>());
+}
+}  // namespace
+
+void MergeStats(UniquePtr<RTCStatsCollection> aFromStats,
+                RTCStatsCollection* aIntoStats) {
+  auto fromTuple = ForAllRTCStatsCollectionMembers(
+      *aFromStats,
+      [&](auto&... aMember) { return std::make_tuple(std::move(aMember)...); });
+  auto intoTuple = ForAllRTCStatsCollectionMembers(
+      *aIntoStats,
+      [&](auto&... aMember) { return std::make_tuple(&aMember...); });
+  if (!MoveInto(std::move(fromTuple), intoTuple)) {
+    mozalloc_handle_oom(0);
+  }
+}
+
+void FlattenStats(nsTArray<UniquePtr<RTCStatsCollection>> aFromStats,
+                  RTCStatsCollection* aIntoStats) {
+  for (auto& stats : aFromStats) {
+    MergeStats(std::move(stats), aIntoStats);
+  }
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webrtc/jsapi/RTCStatsReport.h b/dom/media/webrtc/jsapi/RTCStatsReport.h
new file mode 100644
index 0000000000..97bf3daa52
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCStatsReport.h
@@ -0,0 +1,205 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/. */
+
+#ifndef RTCStatsReport_h_
+#define RTCStatsReport_h_
+
+#include "api/units/timestamp.h"  // webrtc::Timestamp
+#include "js/RootingAPI.h"        // JS::Rooted
+#include "js/Value.h"
+#include "mozilla/dom/AutoEntryScript.h"
+#include "mozilla/MozPromise.h"
+#include "mozilla/TimeStamp.h"
+#include "mozilla/dom/PerformanceService.h"
+#include "mozilla/dom/RTCStatsReportBinding.h"  // RTCStatsCollection
+#include "mozilla/dom/ToJSValue.h"
+#include "mozilla/ErrorResult.h"
+#include "mozilla/UniquePtr.h"
+#include "nsCOMPtr.h"
+#include "nsIGlobalObject.h"
+#include "nsPIDOMWindow.h"  // nsPIDOMWindowInner
+#include "nsContentUtils.h"
+#include "nsWrapperCache.h"
+#include "prtime.h"  // PR_Now
+
+namespace mozilla {
+
+extern TimeStamp WebrtcSystemTimeBase();
+
+namespace dom {
+
+/**
+ * Keeps the state needed to convert RTCStatsTimestamps.
+ */
+struct RTCStatsTimestampState {
+  RTCStatsTimestampState();
+  explicit RTCStatsTimestampState(Performance& aPerformance);
+
+  RTCStatsTimestampState(const RTCStatsTimestampState&) = default;
+
+  // These members are sampled when a non-copy constructor is called.
+
+  // Performance's random timeline seed.
+  const uint64_t mRandomTimelineSeed;
+  // TimeStamp::Now() when the members were sampled. This is equivalent to time
+  // 0 in DomRealtime.
+  const TimeStamp mStartDomRealtime;
+  // WebrtcSystemTime() when the members were sampled. This represents the same
+  // point in time as mStartDomRealtime, but as a webrtc timestamp.
+  const webrtc::Timestamp mStartRealtime;
+  // Performance's RTPCallerType.
+  const RTPCallerType mRTPCallerType;
+  // Performance.timeOrigin for mStartDomRealtime when the members were sampled.
+  const DOMHighResTimeStamp mStartWallClockRaw;
+};
+
+/**
+ * Classes that facilitate creating timestamps for webrtc stats by mimicking
+ * dom::Performance, as well as getting and converting timestamps for libwebrtc
+ * and our integration with it.
+ *
+ * They use the same clock to avoid drift and inconsistencies, base on
+ * mozilla::TimeStamp, and convert to and from these time bases:
+ * - Moz        : Monotonic, unspecified (but constant) and inaccessible epoch,
+ *                as implemented by mozilla::TimeStamp
+ * - Realtime   : Monotonic, unspecified (but constant) epoch.
+ * - 1Jan1970   : Monotonic, unix epoch (00:00:00 UTC on 1 January 1970).
+ * - Ntp        : Monotonic, ntp epoch (00:00:00 UTC on 1 January 1900).
+ * - Dom        : Monotonic, milliseconds since unix epoch, as the timestamps
+ *                defined by webrtc-pc. Corresponds to Performance.timeOrigin +
+ *                Performance.now(). Has reduced precision.
+ * - DomRealtime: Like Dom, but with full precision.
+ * - WallClock  : Non-monotonic, unix epoch. Not used here since it is
+ *                non-monotonic and cannot be correlated to the other time
+ *                bases.
+ */
+class RTCStatsTimestampMaker;
+class RTCStatsTimestamp {
+ public:
+  TimeStamp ToMozTime() const;
+  webrtc::Timestamp ToRealtime() const;
+  webrtc::Timestamp To1Jan1970() const;
+  webrtc::Timestamp ToNtp() const;
+  webrtc::Timestamp ToDomRealtime() const;
+  DOMHighResTimeStamp ToDom() const;
+
+  static RTCStatsTimestamp FromMozTime(const RTCStatsTimestampMaker& aMaker,
+                                       TimeStamp aMozTime);
+  static RTCStatsTimestamp FromRealtime(const RTCStatsTimestampMaker& aMaker,
+                                        webrtc::Timestamp aRealtime);
+  static RTCStatsTimestamp From1Jan1970(const RTCStatsTimestampMaker& aMaker,
+                                        webrtc::Timestamp aRealtime);
+  static RTCStatsTimestamp FromNtp(const RTCStatsTimestampMaker& aMaker,
+                                   webrtc::Timestamp aRealtime);
+  static RTCStatsTimestamp FromDomRealtime(const RTCStatsTimestampMaker& aMaker,
+                                           webrtc::Timestamp aDomRealtime);
+  // There is on purpose no conversion functions from DOMHighResTimeStamp
+  // because of the loss in precision of a floating point to integer conversion.
+
+ private:
+  RTCStatsTimestamp(RTCStatsTimestampState aState, TimeStamp aMozTime);
+
+  const RTCStatsTimestampState mState;
+  const TimeStamp mMozTime;
+};
+
+class RTCStatsTimestampMaker {
+ public:
+  static RTCStatsTimestampMaker Create(nsPIDOMWindowInner* aWindow = nullptr);
+
+  RTCStatsTimestamp GetNow() const;
+
+  const RTCStatsTimestampState mState;
+
+ private:
+  explicit RTCStatsTimestampMaker(RTCStatsTimestampState aState);
+};
+
+// TODO(bug 1588303): If we ever get move semantics for webidl dictionaries, we
+// can stop wrapping these in UniquePtr, which will allow us to simplify code
+// in several places.
+typedef MozPromise<UniquePtr<RTCStatsCollection>, nsresult, true>
+    RTCStatsPromise;
+
+typedef MozPromise<UniquePtr<RTCStatsReportInternal>, nsresult, true>
+    RTCStatsReportPromise;
+
+class RTCStatsReport final : public nsWrapperCache {
+ public:
+  NS_INLINE_DECL_CYCLE_COLLECTING_NATIVE_REFCOUNTING(RTCStatsReport)
+  NS_DECL_CYCLE_COLLECTION_NATIVE_WRAPPERCACHE_CLASS(RTCStatsReport)
+
+  explicit RTCStatsReport(nsPIDOMWindowInner* aParent);
+
+  // TODO(bug 1586109): Remove this once we no longer have to create empty
+  // RTCStatsReports from JS.
+  static already_AddRefed<RTCStatsReport> Constructor(
+      const GlobalObject& aGlobal);
+
+  void Incorporate(RTCStatsCollection& aStats);
+
+  nsPIDOMWindowInner* GetParentObject() const { return mParent; }
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+ private:
+  ~RTCStatsReport() = default;
+  void Set(const nsAString& aKey, JS::Handle<JSObject*> aValue,
+           ErrorResult& aRv);
+
+  template <typename T>
+  nsresult SetRTCStats(Sequence<T>& aValues) {
+    for (T& value : aValues) {
+      nsresult rv = SetRTCStats(value);
+      if (NS_FAILED(rv)) {
+        return rv;
+      }
+    }
+    return NS_OK;
+  }
+
+  // We cannot just declare this as SetRTCStats(RTCStats&), because the
+  // conversion function that ToJSValue uses is non-virtual.
+  template <typename T>
+  nsresult SetRTCStats(T& aValue) {
+    static_assert(std::is_base_of<RTCStats, T>::value,
+                  "SetRTCStats is for setting RTCStats only");
+
+    if (!aValue.mId.WasPassed()) {
+      return NS_OK;
+    }
+
+    const nsString key(aValue.mId.Value());
+
+    // Cargo-culted from dom::Promise; converts aValue to a JSObject
+    AutoEntryScript aes(mParent->AsGlobal()->GetGlobalJSObject(),
+                        "RTCStatsReport::SetRTCStats");
+    JSContext* cx = aes.cx();
+    JS::Rooted<JS::Value> val(cx);
+    if (!ToJSValue(cx, std::forward<T>(aValue), &val)) {
+      return NS_ERROR_FAILURE;
+    }
+    JS::Rooted<JSObject*> jsObject(cx, &val.toObject());
+
+    ErrorResult rv;
+    Set(key, jsObject, rv);
+    return rv.StealNSResult();
+  }
+
+  nsCOMPtr<nsPIDOMWindowInner> mParent;
+};
+
+void MergeStats(UniquePtr<dom::RTCStatsCollection> aFromStats,
+                dom::RTCStatsCollection* aIntoStats);
+
+void FlattenStats(nsTArray<UniquePtr<dom::RTCStatsCollection>> aFromStats,
+                  dom::RTCStatsCollection* aIntoStats);
+
+}  // namespace dom
+}  // namespace mozilla
+
+#endif  // RTCStatsReport_h_
diff --git a/dom/media/webrtc/jsapi/RemoteTrackSource.cpp b/dom/media/webrtc/jsapi/RemoteTrackSource.cpp
new file mode 100644
index 0000000000..41c679e431
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RemoteTrackSource.cpp
@@ -0,0 +1,73 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "RemoteTrackSource.h"
+
+#include "MediaStreamError.h"
+#include "MediaTrackGraph.h"
+#include "RTCRtpReceiver.h"
+
+namespace mozilla {
+
+NS_IMPL_ADDREF_INHERITED(RemoteTrackSource, dom::MediaStreamTrackSource)
+NS_IMPL_RELEASE_INHERITED(RemoteTrackSource, dom::MediaStreamTrackSource)
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(RemoteTrackSource)
+NS_INTERFACE_MAP_END_INHERITING(dom::MediaStreamTrackSource)
+NS_IMPL_CYCLE_COLLECTION_CLASS(RemoteTrackSource)
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN_INHERITED(RemoteTrackSource,
+                                                dom::MediaStreamTrackSource)
+  tmp->Destroy();
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mReceiver)
+NS_IMPL_CYCLE_COLLECTION_UNLINK_END
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN_INHERITED(RemoteTrackSource,
+                                                  dom::MediaStreamTrackSource)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mReceiver)
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
+
+RemoteTrackSource::RemoteTrackSource(SourceMediaTrack* aStream,
+                                     dom::RTCRtpReceiver* aReceiver,
+                                     nsIPrincipal* aPrincipal,
+                                     const nsString& aLabel,
+                                     TrackingId aTrackingId)
+    : dom::MediaStreamTrackSource(aPrincipal, aLabel, std::move(aTrackingId)),
+      mStream(aStream),
+      mReceiver(aReceiver) {}
+
+RemoteTrackSource::~RemoteTrackSource() { Destroy(); }
+
+void RemoteTrackSource::Destroy() {
+  if (mStream) {
+    MOZ_ASSERT(!mStream->IsDestroyed());
+    mStream->End();
+    mStream->Destroy();
+    mStream = nullptr;
+
+    GetMainThreadSerialEventTarget()->Dispatch(NewRunnableMethod(
+        "RemoteTrackSource::ForceEnded", this, &RemoteTrackSource::ForceEnded));
+  }
+}
+
+auto RemoteTrackSource::ApplyConstraints(
+    const dom::MediaTrackConstraints& aConstraints, dom::CallerType aCallerType)
+    -> RefPtr<ApplyConstraintsPromise> {
+  return ApplyConstraintsPromise::CreateAndReject(
+      MakeRefPtr<MediaMgrError>(
+          dom::MediaStreamError::Name::OverconstrainedError, ""),
+      __func__);
+}
+
+void RemoteTrackSource::SetPrincipal(nsIPrincipal* aPrincipal) {
+  mPrincipal = aPrincipal;
+  PrincipalChanged();
+}
+
+void RemoteTrackSource::SetMuted(bool aMuted) { MutedChanged(aMuted); }
+
+void RemoteTrackSource::ForceEnded() { OverrideEnded(); }
+
+SourceMediaTrack* RemoteTrackSource::Stream() const { return mStream; }
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/jsapi/RemoteTrackSource.h b/dom/media/webrtc/jsapi/RemoteTrackSource.h
new file mode 100644
index 0000000000..be730f0030
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RemoteTrackSource.h
@@ -0,0 +1,64 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef DOM_MEDIA_WEBRTC_JSAPI_REMOTETRACKSOURCE_H_
+#define DOM_MEDIA_WEBRTC_JSAPI_REMOTETRACKSOURCE_H_
+
+#include "MediaStreamTrack.h"
+
+namespace mozilla {
+
+namespace dom {
+class RTCRtpReceiver;
+}
+
+class SourceMediaTrack;
+
+class RemoteTrackSource : public dom::MediaStreamTrackSource {
+ public:
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(RemoteTrackSource,
+                                           dom::MediaStreamTrackSource)
+
+  RemoteTrackSource(SourceMediaTrack* aStream, dom::RTCRtpReceiver* aReceiver,
+                    nsIPrincipal* aPrincipal, const nsString& aLabel,
+                    TrackingId aTrackingId);
+
+  void Destroy() override;
+
+  dom::MediaSourceEnum GetMediaSource() const override {
+    return dom::MediaSourceEnum::Other;
+  }
+
+  RefPtr<ApplyConstraintsPromise> ApplyConstraints(
+      const dom::MediaTrackConstraints& aConstraints,
+      dom::CallerType aCallerType) override;
+
+  void Stop() override {
+    // XXX (Bug 1314270): Implement rejection logic if necessary when we have
+    //                    clarity in the spec.
+  }
+
+  void Disable() override {}
+
+  void Enable() override {}
+
+  void SetPrincipal(nsIPrincipal* aPrincipal);
+  void SetMuted(bool aMuted);
+  void ForceEnded();
+
+  SourceMediaTrack* Stream() const;
+
+ private:
+  virtual ~RemoteTrackSource();
+
+  RefPtr<SourceMediaTrack> mStream;
+  RefPtr<dom::RTCRtpReceiver> mReceiver;
+};
+
+}  // namespace mozilla
+
+#endif  // DOM_MEDIA_WEBRTC_JSAPI_REMOTETRACKSOURCE_H_
diff --git a/dom/media/webrtc/jsapi/WebrtcGlobalChild.h b/dom/media/webrtc/jsapi/WebrtcGlobalChild.h
new file mode 100644
index 0000000000..147b4b44f0
--- /dev/null
+++ b/dom/media/webrtc/jsapi/WebrtcGlobalChild.h
@@ -0,0 +1,42 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _WEBRTC_GLOBAL_CHILD_H_
+#define _WEBRTC_GLOBAL_CHILD_H_
+
+#include "mozilla/dom/PWebrtcGlobalChild.h"
+
+namespace mozilla::dom {
+
+class WebrtcGlobalChild : public PWebrtcGlobalChild {
+  friend class ContentChild;
+
+  bool mShutdown;
+
+  MOZ_IMPLICIT WebrtcGlobalChild();
+  virtual void ActorDestroy(ActorDestroyReason aWhy) override;
+
+  virtual mozilla::ipc::IPCResult RecvGetStats(
+      const nsAString& aPcIdFilter, GetStatsResolver&& aResolve) override;
+  virtual mozilla::ipc::IPCResult RecvClearStats() override;
+  // MOZ_CAN_RUN_SCRIPT_BOUNDARY because we can't do MOZ_CAN_RUN_SCRIPT in
+  // ipdl-generated things yet.
+  MOZ_CAN_RUN_SCRIPT_BOUNDARY
+  virtual mozilla::ipc::IPCResult RecvGetLog(
+      GetLogResolver&& aResolve) override;
+  virtual mozilla::ipc::IPCResult RecvClearLog() override;
+  virtual mozilla::ipc::IPCResult RecvSetAecLogging(
+      const bool& aEnable) override;
+  virtual mozilla::ipc::IPCResult RecvSetDebugMode(const int& aLevel) override;
+
+  static WebrtcGlobalChild* GetOrSet(const Maybe<WebrtcGlobalChild*>& aChild);
+
+ public:
+  virtual ~WebrtcGlobalChild();
+  static WebrtcGlobalChild* Get();
+};
+
+}  // namespace mozilla::dom
+
+#endif  // _WEBRTC_GLOBAL_CHILD_H_
diff --git a/dom/media/webrtc/jsapi/WebrtcGlobalInformation.cpp b/dom/media/webrtc/jsapi/WebrtcGlobalInformation.cpp
new file mode 100644
index 0000000000..7d0a9e64b1
--- /dev/null
+++ b/dom/media/webrtc/jsapi/WebrtcGlobalInformation.cpp
@@ -0,0 +1,829 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "WebrtcGlobalInformation.h"
+#include "WebrtcGlobalStatsHistory.h"
+#include "mozilla/Assertions.h"
+#include "mozilla/StaticPtr.h"
+#include "mozilla/dom/PWebrtcGlobal.h"
+#include "mozilla/dom/PWebrtcGlobalChild.h"
+#include "mozilla/media/webrtc/WebrtcGlobal.h"
+#include "WebrtcGlobalChild.h"
+#include "WebrtcGlobalParent.h"
+
+#include <algorithm>
+#include <vector>
+#include <type_traits>
+
+#include "mozilla/dom/WebrtcGlobalInformationBinding.h"
+#include "mozilla/dom/RTCStatsReportBinding.h"  // for RTCStatsReportInternal
+#include "mozilla/dom/ContentChild.h"
+
+#include "nsISupports.h"
+#include "nsITimer.h"
+#include "nsLiteralString.h"
+#include "nsNetCID.h"               // NS_SOCKETTRANSPORTSERVICE_CONTRACTID
+#include "nsServiceManagerUtils.h"  // do_GetService
+#include "nsXULAppAPI.h"
+#include "mozilla/ErrorResult.h"
+#include "nsProxyRelease.h"  // nsMainThreadPtrHolder
+#include "mozilla/Telemetry.h"
+#include "mozilla/Unused.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/ClearOnShutdown.h"
+
+#include "common/browser_logging/WebRtcLog.h"
+#include "nsString.h"
+#include "transport/runnable_utils.h"
+#include "MediaTransportHandler.h"
+#include "PeerConnectionCtx.h"
+#include "PeerConnectionImpl.h"
+
+#ifdef XP_WIN
+#  include <process.h>
+#endif
+
+namespace mozilla::dom {
+
+using StatsRequestCallback =
+    nsMainThreadPtrHandle<WebrtcGlobalStatisticsCallback>;
+
+using LogRequestCallback = nsMainThreadPtrHandle<WebrtcGlobalLoggingCallback>;
+
+class WebrtcContentParents {
+ public:
+  static WebrtcGlobalParent* Alloc();
+  static void Dealloc(WebrtcGlobalParent* aParent);
+  static bool Empty() { return sContentParents.empty(); }
+  static const std::vector<RefPtr<WebrtcGlobalParent>>& GetAll() {
+    return sContentParents;
+  }
+
+  WebrtcContentParents() = delete;
+  WebrtcContentParents(const WebrtcContentParents&) = delete;
+  WebrtcContentParents& operator=(const WebrtcContentParents&) = delete;
+
+ private:
+  static std::vector<RefPtr<WebrtcGlobalParent>> sContentParents;
+};
+
+std::vector<RefPtr<WebrtcGlobalParent>> WebrtcContentParents::sContentParents;
+
+WebrtcGlobalParent* WebrtcContentParents::Alloc() {
+  RefPtr<WebrtcGlobalParent> cp = new WebrtcGlobalParent;
+  sContentParents.push_back(cp);
+  return cp.get();
+}
+
+void WebrtcContentParents::Dealloc(WebrtcGlobalParent* aParent) {
+  if (aParent) {
+    aParent->mShutdown = true;
+    auto cp =
+        std::find(sContentParents.begin(), sContentParents.end(), aParent);
+    if (cp != sContentParents.end()) {
+      sContentParents.erase(cp);
+    }
+  }
+}
+
+static PeerConnectionCtx* GetPeerConnectionCtx() {
+  if (PeerConnectionCtx::isActive()) {
+    MOZ_ASSERT(PeerConnectionCtx::GetInstance());
+    return PeerConnectionCtx::GetInstance();
+  }
+  return nullptr;
+}
+
+static nsTArray<dom::RTCStatsReportInternal>& GetWebrtcGlobalStatsStash() {
+  static StaticAutoPtr<nsTArray<dom::RTCStatsReportInternal>> sStash;
+  if (!sStash) {
+    sStash = new nsTArray<dom::RTCStatsReportInternal>();
+    ClearOnShutdown(&sStash);
+  }
+  return *sStash;
+}
+
+static RefPtr<PWebrtcGlobalParent::GetStatsPromise>
+GetStatsPromiseForThisProcess(const nsAString& aPcIdFilter) {
+  nsTArray<RefPtr<dom::RTCStatsReportPromise>> promises;
+
+  std::set<nsString> pcids;
+  if (auto* ctx = GetPeerConnectionCtx()) {
+    // Grab stats for PCs that still exist
+    ctx->ForEachPeerConnection([&](PeerConnectionImpl* aPc) {
+      if (!aPcIdFilter.IsEmpty() &&
+          !aPcIdFilter.EqualsASCII(aPc->GetIdAsAscii().c_str())) {
+        return;
+      }
+      if (!aPc->IsClosed() || !aPc->LongTermStatsIsDisabled()) {
+        nsString id;
+        aPc->GetId(id);
+        pcids.insert(id);
+        promises.AppendElement(aPc->GetStats(nullptr, true));
+      }
+    });
+
+    // Grab previously stashed stats, if they aren't dupes, and ensure they
+    // are marked closed. (In a content process, this should already have
+    // happened, but in the parent process, the stash will contain the last
+    // observed stats from the content processes. From the perspective of the
+    // parent process, these are assumed closed unless we see new stats from the
+    // content process that say otherwise.)
+    for (auto& report : GetWebrtcGlobalStatsStash()) {
+      report.mClosed = true;
+      if ((aPcIdFilter.IsEmpty() || aPcIdFilter == report.mPcid) &&
+          !pcids.count(report.mPcid)) {
+        promises.AppendElement(dom::RTCStatsReportPromise::CreateAndResolve(
+            MakeUnique<dom::RTCStatsReportInternal>(report), __func__));
+      }
+    }
+  }
+
+  auto UnwrapUniquePtrs = [](dom::RTCStatsReportPromise::AllSettledPromiseType::
+                                 ResolveOrRejectValue&& aResult) {
+    nsTArray<dom::RTCStatsReportInternal> reports;
+    MOZ_RELEASE_ASSERT(aResult.IsResolve(), "AllSettled should never reject!");
+    for (auto& reportResult : aResult.ResolveValue()) {
+      if (reportResult.IsResolve()) {
+        reports.AppendElement(*reportResult.ResolveValue());
+      }
+    }
+    return PWebrtcGlobalParent::GetStatsPromise::CreateAndResolve(
+        std::move(reports), __func__);
+  };
+
+  return dom::RTCStatsReportPromise::AllSettled(
+             GetMainThreadSerialEventTarget(), promises)
+      ->Then(GetMainThreadSerialEventTarget(), __func__,
+             std::move(UnwrapUniquePtrs));
+}
+
+static std::map<int32_t, dom::Sequence<nsString>>& GetWebrtcGlobalLogStash() {
+  static StaticAutoPtr<std::map<int32_t, dom::Sequence<nsString>>> sStash;
+  if (!sStash) {
+    sStash = new std::map<int32_t, dom::Sequence<nsString>>();
+    ClearOnShutdown(&sStash);
+  }
+  return *sStash;
+}
+
+static void ClearLongTermStats() {
+  if (!NS_IsMainThread()) {
+    MOZ_ASSERT(NS_IsMainThread());
+    return;
+  }
+
+  GetWebrtcGlobalStatsStash().Clear();
+  if (XRE_IsParentProcess()) {
+    WebrtcGlobalStatsHistory::Clear();
+  }
+  if (auto* ctx = GetPeerConnectionCtx()) {
+    ctx->ClearClosedStats();
+  }
+}
+
+void WebrtcGlobalInformation::ClearAllStats(const GlobalObject& aGlobal) {
+  if (!NS_IsMainThread()) {
+    return;
+  }
+
+  // Chrome-only API
+  MOZ_ASSERT(XRE_IsParentProcess());
+
+  if (!WebrtcContentParents::Empty()) {
+    // Pass on the request to any content process based PeerConnections.
+    for (const auto& cp : WebrtcContentParents::GetAll()) {
+      Unused << cp->SendClearStats();
+    }
+  }
+
+  // Flush the history for the chrome process
+  ClearLongTermStats();
+}
+
+void WebrtcGlobalInformation::GetStatsHistoryPcIds(
+    const GlobalObject& aGlobal,
+    WebrtcGlobalStatisticsHistoryPcIdsCallback& aPcIdsCallback,
+    ErrorResult& aRv) {
+  if (!NS_IsMainThread()) {
+    aRv.Throw(NS_ERROR_NOT_SAME_THREAD);
+    return;
+  }
+
+  MOZ_ASSERT(XRE_IsParentProcess());
+
+  IgnoredErrorResult rv;
+  aPcIdsCallback.Call(WebrtcGlobalStatsHistory::PcIds(), rv);
+  aRv = NS_OK;
+}
+
+void WebrtcGlobalInformation::GetStatsHistorySince(
+    const GlobalObject& aGlobal,
+    WebrtcGlobalStatisticsHistoryCallback& aStatsCallback,
+    const nsAString& pcIdFilter, const Optional<DOMHighResTimeStamp>& aAfter,
+    const Optional<DOMHighResTimeStamp>& aSdpAfter, ErrorResult& aRv) {
+  if (!NS_IsMainThread()) {
+    aRv.Throw(NS_ERROR_NOT_SAME_THREAD);
+    return;
+  }
+
+  MOZ_ASSERT(XRE_IsParentProcess());
+
+  WebrtcGlobalStatisticsReport history;
+
+  auto statsAfter = aAfter.WasPassed() ? Some(aAfter.Value()) : Nothing();
+  auto sdpAfter = aSdpAfter.WasPassed() ? Some(aSdpAfter.Value()) : Nothing();
+
+  WebrtcGlobalStatsHistory::GetHistory(pcIdFilter).apply([&](auto& hist) {
+    if (!history.mReports.AppendElements(hist->Since(statsAfter), fallible)) {
+      mozalloc_handle_oom(0);
+    }
+    if (!history.mSdpHistories.AppendElement(hist->SdpSince(sdpAfter),
+                                             fallible)) {
+      mozalloc_handle_oom(0);
+    }
+  });
+
+  IgnoredErrorResult rv;
+  aStatsCallback.Call(history, rv);
+  aRv = NS_OK;
+}
+
+using StatsPromiseArray =
+    nsTArray<RefPtr<PWebrtcGlobalParent::GetStatsPromise>>;
+
+void WebrtcGlobalInformation::GatherHistory() {
+  const nsString emptyFilter;
+  if (!NS_IsMainThread()) {
+    MOZ_ASSERT(NS_IsMainThread());
+    return;
+  }
+
+  MOZ_ASSERT(XRE_IsParentProcess());
+  using StatsPromise = PWebrtcGlobalParent::GetStatsPromise;
+  auto resolveThenAppendStatsHistory = [](RefPtr<StatsPromise>&& promise) {
+    auto AppendStatsHistory = [](StatsPromise::ResolveOrRejectValue&& result) {
+      if (result.IsReject()) {
+        return;
+      }
+      for (const auto& report : result.ResolveValue()) {
+        WebrtcGlobalStatsHistory::Record(
+            MakeUnique<RTCStatsReportInternal>(report));
+      }
+    };
+    promise->Then(GetMainThreadSerialEventTarget(), __func__,
+                  std::move(AppendStatsHistory));
+  };
+  for (const auto& cp : WebrtcContentParents::GetAll()) {
+    resolveThenAppendStatsHistory(cp->SendGetStats(emptyFilter));
+  }
+  resolveThenAppendStatsHistory(GetStatsPromiseForThisProcess(emptyFilter));
+}
+
+void WebrtcGlobalInformation::GetAllStats(
+    const GlobalObject& aGlobal, WebrtcGlobalStatisticsCallback& aStatsCallback,
+    const Optional<nsAString>& aPcIdFilter, ErrorResult& aRv) {
+  if (!NS_IsMainThread()) {
+    aRv.Throw(NS_ERROR_NOT_SAME_THREAD);
+    return;
+  }
+
+  MOZ_ASSERT(XRE_IsParentProcess());
+
+  StatsPromiseArray statsPromises;
+
+  nsString filter;
+  if (aPcIdFilter.WasPassed()) {
+    filter = aPcIdFilter.Value();
+  }
+
+  for (const auto& cp : WebrtcContentParents::GetAll()) {
+    statsPromises.AppendElement(cp->SendGetStats(filter));
+  }
+
+  // Stats from this (the parent) process. How long do we keep supporting this?
+  statsPromises.AppendElement(GetStatsPromiseForThisProcess(filter));
+
+  // CallbackObject does not support threadsafe refcounting, and must be
+  // used and destroyed on main.
+  StatsRequestCallback callbackHandle(
+      new nsMainThreadPtrHolder<WebrtcGlobalStatisticsCallback>(
+          "WebrtcGlobalStatisticsCallback", &aStatsCallback));
+
+  auto FlattenThenStashThenCallback =
+      [callbackHandle,
+       filter](PWebrtcGlobalParent::GetStatsPromise::AllSettledPromiseType::
+                   ResolveOrRejectValue&& aResult) MOZ_CAN_RUN_SCRIPT_BOUNDARY {
+        std::set<nsString> pcids;
+        WebrtcGlobalStatisticsReport flattened;
+        MOZ_RELEASE_ASSERT(aResult.IsResolve(),
+                           "AllSettled should never reject!");
+        // Flatten stats from content processes and parent process.
+        // The stats from the parent process (which will come last) might
+        // contain some stale content-process stats, so skip those.
+        for (auto& processResult : aResult.ResolveValue()) {
+          // TODO: Report rejection on individual content processes someday?
+          if (processResult.IsResolve()) {
+            for (auto& pcStats : processResult.ResolveValue()) {
+              if (!pcids.count(pcStats.mPcid)) {
+                pcids.insert(pcStats.mPcid);
+                if (!flattened.mReports.AppendElement(std::move(pcStats),
+                                                      fallible)) {
+                  mozalloc_handle_oom(0);
+                }
+              }
+            }
+          }
+        }
+
+        if (filter.IsEmpty()) {
+          // Unfiltered is simple; the flattened result becomes the new stash.
+          GetWebrtcGlobalStatsStash() = flattened.mReports;
+        } else if (!flattened.mReports.IsEmpty()) {
+          // Update our stash with the single result.
+          MOZ_ASSERT(flattened.mReports.Length() == 1);
+          StashStats(flattened.mReports[0]);
+        }
+
+        IgnoredErrorResult rv;
+        callbackHandle->Call(flattened, rv);
+      };
+
+  PWebrtcGlobalParent::GetStatsPromise::AllSettled(
+      GetMainThreadSerialEventTarget(), statsPromises)
+      ->Then(GetMainThreadSerialEventTarget(), __func__,
+             std::move(FlattenThenStashThenCallback));
+
+  aRv = NS_OK;
+}
+
+static RefPtr<PWebrtcGlobalParent::GetLogPromise> GetLogPromise() {
+  PeerConnectionCtx* ctx = GetPeerConnectionCtx();
+  if (!ctx) {
+    // This process has never created a PeerConnection, so no ICE logging.
+    return PWebrtcGlobalParent::GetLogPromise::CreateAndResolve(
+        Sequence<nsString>(), __func__);
+  }
+
+  nsresult rv;
+  nsCOMPtr<nsISerialEventTarget> stsThread =
+      do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &rv);
+
+  if (NS_WARN_IF(NS_FAILED(rv) || !stsThread)) {
+    return PWebrtcGlobalParent::GetLogPromise::CreateAndResolve(
+        Sequence<nsString>(), __func__);
+  }
+
+  RefPtr<MediaTransportHandler> transportHandler = ctx->GetTransportHandler();
+
+  auto AddMarkers =
+      [](MediaTransportHandler::IceLogPromise::ResolveOrRejectValue&& aValue) {
+        nsString pid;
+        pid.AppendInt(getpid());
+        Sequence<nsString> logs;
+        if (aValue.IsResolve() && !aValue.ResolveValue().IsEmpty()) {
+          bool ok = logs.AppendElement(
+              u"+++++++ BEGIN (process id "_ns + pid + u") ++++++++"_ns,
+              fallible);
+          ok &=
+              !!logs.AppendElements(std::move(aValue.ResolveValue()), fallible);
+          ok &= !!logs.AppendElement(
+              u"+++++++ END (process id "_ns + pid + u") ++++++++"_ns,
+              fallible);
+          if (!ok) {
+            mozalloc_handle_oom(0);
+          }
+        }
+        return PWebrtcGlobalParent::GetLogPromise::CreateAndResolve(
+            std::move(logs), __func__);
+      };
+
+  return transportHandler->GetIceLog(nsCString())
+      ->Then(GetMainThreadSerialEventTarget(), __func__, std::move(AddMarkers));
+}
+
+static nsresult RunLogClear() {
+  PeerConnectionCtx* ctx = GetPeerConnectionCtx();
+  if (!ctx) {
+    // This process has never created a PeerConnection, so no ICE logging.
+    return NS_OK;
+  }
+
+  nsresult rv;
+  nsCOMPtr<nsISerialEventTarget> stsThread =
+      do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &rv);
+
+  if (NS_FAILED(rv)) {
+    return rv;
+  }
+  if (!stsThread) {
+    return NS_ERROR_FAILURE;
+  }
+
+  RefPtr<MediaTransportHandler> transportHandler = ctx->GetTransportHandler();
+
+  return RUN_ON_THREAD(
+      stsThread,
+      WrapRunnable(transportHandler, &MediaTransportHandler::ClearIceLog),
+      NS_DISPATCH_NORMAL);
+}
+
+void WebrtcGlobalInformation::ClearLogging(const GlobalObject& aGlobal) {
+  if (!NS_IsMainThread()) {
+    return;
+  }
+
+  // Chrome-only API
+  MOZ_ASSERT(XRE_IsParentProcess());
+  GetWebrtcGlobalLogStash().clear();
+
+  if (!WebrtcContentParents::Empty()) {
+    // Clear content process signaling logs
+    for (const auto& cp : WebrtcContentParents::GetAll()) {
+      Unused << cp->SendClearLog();
+    }
+  }
+
+  // Clear chrome process signaling logs
+  Unused << RunLogClear();
+}
+
+static RefPtr<GenericPromise> UpdateLogStash() {
+  nsTArray<RefPtr<GenericPromise>> logPromises;
+  MOZ_ASSERT(XRE_IsParentProcess());
+  for (const auto& cp : WebrtcContentParents::GetAll()) {
+    auto StashLog =
+        [id = cp->Id() * 2 /* Make sure 1 isn't used */](
+            PWebrtcGlobalParent::GetLogPromise::ResolveOrRejectValue&& aValue) {
+          if (aValue.IsResolve() && !aValue.ResolveValue().IsEmpty()) {
+            GetWebrtcGlobalLogStash()[id] = aValue.ResolveValue();
+          }
+          return GenericPromise::CreateAndResolve(true, __func__);
+        };
+    logPromises.AppendElement(cp->SendGetLog()->Then(
+        GetMainThreadSerialEventTarget(), __func__, std::move(StashLog)));
+  }
+
+  // Get ICE logging for this (the parent) process. How long do we support this?
+  logPromises.AppendElement(GetLogPromise()->Then(
+      GetMainThreadSerialEventTarget(), __func__,
+      [](PWebrtcGlobalParent::GetLogPromise::ResolveOrRejectValue&& aValue) {
+        if (aValue.IsResolve()) {
+          GetWebrtcGlobalLogStash()[1] = aValue.ResolveValue();
+        }
+        return GenericPromise::CreateAndResolve(true, __func__);
+      }));
+
+  return GenericPromise::AllSettled(GetMainThreadSerialEventTarget(),
+                                    logPromises)
+      ->Then(GetMainThreadSerialEventTarget(), __func__,
+             [](GenericPromise::AllSettledPromiseType::ResolveOrRejectValue&&
+                    aValue) {
+               // We don't care about the value, since we're just going to copy
+               // what is in the stash. This ignores failures too, which is what
+               // we want.
+               return GenericPromise::CreateAndResolve(true, __func__);
+             });
+}
+
+void WebrtcGlobalInformation::GetLogging(
+    const GlobalObject& aGlobal, const nsAString& aPattern,
+    WebrtcGlobalLoggingCallback& aLoggingCallback, ErrorResult& aRv) {
+  if (!NS_IsMainThread()) {
+    aRv.Throw(NS_ERROR_NOT_SAME_THREAD);
+    return;
+  }
+
+  MOZ_ASSERT(XRE_IsParentProcess());
+
+  nsAutoString pattern(aPattern);
+
+  // CallbackObject does not support threadsafe refcounting, and must be
+  // destroyed on main.
+  LogRequestCallback callbackHandle(
+      new nsMainThreadPtrHolder<WebrtcGlobalLoggingCallback>(
+          "WebrtcGlobalLoggingCallback", &aLoggingCallback));
+
+  auto FilterThenCallback =
+      [pattern, callbackHandle](GenericPromise::ResolveOrRejectValue&& aValue)
+          MOZ_CAN_RUN_SCRIPT_BOUNDARY {
+            dom::Sequence<nsString> flattened;
+            for (const auto& [id, log] : GetWebrtcGlobalLogStash()) {
+              (void)id;
+              for (const auto& line : log) {
+                if (pattern.IsEmpty() || (line.Find(pattern) != kNotFound)) {
+                  if (!flattened.AppendElement(line, fallible)) {
+                    mozalloc_handle_oom(0);
+                  }
+                }
+              }
+            }
+            IgnoredErrorResult rv;
+            callbackHandle->Call(flattened, rv);
+          };
+
+  UpdateLogStash()->Then(GetMainThreadSerialEventTarget(), __func__,
+                         std::move(FilterThenCallback));
+  aRv = NS_OK;
+}
+
+static int32_t sLastSetLevel = 0;
+static bool sLastAECDebug = false;
+static Maybe<nsCString> sAecDebugLogDir;
+
+void WebrtcGlobalInformation::SetDebugLevel(const GlobalObject& aGlobal,
+                                            int32_t aLevel) {
+  if (aLevel) {
+    StartWebRtcLog(mozilla::LogLevel(aLevel));
+  } else {
+    StopWebRtcLog();
+  }
+  sLastSetLevel = aLevel;
+
+  for (const auto& cp : WebrtcContentParents::GetAll()) {
+    Unused << cp->SendSetDebugMode(aLevel);
+  }
+}
+
+int32_t WebrtcGlobalInformation::DebugLevel(const GlobalObject& aGlobal) {
+  return sLastSetLevel;
+}
+
+void WebrtcGlobalInformation::SetAecDebug(const GlobalObject& aGlobal,
+                                          bool aEnable) {
+  if (aEnable) {
+    sAecDebugLogDir = Some(StartAecLog());
+  } else {
+    StopAecLog();
+  }
+
+  sLastAECDebug = aEnable;
+
+  for (const auto& cp : WebrtcContentParents::GetAll()) {
+    Unused << cp->SendSetAecLogging(aEnable);
+  }
+}
+
+bool WebrtcGlobalInformation::AecDebug(const GlobalObject& aGlobal) {
+  return sLastAECDebug;
+}
+
+void WebrtcGlobalInformation::GetAecDebugLogDir(const GlobalObject& aGlobal,
+                                                nsAString& aDir) {
+  aDir = NS_ConvertASCIItoUTF16(sAecDebugLogDir.valueOr(""_ns));
+}
+
+/*static*/
+void WebrtcGlobalInformation::StashStats(
+    const dom::RTCStatsReportInternal& aReport) {
+  // Remove previous report, if present
+  // TODO: Make this a map instead of an array?
+  for (size_t i = 0; i < GetWebrtcGlobalStatsStash().Length();) {
+    auto& pcStats = GetWebrtcGlobalStatsStash()[i];
+    if (pcStats.mPcid == aReport.mPcid) {
+      GetWebrtcGlobalStatsStash().RemoveElementAt(i);
+      break;
+    }
+    ++i;
+  }
+  // Stash final stats
+  GetWebrtcGlobalStatsStash().AppendElement(aReport);
+}
+
+void WebrtcGlobalInformation::AdjustTimerReferences(
+    PcTrackingUpdate&& aUpdate) {
+  static StaticRefPtr<nsITimer> sHistoryTimer;
+  static StaticAutoPtr<nsTHashSet<nsString>> sPcids;
+
+  MOZ_ASSERT(NS_IsMainThread());
+
+  auto HandleAdd = [&](nsString&& aPcid, bool aIsLongTermStatsDisabled) {
+    if (!sPcids) {
+      sPcids = new nsTHashSet<nsString>();
+      ClearOnShutdown(&sPcids);
+    }
+    sPcids->EnsureInserted(aPcid);
+    // Reserve a stats history
+    WebrtcGlobalStatsHistory::InitHistory(nsString(aPcid),
+                                          aIsLongTermStatsDisabled);
+    if (!sHistoryTimer) {
+      sHistoryTimer = NS_NewTimer(GetMainThreadSerialEventTarget());
+      if (sHistoryTimer) {
+        sHistoryTimer->InitWithNamedFuncCallback(
+            [](nsITimer* aTimer, void* aClosure) {
+              if (WebrtcGlobalStatsHistory::Pref::Enabled()) {
+                WebrtcGlobalInformation::GatherHistory();
+              }
+            },
+            nullptr, WebrtcGlobalStatsHistory::Pref::PollIntervalMs(),
+            nsITimer::TYPE_REPEATING_SLACK,
+            "WebrtcGlobalInformation::GatherHistory");
+      }
+      ClearOnShutdown(&sHistoryTimer);
+    }
+  };
+
+  auto HandleRemove = [&](const nsString& aRemoved) {
+    WebrtcGlobalStatsHistory::CloseHistory(nsString(aRemoved));
+    if (!sPcids || !sPcids->Count()) {
+      return;
+    }
+    if (!sPcids->Contains(aRemoved)) {
+      return;
+    }
+    sPcids->Remove(aRemoved);
+    if (!sPcids->Count() && sHistoryTimer) {
+      sHistoryTimer->Cancel();
+      sHistoryTimer = nullptr;
+    }
+  };
+
+  switch (aUpdate.Type()) {
+    case PcTrackingUpdate::Type::Add: {
+      HandleAdd(std::move(aUpdate.mPcid),
+                aUpdate.mLongTermStatsDisabled.valueOrFrom([&]() {
+                  MOZ_ASSERT(aUpdate.mLongTermStatsDisabled.isNothing());
+                  return true;
+                }));
+      return;
+    }
+    case PcTrackingUpdate::Type::Remove: {
+      HandleRemove(aUpdate.mPcid);
+      return;
+    }
+    default: {
+      MOZ_ASSERT(false, "Invalid PcCount operation");
+    }
+  }
+}
+
+WebrtcGlobalParent* WebrtcGlobalParent::Alloc() {
+  return WebrtcContentParents::Alloc();
+}
+
+bool WebrtcGlobalParent::Dealloc(WebrtcGlobalParent* aActor) {
+  WebrtcContentParents::Dealloc(aActor);
+  return true;
+}
+
+void WebrtcGlobalParent::ActorDestroy(ActorDestroyReason aWhy) {
+  mShutdown = true;
+  for (const auto& pcId : mPcids) {
+    using Update = WebrtcGlobalInformation::PcTrackingUpdate;
+    auto update = Update::Remove(nsString(pcId));
+    WebrtcGlobalInformation::PeerConnectionTracking(update);
+  }
+}
+
+mozilla::ipc::IPCResult WebrtcGlobalParent::Recv__delete__() {
+  return IPC_OK();
+}
+
+mozilla::ipc::IPCResult WebrtcGlobalParent::RecvPeerConnectionCreated(
+    const nsAString& aPcId, const bool& aIsLongTermStatsDisabled) {
+  if (mShutdown) {
+    return IPC_OK();
+  }
+  mPcids.EnsureInserted(aPcId);
+  using Update = WebrtcGlobalInformation::PcTrackingUpdate;
+  auto update = Update::Add(nsString(aPcId), aIsLongTermStatsDisabled);
+  WebrtcGlobalInformation::PeerConnectionTracking(update);
+  return IPC_OK();
+}
+
+mozilla::ipc::IPCResult WebrtcGlobalParent::RecvPeerConnectionDestroyed(
+    const nsAString& aPcId) {
+  mPcids.EnsureRemoved(aPcId);
+  using Update = WebrtcGlobalInformation::PcTrackingUpdate;
+  auto update = Update::Remove(nsString(aPcId));
+  WebrtcGlobalStatsHistory::CloseHistory(aPcId);
+  WebrtcGlobalInformation::PeerConnectionTracking(update);
+  return IPC_OK();
+}
+
+mozilla::ipc::IPCResult WebrtcGlobalParent::RecvPeerConnectionFinalStats(
+    const RTCStatsReportInternal& aFinalStats) {
+  auto finalStats = MakeUnique<RTCStatsReportInternal>(aFinalStats);
+  WebrtcGlobalStatsHistory::Record(std::move(finalStats));
+  WebrtcGlobalStatsHistory::CloseHistory(aFinalStats.mPcid);
+  return IPC_OK();
+}
+
+MOZ_IMPLICIT WebrtcGlobalParent::WebrtcGlobalParent() : mShutdown(false) {
+  MOZ_COUNT_CTOR(WebrtcGlobalParent);
+}
+
+MOZ_IMPLICIT WebrtcGlobalParent::~WebrtcGlobalParent() {
+  MOZ_COUNT_DTOR(WebrtcGlobalParent);
+}
+
+mozilla::ipc::IPCResult WebrtcGlobalChild::RecvGetStats(
+    const nsAString& aPcIdFilter, GetStatsResolver&& aResolve) {
+  if (!mShutdown) {
+    GetStatsPromiseForThisProcess(aPcIdFilter)
+        ->Then(
+            GetMainThreadSerialEventTarget(), __func__,
+            [resolve = std::move(aResolve)](
+                nsTArray<dom::RTCStatsReportInternal>&& aReports) {
+              resolve(std::move(aReports));
+            },
+            []() { MOZ_CRASH(); });
+    return IPC_OK();
+  }
+
+  aResolve(nsTArray<RTCStatsReportInternal>());
+  return IPC_OK();
+}
+
+mozilla::ipc::IPCResult WebrtcGlobalChild::RecvClearStats() {
+  if (mShutdown) {
+    return IPC_OK();
+  }
+
+  ClearLongTermStats();
+  return IPC_OK();
+}
+
+mozilla::ipc::IPCResult WebrtcGlobalChild::RecvGetLog(
+    GetLogResolver&& aResolve) {
+  if (mShutdown) {
+    aResolve(Sequence<nsString>());
+    return IPC_OK();
+  }
+
+  GetLogPromise()->Then(
+      GetMainThreadSerialEventTarget(), __func__,
+      [aResolve = std::move(aResolve)](
+          PWebrtcGlobalParent::GetLogPromise::ResolveOrRejectValue&& aValue) {
+        if (aValue.IsResolve()) {
+          aResolve(aValue.ResolveValue());
+        } else {
+          aResolve(Sequence<nsString>());
+        }
+      });
+
+  return IPC_OK();
+}
+
+mozilla::ipc::IPCResult WebrtcGlobalChild::RecvClearLog() {
+  if (mShutdown) {
+    return IPC_OK();
+  }
+
+  RunLogClear();
+  return IPC_OK();
+}
+
+mozilla::ipc::IPCResult WebrtcGlobalChild::RecvSetAecLogging(
+    const bool& aEnable) {
+  if (!mShutdown) {
+    if (aEnable) {
+      StartAecLog();
+    } else {
+      StopAecLog();
+    }
+  }
+  return IPC_OK();
+}
+
+mozilla::ipc::IPCResult WebrtcGlobalChild::RecvSetDebugMode(const int& aLevel) {
+  if (!mShutdown) {
+    if (aLevel) {
+      StartWebRtcLog(mozilla::LogLevel(aLevel));
+    } else {
+      StopWebRtcLog();
+    }
+  }
+  return IPC_OK();
+}
+
+WebrtcGlobalChild* WebrtcGlobalChild::GetOrSet(
+    const Maybe<WebrtcGlobalChild*>& aChild) {
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_ASSERT(XRE_IsContentProcess());
+  static WebrtcGlobalChild* sChild;
+  if (!sChild && !aChild) {
+    sChild = static_cast<WebrtcGlobalChild*>(
+        ContentChild::GetSingleton()->SendPWebrtcGlobalConstructor());
+  }
+  aChild.apply([](auto* child) { sChild = child; });
+  return sChild;
+}
+
+WebrtcGlobalChild* WebrtcGlobalChild::Get() { return GetOrSet(Nothing()); }
+
+void WebrtcGlobalChild::ActorDestroy(ActorDestroyReason aWhy) {
+  mShutdown = true;
+}
+
+MOZ_IMPLICIT WebrtcGlobalChild::WebrtcGlobalChild() : mShutdown(false) {
+  MOZ_COUNT_CTOR(WebrtcGlobalChild);
+}
+
+MOZ_IMPLICIT WebrtcGlobalChild::~WebrtcGlobalChild() {
+  MOZ_COUNT_DTOR(WebrtcGlobalChild);
+  GetOrSet(Some(nullptr));
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webrtc/jsapi/WebrtcGlobalInformation.h b/dom/media/webrtc/jsapi/WebrtcGlobalInformation.h
new file mode 100644
index 0000000000..d43013bb43
--- /dev/null
+++ b/dom/media/webrtc/jsapi/WebrtcGlobalInformation.h
@@ -0,0 +1,102 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _WEBRTC_GLOBAL_INFORMATION_H_
+#define _WEBRTC_GLOBAL_INFORMATION_H_
+
+#include <tuple>
+#include "mozilla/dom/WebrtcGlobalInformationBinding.h"
+#include "nsString.h"
+#include "mozilla/dom/BindingDeclarations.h"  // for Optional
+#include "nsDOMNavigationTiming.h"
+#include "WebrtcGlobalStatsHistory.h"
+
+namespace mozilla {
+class PeerConnectionImpl;
+class ErrorResult;
+
+namespace dom {
+
+class GlobalObject;
+class WebrtcGlobalStatisticsCallback;
+class WebrtcGlobalStatisticsHistoryPcIdsCallback;
+class WebrtcGlobalLoggingCallback;
+struct RTCStatsReportInternal;
+
+class WebrtcGlobalInformation {
+ public:
+  MOZ_CAN_RUN_SCRIPT
+  static void GetAllStats(const GlobalObject& aGlobal,
+                          WebrtcGlobalStatisticsCallback& aStatsCallback,
+                          const Optional<nsAString>& aPcIdFilter,
+                          ErrorResult& aRv);
+
+  MOZ_CAN_RUN_SCRIPT
+  static void GetStatsHistoryPcIds(
+      const GlobalObject& aGlobal,
+      WebrtcGlobalStatisticsHistoryPcIdsCallback& aPcIdsCallback,
+      ErrorResult& aRv);
+
+  MOZ_CAN_RUN_SCRIPT
+  static void GetStatsHistorySince(
+      const GlobalObject& aGlobal,
+      WebrtcGlobalStatisticsHistoryCallback& aStatsCallback,
+      const nsAString& aPcIdFilter, const Optional<DOMHighResTimeStamp>& aAfter,
+      const Optional<DOMHighResTimeStamp>& aSdpAfter, ErrorResult& aRv);
+
+  static void GatherHistory();
+
+  static void ClearAllStats(const GlobalObject& aGlobal);
+
+  MOZ_CAN_RUN_SCRIPT
+  static void GetLogging(const GlobalObject& aGlobal, const nsAString& aPattern,
+                         WebrtcGlobalLoggingCallback& aLoggingCallback,
+                         ErrorResult& aRv);
+
+  static void ClearLogging(const GlobalObject& aGlobal);
+
+  static void SetDebugLevel(const GlobalObject& aGlobal, int32_t aLevel);
+  static int32_t DebugLevel(const GlobalObject& aGlobal);
+
+  static void SetAecDebug(const GlobalObject& aGlobal, bool aEnable);
+  static bool AecDebug(const GlobalObject& aGlobal);
+  static void GetAecDebugLogDir(const GlobalObject& aGlobal, nsAString& aDir);
+
+  static void StashStats(const RTCStatsReportInternal& aReport);
+
+  WebrtcGlobalInformation() = delete;
+  WebrtcGlobalInformation(const WebrtcGlobalInformation& aOrig) = delete;
+  WebrtcGlobalInformation& operator=(const WebrtcGlobalInformation& aRhs) =
+      delete;
+
+  struct PcTrackingUpdate {
+    static PcTrackingUpdate Add(const nsString& aPcid,
+                                const bool& aLongTermStatsDisabled) {
+      return PcTrackingUpdate{aPcid, Some(aLongTermStatsDisabled)};
+    }
+    static PcTrackingUpdate Remove(const nsString& aPcid) {
+      return PcTrackingUpdate{aPcid, Nothing()};
+    }
+    nsString mPcid;
+    Maybe<bool> mLongTermStatsDisabled;
+    enum class Type {
+      Add,
+      Remove,
+    };
+    Type Type() const {
+      return mLongTermStatsDisabled ? Type::Add : Type::Remove;
+    }
+  };
+  static void PeerConnectionTracking(PcTrackingUpdate& aUpdate) {
+    AdjustTimerReferences(std::move(aUpdate));
+  }
+
+ private:
+  static void AdjustTimerReferences(PcTrackingUpdate&& aUpdate);
+};
+
+}  // namespace dom
+}  // namespace mozilla
+
+#endif  // _WEBRTC_GLOBAL_INFORMATION_H_
diff --git a/dom/media/webrtc/jsapi/WebrtcGlobalParent.h b/dom/media/webrtc/jsapi/WebrtcGlobalParent.h
new file mode 100644
index 0000000000..8372c4182f
--- /dev/null
+++ b/dom/media/webrtc/jsapi/WebrtcGlobalParent.h
@@ -0,0 +1,52 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _WEBRTC_GLOBAL_PARENT_H_
+#define _WEBRTC_GLOBAL_PARENT_H_
+
+#include "mozilla/dom/PWebrtcGlobalParent.h"
+#include "mozilla/dom/BindingDeclarations.h"
+#include "nsISupportsImpl.h"
+
+namespace mozilla::dom {
+
+class WebrtcParents;
+
+class WebrtcGlobalParent : public PWebrtcGlobalParent {
+  friend class ContentParent;
+  friend class WebrtcGlobalInformation;
+  friend class WebrtcContentParents;
+
+  bool mShutdown;
+  nsTHashSet<nsString> mPcids;
+
+  MOZ_IMPLICIT WebrtcGlobalParent();
+
+  static WebrtcGlobalParent* Alloc();
+  static bool Dealloc(WebrtcGlobalParent* aActor);
+
+  virtual void ActorDestroy(ActorDestroyReason aWhy) override;
+  virtual mozilla::ipc::IPCResult Recv__delete__() override;
+  // Notification that a PeerConnection exists, and stats polling can begin
+  // if it hasn't already begun due to a previously created PeerConnection.
+  virtual mozilla::ipc::IPCResult RecvPeerConnectionCreated(
+      const nsAString& aPcId, const bool& aIsLongTermStatsDisabled) override;
+  // Notification that a PeerConnection no longer exists, and stats polling
+  // can end if there are no other PeerConnections.
+  virtual mozilla::ipc::IPCResult RecvPeerConnectionDestroyed(
+      const nsAString& aPcid) override;
+  // Ditto but we have final stats
+  virtual mozilla::ipc::IPCResult RecvPeerConnectionFinalStats(
+      const RTCStatsReportInternal& aFinalStats) override;
+  virtual ~WebrtcGlobalParent();
+
+ public:
+  NS_INLINE_DECL_REFCOUNTING(WebrtcGlobalParent)
+
+  bool IsActive() { return !mShutdown; }
+};
+
+}  // namespace mozilla::dom
+
+#endif  // _WEBRTC_GLOBAL_PARENT_H_
diff --git a/dom/media/webrtc/jsapi/WebrtcGlobalStatsHistory.cpp b/dom/media/webrtc/jsapi/WebrtcGlobalStatsHistory.cpp
new file mode 100644
index 0000000000..1d576b5dca
--- /dev/null
+++ b/dom/media/webrtc/jsapi/WebrtcGlobalStatsHistory.cpp
@@ -0,0 +1,282 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "WebrtcGlobalStatsHistory.h"
+#include <memory>
+
+#include "domstubs.h"
+#include "mozilla/LinkedList.h"
+#include "mozilla/dom/BindingUtils.h"
+#include "mozilla/dom/RTCStatsReportBinding.h"  // for RTCStatsReportInternal
+#include "mozilla/ClearOnShutdown.h"
+#include "mozilla/StaticPrefs_media.h"
+#include "mozilla/fallible.h"
+#include "mozilla/mozalloc_oom.h"
+#include "nsDOMNavigationTiming.h"
+
+namespace mozilla::dom {
+
+constexpr auto SEC_TO_MS(const DOMHighResTimeStamp sec) -> DOMHighResTimeStamp {
+  return sec * 1000.0;
+}
+
+constexpr auto MIN_TO_MS(const DOMHighResTimeStamp min) -> DOMHighResTimeStamp {
+  return SEC_TO_MS(min * 60.0);
+}
+
+// Prefs
+auto WebrtcGlobalStatsHistory::Pref::Enabled() -> bool {
+  return mozilla::StaticPrefs::media_aboutwebrtc_hist_enabled();
+}
+
+auto WebrtcGlobalStatsHistory::Pref::PollIntervalMs() -> uint32_t {
+  return mozilla::StaticPrefs::media_aboutwebrtc_hist_poll_interval_ms();
+}
+
+auto WebrtcGlobalStatsHistory::Pref::StorageWindowS() -> uint32_t {
+  return mozilla::StaticPrefs::media_aboutwebrtc_hist_storage_window_s();
+}
+
+auto WebrtcGlobalStatsHistory::Pref::PruneAfterM() -> uint32_t {
+  return mozilla::StaticPrefs::media_aboutwebrtc_hist_prune_after_m();
+}
+
+auto WebrtcGlobalStatsHistory::Pref::ClosedStatsToRetain() -> uint32_t {
+  return mozilla::StaticPrefs::media_aboutwebrtc_hist_closed_stats_to_retain();
+}
+
+auto WebrtcGlobalStatsHistory::Get() -> WebrtcGlobalStatsHistory::StatsMap& {
+  static StaticAutoPtr<StatsMap> sHist;
+  if (!sHist) {
+    sHist = new StatsMap();
+    ClearOnShutdown(&sHist);
+  }
+  return *sHist;
+}
+
+auto WebrtcGlobalStatsHistory::Entry::ReportElement::Timestamp() const
+    -> DOMHighResTimeStamp {
+  return report->mTimestamp;
+}
+
+auto WebrtcGlobalStatsHistory::Entry::SdpElement::Timestamp() const
+    -> DOMHighResTimeStamp {
+  return sdp.mTimestamp;
+}
+
+auto WebrtcGlobalStatsHistory::Entry::MakeReportElement(
+    UniquePtr<RTCStatsReportInternal> aReport)
+    -> WebrtcGlobalStatsHistory::Entry::ReportElement* {
+  auto* elem = new ReportElement();
+  elem->report = std::move(aReport);
+  // We don't want to store a copy of the SDP history with each stats entry.
+  // SDP History is stored seperately, see MakeSdpElements.
+  elem->report->mSdpHistory.Clear();
+  return elem;
+}
+
+auto WebrtcGlobalStatsHistory::Entry::MakeSdpElementsSince(
+    Sequence<RTCSdpHistoryEntryInternal>&& aSdpHistory,
+    const Maybe<DOMHighResTimeStamp>& aSdpAfter)
+    -> AutoCleanLinkedList<WebrtcGlobalStatsHistory::Entry::SdpElement> {
+  AutoCleanLinkedList<WebrtcGlobalStatsHistory::Entry::SdpElement> result;
+  for (auto& sdpHist : aSdpHistory) {
+    if (!aSdpAfter || aSdpAfter.value() < sdpHist.mTimestamp) {
+      auto* element = new SdpElement();
+      element->sdp = sdpHist;
+      result.insertBack(element);
+    }
+  }
+  return result;
+}
+
+template <typename T>
+auto FindFirstEntryAfter(const T* first,
+                         const Maybe<DOMHighResTimeStamp>& aAfter) -> const T* {
+  const auto* current = first;
+  while (aAfter && current && current->Timestamp() <= aAfter.value()) {
+    current = current->getNext();
+  }
+  return current;
+}
+
+template <typename T>
+auto CountElementsToEndInclusive(const LinkedListElement<T>* elem) -> size_t {
+  size_t count = 0;
+  const auto* cursor = elem;
+  while (cursor && cursor->isInList()) {
+    count++;
+    cursor = cursor->getNext();
+  }
+  return count;
+}
+
+auto WebrtcGlobalStatsHistory::Entry::Since(
+    const Maybe<DOMHighResTimeStamp>& aAfter) const
+    -> nsTArray<RTCStatsReportInternal> {
+  nsTArray<RTCStatsReportInternal> results;
+  const auto* cursor = FindFirstEntryAfter(mReports.getFirst(), aAfter);
+  const auto count = CountElementsToEndInclusive(cursor);
+  if (!results.SetCapacity(count, fallible)) {
+    mozalloc_handle_oom(0);
+  }
+  while (cursor) {
+    results.AppendElement(RTCStatsReportInternal(*cursor->report));
+    cursor = cursor->getNext();
+  }
+  return results;
+}
+
+auto WebrtcGlobalStatsHistory::Entry::SdpSince(
+    const Maybe<DOMHighResTimeStamp>& aAfter) const -> RTCSdpHistoryInternal {
+  RTCSdpHistoryInternal results;
+  results.mPcid = mPcid;
+  // If no timestamp was passed copy the entire history
+  const auto* cursor = FindFirstEntryAfter(mSdp.getFirst(), aAfter);
+  const auto count = CountElementsToEndInclusive(cursor);
+  if (!results.mSdpHistory.SetCapacity(count, fallible)) {
+    mozalloc_handle_oom(0);
+  }
+  while (cursor) {
+    if (!results.mSdpHistory.AppendElement(
+            RTCSdpHistoryEntryInternal(cursor->sdp), fallible)) {
+      mozalloc_handle_oom(0);
+    }
+    cursor = cursor->getNext();
+  }
+  return results;
+}
+
+auto WebrtcGlobalStatsHistory::Entry::Prune(const DOMHighResTimeStamp aBefore)
+    -> void {
+  // Clear everything in the case that we don't keep stats
+  if (mIsLongTermStatsDisabled) {
+    mReports.clear();
+  }
+  // Clear everything before the cutoff
+  for (auto* element = mReports.getFirst();
+       element && element->report->mTimestamp < aBefore;
+       element = mReports.getFirst()) {
+    delete mReports.popFirst();
+  }
+  // I don't think we should prune SDPs but if we did it would look like this:
+  // Note: we always keep the most recent SDP
+}
+
+auto WebrtcGlobalStatsHistory::InitHistory(const nsAString& aPcId,
+                                           const bool aIsLongTermStatsDisabled)
+    -> void {
+  MOZ_ASSERT(XRE_IsParentProcess());
+  if (WebrtcGlobalStatsHistory::Get().MaybeGet(aPcId)) {
+    return;
+  }
+  WebrtcGlobalStatsHistory::Get().GetOrInsertNew(aPcId, nsString(aPcId),
+                                                 aIsLongTermStatsDisabled);
+};
+
+auto WebrtcGlobalStatsHistory::Record(UniquePtr<RTCStatsReportInternal> aReport)
+    -> void {
+  MOZ_ASSERT(XRE_IsParentProcess());
+  // Use the report timestamp as "now" for determining time depth
+  // based pruning.
+  const auto now = aReport->mTimestamp;
+  const auto earliest = now - SEC_TO_MS(Pref::StorageWindowS());
+
+  // Store closed state before moving the report
+  const auto closed = aReport->mClosed;
+  const auto pcId = aReport->mPcid;
+
+  auto history = WebrtcGlobalStatsHistory::GetHistory(aReport->mPcid);
+  if (history && Pref::Enabled()) {
+    auto entry = history.value();
+    // Remove expired entries
+    entry->Prune(earliest);
+    // Find new SDP entries
+    auto sdpAfter = Maybe<DOMHighResTimeStamp>(Nothing());
+    if (auto* lastSdp = entry->mSdp.getLast(); lastSdp) {
+      sdpAfter = Some(lastSdp->Timestamp());
+    }
+    entry->mSdp.extendBack(
+        Entry::MakeSdpElementsSince(std::move(aReport->mSdpHistory), sdpAfter));
+    // Reports must be in ascending order by mTimestamp
+    const auto* latest = entry->mReports.getLast();
+    // Maintain sorted order
+    if (!latest || latest->report->mTimestamp < aReport->mTimestamp) {
+      entry->mReports.insertBack(Entry::MakeReportElement(std::move(aReport)));
+    }
+  }
+  // Close the history if needed
+  if (closed) {
+    CloseHistory(pcId);
+  }
+}
+
+auto WebrtcGlobalStatsHistory::CloseHistory(const nsAString& aPcId) -> void {
+  MOZ_ASSERT(XRE_IsParentProcess());
+  auto maybeHist = WebrtcGlobalStatsHistory::Get().MaybeGet(aPcId);
+  if (!maybeHist) {
+    return;
+  }
+  {
+    auto&& hist = maybeHist.value();
+    hist->mIsClosed = true;
+
+    if (hist->mIsLongTermStatsDisabled) {
+      WebrtcGlobalStatsHistory::Get().Remove(aPcId);
+      return;
+    }
+  }
+  size_t remainingClosedStatsToRetain =
+      WebrtcGlobalStatsHistory::Pref::ClosedStatsToRetain();
+  WebrtcGlobalStatsHistory::Get().RemoveIf([&](auto& iter) {
+    auto& entry = iter.Data();
+    if (!entry->mIsClosed) {
+      return false;
+    }
+    if (entry->mIsLongTermStatsDisabled) {
+      return true;
+    }
+    if (remainingClosedStatsToRetain > 0) {
+      remainingClosedStatsToRetain -= 1;
+      return false;
+    }
+    return true;
+  });
+}
+
+auto WebrtcGlobalStatsHistory::Clear() -> void {
+  MOZ_ASSERT(XRE_IsParentProcess());
+  WebrtcGlobalStatsHistory::Get().RemoveIf([](auto& aIter) {
+    // First clear all the closed histories.
+    if (aIter.Data()->mIsClosed) {
+      return true;
+    }
+    // For all remaining histories clear their stored reports
+    aIter.Data()->mReports.clear();
+    // As an optimization we don't clear the SDP, because that would
+    // be reconstitued in the very next stats gathering polling period.
+    // Those are potentially large allocations which we can skip.
+    return false;
+  });
+}
+
+auto WebrtcGlobalStatsHistory::PcIds() -> dom::Sequence<nsString> {
+  MOZ_ASSERT(XRE_IsParentProcess());
+  dom::Sequence<nsString> pcIds;
+  for (const auto& pcId : WebrtcGlobalStatsHistory::Get().Keys()) {
+    if (!pcIds.AppendElement(pcId, fallible)) {
+      mozalloc_handle_oom(0);
+    }
+  }
+  return pcIds;
+}
+
+auto WebrtcGlobalStatsHistory::GetHistory(const nsAString& aPcId)
+    -> Maybe<RefPtr<Entry> > {
+  MOZ_ASSERT(XRE_IsParentProcess());
+  const auto pcid = NS_ConvertUTF16toUTF8(aPcId);
+
+  return WebrtcGlobalStatsHistory::Get().MaybeGet(aPcId);
+}
+}  // namespace mozilla::dom
diff --git a/dom/media/webrtc/jsapi/WebrtcGlobalStatsHistory.h b/dom/media/webrtc/jsapi/WebrtcGlobalStatsHistory.h
new file mode 100644
index 0000000000..aa820e787d
--- /dev/null
+++ b/dom/media/webrtc/jsapi/WebrtcGlobalStatsHistory.h
@@ -0,0 +1,88 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#pragma once
+
+#include <memory>
+#include "domstubs.h"
+#include "mozilla/LinkedList.h"
+#include "mozilla/UniquePtr.h"
+#include "mozilla/dom/BindingDeclarations.h"
+#include "mozilla/dom/DOMString.h"
+#include "mozilla/dom/RTCStatsReportBinding.h"
+#include "nsDOMNavigationTiming.h"
+#include "nsHashKeys.h"
+#include "nsTHashMap.h"
+
+namespace mozilla::dom {
+class WebrtcGlobalStatisticsHistoryCallback;
+struct RTCStatsReportInternal;
+
+struct WebrtcGlobalStatsHistory {
+  // History preferences
+  struct Pref {
+    static auto Enabled() -> bool;
+    static auto PollIntervalMs() -> uint32_t;
+    static auto StorageWindowS() -> uint32_t;
+    static auto PruneAfterM() -> uint32_t;
+    static auto ClosedStatsToRetain() -> uint32_t;
+    Pref() = delete;
+    ~Pref() = delete;
+  };
+
+  struct Entry {
+    NS_INLINE_DECL_THREADSAFE_REFCOUNTING(Entry)
+    // We need to wrap the report in an element
+    struct ReportElement : public LinkedListElement<ReportElement> {
+      UniquePtr<RTCStatsReportInternal> report;
+      auto Timestamp() const -> DOMHighResTimeStamp;
+      virtual ~ReportElement() = default;
+    };
+    // And likewise for the SDP history
+    struct SdpElement : public LinkedListElement<SdpElement> {
+      RTCSdpHistoryEntryInternal sdp;
+      auto Timestamp() const -> DOMHighResTimeStamp;
+      virtual ~SdpElement() = default;
+    };
+
+    explicit Entry(const nsString& aPcid, const bool aIsLongTermStatsDisabled)
+        : mPcid(aPcid), mIsLongTermStatsDisabled(aIsLongTermStatsDisabled) {}
+
+    nsString mPcid;
+    AutoCleanLinkedList<ReportElement> mReports;
+    AutoCleanLinkedList<SdpElement> mSdp;
+    bool mIsLongTermStatsDisabled;
+    bool mIsClosed = false;
+
+    auto Since(const Maybe<DOMHighResTimeStamp>& aAfter) const
+        -> nsTArray<RTCStatsReportInternal>;
+    auto SdpSince(const Maybe<DOMHighResTimeStamp>& aAfter) const
+        -> RTCSdpHistoryInternal;
+
+    static auto MakeReportElement(UniquePtr<RTCStatsReportInternal> aReport)
+        -> ReportElement*;
+    static auto MakeSdpElementsSince(
+        Sequence<RTCSdpHistoryEntryInternal>&& aSdpHistory,
+        const Maybe<DOMHighResTimeStamp>& aSdpAfter)
+        -> AutoCleanLinkedList<SdpElement>;
+    auto Prune(const DOMHighResTimeStamp aBefore) -> void;
+
+   private:
+    virtual ~Entry() = default;
+  };
+  using StatsMap = nsTHashMap<nsStringHashKey, RefPtr<Entry> >;
+  static auto InitHistory(const nsAString& aPcId,
+                          const bool aIsLongTermStatsDisabled) -> void;
+  static auto Record(UniquePtr<RTCStatsReportInternal> aReport) -> void;
+  static auto CloseHistory(const nsAString& aPcId) -> void;
+  static auto GetHistory(const nsAString& aPcId) -> Maybe<RefPtr<Entry> >;
+  static auto Clear() -> void;
+  static auto PcIds() -> dom::Sequence<nsString>;
+
+  WebrtcGlobalStatsHistory() = delete;
+
+ private:
+  static auto Get() -> StatsMap&;
+};
+}  // namespace mozilla::dom
diff --git a/dom/media/webrtc/jsapi/moz.build b/dom/media/webrtc/jsapi/moz.build
new file mode 100644
index 0000000000..2c1dbe79f1
--- /dev/null
+++ b/dom/media/webrtc/jsapi/moz.build
@@ -0,0 +1,51 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+include("/dom/media/webrtc/third_party_build/webrtc.mozbuild")
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "/dom/base",
+    "/dom/media",
+    "/dom/media/webrtc",
+    "/ipc/chromium/src",
+    "/media/webrtc",
+    "/netwerk/dns",  # For nsDNSService2.h
+    "/third_party/libsrtp/src/include",
+    "/third_party/libwebrtc",
+    "/third_party/libwebrtc/third_party/abseil-cpp",
+]
+
+UNIFIED_SOURCES += [
+    "MediaTransportHandler.cpp",
+    "MediaTransportHandlerIPC.cpp",
+    "MediaTransportParent.cpp",
+    "PacketDumper.cpp",
+    "PeerConnectionCtx.cpp",
+    "PeerConnectionImpl.cpp",
+    "RemoteTrackSource.cpp",
+    "RTCDtlsTransport.cpp",
+    "RTCDTMFSender.cpp",
+    "RTCRtpReceiver.cpp",
+    "RTCRtpSender.cpp",
+    "RTCRtpTransceiver.cpp",
+    "RTCSctpTransport.cpp",
+    "RTCStatsIdGenerator.cpp",
+    "RTCStatsReport.cpp",
+    "WebrtcGlobalInformation.cpp",
+    "WebrtcGlobalStatsHistory.cpp",
+]
+
+EXPORTS.mozilla.dom += [
+    "RTCDtlsTransport.h",
+    "RTCDTMFSender.h",
+    "RTCRtpReceiver.h",
+    "RTCRtpSender.h",
+    "RTCRtpTransceiver.h",
+    "RTCSctpTransport.h",
+    "RTCStatsReport.h",
+]
+
+FINAL_LIBRARY = "xul"
diff --git a/dom/media/webrtc/jsep/JsepCodecDescription.h b/dom/media/webrtc/jsep/JsepCodecDescription.h
new file mode 100644
index 0000000000..5336182b77
--- /dev/null
+++ b/dom/media/webrtc/jsep/JsepCodecDescription.h
@@ -0,0 +1,1196 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _JSEPCODECDESCRIPTION_H_
+#define _JSEPCODECDESCRIPTION_H_
+
+#include <cmath>
+#include <string>
+#include "sdp/SdpMediaSection.h"
+#include "sdp/SdpHelper.h"
+#include "nsCRT.h"
+#include "nsString.h"
+#include "mozilla/net/DataChannelProtocol.h"
+#include "mozilla/Preferences.h"
+
+namespace mozilla {
+
+#define JSEP_CODEC_CLONE(T) \
+  virtual JsepCodecDescription* Clone() const override { return new T(*this); }
+
+// A single entry in our list of known codecs.
+class JsepCodecDescription {
+ public:
+  JsepCodecDescription(const std::string& defaultPt, const std::string& name,
+                       uint32_t clock, uint32_t channels, bool enabled)
+      : mDefaultPt(defaultPt),
+        mName(name),
+        mClock(clock),
+        mChannels(channels),
+        mEnabled(enabled),
+        mStronglyPreferred(false),
+        mDirection(sdp::kSend) {}
+  virtual ~JsepCodecDescription() {}
+
+  virtual SdpMediaSection::MediaType Type() const = 0;
+
+  virtual JsepCodecDescription* Clone() const = 0;
+
+  bool GetPtAsInt(uint16_t* ptOutparam) const {
+    return SdpHelper::GetPtAsInt(mDefaultPt, ptOutparam);
+  }
+
+  // The id used for codec stats, to uniquely identify this codec configuration
+  // within a transport.
+  const nsString& StatsId() const {
+    if (!mStatsId) {
+      mStatsId.emplace();
+      mStatsId->AppendPrintf(
+          "_%s_%s/%s_%u_%u_%s", mDefaultPt.c_str(),
+          Type() == SdpMediaSection::kVideo ? "video" : "audio", mName.c_str(),
+          mClock, mChannels, mSdpFmtpLine ? mSdpFmtpLine->c_str() : "nothing");
+    }
+    return *mStatsId;
+  }
+
+  virtual bool Matches(const std::string& fmt,
+                       const SdpMediaSection& remoteMsection) const {
+    // note: fmt here is remote fmt (to go with remoteMsection)
+    if (Type() != remoteMsection.GetMediaType()) {
+      return false;
+    }
+
+    const SdpRtpmapAttributeList::Rtpmap* entry(remoteMsection.FindRtpmap(fmt));
+
+    if (entry) {
+      if (!nsCRT::strcasecmp(mName.c_str(), entry->name.c_str()) &&
+          (mClock == entry->clock) && (mChannels == entry->channels)) {
+        return ParametersMatch(fmt, remoteMsection);
+      }
+    } else if (!fmt.compare("9") && mName == "G722") {
+      return true;
+    } else if (!fmt.compare("0") && mName == "PCMU") {
+      return true;
+    } else if (!fmt.compare("8") && mName == "PCMA") {
+      return true;
+    }
+    return false;
+  }
+
+  virtual bool ParametersMatch(const std::string& fmt,
+                               const SdpMediaSection& remoteMsection) const {
+    return true;
+  }
+
+  Maybe<std::string> GetMatchingFormat(
+      const SdpMediaSection& remoteMsection) const {
+    for (const auto& fmt : remoteMsection.GetFormats()) {
+      if (Matches(fmt, remoteMsection)) {
+        return Some(fmt);
+      }
+    }
+    return Nothing();
+  }
+
+  virtual bool Negotiate(const std::string& pt,
+                         const SdpMediaSection& remoteMsection,
+                         bool remoteIsOffer,
+                         Maybe<const SdpMediaSection&> localMsection) {
+    // Configuration might change. Invalidate the stats id.
+    mStatsId = Nothing();
+    if (mDirection == sdp::kSend || remoteIsOffer) {
+      mDefaultPt = pt;
+    }
+    if (localMsection) {
+      // Offer/answer is concluding. Update the sdpFmtpLine.
+      MOZ_ASSERT(mDirection == sdp::kSend || mDirection == sdp::kRecv);
+      const SdpMediaSection& msection =
+          mDirection == sdp::kSend ? remoteMsection : *localMsection;
+      UpdateSdpFmtpLine(ToMaybeRef(msection.FindFmtp(mDefaultPt)));
+    }
+    return true;
+  }
+
+  virtual void ApplyConfigToFmtp(
+      UniquePtr<SdpFmtpAttributeList::Parameters>& aFmtp) const = 0;
+
+  virtual void AddToMediaSection(SdpMediaSection& msection) const {
+    if (mEnabled && msection.GetMediaType() == Type()) {
+      if (mDirection == sdp::kRecv) {
+        msection.AddCodec(mDefaultPt, mName, mClock, mChannels);
+      }
+
+      AddParametersToMSection(msection);
+    }
+  }
+
+  virtual void AddParametersToMSection(SdpMediaSection& msection) const {}
+
+  virtual void EnsureNoDuplicatePayloadTypes(std::set<std::string>& aUsedPts) {
+    mEnabled = EnsurePayloadTypeNotDuplicate(aUsedPts, mDefaultPt);
+  }
+
+  bool EnsurePayloadTypeNotDuplicate(std::set<std::string>& aUsedPts,
+                                     std::string& aPtToCheck) {
+    if (!mEnabled) {
+      return false;
+    }
+
+    if (!aUsedPts.count(aPtToCheck)) {
+      aUsedPts.insert(aPtToCheck);
+      return true;
+    }
+
+    // |codec| cannot use its current payload type. Try to find another.
+    for (uint16_t freePt = 96; freePt <= 127; ++freePt) {
+      // Not super efficient, but readability is probably more important.
+      std::ostringstream os;
+      os << freePt;
+      std::string freePtAsString = os.str();
+
+      if (!aUsedPts.count(freePtAsString)) {
+        aUsedPts.insert(freePtAsString);
+        aPtToCheck = freePtAsString;
+        return true;
+      }
+    }
+
+    return false;
+  }
+
+  // TODO Bug 1751671: Take a verbatim fmtp line (std::string or eq.) instead
+  // of fmtp parameters that have to be (re-)serialized.
+  void UpdateSdpFmtpLine(
+      const Maybe<const SdpFmtpAttributeList::Parameters&> aParams) {
+    mSdpFmtpLine = aParams.map([](const auto& aFmtp) {
+      std::stringstream ss;
+      aFmtp.Serialize(ss);
+      return ss.str();
+    });
+  }
+
+  std::string mDefaultPt;
+  std::string mName;
+  Maybe<std::string> mSdpFmtpLine;
+  mutable Maybe<nsString> mStatsId;
+  uint32_t mClock;
+  uint32_t mChannels;
+  bool mEnabled;
+  bool mStronglyPreferred;
+  sdp::Direction mDirection;
+  // Will hold constraints from both fmtp and rid
+  EncodingConstraints mConstraints;
+};
+
+class JsepAudioCodecDescription : public JsepCodecDescription {
+ public:
+  JsepAudioCodecDescription(const std::string& defaultPt,
+                            const std::string& name, uint32_t clock,
+                            uint32_t channels, bool enabled = true)
+      : JsepCodecDescription(defaultPt, name, clock, channels, enabled),
+        mMaxPlaybackRate(0),
+        mForceMono(false),
+        mFECEnabled(false),
+        mDtmfEnabled(false),
+        mMaxAverageBitrate(0),
+        mDTXEnabled(false),
+        mFrameSizeMs(0),
+        mMinFrameSizeMs(0),
+        mMaxFrameSizeMs(0),
+        mCbrEnabled(false) {}
+
+  static constexpr SdpMediaSection::MediaType type = SdpMediaSection::kAudio;
+
+  SdpMediaSection::MediaType Type() const override { return type; }
+
+  JSEP_CODEC_CLONE(JsepAudioCodecDescription)
+
+  static UniquePtr<JsepAudioCodecDescription> CreateDefaultOpus() {
+    // Per jmspeex on IRC:
+    // For 32KHz sampling, 28 is ok, 32 is good, 40 should be really good
+    // quality.  Note that 1-2Kbps will be wasted on a stereo Opus channel
+    // with mono input compared to configuring it for mono.
+    // If we reduce bitrate enough Opus will low-pass us; 16000 will kill a
+    // 9KHz tone.  This should be adaptive when we're at the low-end of video
+    // bandwidth (say <100Kbps), and if we're audio-only, down to 8 or
+    // 12Kbps.
+    return MakeUnique<JsepAudioCodecDescription>("109", "opus", 48000, 2);
+  }
+
+  static UniquePtr<JsepAudioCodecDescription> CreateDefaultG722() {
+    return MakeUnique<JsepAudioCodecDescription>("9", "G722", 8000, 1);
+  }
+
+  static UniquePtr<JsepAudioCodecDescription> CreateDefaultPCMU() {
+    return MakeUnique<JsepAudioCodecDescription>("0", "PCMU", 8000, 1);
+  }
+
+  static UniquePtr<JsepAudioCodecDescription> CreateDefaultPCMA() {
+    return MakeUnique<JsepAudioCodecDescription>("8", "PCMA", 8000, 1);
+  }
+
+  static UniquePtr<JsepAudioCodecDescription> CreateDefaultTelephoneEvent() {
+    return MakeUnique<JsepAudioCodecDescription>("101", "telephone-event", 8000,
+                                                 1);
+  }
+
+  SdpFmtpAttributeList::OpusParameters GetOpusParameters(
+      const std::string& pt, const SdpMediaSection& msection) const {
+    // Will contain defaults if nothing else
+    SdpFmtpAttributeList::OpusParameters result;
+    auto* params = msection.FindFmtp(pt);
+
+    if (params && params->codec_type == SdpRtpmapAttributeList::kOpus) {
+      result =
+          static_cast<const SdpFmtpAttributeList::OpusParameters&>(*params);
+    }
+
+    return result;
+  }
+
+  SdpFmtpAttributeList::TelephoneEventParameters GetTelephoneEventParameters(
+      const std::string& pt, const SdpMediaSection& msection) const {
+    // Will contain defaults if nothing else
+    SdpFmtpAttributeList::TelephoneEventParameters result;
+    auto* params = msection.FindFmtp(pt);
+
+    if (params &&
+        params->codec_type == SdpRtpmapAttributeList::kTelephoneEvent) {
+      result =
+          static_cast<const SdpFmtpAttributeList::TelephoneEventParameters&>(
+              *params);
+    }
+
+    return result;
+  }
+
+  void AddParametersToMSection(SdpMediaSection& msection) const override {
+    if (mDirection == sdp::kSend) {
+      return;
+    }
+
+    if (mName == "opus") {
+      UniquePtr<SdpFmtpAttributeList::Parameters> opusParams =
+          MakeUnique<SdpFmtpAttributeList::OpusParameters>(
+              GetOpusParameters(mDefaultPt, msection));
+
+      ApplyConfigToFmtp(opusParams);
+
+      msection.SetFmtp(SdpFmtpAttributeList::Fmtp(mDefaultPt, *opusParams));
+    } else if (mName == "telephone-event") {
+      // add the default dtmf tones
+      SdpFmtpAttributeList::TelephoneEventParameters teParams(
+          GetTelephoneEventParameters(mDefaultPt, msection));
+      msection.SetFmtp(SdpFmtpAttributeList::Fmtp(mDefaultPt, teParams));
+    }
+  }
+
+  bool Negotiate(const std::string& pt, const SdpMediaSection& remoteMsection,
+                 bool remoteIsOffer,
+                 Maybe<const SdpMediaSection&> localMsection) override {
+    JsepCodecDescription::Negotiate(pt, remoteMsection, remoteIsOffer,
+                                    localMsection);
+    if (mName == "opus" && mDirection == sdp::kSend) {
+      SdpFmtpAttributeList::OpusParameters opusParams(
+          GetOpusParameters(mDefaultPt, remoteMsection));
+
+      mMaxPlaybackRate = opusParams.maxplaybackrate;
+      mForceMono = !opusParams.stereo;
+      // draft-ietf-rtcweb-fec-03.txt section 4.2 says support for FEC
+      // at the received side is declarative and can be negotiated
+      // separately for either media direction.
+      mFECEnabled = opusParams.useInBandFec;
+      if ((opusParams.maxAverageBitrate >= 6000) &&
+          (opusParams.maxAverageBitrate <= 510000)) {
+        mMaxAverageBitrate = opusParams.maxAverageBitrate;
+      }
+      mDTXEnabled = opusParams.useDTX;
+      if (remoteMsection.GetAttributeList().HasAttribute(
+              SdpAttribute::kPtimeAttribute)) {
+        mFrameSizeMs = remoteMsection.GetAttributeList().GetPtime();
+      } else {
+        mFrameSizeMs = opusParams.frameSizeMs;
+      }
+      mMinFrameSizeMs = opusParams.minFrameSizeMs;
+      if (remoteMsection.GetAttributeList().HasAttribute(
+              SdpAttribute::kMaxptimeAttribute)) {
+        mMaxFrameSizeMs = remoteMsection.GetAttributeList().GetMaxptime();
+      } else {
+        mMaxFrameSizeMs = opusParams.maxFrameSizeMs;
+      }
+      mCbrEnabled = opusParams.useCbr;
+    }
+
+    return true;
+  }
+
+  void ApplyConfigToFmtp(
+      UniquePtr<SdpFmtpAttributeList::Parameters>& aFmtp) const override {
+    if (mName == "opus") {
+      SdpFmtpAttributeList::OpusParameters opusParams;
+      if (aFmtp) {
+        MOZ_RELEASE_ASSERT(aFmtp->codec_type == SdpRtpmapAttributeList::kOpus);
+        opusParams =
+            static_cast<const SdpFmtpAttributeList::OpusParameters&>(*aFmtp);
+        opusParams.useInBandFec = mFECEnabled ? 1 : 0;
+      } else {
+        // If we weren't passed a fmtp to use then show we can do in band FEC
+        // for getCapabilities queries.
+        opusParams.useInBandFec = 1;
+      }
+      if (mMaxPlaybackRate) {
+        opusParams.maxplaybackrate = mMaxPlaybackRate;
+      }
+      opusParams.maxAverageBitrate = mMaxAverageBitrate;
+
+      if (mChannels == 2 &&
+          !Preferences::GetBool("media.peerconnection.sdp.disable_stereo_fmtp",
+                                false) &&
+          !mForceMono) {
+        // We prefer to receive stereo, if available.
+        opusParams.stereo = 1;
+      }
+      opusParams.useDTX = mDTXEnabled;
+      opusParams.frameSizeMs = mFrameSizeMs;
+      opusParams.minFrameSizeMs = mMinFrameSizeMs;
+      opusParams.maxFrameSizeMs = mMaxFrameSizeMs;
+      opusParams.useCbr = mCbrEnabled;
+      aFmtp.reset(opusParams.Clone());
+    }
+  };
+
+  uint32_t mMaxPlaybackRate;
+  bool mForceMono;
+  bool mFECEnabled;
+  bool mDtmfEnabled;
+  uint32_t mMaxAverageBitrate;
+  bool mDTXEnabled;
+  uint32_t mFrameSizeMs;
+  uint32_t mMinFrameSizeMs;
+  uint32_t mMaxFrameSizeMs;
+  bool mCbrEnabled;
+};
+
+class JsepVideoCodecDescription : public JsepCodecDescription {
+ public:
+  JsepVideoCodecDescription(const std::string& defaultPt,
+                            const std::string& name, uint32_t clock,
+                            bool enabled = true)
+      : JsepCodecDescription(defaultPt, name, clock, 0, enabled),
+        mTmmbrEnabled(false),
+        mRembEnabled(false),
+        mFECEnabled(false),
+        mTransportCCEnabled(false),
+        mRtxEnabled(false),
+        mProfileLevelId(0),
+        mPacketizationMode(0) {
+    // Add supported rtcp-fb types
+    mNackFbTypes.push_back("");
+    mNackFbTypes.push_back(SdpRtcpFbAttributeList::pli);
+    mCcmFbTypes.push_back(SdpRtcpFbAttributeList::fir);
+  }
+
+  static constexpr SdpMediaSection::MediaType type = SdpMediaSection::kVideo;
+
+  SdpMediaSection::MediaType Type() const override { return type; }
+
+  static UniquePtr<JsepVideoCodecDescription> CreateDefaultVP8(bool aUseRtx) {
+    auto codec = MakeUnique<JsepVideoCodecDescription>("120", "VP8", 90000);
+    // Defaults for mandatory params
+    codec->mConstraints.maxFs = 12288;  // Enough for 2048x1536
+    codec->mConstraints.maxFps = Some(60);
+    if (aUseRtx) {
+      codec->EnableRtx("124");
+    }
+    return codec;
+  }
+
+  static UniquePtr<JsepVideoCodecDescription> CreateDefaultVP9(bool aUseRtx) {
+    auto codec = MakeUnique<JsepVideoCodecDescription>("121", "VP9", 90000);
+    // Defaults for mandatory params
+    codec->mConstraints.maxFs = 12288;  // Enough for 2048x1536
+    codec->mConstraints.maxFps = Some(60);
+    if (aUseRtx) {
+      codec->EnableRtx("125");
+    }
+    return codec;
+  }
+
+  static UniquePtr<JsepVideoCodecDescription> CreateDefaultH264_0(
+      bool aUseRtx) {
+    auto codec = MakeUnique<JsepVideoCodecDescription>("97", "H264", 90000);
+    codec->mPacketizationMode = 0;
+    // Defaults for mandatory params
+    codec->mProfileLevelId = 0x42E00D;
+    if (aUseRtx) {
+      codec->EnableRtx("98");
+    }
+    return codec;
+  }
+
+  static UniquePtr<JsepVideoCodecDescription> CreateDefaultH264_1(
+      bool aUseRtx) {
+    auto codec = MakeUnique<JsepVideoCodecDescription>("126", "H264", 90000);
+    codec->mPacketizationMode = 1;
+    // Defaults for mandatory params
+    codec->mProfileLevelId = 0x42E00D;
+    if (aUseRtx) {
+      codec->EnableRtx("127");
+    }
+    return codec;
+  }
+
+  static UniquePtr<JsepVideoCodecDescription> CreateDefaultUlpFec() {
+    return MakeUnique<JsepVideoCodecDescription>(
+        "123",     // payload type
+        "ulpfec",  // codec name
+        90000      // clock rate (match other video codecs)
+    );
+  }
+
+  static UniquePtr<JsepVideoCodecDescription> CreateDefaultRed() {
+    return MakeUnique<JsepVideoCodecDescription>(
+        "122",  // payload type
+        "red",  // codec name
+        90000   // clock rate (match other video codecs)
+    );
+  }
+
+  void ApplyConfigToFmtp(
+      UniquePtr<SdpFmtpAttributeList::Parameters>& aFmtp) const override {
+    if (mName == "H264") {
+      SdpFmtpAttributeList::H264Parameters h264Params;
+      if (aFmtp) {
+        MOZ_RELEASE_ASSERT(aFmtp->codec_type == SdpRtpmapAttributeList::kH264);
+        h264Params =
+            static_cast<const SdpFmtpAttributeList::H264Parameters&>(*aFmtp);
+      }
+
+      if (mDirection == sdp::kSend) {
+        if (!h264Params.level_asymmetry_allowed) {
+          // First time the fmtp has been set; set just in case this is for a
+          // sendonly m-line, since even though we aren't receiving the level
+          // negotiation still needs to happen (sigh).
+          h264Params.profile_level_id = mProfileLevelId;
+        }
+      } else {
+        // Parameters that only apply to what we receive
+        h264Params.max_mbps = mConstraints.maxMbps;
+        h264Params.max_fs = mConstraints.maxFs;
+        h264Params.max_cpb = mConstraints.maxCpb;
+        h264Params.max_dpb = mConstraints.maxDpb;
+        h264Params.max_br = mConstraints.maxBr;
+        strncpy(h264Params.sprop_parameter_sets, mSpropParameterSets.c_str(),
+                sizeof(h264Params.sprop_parameter_sets) - 1);
+        h264Params.profile_level_id = mProfileLevelId;
+      }
+
+      // Parameters that apply to both the send and recv directions
+      h264Params.packetization_mode = mPacketizationMode;
+      // Hard-coded, may need to change someday?
+      h264Params.level_asymmetry_allowed = true;
+
+      // Parameters that apply to both the send and recv directions
+      h264Params.packetization_mode = mPacketizationMode;
+      // Hard-coded, may need to change someday?
+      h264Params.level_asymmetry_allowed = true;
+      aFmtp.reset(h264Params.Clone());
+    } else if (mName == "VP8" || mName == "VP9") {
+      SdpRtpmapAttributeList::CodecType type =
+          mName == "VP8" ? SdpRtpmapAttributeList::CodecType::kVP8
+                         : SdpRtpmapAttributeList::CodecType::kVP9;
+      auto vp8Params = SdpFmtpAttributeList::VP8Parameters(type);
+
+      if (aFmtp) {
+        MOZ_RELEASE_ASSERT(aFmtp->codec_type == type);
+        vp8Params =
+            static_cast<const SdpFmtpAttributeList::VP8Parameters&>(*aFmtp);
+      }
+      // VP8 and VP9 share the same SDP parameters thus far
+      vp8Params.max_fs = mConstraints.maxFs;
+      if (mConstraints.maxFps.isSome()) {
+        vp8Params.max_fr =
+            static_cast<unsigned int>(std::round(*mConstraints.maxFps));
+      } else {
+        vp8Params.max_fr = 60;
+      }
+      aFmtp.reset(vp8Params.Clone());
+    }
+  }
+
+  virtual void EnableTmmbr() {
+    // EnableTmmbr can be called multiple times due to multiple calls to
+    // PeerConnectionImpl::ConfigureJsepSessionCodecs
+    if (!mTmmbrEnabled) {
+      mTmmbrEnabled = true;
+      mCcmFbTypes.push_back(SdpRtcpFbAttributeList::tmmbr);
+    }
+  }
+
+  virtual void EnableRemb() {
+    // EnableRemb can be called multiple times due to multiple calls to
+    // PeerConnectionImpl::ConfigureJsepSessionCodecs
+    if (!mRembEnabled) {
+      mRembEnabled = true;
+      mOtherFbTypes.push_back({"", SdpRtcpFbAttributeList::kRemb, "", ""});
+    }
+  }
+
+  virtual void EnableFec(std::string redPayloadType,
+                         std::string ulpfecPayloadType) {
+    // Enabling FEC for video works a little differently than enabling
+    // REMB or TMMBR.  Support for FEC is indicated by the presence of
+    // particular codes (red and ulpfec) instead of using rtcpfb
+    // attributes on a given codec.  There is no rtcpfb to push for FEC
+    // as can be seen above when REMB or TMMBR are enabled.
+
+    // Ensure we have valid payload types. This returns zero on failure, which
+    // is a valid payload type.
+    uint16_t redPt, ulpfecPt;
+    if (!SdpHelper::GetPtAsInt(redPayloadType, &redPt) ||
+        !SdpHelper::GetPtAsInt(ulpfecPayloadType, &ulpfecPt)) {
+      return;
+    }
+
+    mFECEnabled = true;
+    mREDPayloadType = redPayloadType;
+    mULPFECPayloadType = ulpfecPayloadType;
+  }
+
+  virtual void EnableTransportCC() {
+    if (!mTransportCCEnabled) {
+      mTransportCCEnabled = true;
+      mOtherFbTypes.push_back(
+          {"", SdpRtcpFbAttributeList::kTransportCC, "", ""});
+    }
+  }
+
+  void EnableRtx(const std::string& rtxPayloadType) {
+    mRtxEnabled = true;
+    mRtxPayloadType = rtxPayloadType;
+  }
+
+  void AddParametersToMSection(SdpMediaSection& msection) const override {
+    AddFmtpsToMSection(msection);
+    AddRtcpFbsToMSection(msection);
+  }
+
+  void AddFmtpsToMSection(SdpMediaSection& msection) const {
+    if (mName == "H264") {
+      UniquePtr<SdpFmtpAttributeList::Parameters> h264Params =
+          MakeUnique<SdpFmtpAttributeList::H264Parameters>(
+              GetH264Parameters(mDefaultPt, msection));
+
+      ApplyConfigToFmtp(h264Params);
+
+      msection.SetFmtp(SdpFmtpAttributeList::Fmtp(mDefaultPt, *h264Params));
+    } else if (mName == "red" && !mRedundantEncodings.empty()) {
+      SdpFmtpAttributeList::RedParameters redParams(
+          GetRedParameters(mDefaultPt, msection));
+      redParams.encodings = mRedundantEncodings;
+      msection.SetFmtp(SdpFmtpAttributeList::Fmtp(mDefaultPt, redParams));
+    } else if (mName == "VP8" || mName == "VP9") {
+      if (mDirection == sdp::kRecv) {
+        // VP8 and VP9 share the same SDP parameters thus far
+        UniquePtr<SdpFmtpAttributeList::Parameters> vp8Params =
+            MakeUnique<SdpFmtpAttributeList::VP8Parameters>(
+                GetVP8Parameters(mDefaultPt, msection));
+        ApplyConfigToFmtp(vp8Params);
+        msection.SetFmtp(SdpFmtpAttributeList::Fmtp(mDefaultPt, *vp8Params));
+      }
+    }
+
+    if (mRtxEnabled && mDirection == sdp::kRecv) {
+      SdpFmtpAttributeList::RtxParameters params(
+          GetRtxParameters(mDefaultPt, msection));
+      uint16_t apt;
+      if (SdpHelper::GetPtAsInt(mDefaultPt, &apt)) {
+        if (apt <= 127) {
+          msection.AddCodec(mRtxPayloadType, "rtx", mClock, mChannels);
+
+          params.apt = apt;
+          msection.SetFmtp(SdpFmtpAttributeList::Fmtp(mRtxPayloadType, params));
+        }
+      }
+    }
+  }
+
+  void AddRtcpFbsToMSection(SdpMediaSection& msection) const {
+    SdpRtcpFbAttributeList rtcpfbs(msection.GetRtcpFbs());
+    for (const auto& rtcpfb : rtcpfbs.mFeedbacks) {
+      if (rtcpfb.pt == mDefaultPt) {
+        // Already set by the codec for the other direction.
+        return;
+      }
+    }
+
+    for (const std::string& type : mAckFbTypes) {
+      rtcpfbs.PushEntry(mDefaultPt, SdpRtcpFbAttributeList::kAck, type);
+    }
+    for (const std::string& type : mNackFbTypes) {
+      rtcpfbs.PushEntry(mDefaultPt, SdpRtcpFbAttributeList::kNack, type);
+    }
+    for (const std::string& type : mCcmFbTypes) {
+      rtcpfbs.PushEntry(mDefaultPt, SdpRtcpFbAttributeList::kCcm, type);
+    }
+    for (const auto& fb : mOtherFbTypes) {
+      rtcpfbs.PushEntry(mDefaultPt, fb.type, fb.parameter, fb.extra);
+    }
+
+    msection.SetRtcpFbs(rtcpfbs);
+  }
+
+  SdpFmtpAttributeList::H264Parameters GetH264Parameters(
+      const std::string& pt, const SdpMediaSection& msection) const {
+    // Will contain defaults if nothing else
+    SdpFmtpAttributeList::H264Parameters result;
+    auto* params = msection.FindFmtp(pt);
+
+    if (params && params->codec_type == SdpRtpmapAttributeList::kH264) {
+      result =
+          static_cast<const SdpFmtpAttributeList::H264Parameters&>(*params);
+    }
+
+    return result;
+  }
+
+  SdpFmtpAttributeList::RedParameters GetRedParameters(
+      const std::string& pt, const SdpMediaSection& msection) const {
+    SdpFmtpAttributeList::RedParameters result;
+    auto* params = msection.FindFmtp(pt);
+
+    if (params && params->codec_type == SdpRtpmapAttributeList::kRed) {
+      result = static_cast<const SdpFmtpAttributeList::RedParameters&>(*params);
+    }
+
+    return result;
+  }
+
+  SdpFmtpAttributeList::RtxParameters GetRtxParameters(
+      const std::string& pt, const SdpMediaSection& msection) const {
+    SdpFmtpAttributeList::RtxParameters result;
+    const auto* params = msection.FindFmtp(pt);
+
+    if (params && params->codec_type == SdpRtpmapAttributeList::kRtx) {
+      result = static_cast<const SdpFmtpAttributeList::RtxParameters&>(*params);
+    }
+
+    return result;
+  }
+
+  Maybe<std::string> GetRtxPtByApt(const std::string& apt,
+                                   const SdpMediaSection& msection) const {
+    Maybe<std::string> result;
+    uint16_t aptAsInt;
+    if (!SdpHelper::GetPtAsInt(apt, &aptAsInt)) {
+      return result;
+    }
+
+    const SdpAttributeList& attrs = msection.GetAttributeList();
+    if (attrs.HasAttribute(SdpAttribute::kFmtpAttribute)) {
+      for (const auto& fmtpAttr : attrs.GetFmtp().mFmtps) {
+        if (fmtpAttr.parameters) {
+          auto* params = fmtpAttr.parameters.get();
+          if (params && params->codec_type == SdpRtpmapAttributeList::kRtx) {
+            const SdpFmtpAttributeList::RtxParameters* rtxParams =
+                static_cast<const SdpFmtpAttributeList::RtxParameters*>(params);
+            if (rtxParams->apt == aptAsInt) {
+              result = Some(fmtpAttr.format);
+              break;
+            }
+          }
+        }
+      }
+    }
+    return result;
+  }
+
+  SdpFmtpAttributeList::VP8Parameters GetVP8Parameters(
+      const std::string& pt, const SdpMediaSection& msection) const {
+    SdpRtpmapAttributeList::CodecType expectedType(
+        mName == "VP8" ? SdpRtpmapAttributeList::kVP8
+                       : SdpRtpmapAttributeList::kVP9);
+
+    // Will contain defaults if nothing else
+    SdpFmtpAttributeList::VP8Parameters result(expectedType);
+    auto* params = msection.FindFmtp(pt);
+
+    if (params && params->codec_type == expectedType) {
+      result = static_cast<const SdpFmtpAttributeList::VP8Parameters&>(*params);
+    }
+
+    return result;
+  }
+
+  void NegotiateRtcpFb(const SdpMediaSection& remoteMsection,
+                       SdpRtcpFbAttributeList::Type type,
+                       std::vector<std::string>* supportedTypes) {
+    Maybe<std::string> remoteFmt = GetMatchingFormat(remoteMsection);
+    if (!remoteFmt) {
+      return;
+    }
+    std::vector<std::string> temp;
+    for (auto& subType : *supportedTypes) {
+      if (remoteMsection.HasRtcpFb(*remoteFmt, type, subType)) {
+        temp.push_back(subType);
+      }
+    }
+    *supportedTypes = temp;
+  }
+
+  void NegotiateRtcpFb(
+      const SdpMediaSection& remoteMsection,
+      std::vector<SdpRtcpFbAttributeList::Feedback>* supportedFbs) {
+    Maybe<std::string> remoteFmt = GetMatchingFormat(remoteMsection);
+    if (!remoteFmt) {
+      return;
+    }
+    std::vector<SdpRtcpFbAttributeList::Feedback> temp;
+    for (auto& fb : *supportedFbs) {
+      if (remoteMsection.HasRtcpFb(*remoteFmt, fb.type, fb.parameter)) {
+        temp.push_back(fb);
+      }
+    }
+    *supportedFbs = temp;
+  }
+
+  void NegotiateRtcpFb(const SdpMediaSection& remote) {
+    // Removes rtcp-fb types that the other side doesn't support
+    NegotiateRtcpFb(remote, SdpRtcpFbAttributeList::kAck, &mAckFbTypes);
+    NegotiateRtcpFb(remote, SdpRtcpFbAttributeList::kNack, &mNackFbTypes);
+    NegotiateRtcpFb(remote, SdpRtcpFbAttributeList::kCcm, &mCcmFbTypes);
+    NegotiateRtcpFb(remote, &mOtherFbTypes);
+  }
+
+  virtual bool Negotiate(const std::string& pt,
+                         const SdpMediaSection& remoteMsection,
+                         bool remoteIsOffer,
+                         Maybe<const SdpMediaSection&> localMsection) override {
+    JsepCodecDescription::Negotiate(pt, remoteMsection, remoteIsOffer,
+                                    localMsection);
+    if (mName == "H264") {
+      SdpFmtpAttributeList::H264Parameters h264Params(
+          GetH264Parameters(mDefaultPt, remoteMsection));
+
+      // Level is negotiated symmetrically if level asymmetry is disallowed
+      if (!h264Params.level_asymmetry_allowed) {
+        SetSaneH264Level(std::min(GetSaneH264Level(h264Params.profile_level_id),
+                                  GetSaneH264Level(mProfileLevelId)),
+                         &mProfileLevelId);
+      }
+
+      if (mDirection == sdp::kSend) {
+        // Remote values of these apply only to the send codec.
+        mConstraints.maxFs = h264Params.max_fs;
+        mConstraints.maxMbps = h264Params.max_mbps;
+        mConstraints.maxCpb = h264Params.max_cpb;
+        mConstraints.maxDpb = h264Params.max_dpb;
+        mConstraints.maxBr = h264Params.max_br;
+        mSpropParameterSets = h264Params.sprop_parameter_sets;
+        // Only do this if we didn't symmetrically negotiate above
+        if (h264Params.level_asymmetry_allowed) {
+          SetSaneH264Level(GetSaneH264Level(h264Params.profile_level_id),
+                           &mProfileLevelId);
+        }
+      } else {
+        // TODO(bug 1143709): max-recv-level support
+      }
+    } else if (mName == "red") {
+      SdpFmtpAttributeList::RedParameters redParams(
+          GetRedParameters(mDefaultPt, remoteMsection));
+      mRedundantEncodings = redParams.encodings;
+    } else if (mName == "VP8" || mName == "VP9") {
+      if (mDirection == sdp::kSend) {
+        SdpFmtpAttributeList::VP8Parameters vp8Params(
+            GetVP8Parameters(mDefaultPt, remoteMsection));
+
+        mConstraints.maxFs = vp8Params.max_fs;
+        // Right now, we treat max-fr=0 (or the absence of max-fr) as no limit.
+        // We will eventually want to stop doing this (bug 1762600).
+        if (vp8Params.max_fr) {
+          mConstraints.maxFps = Some(vp8Params.max_fr);
+        }
+      }
+    }
+
+    if (mRtxEnabled && (mDirection == sdp::kSend || remoteIsOffer)) {
+      Maybe<std::string> rtxPt = GetRtxPtByApt(mDefaultPt, remoteMsection);
+      if (rtxPt.isSome()) {
+        EnableRtx(*rtxPt);
+      } else {
+        mRtxEnabled = false;
+        mRtxPayloadType = "";
+      }
+    }
+
+    NegotiateRtcpFb(remoteMsection);
+    return true;
+  }
+
+  // Maps the not-so-sane encoding of H264 level into something that is
+  // ordered in the way one would expect
+  // 1b is 0xAB, everything else is the level left-shifted one half-byte
+  // (eg; 1.0 is 0xA0, 1.1 is 0xB0, 3.1 is 0x1F0)
+  static uint32_t GetSaneH264Level(uint32_t profileLevelId) {
+    uint32_t profileIdc = (profileLevelId >> 16);
+
+    if (profileIdc == 0x42 || profileIdc == 0x4D || profileIdc == 0x58) {
+      if ((profileLevelId & 0x10FF) == 0x100B) {
+        // Level 1b
+        return 0xAB;
+      }
+    }
+
+    uint32_t level = profileLevelId & 0xFF;
+
+    if (level == 0x09) {
+      // Another way to encode level 1b
+      return 0xAB;
+    }
+
+    return level << 4;
+  }
+
+  static void SetSaneH264Level(uint32_t level, uint32_t* profileLevelId) {
+    uint32_t profileIdc = (*profileLevelId >> 16);
+    uint32_t levelMask = 0xFF;
+
+    if (profileIdc == 0x42 || profileIdc == 0x4d || profileIdc == 0x58) {
+      levelMask = 0x10FF;
+      if (level == 0xAB) {
+        // Level 1b
+        level = 0x100B;
+      } else {
+        // Not 1b, just shift
+        level = level >> 4;
+      }
+    } else if (level == 0xAB) {
+      // Another way to encode 1b
+      level = 0x09;
+    } else {
+      // Not 1b, just shift
+      level = level >> 4;
+    }
+
+    *profileLevelId = (*profileLevelId & ~levelMask) | level;
+  }
+
+  enum Subprofile {
+    kH264ConstrainedBaseline,
+    kH264Baseline,
+    kH264Main,
+    kH264Extended,
+    kH264High,
+    kH264High10,
+    kH264High42,
+    kH264High44,
+    kH264High10I,
+    kH264High42I,
+    kH264High44I,
+    kH264CALVC44,
+    kH264UnknownSubprofile
+  };
+
+  static Subprofile GetSubprofile(uint32_t profileLevelId) {
+    // Based on Table 5 from RFC 6184:
+    //        Profile     profile_idc        profile-iop
+    //                    (hexadecimal)      (binary)
+
+    //        CB          42 (B)             x1xx0000
+    //           same as: 4D (M)             1xxx0000
+    //           same as: 58 (E)             11xx0000
+    //        B           42 (B)             x0xx0000
+    //           same as: 58 (E)             10xx0000
+    //        M           4D (M)             0x0x0000
+    //        E           58                 00xx0000
+    //        H           64                 00000000
+    //        H10         6E                 00000000
+    //        H42         7A                 00000000
+    //        H44         F4                 00000000
+    //        H10I        6E                 00010000
+    //        H42I        7A                 00010000
+    //        H44I        F4                 00010000
+    //        C44I        2C                 00010000
+
+    if ((profileLevelId & 0xFF4F00) == 0x424000) {
+      // 01001111 (mask, 0x4F)
+      // x1xx0000 (from table)
+      // 01000000 (expected value, 0x40)
+      return kH264ConstrainedBaseline;
+    }
+
+    if ((profileLevelId & 0xFF8F00) == 0x4D8000) {
+      // 10001111 (mask, 0x8F)
+      // 1xxx0000 (from table)
+      // 10000000 (expected value, 0x80)
+      return kH264ConstrainedBaseline;
+    }
+
+    if ((profileLevelId & 0xFFCF00) == 0x58C000) {
+      // 11001111 (mask, 0xCF)
+      // 11xx0000 (from table)
+      // 11000000 (expected value, 0xC0)
+      return kH264ConstrainedBaseline;
+    }
+
+    if ((profileLevelId & 0xFF4F00) == 0x420000) {
+      // 01001111 (mask, 0x4F)
+      // x0xx0000 (from table)
+      // 00000000 (expected value)
+      return kH264Baseline;
+    }
+
+    if ((profileLevelId & 0xFFCF00) == 0x588000) {
+      // 11001111 (mask, 0xCF)
+      // 10xx0000 (from table)
+      // 10000000 (expected value, 0x80)
+      return kH264Baseline;
+    }
+
+    if ((profileLevelId & 0xFFAF00) == 0x4D0000) {
+      // 10101111 (mask, 0xAF)
+      // 0x0x0000 (from table)
+      // 00000000 (expected value)
+      return kH264Main;
+    }
+
+    if ((profileLevelId & 0xFF0000) == 0x580000) {
+      // 11001111 (mask, 0xCF)
+      // 00xx0000 (from table)
+      // 00000000 (expected value)
+      return kH264Extended;
+    }
+
+    if ((profileLevelId & 0xFFFF00) == 0x640000) {
+      return kH264High;
+    }
+
+    if ((profileLevelId & 0xFFFF00) == 0x6E0000) {
+      return kH264High10;
+    }
+
+    if ((profileLevelId & 0xFFFF00) == 0x7A0000) {
+      return kH264High42;
+    }
+
+    if ((profileLevelId & 0xFFFF00) == 0xF40000) {
+      return kH264High44;
+    }
+
+    if ((profileLevelId & 0xFFFF00) == 0x6E1000) {
+      return kH264High10I;
+    }
+
+    if ((profileLevelId & 0xFFFF00) == 0x7A1000) {
+      return kH264High42I;
+    }
+
+    if ((profileLevelId & 0xFFFF00) == 0xF41000) {
+      return kH264High44I;
+    }
+
+    if ((profileLevelId & 0xFFFF00) == 0x2C1000) {
+      return kH264CALVC44;
+    }
+
+    return kH264UnknownSubprofile;
+  }
+
+  virtual bool ParametersMatch(
+      const std::string& fmt,
+      const SdpMediaSection& remoteMsection) const override {
+    if (mName == "H264") {
+      SdpFmtpAttributeList::H264Parameters h264Params(
+          GetH264Parameters(fmt, remoteMsection));
+
+      if (h264Params.packetization_mode != mPacketizationMode) {
+        return false;
+      }
+
+      if (GetSubprofile(h264Params.profile_level_id) !=
+          GetSubprofile(mProfileLevelId)) {
+        return false;
+      }
+    }
+
+    return true;
+  }
+
+  virtual bool RtcpFbRembIsSet() const {
+    for (const auto& fb : mOtherFbTypes) {
+      if (fb.type == SdpRtcpFbAttributeList::kRemb) {
+        return true;
+      }
+    }
+    return false;
+  }
+
+  virtual bool RtcpFbTransportCCIsSet() const {
+    for (const auto& fb : mOtherFbTypes) {
+      if (fb.type == SdpRtcpFbAttributeList::kTransportCC) {
+        return true;
+      }
+    }
+    return false;
+  }
+
+  virtual void UpdateRedundantEncodings(
+      const std::vector<UniquePtr<JsepCodecDescription>>& codecs) {
+    for (const auto& codec : codecs) {
+      if (codec->Type() == type && codec->mEnabled && codec->mName != "red") {
+        uint16_t pt;
+        if (!SdpHelper::GetPtAsInt(codec->mDefaultPt, &pt)) {
+          continue;
+        }
+        mRedundantEncodings.push_back(pt);
+      }
+    }
+  }
+
+  void EnsureNoDuplicatePayloadTypes(std::set<std::string>& aUsedPts) override {
+    JsepCodecDescription::EnsureNoDuplicatePayloadTypes(aUsedPts);
+    if (mFECEnabled) {
+      mFECEnabled = EnsurePayloadTypeNotDuplicate(aUsedPts, mREDPayloadType) &&
+                    EnsurePayloadTypeNotDuplicate(aUsedPts, mULPFECPayloadType);
+    }
+    if (mRtxEnabled) {
+      mRtxEnabled = EnsurePayloadTypeNotDuplicate(aUsedPts, mRtxPayloadType);
+    }
+  }
+
+  JSEP_CODEC_CLONE(JsepVideoCodecDescription)
+
+  std::vector<std::string> mAckFbTypes;
+  std::vector<std::string> mNackFbTypes;
+  std::vector<std::string> mCcmFbTypes;
+  std::vector<SdpRtcpFbAttributeList::Feedback> mOtherFbTypes;
+  bool mTmmbrEnabled;
+  bool mRembEnabled;
+  bool mFECEnabled;
+  bool mTransportCCEnabled;
+  bool mRtxEnabled;
+  std::string mREDPayloadType;
+  std::string mULPFECPayloadType;
+  std::string mRtxPayloadType;
+  std::vector<uint8_t> mRedundantEncodings;
+
+  // H264-specific stuff
+  uint32_t mProfileLevelId;
+  uint32_t mPacketizationMode;
+  std::string mSpropParameterSets;
+};
+
+class JsepApplicationCodecDescription : public JsepCodecDescription {
+  // This is the new draft-21 implementation
+ public:
+  JsepApplicationCodecDescription(const std::string& name, uint16_t channels,
+                                  uint16_t localPort,
+                                  uint32_t localMaxMessageSize,
+                                  bool enabled = true)
+      : JsepCodecDescription("", name, 0, channels, enabled),
+        mLocalPort(localPort),
+        mLocalMaxMessageSize(localMaxMessageSize),
+        mRemotePort(0),
+        mRemoteMaxMessageSize(0),
+        mRemoteMMSSet(false) {}
+
+  static constexpr SdpMediaSection::MediaType type =
+      SdpMediaSection::kApplication;
+
+  SdpMediaSection::MediaType Type() const override { return type; }
+
+  JSEP_CODEC_CLONE(JsepApplicationCodecDescription)
+
+  static UniquePtr<JsepApplicationCodecDescription> CreateDefault() {
+    return MakeUnique<JsepApplicationCodecDescription>(
+        "webrtc-datachannel", WEBRTC_DATACHANNEL_STREAMS_DEFAULT,
+        WEBRTC_DATACHANNEL_PORT_DEFAULT,
+        WEBRTC_DATACHANNEL_MAX_MESSAGE_SIZE_LOCAL);
+  }
+
+  // Override, uses sctpport or sctpmap instead of rtpmap
+  virtual bool Matches(const std::string& fmt,
+                       const SdpMediaSection& remoteMsection) const override {
+    if (type != remoteMsection.GetMediaType()) {
+      return false;
+    }
+
+    int sctp_port = remoteMsection.GetSctpPort();
+    bool fmt_matches =
+        nsCRT::strcasecmp(mName.c_str(),
+                          remoteMsection.GetFormats()[0].c_str()) == 0;
+    if (sctp_port && fmt_matches) {
+      // New sctp draft 21 format
+      return true;
+    }
+
+    const SdpSctpmapAttributeList::Sctpmap* sctp_map(
+        remoteMsection.GetSctpmap());
+    if (sctp_map) {
+      // Old sctp draft 05 format
+      return nsCRT::strcasecmp(mName.c_str(), sctp_map->name.c_str()) == 0;
+    }
+
+    return false;
+  }
+
+  virtual void AddToMediaSection(SdpMediaSection& msection) const override {
+    if (mEnabled && msection.GetMediaType() == type) {
+      if (mDirection == sdp::kRecv) {
+        msection.AddDataChannel(mName, mLocalPort, mChannels,
+                                mLocalMaxMessageSize);
+      }
+
+      AddParametersToMSection(msection);
+    }
+  }
+
+  bool Negotiate(const std::string& pt, const SdpMediaSection& remoteMsection,
+                 bool remoteIsOffer,
+                 Maybe<const SdpMediaSection&> localMsection) override {
+    JsepCodecDescription::Negotiate(pt, remoteMsection, remoteIsOffer,
+                                    localMsection);
+
+    uint32_t message_size;
+    mRemoteMMSSet = remoteMsection.GetMaxMessageSize(&message_size);
+    if (mRemoteMMSSet) {
+      mRemoteMaxMessageSize = message_size;
+    } else {
+      mRemoteMaxMessageSize =
+          WEBRTC_DATACHANNEL_MAX_MESSAGE_SIZE_REMOTE_DEFAULT;
+    }
+
+    int sctp_port = remoteMsection.GetSctpPort();
+    if (sctp_port) {
+      mRemotePort = sctp_port;
+      return true;
+    }
+
+    const SdpSctpmapAttributeList::Sctpmap* sctp_map(
+        remoteMsection.GetSctpmap());
+    if (sctp_map) {
+      mRemotePort = std::stoi(sctp_map->pt);
+      return true;
+    }
+
+    return false;
+  }
+
+  // We only support one datachannel per m-section
+  void EnsureNoDuplicatePayloadTypes(std::set<std::string>& aUsedPts) override {
+  }
+
+  void ApplyConfigToFmtp(
+      UniquePtr<SdpFmtpAttributeList::Parameters>& aFmtp) const override{};
+
+  uint16_t mLocalPort;
+  uint32_t mLocalMaxMessageSize;
+  uint16_t mRemotePort;
+  uint32_t mRemoteMaxMessageSize;
+  bool mRemoteMMSSet;
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/jsep/JsepSession.h b/dom/media/webrtc/jsep/JsepSession.h
new file mode 100644
index 0000000000..bf68da900b
--- /dev/null
+++ b/dom/media/webrtc/jsep/JsepSession.h
@@ -0,0 +1,287 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _JSEPSESSION_H_
+#define _JSEPSESSION_H_
+
+#include <map>
+#include <string>
+#include <vector>
+#include "mozilla/Attributes.h"
+#include "mozilla/Maybe.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/UniquePtr.h"
+#include "nsError.h"
+
+#include "jsep/JsepTransport.h"
+#include "sdp/Sdp.h"
+
+#include "jsep/JsepTransceiver.h"
+
+#include "mozilla/dom/PeerConnectionObserverEnumsBinding.h"
+
+namespace mozilla {
+
+// Forward declarations
+class JsepCodecDescription;
+
+enum JsepSignalingState {
+  kJsepStateStable,
+  kJsepStateHaveLocalOffer,
+  kJsepStateHaveRemoteOffer,
+  kJsepStateHaveLocalPranswer,
+  kJsepStateHaveRemotePranswer,
+  kJsepStateClosed
+};
+
+enum JsepSdpType {
+  kJsepSdpOffer,
+  kJsepSdpAnswer,
+  kJsepSdpPranswer,
+  kJsepSdpRollback
+};
+
+enum JsepDescriptionPendingOrCurrent {
+  kJsepDescriptionCurrent,
+  kJsepDescriptionPending,
+  kJsepDescriptionPendingOrCurrent
+};
+
+struct JsepOAOptions {};
+struct JsepOfferOptions : public JsepOAOptions {
+  Maybe<size_t> mOfferToReceiveAudio;
+  Maybe<size_t> mOfferToReceiveVideo;
+  Maybe<bool> mIceRestart;  // currently ignored by JsepSession
+};
+struct JsepAnswerOptions : public JsepOAOptions {};
+
+enum JsepBundlePolicy { kBundleBalanced, kBundleMaxCompat, kBundleMaxBundle };
+
+enum JsepMediaType { kNone = 0, kAudio, kVideo, kAudioVideo };
+
+struct JsepExtmapMediaType {
+  JsepMediaType mMediaType;
+  SdpExtmapAttributeList::Extmap mExtmap;
+};
+
+class JsepSession {
+ public:
+  explicit JsepSession(const std::string& name)
+      : mName(name), mState(kJsepStateStable), mNegotiations(0) {}
+  virtual ~JsepSession() {}
+
+  virtual JsepSession* Clone() const = 0;
+
+  virtual nsresult Init() = 0;
+
+  // Accessors for basic properties.
+  virtual const std::string& GetName() const { return mName; }
+  virtual JsepSignalingState GetState() const { return mState; }
+  virtual uint32_t GetNegotiations() const { return mNegotiations; }
+
+  // Set up the ICE And DTLS data.
+  virtual nsresult SetBundlePolicy(JsepBundlePolicy policy) = 0;
+  virtual bool RemoteIsIceLite() const = 0;
+  virtual std::vector<std::string> GetIceOptions() const = 0;
+
+  virtual nsresult AddDtlsFingerprint(const std::string& algorithm,
+                                      const std::vector<uint8_t>& value) = 0;
+
+  virtual nsresult AddRtpExtension(
+      JsepMediaType mediaType, const std::string& extensionName,
+      SdpDirectionAttribute::Direction direction) = 0;
+  virtual nsresult AddAudioRtpExtension(
+      const std::string& extensionName,
+      SdpDirectionAttribute::Direction direction) = 0;
+  virtual nsresult AddVideoRtpExtension(
+      const std::string& extensionName,
+      SdpDirectionAttribute::Direction direction) = 0;
+  virtual nsresult AddAudioVideoRtpExtension(
+      const std::string& extensionName,
+      SdpDirectionAttribute::Direction direction) = 0;
+
+  // Kinda gross to be locking down the data structure type like this, but
+  // returning by value is problematic due to the lack of stl move semantics in
+  // our build config, since we can't use UniquePtr in the container. The
+  // alternative is writing a raft of accessor functions that allow arbitrary
+  // manipulation (which will be unwieldy), or allowing functors to be injected
+  // that manipulate the data structure (still pretty unwieldy).
+  virtual std::vector<UniquePtr<JsepCodecDescription>>& Codecs() = 0;
+
+  template <class UnaryFunction>
+  void ForEachCodec(UnaryFunction& function) {
+    std::for_each(Codecs().begin(), Codecs().end(), function);
+    for (auto& transceiver : GetTransceivers()) {
+      transceiver.mSendTrack.ForEachCodec(function);
+      transceiver.mRecvTrack.ForEachCodec(function);
+    }
+  }
+
+  template <class BinaryPredicate>
+  void SortCodecs(BinaryPredicate& sorter) {
+    std::stable_sort(Codecs().begin(), Codecs().end(), sorter);
+    for (auto& transceiver : GetTransceivers()) {
+      transceiver.mSendTrack.SortCodecs(sorter);
+      transceiver.mRecvTrack.SortCodecs(sorter);
+    }
+  }
+
+  // Would be nice to have this return a Maybe containing the return of
+  // |aFunction|, but Maybe cannot contain a void.
+  template <typename UnaryFunction>
+  bool ApplyToTransceiver(const std::string& aId, UnaryFunction&& aFunction) {
+    for (auto& transceiver : GetTransceivers()) {
+      if (transceiver.GetUuid() == aId) {
+        std::forward<UnaryFunction>(aFunction)(transceiver);
+        return true;
+      }
+    }
+    return false;
+  }
+
+  template <typename UnaryFunction>
+  void ForEachTransceiver(UnaryFunction&& aFunction) {
+    for (auto& transceiver : GetTransceivers()) {
+      std::forward<UnaryFunction>(aFunction)(transceiver);
+    }
+  }
+
+  template <typename UnaryFunction>
+  void ForEachTransceiver(UnaryFunction&& aFunction) const {
+    for (const auto& transceiver : GetTransceivers()) {
+      std::forward<UnaryFunction>(aFunction)(transceiver);
+    }
+  }
+
+  Maybe<const JsepTransceiver> GetTransceiver(const std::string& aId) const {
+    for (const auto& transceiver : GetTransceivers()) {
+      if (transceiver.GetUuid() == aId) {
+        return Some(transceiver);
+      }
+    }
+    return Nothing();
+  }
+
+  template <typename MatchFunction>
+  Maybe<const JsepTransceiver> FindTransceiver(MatchFunction&& aFunc) const {
+    for (const auto& transceiver : GetTransceivers()) {
+      if (std::forward<MatchFunction>(aFunc)(transceiver)) {
+        return Some(transceiver);
+      }
+    }
+    return Nothing();
+  }
+
+  bool SetTransceiver(const JsepTransceiver& aNew) {
+    return ApplyToTransceiver(aNew.GetUuid(),
+                              [aNew](JsepTransceiver& aOld) { aOld = aNew; });
+  }
+
+  virtual void AddTransceiver(const JsepTransceiver& transceiver) = 0;
+
+  class Result {
+   public:
+    Result() = default;
+    MOZ_IMPLICIT Result(dom::PCError aError) : mError(Some(aError)) {}
+    // TODO(bug 1527916): Need c'tor and members for handling RTCError.
+    Maybe<dom::PCError> mError;
+  };
+
+  // Basic JSEP operations.
+  virtual Result CreateOffer(const JsepOfferOptions& options,
+                             std::string* offer) = 0;
+  virtual Result CreateAnswer(const JsepAnswerOptions& options,
+                              std::string* answer) = 0;
+  virtual std::string GetLocalDescription(
+      JsepDescriptionPendingOrCurrent type) const = 0;
+  virtual std::string GetRemoteDescription(
+      JsepDescriptionPendingOrCurrent type) const = 0;
+  virtual Result SetLocalDescription(JsepSdpType type,
+                                     const std::string& sdp) = 0;
+  virtual Result SetRemoteDescription(JsepSdpType type,
+                                      const std::string& sdp) = 0;
+  virtual Result AddRemoteIceCandidate(const std::string& candidate,
+                                       const std::string& mid,
+                                       const Maybe<uint16_t>& level,
+                                       const std::string& ufrag,
+                                       std::string* transportId) = 0;
+  virtual nsresult AddLocalIceCandidate(const std::string& candidate,
+                                        const std::string& transportId,
+                                        const std::string& ufrag,
+                                        uint16_t* level, std::string* mid,
+                                        bool* skipped) = 0;
+  virtual nsresult UpdateDefaultCandidate(
+      const std::string& defaultCandidateAddr, uint16_t defaultCandidatePort,
+      const std::string& defaultRtcpCandidateAddr,
+      uint16_t defaultRtcpCandidatePort, const std::string& transportId) = 0;
+  virtual nsresult Close() = 0;
+
+  // ICE controlling or controlled
+  virtual bool IsIceControlling() const = 0;
+  virtual Maybe<bool> IsPendingOfferer() const = 0;
+  virtual Maybe<bool> IsCurrentOfferer() const = 0;
+  virtual bool IsIceRestarting() const = 0;
+  virtual std::set<std::pair<std::string, std::string>> GetLocalIceCredentials()
+      const = 0;
+
+  virtual const std::string GetLastError() const { return "Error"; }
+
+  virtual const std::vector<std::pair<size_t, std::string>>&
+  GetLastSdpParsingErrors() const = 0;
+
+  static const char* GetStateStr(JsepSignalingState state) {
+    static const char* states[] = {"stable",
+                                   "have-local-offer",
+                                   "have-remote-offer",
+                                   "have-local-pranswer",
+                                   "have-remote-pranswer",
+                                   "closed"};
+
+    return states[state];
+  }
+
+  virtual bool CheckNegotiationNeeded() const = 0;
+
+  void CountTracksAndDatachannels(
+      uint16_t (&receiving)[SdpMediaSection::kMediaTypes],
+      uint16_t (&sending)[SdpMediaSection::kMediaTypes]) const {
+    memset(receiving, 0, sizeof(receiving));
+    memset(sending, 0, sizeof(sending));
+
+    for (const auto& transceiver : GetTransceivers()) {
+      if (transceiver.mRecvTrack.GetActive() ||
+          transceiver.GetMediaType() == SdpMediaSection::kApplication) {
+        receiving[transceiver.mRecvTrack.GetMediaType()]++;
+      }
+
+      if (transceiver.mSendTrack.GetActive() ||
+          transceiver.GetMediaType() == SdpMediaSection::kApplication) {
+        sending[transceiver.mSendTrack.GetMediaType()]++;
+      }
+    }
+  }
+
+  virtual void SetDefaultCodecs(
+      const std::vector<UniquePtr<JsepCodecDescription>>& aPreferredCodecs) = 0;
+
+  // See Bug 1642419, this can be removed when all sites are working with RTX.
+  void SetRtxIsAllowed(bool aRtxIsAllowed) { mRtxIsAllowed = aRtxIsAllowed; }
+
+ protected:
+  friend class JsepSessionTest;
+  // Returns transceivers in the order they were added.
+  virtual std::vector<JsepTransceiver>& GetTransceivers() = 0;
+  virtual const std::vector<JsepTransceiver>& GetTransceivers() const = 0;
+
+  const std::string mName;
+  JsepSignalingState mState;
+  uint32_t mNegotiations;
+
+  // See Bug 1642419, this can be removed when all sites are working with RTX.
+  bool mRtxIsAllowed = true;
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/jsep/JsepSessionImpl.cpp b/dom/media/webrtc/jsep/JsepSessionImpl.cpp
new file mode 100644
index 0000000000..bb792e7764
--- /dev/null
+++ b/dom/media/webrtc/jsep/JsepSessionImpl.cpp
@@ -0,0 +1,2457 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "jsep/JsepSessionImpl.h"
+
+#include <stdlib.h>
+
+#include <bitset>
+#include <iterator>
+#include <set>
+#include <string>
+#include <utility>
+
+#include "transport/logging.h"
+#include "mozilla/Preferences.h"
+#include "mozilla/Telemetry.h"
+#include "mozilla/UniquePtr.h"
+#include "mozilla/net/DataChannelProtocol.h"
+#include "nsDebug.h"
+#include "nspr.h"
+#include "nss.h"
+#include "pk11pub.h"
+
+#include "api/rtp_parameters.h"
+
+#include "jsep/JsepTrack.h"
+#include "jsep/JsepTransport.h"
+#include "sdp/HybridSdpParser.h"
+#include "sdp/SipccSdp.h"
+
+namespace mozilla {
+
+MOZ_MTLOG_MODULE("jsep")
+
+#define JSEP_SET_ERROR(error)                                 \
+  do {                                                        \
+    std::ostringstream os;                                    \
+    os << error;                                              \
+    mLastError = os.str();                                    \
+    MOZ_MTLOG(ML_ERROR, "[" << mName << "]: " << mLastError); \
+  } while (0);
+
+static std::bitset<128> GetForbiddenSdpPayloadTypes() {
+  std::bitset<128> forbidden(0);
+  forbidden[1] = true;
+  forbidden[2] = true;
+  forbidden[19] = true;
+  for (uint16_t i = 64; i < 96; ++i) {
+    forbidden[i] = true;
+  }
+  return forbidden;
+}
+
+static std::string GetRandomHex(size_t words) {
+  std::ostringstream os;
+
+  for (size_t i = 0; i < words; ++i) {
+    uint32_t rand;
+    SECStatus rv = PK11_GenerateRandom(reinterpret_cast<unsigned char*>(&rand),
+                                       sizeof(rand));
+    if (rv != SECSuccess) {
+      MOZ_CRASH();
+      return "";
+    }
+
+    os << std::hex << std::setfill('0') << std::setw(8) << rand;
+  }
+  return os.str();
+}
+
+JsepSessionImpl::JsepSessionImpl(const JsepSessionImpl& aOrig)
+    : JsepSession(aOrig),
+      JsepSessionCopyableStuff(aOrig),
+      mUuidGen(aOrig.mUuidGen->Clone()),
+      mGeneratedOffer(aOrig.mGeneratedOffer ? aOrig.mGeneratedOffer->Clone()
+                                            : nullptr),
+      mGeneratedAnswer(aOrig.mGeneratedAnswer ? aOrig.mGeneratedAnswer->Clone()
+                                              : nullptr),
+      mCurrentLocalDescription(aOrig.mCurrentLocalDescription
+                                   ? aOrig.mCurrentLocalDescription->Clone()
+                                   : nullptr),
+      mCurrentRemoteDescription(aOrig.mCurrentRemoteDescription
+                                    ? aOrig.mCurrentRemoteDescription->Clone()
+                                    : nullptr),
+      mPendingLocalDescription(aOrig.mPendingLocalDescription
+                                   ? aOrig.mPendingLocalDescription->Clone()
+                                   : nullptr),
+      mPendingRemoteDescription(aOrig.mPendingRemoteDescription
+                                    ? aOrig.mPendingRemoteDescription->Clone()
+                                    : nullptr),
+      mSdpHelper(&mLastError),
+      mParser(new HybridSdpParser()) {
+  for (const auto& codec : aOrig.mSupportedCodecs) {
+    mSupportedCodecs.emplace_back(codec->Clone());
+  }
+}
+
+nsresult JsepSessionImpl::Init() {
+  mLastError.clear();
+
+  MOZ_ASSERT(!mSessionId, "Init called more than once");
+
+  nsresult rv = SetupIds();
+  NS_ENSURE_SUCCESS(rv, rv);
+
+  mEncodeTrackId =
+      Preferences::GetBool("media.peerconnection.sdp.encode_track_id", true);
+
+  mIceUfrag = GetRandomHex(1);
+  mIcePwd = GetRandomHex(4);
+  return NS_OK;
+}
+
+static void GetIceCredentials(
+    const Sdp& aSdp,
+    std::set<std::pair<std::string, std::string>>* aCredentials) {
+  for (size_t i = 0; i < aSdp.GetMediaSectionCount(); ++i) {
+    const SdpAttributeList& attrs = aSdp.GetMediaSection(i).GetAttributeList();
+    if (attrs.HasAttribute(SdpAttribute::kIceUfragAttribute) &&
+        attrs.HasAttribute(SdpAttribute::kIcePwdAttribute)) {
+      aCredentials->insert(
+          std::make_pair(attrs.GetIceUfrag(), attrs.GetIcePwd()));
+    }
+  }
+}
+
+std::set<std::pair<std::string, std::string>>
+JsepSessionImpl::GetLocalIceCredentials() const {
+  std::set<std::pair<std::string, std::string>> result;
+  if (mCurrentLocalDescription) {
+    GetIceCredentials(*mCurrentLocalDescription, &result);
+  }
+  if (mPendingLocalDescription) {
+    GetIceCredentials(*mPendingLocalDescription, &result);
+  }
+  return result;
+}
+
+void JsepSessionImpl::AddTransceiver(const JsepTransceiver& aTransceiver) {
+  mLastError.clear();
+  MOZ_MTLOG(ML_DEBUG,
+            "[" << mName << "]: Adding transceiver " << aTransceiver.GetUuid());
+#ifdef DEBUG
+  if (aTransceiver.GetMediaType() == SdpMediaSection::kApplication) {
+    // Make sure we don't add more than one DataChannel transceiver
+    for (const auto& transceiver : mTransceivers) {
+      MOZ_ASSERT(transceiver.GetMediaType() != SdpMediaSection::kApplication);
+    }
+  }
+#endif
+  mTransceivers.push_back(aTransceiver);
+  InitTransceiver(mTransceivers.back());
+}
+
+void JsepSessionImpl::InitTransceiver(JsepTransceiver& aTransceiver) {
+  mLastError.clear();
+
+  if (aTransceiver.GetMediaType() != SdpMediaSection::kApplication) {
+    // Make sure we have an ssrc. Might already be set.
+    aTransceiver.mSendTrack.EnsureSsrcs(mSsrcGenerator, 1U);
+    aTransceiver.mSendTrack.SetCNAME(mCNAME);
+
+    // Make sure we have identifiers for send track, just in case.
+    // (man I hate this)
+    if (mEncodeTrackId) {
+      aTransceiver.mSendTrack.SetTrackId(aTransceiver.GetUuid());
+    }
+  } else {
+    // Datachannel transceivers should always be sendrecv. Just set it instead
+    // of asserting.
+    aTransceiver.mJsDirection = SdpDirectionAttribute::kSendrecv;
+  }
+
+  aTransceiver.mSendTrack.PopulateCodecs(mSupportedCodecs);
+  aTransceiver.mRecvTrack.PopulateCodecs(mSupportedCodecs);
+  // We do not set mLevel yet, we do that either on createOffer, or setRemote
+}
+
+nsresult JsepSessionImpl::SetBundlePolicy(JsepBundlePolicy policy) {
+  mLastError.clear();
+
+  if (mBundlePolicy == policy) {
+    return NS_OK;
+  }
+
+  if (mCurrentLocalDescription) {
+    JSEP_SET_ERROR(
+        "Changing the bundle policy is only supported before the "
+        "first SetLocalDescription.");
+    return NS_ERROR_UNEXPECTED;
+  }
+
+  mBundlePolicy = policy;
+  return NS_OK;
+}
+
+nsresult JsepSessionImpl::AddDtlsFingerprint(
+    const std::string& algorithm, const std::vector<uint8_t>& value) {
+  mLastError.clear();
+  JsepDtlsFingerprint fp;
+
+  fp.mAlgorithm = algorithm;
+  fp.mValue = value;
+
+  mDtlsFingerprints.push_back(fp);
+
+  return NS_OK;
+}
+
+nsresult JsepSessionImpl::AddRtpExtension(
+    JsepMediaType mediaType, const std::string& extensionName,
+    SdpDirectionAttribute::Direction direction) {
+  mLastError.clear();
+
+  for (auto& ext : mRtpExtensions) {
+    if (ext.mExtmap.direction == direction &&
+        ext.mExtmap.extensionname == extensionName) {
+      if (ext.mMediaType != mediaType) {
+        ext.mMediaType = JsepMediaType::kAudioVideo;
+      }
+      return NS_OK;
+    }
+  }
+
+  uint16_t freeEntry = GetNeverUsedExtmapEntry();
+
+  if (freeEntry == 0) {
+    return NS_ERROR_FAILURE;
+  }
+
+  JsepExtmapMediaType extMediaType = {
+      mediaType,
+      {freeEntry, direction,
+       // do we want to specify direction?
+       direction != SdpDirectionAttribute::kSendrecv, extensionName, ""}};
+
+  mRtpExtensions.push_back(extMediaType);
+  return NS_OK;
+}
+
+nsresult JsepSessionImpl::AddAudioRtpExtension(
+    const std::string& extensionName,
+    SdpDirectionAttribute::Direction direction) {
+  return AddRtpExtension(JsepMediaType::kAudio, extensionName, direction);
+}
+
+nsresult JsepSessionImpl::AddVideoRtpExtension(
+    const std::string& extensionName,
+    SdpDirectionAttribute::Direction direction) {
+  return AddRtpExtension(JsepMediaType::kVideo, extensionName, direction);
+}
+
+nsresult JsepSessionImpl::AddAudioVideoRtpExtension(
+    const std::string& extensionName,
+    SdpDirectionAttribute::Direction direction) {
+  return AddRtpExtension(JsepMediaType::kAudioVideo, extensionName, direction);
+}
+
+nsresult JsepSessionImpl::CreateOfferMsection(const JsepOfferOptions& options,
+                                              JsepTransceiver& transceiver,
+                                              Sdp* local) {
+  SdpMediaSection::Protocol protocol(
+      SdpHelper::GetProtocolForMediaType(transceiver.GetMediaType()));
+
+  const Sdp* answer(GetAnswer());
+  const SdpMediaSection* lastAnswerMsection = nullptr;
+
+  if (answer &&
+      (local->GetMediaSectionCount() < answer->GetMediaSectionCount())) {
+    lastAnswerMsection =
+        &answer->GetMediaSection(local->GetMediaSectionCount());
+    // Use the protocol the answer used, even if it is not what we would have
+    // used.
+    protocol = lastAnswerMsection->GetProtocol();
+  }
+
+  SdpMediaSection* msection = &local->AddMediaSection(
+      transceiver.GetMediaType(), transceiver.mJsDirection, 0, protocol,
+      sdp::kIPv4, "0.0.0.0");
+
+  // Some of this stuff (eg; mid) sticks around even if disabled
+  if (lastAnswerMsection) {
+    MOZ_ASSERT(lastAnswerMsection->GetMediaType() ==
+               transceiver.GetMediaType());
+    nsresult rv = mSdpHelper.CopyStickyParams(*lastAnswerMsection, msection);
+    NS_ENSURE_SUCCESS(rv, rv);
+  }
+
+  if (transceiver.IsStopped()) {
+    SdpHelper::DisableMsection(local, msection);
+    return NS_OK;
+  }
+
+  msection->SetPort(9);
+
+  // We don't do this in AddTransportAttributes because that is also used for
+  // making answers, and we don't want to unconditionally set rtcp-mux or
+  // rtcp-rsize there.
+  if (mSdpHelper.HasRtcp(msection->GetProtocol())) {
+    // Set RTCP-MUX.
+    msection->GetAttributeList().SetAttribute(
+        new SdpFlagAttribute(SdpAttribute::kRtcpMuxAttribute));
+    // Set RTCP-RSIZE
+    if (msection->GetMediaType() == SdpMediaSection::MediaType::kVideo &&
+        Preferences::GetBool("media.navigator.video.offer_rtcp_rsize", false)) {
+      msection->GetAttributeList().SetAttribute(
+          new SdpFlagAttribute(SdpAttribute::kRtcpRsizeAttribute));
+    }
+  }
+
+  nsresult rv = AddTransportAttributes(msection, SdpSetupAttribute::kActpass);
+  NS_ENSURE_SUCCESS(rv, rv);
+
+  transceiver.mSendTrack.AddToOffer(mSsrcGenerator, msection);
+  transceiver.mRecvTrack.AddToOffer(mSsrcGenerator, msection);
+
+  AddExtmap(msection);
+
+  std::string mid;
+  // We do not set the mid on the transceiver, that happens when a description
+  // is set.
+  if (transceiver.IsAssociated()) {
+    mid = transceiver.GetMid();
+  } else {
+    mid = GetNewMid();
+  }
+
+  msection->GetAttributeList().SetAttribute(
+      new SdpStringAttribute(SdpAttribute::kMidAttribute, mid));
+
+  return NS_OK;
+}
+
+void JsepSessionImpl::SetupBundle(Sdp* sdp) const {
+  std::vector<std::string> mids;
+  std::set<SdpMediaSection::MediaType> observedTypes;
+
+  // This has the effect of changing the bundle level if the first m-section
+  // goes from disabled to enabled. This is kinda inefficient.
+
+  for (size_t i = 0; i < sdp->GetMediaSectionCount(); ++i) {
+    auto& attrs = sdp->GetMediaSection(i).GetAttributeList();
+    if ((sdp->GetMediaSection(i).GetPort() != 0) &&
+        attrs.HasAttribute(SdpAttribute::kMidAttribute)) {
+      bool useBundleOnly = false;
+      switch (mBundlePolicy) {
+        case kBundleMaxCompat:
+          // We don't use bundle-only for max-compat
+          break;
+        case kBundleBalanced:
+          // balanced means we use bundle-only on everything but the first
+          // m-section of a given type
+          if (observedTypes.count(sdp->GetMediaSection(i).GetMediaType())) {
+            useBundleOnly = true;
+          }
+          observedTypes.insert(sdp->GetMediaSection(i).GetMediaType());
+          break;
+        case kBundleMaxBundle:
+          // max-bundle means we use bundle-only on everything but the first
+          // m-section
+          useBundleOnly = !mids.empty();
+          break;
+      }
+
+      if (useBundleOnly) {
+        attrs.SetAttribute(
+            new SdpFlagAttribute(SdpAttribute::kBundleOnlyAttribute));
+        // Set port to 0 for sections with bundle-only attribute. (mjf)
+        sdp->GetMediaSection(i).SetPort(0);
+      }
+
+      mids.push_back(attrs.GetMid());
+    }
+  }
+
+  if (!mids.empty()) {
+    UniquePtr<SdpGroupAttributeList> groupAttr(new SdpGroupAttributeList);
+    groupAttr->PushEntry(SdpGroupAttributeList::kBundle, mids);
+    sdp->GetAttributeList().SetAttribute(groupAttr.release());
+  }
+}
+
+JsepSession::Result JsepSessionImpl::CreateOffer(
+    const JsepOfferOptions& options, std::string* offer) {
+  mLastError.clear();
+
+  if (mState != kJsepStateStable && mState != kJsepStateHaveLocalOffer) {
+    JSEP_SET_ERROR("Cannot create offer in state " << GetStateStr(mState));
+    // Spec doesn't seem to say this is an error. It probably should.
+    return dom::PCError::InvalidStateError;
+  }
+
+  // This is one of those places where CreateOffer sets some state.
+  SetIceRestarting(options.mIceRestart.isSome() && *(options.mIceRestart));
+
+  UniquePtr<Sdp> sdp;
+
+  // Make the basic SDP that is common to offer/answer.
+  nsresult rv = CreateGenericSDP(&sdp);
+  NS_ENSURE_SUCCESS(rv, dom::PCError::OperationError);
+
+  for (size_t level = 0;
+       Maybe<JsepTransceiver> transceiver = GetTransceiverForLocal(level);
+       ++level) {
+    rv = CreateOfferMsection(options, *transceiver, sdp.get());
+    NS_ENSURE_SUCCESS(rv, dom::PCError::OperationError);
+    SetTransceiver(*transceiver);
+  }
+
+  SetupBundle(sdp.get());
+
+  if (mCurrentLocalDescription) {
+    rv = CopyPreviousTransportParams(*GetAnswer(), *mCurrentLocalDescription,
+                                     *sdp, sdp.get());
+    NS_ENSURE_SUCCESS(rv, dom::PCError::OperationError);
+  }
+
+  *offer = sdp->ToString();
+  mGeneratedOffer = std::move(sdp);
+  ++mSessionVersion;
+  MOZ_MTLOG(ML_DEBUG, "[" << mName << "]: CreateOffer \nSDP=\n" << *offer);
+
+  return Result();
+}
+
+std::string JsepSessionImpl::GetLocalDescription(
+    JsepDescriptionPendingOrCurrent type) const {
+  std::ostringstream os;
+  mozilla::Sdp* sdp = GetParsedLocalDescription(type);
+  if (sdp) {
+    sdp->Serialize(os);
+  }
+  return os.str();
+}
+
+std::string JsepSessionImpl::GetRemoteDescription(
+    JsepDescriptionPendingOrCurrent type) const {
+  std::ostringstream os;
+  mozilla::Sdp* sdp = GetParsedRemoteDescription(type);
+  if (sdp) {
+    sdp->Serialize(os);
+  }
+  return os.str();
+}
+
+void JsepSessionImpl::AddExtmap(SdpMediaSection* msection) {
+  auto extensions = GetRtpExtensions(*msection);
+
+  if (!extensions.empty()) {
+    SdpExtmapAttributeList* extmap = new SdpExtmapAttributeList;
+    extmap->mExtmaps = extensions;
+    msection->GetAttributeList().SetAttribute(extmap);
+  }
+}
+
+std::vector<SdpExtmapAttributeList::Extmap> JsepSessionImpl::GetRtpExtensions(
+    const SdpMediaSection& msection) {
+  std::vector<SdpExtmapAttributeList::Extmap> result;
+  JsepMediaType mediaType = JsepMediaType::kNone;
+  switch (msection.GetMediaType()) {
+    case SdpMediaSection::kAudio:
+      mediaType = JsepMediaType::kAudio;
+      break;
+    case SdpMediaSection::kVideo:
+      mediaType = JsepMediaType::kVideo;
+      if (msection.GetAttributeList().HasAttribute(
+              SdpAttribute::kRidAttribute)) {
+        // We need RID support
+        // TODO: Would it be worth checking that the direction is sane?
+        AddVideoRtpExtension(webrtc::RtpExtension::kRidUri,
+                             SdpDirectionAttribute::kSendonly);
+
+        if (mRtxIsAllowed &&
+            Preferences::GetBool("media.peerconnection.video.use_rtx", false)) {
+          AddVideoRtpExtension(webrtc::RtpExtension::kRepairedRidUri,
+                               SdpDirectionAttribute::kSendonly);
+        }
+      }
+      break;
+    default:;
+  }
+  if (mediaType != JsepMediaType::kNone) {
+    for (auto ext = mRtpExtensions.begin(); ext != mRtpExtensions.end();
+         ++ext) {
+      if (ext->mMediaType == mediaType ||
+          ext->mMediaType == JsepMediaType::kAudioVideo) {
+        result.push_back(ext->mExtmap);
+      }
+    }
+  }
+  return result;
+}
+
+std::string JsepSessionImpl::GetNewMid() {
+  std::string mid;
+
+  do {
+    std::ostringstream osMid;
+    osMid << mMidCounter++;
+    mid = osMid.str();
+  } while (mUsedMids.count(mid));
+
+  mUsedMids.insert(mid);
+  return mid;
+}
+
+void JsepSessionImpl::AddCommonExtmaps(const SdpMediaSection& remoteMsection,
+                                       SdpMediaSection* msection) {
+  auto negotiatedRtpExtensions = GetRtpExtensions(*msection);
+  mSdpHelper.NegotiateAndAddExtmaps(remoteMsection, negotiatedRtpExtensions,
+                                    msection);
+}
+
+uint16_t JsepSessionImpl::GetNeverUsedExtmapEntry() {
+  uint16_t result = 1;
+
+  // Walk the set in order, and return the first "hole" we find
+  for (const auto used : mExtmapEntriesEverUsed) {
+    if (result != used) {
+      MOZ_ASSERT(result < used);
+      break;
+    }
+
+    // RFC 5285 says entries >= 4096 are used in offers to force the answerer
+    // to pick, so we do not want to actually use these
+    if (used == 4095) {
+      JSEP_SET_ERROR(
+          "Too many rtp extensions have been added. "
+          "That's 4095. Who _does_ that?");
+      return 0;
+    }
+
+    result = used + 1;
+  }
+
+  mExtmapEntriesEverUsed.insert(result);
+  return result;
+}
+
+JsepSession::Result JsepSessionImpl::CreateAnswer(
+    const JsepAnswerOptions& options, std::string* answer) {
+  mLastError.clear();
+
+  if (mState != kJsepStateHaveRemoteOffer) {
+    JSEP_SET_ERROR("Cannot create answer in state " << GetStateStr(mState));
+    return dom::PCError::InvalidStateError;
+  }
+
+  UniquePtr<Sdp> sdp;
+
+  // Make the basic SDP that is common to offer/answer.
+  nsresult rv = CreateGenericSDP(&sdp);
+  NS_ENSURE_SUCCESS(rv, dom::PCError::OperationError);
+
+  const Sdp& offer = *mPendingRemoteDescription;
+
+  // Copy the bundle groups into our answer
+  UniquePtr<SdpGroupAttributeList> groupAttr(new SdpGroupAttributeList);
+  mSdpHelper.GetBundleGroups(offer, &groupAttr->mGroups);
+  sdp->GetAttributeList().SetAttribute(groupAttr.release());
+
+  for (size_t i = 0; i < offer.GetMediaSectionCount(); ++i) {
+    // The transceivers are already in place, due to setRemote
+    Maybe<JsepTransceiver> transceiver(GetTransceiverForLevel(i));
+    if (!transceiver) {
+      JSEP_SET_ERROR("No transceiver for level " << i);
+      MOZ_ASSERT(false);
+      return dom::PCError::OperationError;
+    }
+    rv = CreateAnswerMsection(options, *transceiver, offer.GetMediaSection(i),
+                              sdp.get());
+    NS_ENSURE_SUCCESS(rv, dom::PCError::OperationError);
+    SetTransceiver(*transceiver);
+  }
+
+  // Ensure that each bundle-group starts with a mid that has a transport, in
+  // case we've disabled what the offerer wanted to use. If the group doesn't
+  // contain anything that has a transport, remove it.
+  groupAttr.reset(new SdpGroupAttributeList);
+  std::vector<SdpGroupAttributeList::Group> bundleGroups;
+  mSdpHelper.GetBundleGroups(*sdp, &bundleGroups);
+  for (auto& group : bundleGroups) {
+    for (auto& mid : group.tags) {
+      const SdpMediaSection* msection =
+          mSdpHelper.FindMsectionByMid(offer, mid);
+
+      if (msection && !msection->GetAttributeList().HasAttribute(
+                          SdpAttribute::kBundleOnlyAttribute)) {
+        std::swap(group.tags[0], mid);
+        groupAttr->mGroups.push_back(group);
+        break;
+      }
+    }
+  }
+  sdp->GetAttributeList().SetAttribute(groupAttr.release());
+
+  if (mCurrentLocalDescription) {
+    // per discussion with bwc, 3rd parm here should be offer, not *sdp. (mjf)
+    rv = CopyPreviousTransportParams(*GetAnswer(), *mCurrentRemoteDescription,
+                                     offer, sdp.get());
+    NS_ENSURE_SUCCESS(rv, dom::PCError::OperationError);
+  }
+
+  *answer = sdp->ToString();
+  mGeneratedAnswer = std::move(sdp);
+  ++mSessionVersion;
+  MOZ_MTLOG(ML_DEBUG, "[" << mName << "]: CreateAnswer \nSDP=\n" << *answer);
+
+  return Result();
+}
+
+nsresult JsepSessionImpl::CreateAnswerMsection(
+    const JsepAnswerOptions& options, JsepTransceiver& transceiver,
+    const SdpMediaSection& remoteMsection, Sdp* sdp) {
+  MOZ_ASSERT(transceiver.GetMediaType() == remoteMsection.GetMediaType());
+  SdpDirectionAttribute::Direction direction =
+      reverse(remoteMsection.GetDirection()) & transceiver.mJsDirection;
+  SdpMediaSection& msection =
+      sdp->AddMediaSection(remoteMsection.GetMediaType(), direction, 9,
+                           remoteMsection.GetProtocol(), sdp::kIPv4, "0.0.0.0");
+
+  nsresult rv = mSdpHelper.CopyStickyParams(remoteMsection, &msection);
+  NS_ENSURE_SUCCESS(rv, rv);
+
+  if (mSdpHelper.MsectionIsDisabled(remoteMsection) ||
+      // JS might have stopped this
+      transceiver.IsStopped()) {
+    SdpHelper::DisableMsection(sdp, &msection);
+    return NS_OK;
+  }
+
+  MOZ_ASSERT(transceiver.IsAssociated());
+  if (msection.GetAttributeList().GetMid().empty()) {
+    msection.GetAttributeList().SetAttribute(new SdpStringAttribute(
+        SdpAttribute::kMidAttribute, transceiver.GetMid()));
+  }
+
+  MOZ_ASSERT(transceiver.GetMid() == msection.GetAttributeList().GetMid());
+
+  SdpSetupAttribute::Role role;
+  if (transceiver.mTransport.mDtls && !IsIceRestarting()) {
+    role = (transceiver.mTransport.mDtls->mRole ==
+            JsepDtlsTransport::kJsepDtlsClient)
+               ? SdpSetupAttribute::kActive
+               : SdpSetupAttribute::kPassive;
+  } else {
+    rv = DetermineAnswererSetupRole(remoteMsection, &role);
+    NS_ENSURE_SUCCESS(rv, rv);
+  }
+
+  rv = AddTransportAttributes(&msection, role);
+  NS_ENSURE_SUCCESS(rv, rv);
+
+  transceiver.mSendTrack.AddToAnswer(remoteMsection, mSsrcGenerator, &msection);
+  transceiver.mRecvTrack.AddToAnswer(remoteMsection, mSsrcGenerator, &msection);
+
+  // Add extmap attributes. This logic will probably be moved to the track,
+  // since it can be specified on a per-sender basis in JS.
+  // We will need some validation to ensure that the ids are identical for
+  // RTP streams that are bundled together, though (bug 1406529).
+  AddCommonExtmaps(remoteMsection, &msection);
+
+  if (msection.GetFormats().empty()) {
+    // Could not negotiate anything. Disable m-section.
+    SdpHelper::DisableMsection(sdp, &msection);
+  }
+
+  return NS_OK;
+}
+
+nsresult JsepSessionImpl::DetermineAnswererSetupRole(
+    const SdpMediaSection& remoteMsection, SdpSetupAttribute::Role* rolep) {
+  // Determine the role.
+  // RFC 5763 says:
+  //
+  //   The endpoint MUST use the setup attribute defined in [RFC4145].
+  //   The endpoint that is the offerer MUST use the setup attribute
+  //   value of setup:actpass and be prepared to receive a client_hello
+  //   before it receives the answer.  The answerer MUST use either a
+  //   setup attribute value of setup:active or setup:passive.  Note that
+  //   if the answerer uses setup:passive, then the DTLS handshake will
+  //   not begin until the answerer is received, which adds additional
+  //   latency. setup:active allows the answer and the DTLS handshake to
+  //   occur in parallel.  Thus, setup:active is RECOMMENDED.  Whichever
+  //   party is active MUST initiate a DTLS handshake by sending a
+  //   ClientHello over each flow (host/port quartet).
+  //
+  //   We default to assuming that the offerer is passive and we are active.
+  SdpSetupAttribute::Role role = SdpSetupAttribute::kActive;
+
+  if (remoteMsection.GetAttributeList().HasAttribute(
+          SdpAttribute::kSetupAttribute)) {
+    switch (remoteMsection.GetAttributeList().GetSetup().mRole) {
+      case SdpSetupAttribute::kActive:
+        role = SdpSetupAttribute::kPassive;
+        break;
+      case SdpSetupAttribute::kPassive:
+      case SdpSetupAttribute::kActpass:
+        role = SdpSetupAttribute::kActive;
+        break;
+      case SdpSetupAttribute::kHoldconn:
+        // This should have been caught by ParseSdp
+        MOZ_ASSERT(false);
+        JSEP_SET_ERROR(
+            "The other side used an illegal setup attribute"
+            " (\"holdconn\").");
+        return NS_ERROR_INVALID_ARG;
+    }
+  }
+
+  *rolep = role;
+  return NS_OK;
+}
+
+JsepSession::Result JsepSessionImpl::SetLocalDescription(
+    JsepSdpType type, const std::string& constSdp) {
+  mLastError.clear();
+  std::string sdp = constSdp;
+
+  MOZ_MTLOG(ML_DEBUG, "[" << mName << "]: SetLocalDescription type=" << type
+                          << "\nSDP=\n"
+                          << sdp);
+
+  switch (type) {
+    case kJsepSdpOffer:
+      if (!mGeneratedOffer) {
+        JSEP_SET_ERROR(
+            "Cannot set local offer when createOffer has not been called.");
+        return dom::PCError::InvalidModificationError;
+      }
+      if (sdp.empty()) {
+        sdp = mGeneratedOffer->ToString();
+      }
+      if (mState == kJsepStateHaveLocalOffer) {
+        // Rollback previous offer before applying the new one.
+        SetLocalDescription(kJsepSdpRollback, "");
+        MOZ_ASSERT(mState == kJsepStateStable);
+      }
+      break;
+    case kJsepSdpAnswer:
+    case kJsepSdpPranswer:
+      if (!mGeneratedAnswer) {
+        JSEP_SET_ERROR(
+            "Cannot set local answer when createAnswer has not been called.");
+        return dom::PCError::InvalidModificationError;
+      }
+      if (sdp.empty()) {
+        sdp = mGeneratedAnswer->ToString();
+      }
+      break;
+    case kJsepSdpRollback:
+      if (mState != kJsepStateHaveLocalOffer) {
+        JSEP_SET_ERROR("Cannot rollback local description in "
+                       << GetStateStr(mState));
+        // Currently, spec allows this in any state except stable, and
+        // sRD(rollback) and sLD(rollback) do exactly the same thing.
+        return dom::PCError::InvalidStateError;
+      }
+
+      mPendingLocalDescription.reset();
+      SetState(kJsepStateStable);
+      RollbackLocalOffer();
+      return Result();
+  }
+
+  switch (mState) {
+    case kJsepStateStable:
+      if (type != kJsepSdpOffer) {
+        JSEP_SET_ERROR("Cannot set local answer in state "
+                       << GetStateStr(mState));
+        return dom::PCError::InvalidStateError;
+      }
+      break;
+    case kJsepStateHaveRemoteOffer:
+      if (type != kJsepSdpAnswer && type != kJsepSdpPranswer) {
+        JSEP_SET_ERROR("Cannot set local offer in state "
+                       << GetStateStr(mState));
+        return dom::PCError::InvalidStateError;
+      }
+      break;
+    default:
+      JSEP_SET_ERROR("Cannot set local offer or answer in state "
+                     << GetStateStr(mState));
+      return dom::PCError::InvalidStateError;
+  }
+
+  UniquePtr<Sdp> parsed;
+  nsresult rv = ParseSdp(sdp, &parsed);
+  // Needs to be RTCError with sdp-syntax-error
+  NS_ENSURE_SUCCESS(rv, dom::PCError::OperationError);
+
+  // Check that content hasn't done anything unsupported with the SDP
+  rv = ValidateLocalDescription(*parsed, type);
+  NS_ENSURE_SUCCESS(rv, dom::PCError::InvalidModificationError);
+
+  switch (type) {
+    case kJsepSdpOffer:
+      rv = ValidateOffer(*parsed);
+      break;
+    case kJsepSdpAnswer:
+    case kJsepSdpPranswer:
+      rv = ValidateAnswer(*mPendingRemoteDescription, *parsed);
+      break;
+    case kJsepSdpRollback:
+      MOZ_CRASH();  // Handled above
+  }
+  NS_ENSURE_SUCCESS(rv, dom::PCError::InvalidAccessError);
+
+  if (type == kJsepSdpOffer) {
+    // Save in case we need to rollback
+    mOldTransceivers = mTransceivers;
+  }
+
+  SdpHelper::BundledMids bundledMids;
+  rv = mSdpHelper.GetBundledMids(*parsed, &bundledMids);
+  NS_ENSURE_SUCCESS(rv, dom::PCError::OperationError);
+
+  SdpHelper::BundledMids remoteBundledMids;
+  if (type != kJsepSdpOffer) {
+    rv = mSdpHelper.GetBundledMids(*mPendingRemoteDescription,
+                                   &remoteBundledMids);
+    NS_ENSURE_SUCCESS(rv, dom::PCError::OperationError);
+  }
+
+  for (size_t i = 0; i < parsed->GetMediaSectionCount(); ++i) {
+    Maybe<JsepTransceiver> transceiver(GetTransceiverForLevel(i));
+    if (!transceiver) {
+      MOZ_ASSERT(false);
+      JSEP_SET_ERROR("No transceiver for level " << i);
+      return dom::PCError::OperationError;
+    }
+
+    const auto& msection = parsed->GetMediaSection(i);
+    transceiver->Associate(msection.GetAttributeList().GetMid());
+    transceiver->mRecvTrack.RecvTrackSetLocal(msection);
+
+    if (mSdpHelper.MsectionIsDisabled(msection)) {
+      transceiver->mTransport.Close();
+      SetTransceiver(*transceiver);
+      continue;
+    }
+
+    bool hasOwnTransport = mSdpHelper.OwnsTransport(
+        msection, bundledMids,
+        (type == kJsepSdpOffer) ? sdp::kOffer : sdp::kAnswer);
+    if (type != kJsepSdpOffer) {
+      const auto& remoteMsection =
+          mPendingRemoteDescription->GetMediaSection(i);
+      // Don't allow the answer to override what the offer allowed for
+      hasOwnTransport &= mSdpHelper.OwnsTransport(
+          remoteMsection, remoteBundledMids, sdp::kOffer);
+    }
+
+    if (hasOwnTransport) {
+      EnsureHasOwnTransport(parsed->GetMediaSection(i), *transceiver);
+    }
+
+    if (type == kJsepSdpOffer) {
+      if (!hasOwnTransport) {
+        auto it = bundledMids.find(transceiver->GetMid());
+        if (it != bundledMids.end()) {
+          transceiver->SetBundleLevel(it->second->GetLevel());
+        }
+      }
+    } else {
+      auto it = remoteBundledMids.find(transceiver->GetMid());
+      if (it != remoteBundledMids.end()) {
+        transceiver->SetBundleLevel(it->second->GetLevel());
+      }
+    }
+    SetTransceiver(*transceiver);
+  }
+
+  CopyBundleTransports();
+
+  switch (type) {
+    case kJsepSdpOffer:
+      rv = SetLocalDescriptionOffer(std::move(parsed));
+      break;
+    case kJsepSdpAnswer:
+    case kJsepSdpPranswer:
+      rv = SetLocalDescriptionAnswer(type, std::move(parsed));
+      break;
+    case kJsepSdpRollback:
+      MOZ_CRASH();  // Handled above
+  }
+
+  NS_ENSURE_SUCCESS(rv, dom::PCError::OperationError);
+  return Result();
+}
+
+nsresult JsepSessionImpl::SetLocalDescriptionOffer(UniquePtr<Sdp> offer) {
+  MOZ_ASSERT(mState == kJsepStateStable);
+  mPendingLocalDescription = std::move(offer);
+  mIsPendingOfferer = Some(true);
+  SetState(kJsepStateHaveLocalOffer);
+  return NS_OK;
+}
+
+nsresult JsepSessionImpl::SetLocalDescriptionAnswer(JsepSdpType type,
+                                                    UniquePtr<Sdp> answer) {
+  MOZ_ASSERT(mState == kJsepStateHaveRemoteOffer);
+  mPendingLocalDescription = std::move(answer);
+
+  nsresult rv = HandleNegotiatedSession(mPendingLocalDescription,
+                                        mPendingRemoteDescription);
+  NS_ENSURE_SUCCESS(rv, rv);
+
+  mCurrentRemoteDescription = std::move(mPendingRemoteDescription);
+  mCurrentLocalDescription = std::move(mPendingLocalDescription);
+  MOZ_ASSERT(mIsPendingOfferer.isSome() && !*mIsPendingOfferer);
+  mIsPendingOfferer.reset();
+  mIsCurrentOfferer = Some(false);
+
+  SetState(kJsepStateStable);
+  return NS_OK;
+}
+
+JsepSession::Result JsepSessionImpl::SetRemoteDescription(
+    JsepSdpType type, const std::string& sdp) {
+  mLastError.clear();
+
+  MOZ_MTLOG(ML_DEBUG, "[" << mName << "]: SetRemoteDescription type=" << type
+                          << "\nSDP=\n"
+                          << sdp);
+
+  if (mState == kJsepStateHaveRemoteOffer && type == kJsepSdpOffer) {
+    // Rollback previous offer before applying the new one.
+    SetRemoteDescription(kJsepSdpRollback, "");
+    MOZ_ASSERT(mState == kJsepStateStable);
+  }
+
+  if (type == kJsepSdpRollback) {
+    if (mState != kJsepStateHaveRemoteOffer) {
+      JSEP_SET_ERROR("Cannot rollback remote description in "
+                     << GetStateStr(mState));
+      return dom::PCError::InvalidStateError;
+    }
+
+    mPendingRemoteDescription.reset();
+    SetState(kJsepStateStable);
+    RollbackRemoteOffer();
+
+    return Result();
+  }
+
+  switch (mState) {
+    case kJsepStateStable:
+      if (type != kJsepSdpOffer) {
+        JSEP_SET_ERROR("Cannot set remote answer in state "
+                       << GetStateStr(mState));
+        return dom::PCError::InvalidStateError;
+      }
+      break;
+    case kJsepStateHaveLocalOffer:
+    case kJsepStateHaveRemotePranswer:
+      if (type != kJsepSdpAnswer && type != kJsepSdpPranswer) {
+        JSEP_SET_ERROR("Cannot set remote offer in state "
+                       << GetStateStr(mState));
+        return dom::PCError::InvalidStateError;
+      }
+      break;
+    default:
+      JSEP_SET_ERROR("Cannot set remote offer or answer in current state "
+                     << GetStateStr(mState));
+      return dom::PCError::InvalidStateError;
+  }
+
+  // Parse.
+  UniquePtr<Sdp> parsed;
+  nsresult rv = ParseSdp(sdp, &parsed);
+  // Needs to be RTCError with sdp-syntax-error
+  NS_ENSURE_SUCCESS(rv, dom::PCError::OperationError);
+
+  rv = ValidateRemoteDescription(*parsed);
+  NS_ENSURE_SUCCESS(rv, dom::PCError::InvalidAccessError);
+
+  switch (type) {
+    case kJsepSdpOffer:
+      rv = ValidateOffer(*parsed);
+      break;
+    case kJsepSdpAnswer:
+    case kJsepSdpPranswer:
+      rv = ValidateAnswer(*mPendingLocalDescription, *parsed);
+      break;
+    case kJsepSdpRollback:
+      MOZ_CRASH();  // Handled above
+  }
+  NS_ENSURE_SUCCESS(rv, dom::PCError::InvalidAccessError);
+
+  bool iceLite =
+      parsed->GetAttributeList().HasAttribute(SdpAttribute::kIceLiteAttribute);
+
+  // check for mismatch ufrag/pwd indicating ice restart
+  // can't just check the first one because it might be disabled
+  bool iceRestarting = false;
+  if (mCurrentRemoteDescription.get()) {
+    for (size_t i = 0; !iceRestarting &&
+                       i < mCurrentRemoteDescription->GetMediaSectionCount();
+         ++i) {
+      const SdpMediaSection& newMsection = parsed->GetMediaSection(i);
+      const SdpMediaSection& oldMsection =
+          mCurrentRemoteDescription->GetMediaSection(i);
+
+      if (mSdpHelper.MsectionIsDisabled(newMsection) ||
+          mSdpHelper.MsectionIsDisabled(oldMsection)) {
+        continue;
+      }
+
+      iceRestarting = mSdpHelper.IceCredentialsDiffer(newMsection, oldMsection);
+    }
+  }
+
+  std::vector<std::string> iceOptions;
+  if (parsed->GetAttributeList().HasAttribute(
+          SdpAttribute::kIceOptionsAttribute)) {
+    iceOptions = parsed->GetAttributeList().GetIceOptions().mValues;
+  }
+
+  if (type == kJsepSdpOffer) {
+    // Save in case we need to rollback.
+    mOldTransceivers = mTransceivers;
+    for (auto& transceiver : mTransceivers) {
+      if (!transceiver.IsNegotiated()) {
+        // We chose a level for this transceiver, but never negotiated it.
+        // Discard this state.
+        transceiver.ClearLevel();
+      }
+    }
+  }
+
+  // TODO(bug 1095780): Note that we create remote tracks even when
+  // They contain only codecs we can't negotiate or other craziness.
+  rv = UpdateTransceiversFromRemoteDescription(*parsed);
+  NS_ENSURE_SUCCESS(rv, dom::PCError::OperationError);
+
+  for (size_t i = 0; i < parsed->GetMediaSectionCount(); ++i) {
+    MOZ_ASSERT(GetTransceiverForLevel(i));
+  }
+
+  switch (type) {
+    case kJsepSdpOffer:
+      rv = SetRemoteDescriptionOffer(std::move(parsed));
+      break;
+    case kJsepSdpAnswer:
+    case kJsepSdpPranswer:
+      rv = SetRemoteDescriptionAnswer(type, std::move(parsed));
+      break;
+    case kJsepSdpRollback:
+      MOZ_CRASH();  // Handled above
+  }
+
+  NS_ENSURE_SUCCESS(rv, dom::PCError::OperationError);
+
+  mRemoteIsIceLite = iceLite;
+  mIceOptions = iceOptions;
+  SetIceRestarting(iceRestarting);
+  return Result();
+}
+
+nsresult JsepSessionImpl::HandleNegotiatedSession(
+    const UniquePtr<Sdp>& local, const UniquePtr<Sdp>& remote) {
+  // local ufrag/pwd has been negotiated; we will never go back to the old ones
+  mOldIceUfrag.clear();
+  mOldIcePwd.clear();
+
+  bool remoteIceLite =
+      remote->GetAttributeList().HasAttribute(SdpAttribute::kIceLiteAttribute);
+
+  mIceControlling = remoteIceLite || *mIsPendingOfferer;
+
+  const Sdp& answer = *mIsPendingOfferer ? *remote : *local;
+
+  SdpHelper::BundledMids bundledMids;
+  nsresult rv = mSdpHelper.GetBundledMids(answer, &bundledMids);
+  NS_ENSURE_SUCCESS(rv, rv);
+
+  // First, set the bundle level on the transceivers
+  for (auto& [mid, transportOwner] : bundledMids) {
+    Maybe<JsepTransceiver> bundledTransceiver = GetTransceiverForMid(mid);
+    if (!bundledTransceiver) {
+      JSEP_SET_ERROR("No transceiver for bundled mid " << mid);
+      return NS_ERROR_INVALID_ARG;
+    }
+    bundledTransceiver->SetBundleLevel(transportOwner->GetLevel());
+    SetTransceiver(*bundledTransceiver);
+  }
+
+  // Now walk through the m-sections, perform negotiation, and update the
+  // transceivers.
+  for (size_t i = 0; i < local->GetMediaSectionCount(); ++i) {
+    Maybe<JsepTransceiver> transceiver(GetTransceiverForLevel(i));
+    if (!transceiver) {
+      MOZ_ASSERT(false);
+      JSEP_SET_ERROR("No transceiver for level " << i);
+      return NS_ERROR_FAILURE;
+    }
+
+    // Skip disabled m-sections.
+    if (answer.GetMediaSection(i).GetPort() == 0) {
+      transceiver->mTransport.Close();
+      transceiver->Stop();
+      transceiver->Disassociate();
+      transceiver->ClearBundleLevel();
+      transceiver->mSendTrack.SetActive(false);
+      transceiver->mRecvTrack.SetActive(false);
+      transceiver->SetCanRecycle();
+      SetTransceiver(*transceiver);
+      // Do not clear mLevel yet! That will happen on the next negotiation.
+      continue;
+    }
+
+    rv = MakeNegotiatedTransceiver(remote->GetMediaSection(i),
+                                   local->GetMediaSection(i), *transceiver);
+    NS_ENSURE_SUCCESS(rv, rv);
+    SetTransceiver(*transceiver);
+  }
+
+  CopyBundleTransports();
+
+  std::vector<JsepTrack*> remoteTracks;
+  for (auto& transceiver : mTransceivers) {
+    remoteTracks.push_back(&transceiver.mRecvTrack);
+  }
+  JsepTrack::SetUniquePayloadTypes(remoteTracks);
+
+  mNegotiations++;
+
+  mGeneratedAnswer.reset();
+  mGeneratedOffer.reset();
+
+  return NS_OK;
+}
+
+nsresult JsepSessionImpl::MakeNegotiatedTransceiver(
+    const SdpMediaSection& remote, const SdpMediaSection& local,
+    JsepTransceiver& transceiver) {
+  const SdpMediaSection& answer = *mIsPendingOfferer ? remote : local;
+
+  bool sending = false;
+  bool receiving = false;
+
+  // JS could stop the transceiver after the answer was created.
+  if (!transceiver.IsStopped()) {
+    if (*mIsPendingOfferer) {
+      receiving = answer.IsSending();
+      sending = answer.IsReceiving();
+    } else {
+      sending = answer.IsSending();
+      receiving = answer.IsReceiving();
+    }
+  }
+
+  MOZ_MTLOG(ML_DEBUG, "[" << mName << "]: Negotiated m= line"
+                          << " index=" << local.GetLevel() << " type="
+                          << local.GetMediaType() << " sending=" << sending
+                          << " receiving=" << receiving);
+
+  transceiver.SetNegotiated();
+
+  // Ensure that this is finalized in case we need to copy it below
+  nsresult rv =
+      FinalizeTransport(remote.GetAttributeList(), answer.GetAttributeList(),
+                        &transceiver.mTransport);
+  NS_ENSURE_SUCCESS(rv, rv);
+
+  transceiver.mSendTrack.SetActive(sending);
+  rv = transceiver.mSendTrack.Negotiate(answer, remote, local);
+  if (NS_FAILED(rv)) {
+    JSEP_SET_ERROR("Answer had no codecs in common with offer in m-section "
+                   << local.GetLevel());
+    return rv;
+  }
+
+  JsepTrack& recvTrack = transceiver.mRecvTrack;
+  recvTrack.SetActive(receiving);
+  rv = recvTrack.Negotiate(answer, remote, local);
+  if (NS_FAILED(rv)) {
+    JSEP_SET_ERROR("Answer had no codecs in common with offer in m-section "
+                   << local.GetLevel());
+    return rv;
+  }
+
+  if (transceiver.HasBundleLevel() && recvTrack.GetSsrcs().empty() &&
+      recvTrack.GetMediaType() != SdpMediaSection::kApplication) {
+    // TODO(bug 1105005): Once we have urn:ietf:params:rtp-hdrext:sdes:mid
+    // support, we should only fire this warning if that extension was not
+    // negotiated.
+    MOZ_MTLOG(ML_ERROR, "[" << mName
+                            << "]: Bundled m-section has no ssrc "
+                               "attributes. This may cause media packets to be "
+                               "dropped.");
+  }
+
+  if (transceiver.mTransport.mComponents == 2) {
+    // RTCP MUX or not.
+    // TODO(bug 1095743): verify that the PTs are consistent with mux.
+    MOZ_MTLOG(ML_DEBUG, "[" << mName << "]: RTCP-MUX is off");
+  }
+
+  if (answer.GetAttributeList().HasAttribute(SdpAttribute::kExtmapAttribute)) {
+    const auto extmaps = answer.GetAttributeList().GetExtmap().mExtmaps;
+    for (const auto& negotiatedExtension : extmaps) {
+      if (negotiatedExtension.entry == 0) {
+        MOZ_ASSERT(false, "This should have been caught sooner");
+        continue;
+      }
+
+      mExtmapEntriesEverNegotiated[negotiatedExtension.entry] =
+          negotiatedExtension.extensionname;
+
+      for (auto& originalExtension : mRtpExtensions) {
+        if (negotiatedExtension.extensionname ==
+            originalExtension.mExtmap.extensionname) {
+          // Update extmap to match what was negotiated
+          originalExtension.mExtmap.entry = negotiatedExtension.entry;
+          mExtmapEntriesEverUsed.insert(negotiatedExtension.entry);
+        } else if (originalExtension.mExtmap.entry ==
+                   negotiatedExtension.entry) {
+          // If this extmap entry was claimed for a different extension, update
+          // it to a new value so we don't end up with a duplicate.
+          originalExtension.mExtmap.entry = GetNeverUsedExtmapEntry();
+        }
+      }
+    }
+  }
+
+  return NS_OK;
+}
+
+void JsepSessionImpl::EnsureHasOwnTransport(const SdpMediaSection& msection,
+                                            JsepTransceiver& transceiver) {
+  JsepTransport& transport = transceiver.mTransport;
+
+  if (!transceiver.HasOwnTransport()) {
+    // Transceiver didn't own this transport last time, it won't now either
+    transport.Close();
+  }
+
+  transport.mLocalUfrag = msection.GetAttributeList().GetIceUfrag();
+  transport.mLocalPwd = msection.GetAttributeList().GetIcePwd();
+
+  transceiver.ClearBundleLevel();
+
+  if (!transport.mComponents) {
+    if (mSdpHelper.HasRtcp(msection.GetProtocol())) {
+      transport.mComponents = 2;
+    } else {
+      transport.mComponents = 1;
+    }
+  }
+
+  if (transport.mTransportId.empty()) {
+    // TODO: Once we use different ICE ufrag/pass for each m-section, we can
+    // use that here.
+    std::ostringstream os;
+    os << "transport_" << mTransportIdCounter++;
+    transport.mTransportId = os.str();
+  }
+}
+
+void JsepSessionImpl::CopyBundleTransports() {
+  for (auto& transceiver : mTransceivers) {
+    if (transceiver.HasBundleLevel()) {
+      MOZ_MTLOG(ML_DEBUG,
+                "[" << mName << "] Transceiver " << transceiver.GetLevel()
+                    << " is in a bundle; transceiver "
+                    << transceiver.BundleLevel() << " owns the transport.");
+      Maybe<const JsepTransceiver> transportOwner =
+          GetTransceiverForLevel(transceiver.BundleLevel());
+      MOZ_ASSERT(transportOwner);
+      if (transportOwner) {
+        transceiver.mTransport = transportOwner->mTransport;
+      }
+    } else if (transceiver.HasLevel()) {
+      MOZ_MTLOG(ML_DEBUG, "[" << mName << "] Transceiver "
+                              << transceiver.GetLevel()
+                              << " is not necessarily in a bundle.");
+    }
+    if (transceiver.HasLevel()) {
+      MOZ_MTLOG(ML_DEBUG,
+                "[" << mName << "] Transceiver " << transceiver.GetLevel()
+                    << " transport-id: " << transceiver.mTransport.mTransportId
+                    << " components: " << transceiver.mTransport.mComponents);
+    }
+  }
+}
+
+nsresult JsepSessionImpl::FinalizeTransport(const SdpAttributeList& remote,
+                                            const SdpAttributeList& answer,
+                                            JsepTransport* transport) const {
+  if (!transport->mComponents) {
+    return NS_OK;
+  }
+
+  if (!transport->mIce || transport->mIce->mUfrag != remote.GetIceUfrag() ||
+      transport->mIce->mPwd != remote.GetIcePwd()) {
+    UniquePtr<JsepIceTransport> ice = MakeUnique<JsepIceTransport>();
+    transport->mDtls = nullptr;
+
+    // We do sanity-checking for these in ParseSdp
+    ice->mUfrag = remote.GetIceUfrag();
+    ice->mPwd = remote.GetIcePwd();
+    transport->mIce = std::move(ice);
+  }
+
+  if (remote.HasAttribute(SdpAttribute::kCandidateAttribute)) {
+    transport->mIce->mCandidates = remote.GetCandidate();
+  }
+
+  if (!transport->mDtls) {
+    // RFC 5763 says:
+    //
+    //   The endpoint MUST use the setup attribute defined in [RFC4145].
+    //   The endpoint that is the offerer MUST use the setup attribute
+    //   value of setup:actpass and be prepared to receive a client_hello
+    //   before it receives the answer.  The answerer MUST use either a
+    //   setup attribute value of setup:active or setup:passive.  Note that
+    //   if the answerer uses setup:passive, then the DTLS handshake will
+    //   not begin until the answerer is received, which adds additional
+    //   latency. setup:active allows the answer and the DTLS handshake to
+    //   occur in parallel.  Thus, setup:active is RECOMMENDED.  Whichever
+    //   party is active MUST initiate a DTLS handshake by sending a
+    //   ClientHello over each flow (host/port quartet).
+    UniquePtr<JsepDtlsTransport> dtls = MakeUnique<JsepDtlsTransport>();
+    dtls->mFingerprints = remote.GetFingerprint();
+    if (!answer.HasAttribute(mozilla::SdpAttribute::kSetupAttribute)) {
+      dtls->mRole = *mIsPendingOfferer ? JsepDtlsTransport::kJsepDtlsServer
+                                       : JsepDtlsTransport::kJsepDtlsClient;
+    } else {
+      if (*mIsPendingOfferer) {
+        dtls->mRole = (answer.GetSetup().mRole == SdpSetupAttribute::kActive)
+                          ? JsepDtlsTransport::kJsepDtlsServer
+                          : JsepDtlsTransport::kJsepDtlsClient;
+      } else {
+        dtls->mRole = (answer.GetSetup().mRole == SdpSetupAttribute::kActive)
+                          ? JsepDtlsTransport::kJsepDtlsClient
+                          : JsepDtlsTransport::kJsepDtlsServer;
+      }
+    }
+
+    transport->mDtls = std::move(dtls);
+  }
+
+  if (answer.HasAttribute(SdpAttribute::kRtcpMuxAttribute)) {
+    transport->mComponents = 1;
+  }
+
+  return NS_OK;
+}
+
+nsresult JsepSessionImpl::AddTransportAttributes(
+    SdpMediaSection* msection, SdpSetupAttribute::Role dtlsRole) {
+  if (mIceUfrag.empty() || mIcePwd.empty()) {
+    JSEP_SET_ERROR("Missing ICE ufrag or password");
+    return NS_ERROR_FAILURE;
+  }
+
+  SdpAttributeList& attrList = msection->GetAttributeList();
+  attrList.SetAttribute(
+      new SdpStringAttribute(SdpAttribute::kIceUfragAttribute, mIceUfrag));
+  attrList.SetAttribute(
+      new SdpStringAttribute(SdpAttribute::kIcePwdAttribute, mIcePwd));
+
+  msection->GetAttributeList().SetAttribute(new SdpSetupAttribute(dtlsRole));
+
+  return NS_OK;
+}
+
+nsresult JsepSessionImpl::CopyPreviousTransportParams(
+    const Sdp& oldAnswer, const Sdp& offerersPreviousSdp, const Sdp& newOffer,
+    Sdp* newLocal) {
+  for (size_t i = 0; i < oldAnswer.GetMediaSectionCount(); ++i) {
+    if (!mSdpHelper.MsectionIsDisabled(newLocal->GetMediaSection(i)) &&
+        mSdpHelper.AreOldTransportParamsValid(oldAnswer, offerersPreviousSdp,
+                                              newOffer, i)) {
+      // If newLocal is an offer, this will be the number of components we used
+      // last time, and if it is an answer, this will be the number of
+      // components we've decided we're using now.
+      Maybe<const JsepTransceiver> transceiver(GetTransceiverForLevel(i));
+      if (!transceiver) {
+        MOZ_ASSERT(false);
+        JSEP_SET_ERROR("No transceiver for level " << i);
+        return NS_ERROR_FAILURE;
+      }
+      size_t numComponents = transceiver->mTransport.mComponents;
+      nsresult rv = mSdpHelper.CopyTransportParams(
+          numComponents, mCurrentLocalDescription->GetMediaSection(i),
+          &newLocal->GetMediaSection(i));
+      NS_ENSURE_SUCCESS(rv, rv);
+    }
+  }
+
+  return NS_OK;
+}
+
+nsresult JsepSessionImpl::ParseSdp(const std::string& sdp,
+                                   UniquePtr<Sdp>* parsedp) {
+  auto results = mParser->Parse(sdp);
+  auto parsed = std::move(results->Sdp());
+  mLastSdpParsingErrors = results->Errors();
+  if (!parsed) {
+    std::string error = results->ParserName() + " Failed to parse SDP: ";
+    mSdpHelper.AppendSdpParseErrors(mLastSdpParsingErrors, &error);
+    JSEP_SET_ERROR(error);
+    return NS_ERROR_INVALID_ARG;
+  }
+  // Verify that the JSEP rules for all SDP are followed
+  for (size_t i = 0; i < parsed->GetMediaSectionCount(); ++i) {
+    if (mSdpHelper.MsectionIsDisabled(parsed->GetMediaSection(i))) {
+      // Disabled, let this stuff slide.
+      continue;
+    }
+
+    const SdpMediaSection& msection(parsed->GetMediaSection(i));
+    auto& mediaAttrs = msection.GetAttributeList();
+
+    if (mediaAttrs.HasAttribute(SdpAttribute::kMidAttribute) &&
+        mediaAttrs.GetMid().length() > 16) {
+      JSEP_SET_ERROR(
+          "Invalid description, mid length greater than 16 "
+          "unsupported until 2-byte rtp header extensions are "
+          "supported in webrtc.org");
+      return NS_ERROR_INVALID_ARG;
+    }
+
+    if (mediaAttrs.HasAttribute(SdpAttribute::kExtmapAttribute)) {
+      std::set<uint16_t> extIds;
+      for (const auto& ext : mediaAttrs.GetExtmap().mExtmaps) {
+        uint16_t id = ext.entry;
+
+        if (id < 1 || id > 14) {
+          JSEP_SET_ERROR("Description contains invalid extension id "
+                         << id << " on level " << i
+                         << " which is unsupported until 2-byte rtp"
+                            " header extensions are supported in webrtc.org");
+          return NS_ERROR_INVALID_ARG;
+        }
+
+        if (extIds.find(id) != extIds.end()) {
+          JSEP_SET_ERROR("Description contains duplicate extension id "
+                         << id << " on level " << i);
+          return NS_ERROR_INVALID_ARG;
+        }
+        extIds.insert(id);
+      }
+    }
+
+    static const std::bitset<128> forbidden = GetForbiddenSdpPayloadTypes();
+    if (msection.GetMediaType() == SdpMediaSection::kAudio ||
+        msection.GetMediaType() == SdpMediaSection::kVideo) {
+      // Sanity-check that payload type can work with RTP
+      for (const std::string& fmt : msection.GetFormats()) {
+        uint16_t payloadType;
+        if (!SdpHelper::GetPtAsInt(fmt, &payloadType)) {
+          JSEP_SET_ERROR("Payload type \""
+                         << fmt << "\" is not a 16-bit unsigned int at level "
+                         << i);
+          return NS_ERROR_INVALID_ARG;
+        }
+        if (payloadType > 127) {
+          JSEP_SET_ERROR("audio/video payload type \""
+                         << fmt << "\" is too large at level " << i);
+          return NS_ERROR_INVALID_ARG;
+        }
+        if (forbidden.test(payloadType)) {
+          JSEP_SET_ERROR("Illegal audio/video payload type \""
+                         << fmt << "\" at level " << i);
+          return NS_ERROR_INVALID_ARG;
+        }
+      }
+    }
+  }
+
+  *parsedp = std::move(parsed);
+  return NS_OK;
+}
+
+nsresult JsepSessionImpl::SetRemoteDescriptionOffer(UniquePtr<Sdp> offer) {
+  MOZ_ASSERT(mState == kJsepStateStable);
+
+  mPendingRemoteDescription = std::move(offer);
+  mIsPendingOfferer = Some(false);
+
+  SetState(kJsepStateHaveRemoteOffer);
+  return NS_OK;
+}
+
+nsresult JsepSessionImpl::SetRemoteDescriptionAnswer(JsepSdpType type,
+                                                     UniquePtr<Sdp> answer) {
+  MOZ_ASSERT(mState == kJsepStateHaveLocalOffer ||
+             mState == kJsepStateHaveRemotePranswer);
+
+  mPendingRemoteDescription = std::move(answer);
+
+  nsresult rv = HandleNegotiatedSession(mPendingLocalDescription,
+                                        mPendingRemoteDescription);
+  NS_ENSURE_SUCCESS(rv, rv);
+
+  mCurrentRemoteDescription = std::move(mPendingRemoteDescription);
+  mCurrentLocalDescription = std::move(mPendingLocalDescription);
+  MOZ_ASSERT(mIsPendingOfferer.isSome() && *mIsPendingOfferer);
+  mIsPendingOfferer.reset();
+  mIsCurrentOfferer = Some(true);
+
+  SetState(kJsepStateStable);
+  return NS_OK;
+}
+
+Maybe<JsepTransceiver> JsepSessionImpl::GetTransceiverForLevel(
+    size_t level) const {
+  return FindTransceiver([level](const JsepTransceiver& transceiver) {
+    return transceiver.HasLevel() && (transceiver.GetLevel() == level);
+  });
+}
+
+Maybe<JsepTransceiver> JsepSessionImpl::GetTransceiverForMid(
+    const std::string& mid) const {
+  return FindTransceiver([mid](const JsepTransceiver& transceiver) {
+    return transceiver.IsAssociated() && (transceiver.GetMid() == mid);
+  });
+}
+
+Maybe<JsepTransceiver> JsepSessionImpl::GetTransceiverForLocal(size_t level) {
+  if (Maybe<JsepTransceiver> transceiver = GetTransceiverForLevel(level)) {
+    if (transceiver->CanRecycle() &&
+        transceiver->GetMediaType() != SdpMediaSection::kApplication) {
+      // Attempt to recycle. If this fails, the old transceiver stays put.
+      transceiver->Disassociate();
+      Maybe<JsepTransceiver> newTransceiver =
+          FindUnassociatedTransceiver(transceiver->GetMediaType(), false);
+      if (newTransceiver) {
+        newTransceiver->SetLevel(level);
+        transceiver->ClearLevel();
+        transceiver->mSendTrack.ClearRids();
+        SetTransceiver(*newTransceiver);
+        SetTransceiver(*transceiver);
+        return newTransceiver;
+      }
+    }
+
+    SetTransceiver(*transceiver);
+    return transceiver;
+  }
+
+  // There is no transceiver for |level| right now.
+
+  // Look for an RTP transceiver
+  for (auto& transceiver : mTransceivers) {
+    if (transceiver.GetMediaType() != SdpMediaSection::kApplication &&
+        !transceiver.IsStopped() && !transceiver.HasLevel()) {
+      transceiver.SetLevel(level);
+      return Some(transceiver);
+    }
+  }
+
+  // Ok, look for a datachannel
+  for (auto& transceiver : mTransceivers) {
+    if (!transceiver.IsStopped() && !transceiver.HasLevel()) {
+      transceiver.SetLevel(level);
+      return Some(transceiver);
+    }
+  }
+
+  return Nothing();
+}
+
+Maybe<JsepTransceiver> JsepSessionImpl::GetTransceiverForRemote(
+    const SdpMediaSection& msection) {
+  size_t level = msection.GetLevel();
+  Maybe<JsepTransceiver> transceiver = GetTransceiverForLevel(level);
+  if (transceiver) {
+    if (!transceiver->CanRecycle()) {
+      return transceiver;
+    }
+    transceiver->Disassociate();
+    transceiver->ClearLevel();
+    transceiver->mSendTrack.ClearRids();
+    SetTransceiver(*transceiver);
+  }
+
+  // No transceiver for |level|
+  transceiver = FindUnassociatedTransceiver(msection.GetMediaType(), true);
+  if (transceiver) {
+    transceiver->SetLevel(level);
+    SetTransceiver(*transceiver);
+    return transceiver;
+  }
+
+  // Make a new transceiver
+  JsepTransceiver newTransceiver(msection.GetMediaType(), *mUuidGen,
+                                 SdpDirectionAttribute::kRecvonly);
+  newTransceiver.SetLevel(level);
+  newTransceiver.SetOnlyExistsBecauseOfSetRemote(true);
+  AddTransceiver(newTransceiver);
+  return Some(mTransceivers.back());
+}
+
+Maybe<JsepTransceiver> JsepSessionImpl::GetTransceiverWithTransport(
+    const std::string& transportId) const {
+  for (const auto& transceiver : mTransceivers) {
+    if (transceiver.HasOwnTransport() &&
+        (transceiver.mTransport.mTransportId == transportId)) {
+      MOZ_ASSERT(transceiver.HasLevel(),
+                 "Transceiver has a transport, but no level!");
+      return Some(transceiver);
+    }
+  }
+
+  return Nothing();
+}
+
+nsresult JsepSessionImpl::UpdateTransceiversFromRemoteDescription(
+    const Sdp& remote) {
+  // Iterate over the sdp, updating remote tracks as we go
+  for (size_t i = 0; i < remote.GetMediaSectionCount(); ++i) {
+    const SdpMediaSection& msection = remote.GetMediaSection(i);
+
+    Maybe<JsepTransceiver> transceiver(GetTransceiverForRemote(msection));
+    if (!transceiver) {
+      return NS_ERROR_FAILURE;
+    }
+
+    if (!mSdpHelper.MsectionIsDisabled(msection)) {
+      if (msection.GetAttributeList().HasAttribute(
+              SdpAttribute::kMidAttribute)) {
+        transceiver->Associate(msection.GetAttributeList().GetMid());
+      }
+      if (!transceiver->IsAssociated()) {
+        transceiver->Associate(GetNewMid());
+      } else {
+        mUsedMids.insert(transceiver->GetMid());
+      }
+    } else {
+      transceiver->mTransport.Close();
+      transceiver->Disassociate();
+      // This cannot be rolled back.
+      transceiver->Stop();
+      SetTransceiver(*transceiver);
+      continue;
+    }
+
+    if (msection.GetMediaType() == SdpMediaSection::MediaType::kApplication) {
+      SetTransceiver(*transceiver);
+      continue;
+    }
+
+    transceiver->mSendTrack.SendTrackSetRemote(mSsrcGenerator, msection);
+
+    // Interop workaround for endpoints that don't support msid.
+    // Ensures that there is a default stream id set, provided the remote is
+    // sending.
+    // TODO(bug 1426005): Remove this, or at least move it to JsepTrack.
+    transceiver->mRecvTrack.UpdateStreamIds({mDefaultRemoteStreamId});
+
+    // This will process a=msid if present, or clear the stream ids if the
+    // msection is not sending. If the msection is sending, and there are no
+    // a=msid, the previously set default will stay.
+    transceiver->mRecvTrack.RecvTrackSetRemote(remote, msection);
+    SetTransceiver(*transceiver);
+  }
+
+  return NS_OK;
+}
+
+Maybe<JsepTransceiver> JsepSessionImpl::FindUnassociatedTransceiver(
+    SdpMediaSection::MediaType type, bool magic) {
+  // Look through transceivers that are not mapped to an m-section
+  for (auto& transceiver : mTransceivers) {
+    if (type == SdpMediaSection::kApplication &&
+        type == transceiver.GetMediaType()) {
+      transceiver.RestartDatachannelTransceiver();
+      return Some(transceiver);
+    }
+    if (!transceiver.IsStopped() && !transceiver.HasLevel() &&
+        (!magic || transceiver.HasAddTrackMagic()) &&
+        (transceiver.GetMediaType() == type)) {
+      return Some(transceiver);
+    }
+  }
+
+  return Nothing();
+}
+
+void JsepSessionImpl::RollbackLocalOffer() {
+  for (size_t i = 0; i < mTransceivers.size(); ++i) {
+    auto& transceiver = mTransceivers[i];
+    if (mOldTransceivers.size() > i) {
+      transceiver.Rollback(mOldTransceivers[i], false);
+      mOldTransceivers[i] = transceiver;
+      continue;
+    }
+
+    JsepTransceiver temp(transceiver.GetMediaType(), *mUuidGen);
+    InitTransceiver(temp);
+    transceiver.Rollback(temp, false);
+    mOldTransceivers.push_back(transceiver);
+  }
+
+  mTransceivers = std::move(mOldTransceivers);
+}
+
+void JsepSessionImpl::RollbackRemoteOffer() {
+  for (size_t i = 0; i < mTransceivers.size(); ++i) {
+    auto& transceiver = mTransceivers[i];
+    if (mOldTransceivers.size() > i) {
+      // Some stuff cannot be rolled back. Save this information.
+      transceiver.Rollback(mOldTransceivers[i], true);
+      mOldTransceivers[i] = transceiver;
+      continue;
+    }
+
+    // New transceiver!
+    // We rollback even for transceivers we will remove, just to ensure we end
+    // up at the starting state.
+    JsepTransceiver temp(transceiver.GetMediaType(), *mUuidGen);
+    InitTransceiver(temp);
+    transceiver.Rollback(temp, true);
+
+    if (transceiver.OnlyExistsBecauseOfSetRemote()) {
+      transceiver.Stop();
+      transceiver.SetRemoved();
+    }
+    mOldTransceivers.push_back(transceiver);
+  }
+
+  mTransceivers = std::move(mOldTransceivers);
+}
+
+nsresult JsepSessionImpl::ValidateLocalDescription(const Sdp& description,
+                                                   JsepSdpType type) {
+  Sdp* generated = nullptr;
+  // TODO(bug 1095226): Better checking.
+  if (type == kJsepSdpOffer) {
+    generated = mGeneratedOffer.get();
+  } else {
+    generated = mGeneratedAnswer.get();
+  }
+
+  if (!generated) {
+    JSEP_SET_ERROR(
+        "Calling SetLocal without first calling CreateOffer/Answer"
+        " is not supported.");
+    return NS_ERROR_UNEXPECTED;
+  }
+
+  if (description.GetMediaSectionCount() != generated->GetMediaSectionCount()) {
+    JSEP_SET_ERROR("Changing the number of m-sections is not allowed");
+    return NS_ERROR_INVALID_ARG;
+  }
+
+  for (size_t i = 0; i < description.GetMediaSectionCount(); ++i) {
+    auto& origMsection = generated->GetMediaSection(i);
+    auto& finalMsection = description.GetMediaSection(i);
+    if (origMsection.GetMediaType() != finalMsection.GetMediaType()) {
+      JSEP_SET_ERROR("Changing the media-type of m-sections is not allowed");
+      return NS_ERROR_INVALID_ARG;
+    }
+
+    // These will be present in reoffer
+    if (!mCurrentLocalDescription) {
+      if (finalMsection.GetAttributeList().HasAttribute(
+              SdpAttribute::kCandidateAttribute)) {
+        JSEP_SET_ERROR("Adding your own candidate attributes is not supported");
+        return NS_ERROR_INVALID_ARG;
+      }
+
+      if (finalMsection.GetAttributeList().HasAttribute(
+              SdpAttribute::kEndOfCandidatesAttribute)) {
+        JSEP_SET_ERROR("Why are you trying to set a=end-of-candidates?");
+        return NS_ERROR_INVALID_ARG;
+      }
+    }
+
+    if (mSdpHelper.MsectionIsDisabled(finalMsection)) {
+      continue;
+    }
+
+    if (!finalMsection.GetAttributeList().HasAttribute(
+            SdpAttribute::kMidAttribute)) {
+      JSEP_SET_ERROR("Local descriptions must have a=mid attributes.");
+      return NS_ERROR_INVALID_ARG;
+    }
+
+    if (finalMsection.GetAttributeList().GetMid() !=
+        origMsection.GetAttributeList().GetMid()) {
+      JSEP_SET_ERROR("Changing the mid of m-sections is not allowed.");
+      return NS_ERROR_INVALID_ARG;
+    }
+
+    // TODO(bug 1095218): Check msid
+    // TODO(bug 1095226): Check ice-ufrag and ice-pwd
+    // TODO(bug 1095226): Check fingerprints
+    // TODO(bug 1095226): Check payload types (at least ensure that payload
+    // types we don't actually support weren't added)
+    // TODO(bug 1095226): Check ice-options?
+  }
+
+  if (description.GetAttributeList().HasAttribute(
+          SdpAttribute::kIceLiteAttribute)) {
+    JSEP_SET_ERROR("Running ICE in lite mode is unsupported");
+    return NS_ERROR_INVALID_ARG;
+  }
+
+  return NS_OK;
+}
+
+nsresult JsepSessionImpl::ValidateRemoteDescription(const Sdp& description) {
+  if (!mCurrentLocalDescription) {
+    // Initial offer; nothing to validate besides the stuff in ParseSdp
+    return NS_OK;
+  }
+
+  if (mCurrentLocalDescription->GetMediaSectionCount() >
+      description.GetMediaSectionCount()) {
+    JSEP_SET_ERROR(
+        "New remote description has fewer m-sections than the "
+        "previous remote description.");
+    return NS_ERROR_INVALID_ARG;
+  }
+
+  for (size_t i = 0; i < description.GetMediaSectionCount(); ++i) {
+    const SdpAttributeList& attrs =
+        description.GetMediaSection(i).GetAttributeList();
+
+    if (attrs.HasAttribute(SdpAttribute::kExtmapAttribute)) {
+      for (const auto& ext : attrs.GetExtmap().mExtmaps) {
+        if (mExtmapEntriesEverNegotiated.count(ext.entry) &&
+            mExtmapEntriesEverNegotiated[ext.entry] != ext.extensionname) {
+          JSEP_SET_ERROR(
+              "Remote description attempted to remap RTP extension id "
+              << ext.entry << " from "
+              << mExtmapEntriesEverNegotiated[ext.entry] << " to "
+              << ext.extensionname);
+          return NS_ERROR_INVALID_ARG;
+        }
+      }
+    }
+  }
+
+  if (!mCurrentRemoteDescription) {
+    // No further checking for initial answers
+    return NS_OK;
+  }
+
+  // These are solely to check that bundle is valid
+  SdpHelper::BundledMids bundledMids;
+  nsresult rv = GetNegotiatedBundledMids(&bundledMids);
+  NS_ENSURE_SUCCESS(rv, rv);
+
+  SdpHelper::BundledMids newBundledMids;
+  rv = mSdpHelper.GetBundledMids(description, &newBundledMids);
+  NS_ENSURE_SUCCESS(rv, rv);
+
+  // check for partial ice restart, which is not supported
+  Maybe<bool> iceCredsDiffer;
+  for (size_t i = 0; i < mCurrentRemoteDescription->GetMediaSectionCount();
+       ++i) {
+    const SdpMediaSection& newMsection = description.GetMediaSection(i);
+    const SdpMediaSection& oldMsection =
+        mCurrentRemoteDescription->GetMediaSection(i);
+
+    if (mSdpHelper.MsectionIsDisabled(newMsection) ||
+        mSdpHelper.MsectionIsDisabled(oldMsection)) {
+      continue;
+    }
+
+    if (oldMsection.GetMediaType() != newMsection.GetMediaType()) {
+      JSEP_SET_ERROR("Remote description changes the media type of m-line "
+                     << i);
+      return NS_ERROR_INVALID_ARG;
+    }
+
+    bool differ = mSdpHelper.IceCredentialsDiffer(newMsection, oldMsection);
+
+    if (mIsPendingOfferer.isSome() && *mIsPendingOfferer && differ &&
+        !IsIceRestarting()) {
+      JSEP_SET_ERROR(
+          "Remote description indicates ICE restart but offer did not "
+          "request ICE restart (new remote description changes either "
+          "the ice-ufrag or ice-pwd)");
+      return NS_ERROR_INVALID_ARG;
+    }
+
+    // Detect whether all the creds are the same or all are different
+    if (!iceCredsDiffer.isSome()) {
+      // for the first msection capture whether creds are different or same
+      iceCredsDiffer = mozilla::Some(differ);
+    } else if (iceCredsDiffer.isSome() && *iceCredsDiffer != differ) {
+      // subsequent msections must match the first sections
+      JSEP_SET_ERROR(
+          "Partial ICE restart is unsupported at this time "
+          "(new remote description changes either the ice-ufrag "
+          "or ice-pwd on fewer than all msections)");
+      return NS_ERROR_INVALID_ARG;
+    }
+  }
+
+  return NS_OK;
+}
+
+nsresult JsepSessionImpl::ValidateOffer(const Sdp& offer) {
+  return mSdpHelper.ValidateTransportAttributes(offer, sdp::kOffer);
+}
+
+nsresult JsepSessionImpl::ValidateAnswer(const Sdp& offer, const Sdp& answer) {
+  if (offer.GetMediaSectionCount() != answer.GetMediaSectionCount()) {
+    JSEP_SET_ERROR("Offer and answer have different number of m-lines "
+                   << "(" << offer.GetMediaSectionCount() << " vs "
+                   << answer.GetMediaSectionCount() << ")");
+    return NS_ERROR_INVALID_ARG;
+  }
+
+  nsresult rv = mSdpHelper.ValidateTransportAttributes(answer, sdp::kAnswer);
+  NS_ENSURE_SUCCESS(rv, rv);
+
+  for (size_t i = 0; i < offer.GetMediaSectionCount(); ++i) {
+    const SdpMediaSection& offerMsection = offer.GetMediaSection(i);
+    const SdpMediaSection& answerMsection = answer.GetMediaSection(i);
+
+    if (offerMsection.GetMediaType() != answerMsection.GetMediaType()) {
+      JSEP_SET_ERROR("Answer and offer have different media types at m-line "
+                     << i);
+      return NS_ERROR_INVALID_ARG;
+    }
+
+    if (mSdpHelper.MsectionIsDisabled(answerMsection)) {
+      continue;
+    }
+
+    if (mSdpHelper.MsectionIsDisabled(offerMsection)) {
+      JSEP_SET_ERROR(
+          "Answer tried to enable an m-section that was disabled in the offer");
+      return NS_ERROR_INVALID_ARG;
+    }
+
+    if (!offerMsection.IsSending() && answerMsection.IsReceiving()) {
+      JSEP_SET_ERROR("Answer tried to set recv when offer did not set send");
+      return NS_ERROR_INVALID_ARG;
+    }
+
+    if (!offerMsection.IsReceiving() && answerMsection.IsSending()) {
+      JSEP_SET_ERROR("Answer tried to set send when offer did not set recv");
+      return NS_ERROR_INVALID_ARG;
+    }
+
+    const SdpAttributeList& answerAttrs(answerMsection.GetAttributeList());
+    const SdpAttributeList& offerAttrs(offerMsection.GetAttributeList());
+    if (answerAttrs.HasAttribute(SdpAttribute::kMidAttribute) &&
+        offerAttrs.HasAttribute(SdpAttribute::kMidAttribute) &&
+        offerAttrs.GetMid() != answerAttrs.GetMid()) {
+      JSEP_SET_ERROR("Answer changes mid for level, was \'"
+                     << offerMsection.GetAttributeList().GetMid()
+                     << "\', now \'"
+                     << answerMsection.GetAttributeList().GetMid() << "\'");
+      return NS_ERROR_INVALID_ARG;
+    }
+
+    // Sanity check extmap
+    if (answerAttrs.HasAttribute(SdpAttribute::kExtmapAttribute)) {
+      if (!offerAttrs.HasAttribute(SdpAttribute::kExtmapAttribute)) {
+        JSEP_SET_ERROR("Answer adds extmap attributes to level " << i);
+        return NS_ERROR_INVALID_ARG;
+      }
+
+      for (const auto& ansExt : answerAttrs.GetExtmap().mExtmaps) {
+        bool found = false;
+        for (const auto& offExt : offerAttrs.GetExtmap().mExtmaps) {
+          if (ansExt.extensionname == offExt.extensionname) {
+            if ((ansExt.direction & reverse(offExt.direction)) !=
+                ansExt.direction) {
+              // FIXME we do not return an error here, because Chrome up to
+              // version 57 is actually tripping over this if they are the
+              // answerer. See bug 1355010 for details.
+              MOZ_MTLOG(ML_WARNING,
+                        "[" << mName
+                            << "]: Answer has inconsistent"
+                               " direction on extmap attribute at level "
+                            << i << " (" << ansExt.extensionname
+                            << "). Offer had " << offExt.direction
+                            << ", answer had " << ansExt.direction << ".");
+              // return NS_ERROR_INVALID_ARG;
+            }
+
+            if (offExt.entry < 4096 && (offExt.entry != ansExt.entry)) {
+              JSEP_SET_ERROR("Answer changed id for extmap attribute at level "
+                             << i << " (" << offExt.extensionname << ") from "
+                             << offExt.entry << " to " << ansExt.entry << ".");
+              return NS_ERROR_INVALID_ARG;
+            }
+
+            if (ansExt.entry >= 4096) {
+              JSEP_SET_ERROR("Answer used an invalid id ("
+                             << ansExt.entry
+                             << ") for extmap attribute at level " << i << " ("
+                             << ansExt.extensionname << ").");
+              return NS_ERROR_INVALID_ARG;
+            }
+
+            found = true;
+            break;
+          }
+        }
+
+        if (!found) {
+          JSEP_SET_ERROR("Answer has extmap "
+                         << ansExt.extensionname
+                         << " at "
+                            "level "
+                         << i << " that was not present in offer.");
+          return NS_ERROR_INVALID_ARG;
+        }
+      }
+    }
+  }
+
+  return NS_OK;
+}
+
+nsresult JsepSessionImpl::CreateGenericSDP(UniquePtr<Sdp>* sdpp) {
+  // draft-ietf-rtcweb-jsep-08 Section 5.2.1:
+  //  o  The second SDP line MUST be an "o=" line, as specified in
+  //     [RFC4566], Section 5.2.  The value of the <username> field SHOULD
+  //     be "-".  The value of the <sess-id> field SHOULD be a
+  //     cryptographically random number.  To ensure uniqueness, this
+  //     number SHOULD be at least 64 bits long.  The value of the <sess-
+  //     version> field SHOULD be zero.  The value of the <nettype>
+  //     <addrtype> <unicast-address> tuple SHOULD be set to a non-
+  //     meaningful address, such as IN IP4 0.0.0.0, to prevent leaking the
+  //     local address in this field.  As mentioned in [RFC4566], the
+  //     entire o= line needs to be unique, but selecting a random number
+  //     for <sess-id> is sufficient to accomplish this.
+  //
+  // Historical note: we used to report the actual version number here, after
+  // "SDPARTA-", but that becomes a problem starting with version 100, since
+  // some services parse 100 as "10" and give us legacy/broken behavior. So
+  // we're freezing the version number at 99.0 in this string.
+  auto origin = SdpOrigin("mozilla...THIS_IS_SDPARTA-99.0", mSessionId,
+                          mSessionVersion, sdp::kIPv4, "0.0.0.0");
+
+  UniquePtr<Sdp> sdp = MakeUnique<SipccSdp>(origin);
+
+  if (mDtlsFingerprints.empty()) {
+    JSEP_SET_ERROR("Missing DTLS fingerprint");
+    return NS_ERROR_FAILURE;
+  }
+
+  UniquePtr<SdpFingerprintAttributeList> fpl =
+      MakeUnique<SdpFingerprintAttributeList>();
+  for (auto& dtlsFingerprint : mDtlsFingerprints) {
+    fpl->PushEntry(dtlsFingerprint.mAlgorithm, dtlsFingerprint.mValue);
+  }
+  sdp->GetAttributeList().SetAttribute(fpl.release());
+
+  auto* iceOpts = new SdpOptionsAttribute(SdpAttribute::kIceOptionsAttribute);
+  iceOpts->PushEntry("trickle");
+  sdp->GetAttributeList().SetAttribute(iceOpts);
+
+  // This assumes content doesn't add a bunch of msid attributes with a
+  // different semantic in mind.
+  std::vector<std::string> msids;
+  msids.push_back("*");
+  mSdpHelper.SetupMsidSemantic(msids, sdp.get());
+
+  *sdpp = std::move(sdp);
+  return NS_OK;
+}
+
+nsresult JsepSessionImpl::SetupIds() {
+  SECStatus rv = PK11_GenerateRandom(
+      reinterpret_cast<unsigned char*>(&mSessionId), sizeof(mSessionId));
+  // RFC 3264 says that session-ids MUST be representable as a _signed_
+  // 64 bit number, meaning the MSB cannot be set.
+  mSessionId = mSessionId >> 1;
+  if (rv != SECSuccess) {
+    JSEP_SET_ERROR("Failed to generate session id: " << rv);
+    return NS_ERROR_FAILURE;
+  }
+
+  if (!mUuidGen->Generate(&mDefaultRemoteStreamId)) {
+    JSEP_SET_ERROR("Failed to generate default uuid for streams");
+    return NS_ERROR_FAILURE;
+  }
+
+  if (!mUuidGen->Generate(&mCNAME)) {
+    JSEP_SET_ERROR("Failed to generate CNAME");
+    return NS_ERROR_FAILURE;
+  }
+
+  return NS_OK;
+}
+
+void JsepSessionImpl::SetDefaultCodecs(
+    const std::vector<UniquePtr<JsepCodecDescription>>& aPreferredCodecs) {
+  mSupportedCodecs.clear();
+
+  for (const auto& codec : aPreferredCodecs) {
+    mSupportedCodecs.emplace_back(codec->Clone());
+  }
+}
+
+void JsepSessionImpl::SetState(JsepSignalingState state) {
+  if (state == mState) return;
+
+  MOZ_MTLOG(ML_NOTICE, "[" << mName << "]: " << GetStateStr(mState) << " -> "
+                           << GetStateStr(state));
+  mState = state;
+}
+
+JsepSession::Result JsepSessionImpl::AddRemoteIceCandidate(
+    const std::string& candidate, const std::string& mid,
+    const Maybe<uint16_t>& level, const std::string& ufrag,
+    std::string* transportId) {
+  mLastError.clear();
+  if (!mCurrentRemoteDescription && !mPendingRemoteDescription) {
+    JSEP_SET_ERROR("Cannot add ICE candidate when there is no remote SDP");
+    return dom::PCError::InvalidStateError;
+  }
+
+  if (mid.empty() && !level.isSome() && candidate.empty()) {
+    // Set end-of-candidates on SDP
+    if (mCurrentRemoteDescription) {
+      nsresult rv = mSdpHelper.SetIceGatheringComplete(
+          mCurrentRemoteDescription.get(), ufrag);
+      NS_ENSURE_SUCCESS(rv, dom::PCError::OperationError);
+    }
+
+    if (mPendingRemoteDescription) {
+      // If we had an error when adding the candidate to the current
+      // description, we stomp them here. This is deliberate.
+      nsresult rv = mSdpHelper.SetIceGatheringComplete(
+          mPendingRemoteDescription.get(), ufrag);
+      NS_ENSURE_SUCCESS(rv, dom::PCError::OperationError);
+    }
+    return Result();
+  }
+
+  Maybe<JsepTransceiver> transceiver;
+  if (!mid.empty()) {
+    transceiver = GetTransceiverForMid(mid);
+  } else if (level.isSome()) {
+    transceiver = GetTransceiverForLevel(level.value());
+  }
+
+  if (!transceiver) {
+    JSEP_SET_ERROR("Cannot set ICE candidate for level="
+                   << level << " mid=" << mid << ": No such transceiver.");
+    return dom::PCError::OperationError;
+  }
+
+  if (level.isSome() && transceiver->GetLevel() != level.value()) {
+    MOZ_MTLOG(ML_WARNING, "Mismatch between mid and level - \""
+                              << mid << "\" is not the mid for level "
+                              << level);
+  }
+
+  *transportId = transceiver->mTransport.mTransportId;
+  nsresult rv = NS_ERROR_UNEXPECTED;
+  if (mCurrentRemoteDescription) {
+    rv =
+        mSdpHelper.AddCandidateToSdp(mCurrentRemoteDescription.get(), candidate,
+                                     transceiver->GetLevel(), ufrag);
+  }
+
+  if (mPendingRemoteDescription) {
+    // If we had an error when adding the candidate to the current description,
+    // we stomp them here. This is deliberate.
+    rv =
+        mSdpHelper.AddCandidateToSdp(mPendingRemoteDescription.get(), candidate,
+                                     transceiver->GetLevel(), ufrag);
+  }
+
+  NS_ENSURE_SUCCESS(rv, dom::PCError::OperationError);
+  return Result();
+}
+
+nsresult JsepSessionImpl::AddLocalIceCandidate(const std::string& candidate,
+                                               const std::string& transportId,
+                                               const std::string& ufrag,
+                                               uint16_t* level,
+                                               std::string* mid,
+                                               bool* skipped) {
+  mLastError.clear();
+  *skipped = true;
+  if (!mCurrentLocalDescription && !mPendingLocalDescription) {
+    JSEP_SET_ERROR("Cannot add ICE candidate when there is no local SDP");
+    return NS_ERROR_UNEXPECTED;
+  }
+
+  Maybe<const JsepTransceiver> transceiver =
+      GetTransceiverWithTransport(transportId);
+  if (!transceiver || !transceiver->IsAssociated()) {
+    // mainly here to make some testing less complicated, but also just in case
+    return NS_OK;
+  }
+
+  *level = transceiver->GetLevel();
+  *mid = transceiver->GetMid();
+
+  nsresult rv = NS_ERROR_INVALID_ARG;
+  if (mCurrentLocalDescription) {
+    rv = mSdpHelper.AddCandidateToSdp(mCurrentLocalDescription.get(), candidate,
+                                      *level, ufrag);
+  }
+
+  if (mPendingLocalDescription) {
+    // If we had an error when adding the candidate to the current description,
+    // we stomp them here. This is deliberate.
+    rv = mSdpHelper.AddCandidateToSdp(mPendingLocalDescription.get(), candidate,
+                                      *level, ufrag);
+  }
+
+  *skipped = false;
+  return rv;
+}
+
+nsresult JsepSessionImpl::UpdateDefaultCandidate(
+    const std::string& defaultCandidateAddr, uint16_t defaultCandidatePort,
+    const std::string& defaultRtcpCandidateAddr,
+    uint16_t defaultRtcpCandidatePort, const std::string& transportId) {
+  mLastError.clear();
+
+  mozilla::Sdp* sdp =
+      GetParsedLocalDescription(kJsepDescriptionPendingOrCurrent);
+
+  if (!sdp) {
+    JSEP_SET_ERROR("Cannot add ICE candidate in state " << GetStateStr(mState));
+    return NS_ERROR_UNEXPECTED;
+  }
+
+  for (const auto& transceiver : mTransceivers) {
+    // We set the default address for bundled m-sections, but not candidate
+    // attributes. Ugh.
+    if (transceiver.mTransport.mTransportId == transportId) {
+      MOZ_ASSERT(transceiver.HasLevel(),
+                 "Transceiver has a transport, but no level! "
+                 "This should never happen.");
+      std::string defaultRtcpCandidateAddrCopy(defaultRtcpCandidateAddr);
+      if (mState == kJsepStateStable) {
+        if (transceiver.mTransport.mComponents == 1) {
+          // We know we're doing rtcp-mux by now. Don't create an rtcp attr.
+          defaultRtcpCandidateAddrCopy = "";
+          defaultRtcpCandidatePort = 0;
+        }
+      }
+
+      size_t level = transceiver.GetLevel();
+      if (level >= sdp->GetMediaSectionCount()) {
+        MOZ_ASSERT(false, "Transceiver's level is too large!");
+        JSEP_SET_ERROR("Transceiver's level is too large!");
+        return NS_ERROR_FAILURE;
+      }
+
+      auto& msection = sdp->GetMediaSection(level);
+
+      // Do not add default candidate to a bundle-only m-section, sinice that
+      // might confuse endpoints that do not support bundle-only.
+      if (!msection.GetAttributeList().HasAttribute(
+              SdpAttribute::kBundleOnlyAttribute)) {
+        mSdpHelper.SetDefaultAddresses(
+            defaultCandidateAddr, defaultCandidatePort,
+            defaultRtcpCandidateAddrCopy, defaultRtcpCandidatePort, &msection);
+      }
+    }
+  }
+
+  return NS_OK;
+}
+
+nsresult JsepSessionImpl::GetNegotiatedBundledMids(
+    SdpHelper::BundledMids* bundledMids) {
+  const Sdp* answerSdp = GetAnswer();
+
+  if (!answerSdp) {
+    return NS_OK;
+  }
+
+  return mSdpHelper.GetBundledMids(*answerSdp, bundledMids);
+}
+
+mozilla::Sdp* JsepSessionImpl::GetParsedLocalDescription(
+    JsepDescriptionPendingOrCurrent type) const {
+  if (type == kJsepDescriptionPending) {
+    return mPendingLocalDescription.get();
+  } else if (mPendingLocalDescription &&
+             type == kJsepDescriptionPendingOrCurrent) {
+    return mPendingLocalDescription.get();
+  }
+  return mCurrentLocalDescription.get();
+}
+
+mozilla::Sdp* JsepSessionImpl::GetParsedRemoteDescription(
+    JsepDescriptionPendingOrCurrent type) const {
+  if (type == kJsepDescriptionPending) {
+    return mPendingRemoteDescription.get();
+  } else if (mPendingRemoteDescription &&
+             type == kJsepDescriptionPendingOrCurrent) {
+    return mPendingRemoteDescription.get();
+  }
+  return mCurrentRemoteDescription.get();
+}
+
+const Sdp* JsepSessionImpl::GetAnswer() const {
+  return (mIsCurrentOfferer.isSome() && *mIsCurrentOfferer)
+             ? mCurrentRemoteDescription.get()
+             : mCurrentLocalDescription.get();
+}
+
+void JsepSessionImpl::SetIceRestarting(bool restarting) {
+  if (restarting) {
+    // not restarting -> restarting
+    if (!IsIceRestarting()) {
+      // We don't set this more than once, so the old ufrag/pwd is preserved
+      // even if we CreateOffer({iceRestart:true}) multiple times in a row.
+      mOldIceUfrag = mIceUfrag;
+      mOldIcePwd = mIcePwd;
+    }
+    mIceUfrag = GetRandomHex(1);
+    mIcePwd = GetRandomHex(4);
+  } else if (IsIceRestarting()) {
+    // restarting -> not restarting, restore old ufrag/pwd
+    mIceUfrag = mOldIceUfrag;
+    mIcePwd = mOldIcePwd;
+    mOldIceUfrag.clear();
+    mOldIcePwd.clear();
+  }
+}
+
+nsresult JsepSessionImpl::Close() {
+  mLastError.clear();
+  SetState(kJsepStateClosed);
+  return NS_OK;
+}
+
+const std::string JsepSessionImpl::GetLastError() const { return mLastError; }
+
+const std::vector<std::pair<size_t, std::string>>&
+JsepSessionImpl::GetLastSdpParsingErrors() const {
+  return mLastSdpParsingErrors;
+}
+
+bool JsepSessionImpl::CheckNegotiationNeeded() const {
+  MOZ_ASSERT(mState == kJsepStateStable);
+
+  for (const auto& transceiver : mTransceivers) {
+    if (transceiver.IsStopped()) {
+      if (transceiver.IsAssociated()) {
+        MOZ_MTLOG(ML_DEBUG, "[" << mName
+                                << "]: Negotiation needed because of "
+                                   "stopped transceiver that still has a mid.");
+        return true;
+      }
+      continue;
+    }
+
+    if (!transceiver.IsAssociated()) {
+      MOZ_MTLOG(ML_DEBUG, "[" << mName
+                              << "]: Negotiation needed because of "
+                                 "unassociated (but not stopped) transceiver.");
+      return true;
+    }
+
+    if (!mCurrentLocalDescription || !mCurrentRemoteDescription) {
+      MOZ_CRASH(
+          "Transceivers should not be associated if we're in stable "
+          "before the first negotiation.");
+      continue;
+    }
+
+    if (!transceiver.HasLevel()) {
+      MOZ_CRASH("Associated transceivers should always have a level.");
+      continue;
+    }
+
+    if (transceiver.GetMediaType() == SdpMediaSection::kApplication) {
+      continue;
+    }
+
+    size_t level = transceiver.GetLevel();
+    if (NS_WARN_IF(mCurrentLocalDescription->GetMediaSectionCount() <= level) ||
+        NS_WARN_IF(mCurrentRemoteDescription->GetMediaSectionCount() <=
+                   level)) {
+      MOZ_ASSERT(false);
+      continue;
+    }
+
+    const SdpMediaSection& local =
+        mCurrentLocalDescription->GetMediaSection(level);
+    const SdpMediaSection& remote =
+        mCurrentRemoteDescription->GetMediaSection(level);
+
+    if (transceiver.mJsDirection & sdp::kSend) {
+      std::vector<std::string> sdpMsids;
+      if (local.GetAttributeList().HasAttribute(SdpAttribute::kMsidAttribute)) {
+        for (const auto& msidAttr : local.GetAttributeList().GetMsid().mMsids) {
+          if (msidAttr.identifier != "-") {
+            sdpMsids.push_back(msidAttr.identifier);
+          }
+        }
+      }
+      std::sort(sdpMsids.begin(), sdpMsids.end());
+
+      std::vector<std::string> jsepMsids;
+      for (const auto& jsepMsid : transceiver.mSendTrack.GetStreamIds()) {
+        jsepMsids.push_back(jsepMsid);
+      }
+      std::sort(jsepMsids.begin(), jsepMsids.end());
+
+      if (!std::equal(sdpMsids.begin(), sdpMsids.end(), jsepMsids.begin(),
+                      jsepMsids.end())) {
+        MOZ_MTLOG(ML_DEBUG,
+                  "[" << mName
+                      << "]: Negotiation needed because transceiver "
+                         "is sending, and the local SDP has different "
+                         "msids than the send track");
+        MOZ_MTLOG(ML_DEBUG, "[" << mName << "]: SDP msids = [");
+        for (const auto& msid : sdpMsids) {
+          MOZ_MTLOG(ML_DEBUG, msid << ", ");
+        }
+        MOZ_MTLOG(ML_DEBUG, "]");
+        MOZ_MTLOG(ML_DEBUG, "[" << mName << "]: JSEP msids = [");
+        for (const auto& msid : jsepMsids) {
+          MOZ_MTLOG(ML_DEBUG, msid << ", ");
+        }
+        MOZ_MTLOG(ML_DEBUG, "]");
+        return true;
+      }
+    }
+
+    if (mIsCurrentOfferer.isSome() && *mIsCurrentOfferer) {
+      if ((local.GetDirection() != transceiver.mJsDirection) &&
+          reverse(remote.GetDirection()) != transceiver.mJsDirection) {
+        MOZ_MTLOG(ML_DEBUG, "[" << mName
+                                << "]: Negotiation needed because "
+                                   "the direction on our offer, and the remote "
+                                   "answer, does not "
+                                   "match the direction on a transceiver.");
+        return true;
+      }
+    } else if (local.GetDirection() !=
+               (transceiver.mJsDirection & reverse(remote.GetDirection()))) {
+      MOZ_MTLOG(
+          ML_DEBUG,
+          "[" << mName
+              << "]: Negotiation needed because "
+                 "the direction on our answer doesn't match the direction on a "
+                 "transceiver, even though the remote offer would have allowed "
+                 "it.");
+      return true;
+    }
+  }
+
+  return false;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/jsep/JsepSessionImpl.h b/dom/media/webrtc/jsep/JsepSessionImpl.h
new file mode 100644
index 0000000000..1cdbba9cfc
--- /dev/null
+++ b/dom/media/webrtc/jsep/JsepSessionImpl.h
@@ -0,0 +1,286 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _JSEPSESSIONIMPL_H_
+#define _JSEPSESSIONIMPL_H_
+
+#include <map>
+#include <set>
+#include <string>
+#include <vector>
+
+#include "jsep/JsepCodecDescription.h"
+#include "jsep/JsepSession.h"
+#include "jsep/JsepTrack.h"
+#include "jsep/JsepTransceiver.h"
+#include "jsep/SsrcGenerator.h"
+#include "sdp/HybridSdpParser.h"
+#include "sdp/SdpHelper.h"
+
+namespace mozilla {
+
+// JsepSessionImpl members that have default copy c'tors, to simplify the
+// implementation of the copy c'tor for JsepSessionImpl
+class JsepSessionCopyableStuff {
+ protected:
+  struct JsepDtlsFingerprint {
+    std::string mAlgorithm;
+    std::vector<uint8_t> mValue;
+  };
+
+  Maybe<bool> mIsPendingOfferer;
+  Maybe<bool> mIsCurrentOfferer;
+  bool mIceControlling = false;
+  std::string mIceUfrag;
+  std::string mIcePwd;
+  std::string mOldIceUfrag;
+  std::string mOldIcePwd;
+  bool mRemoteIsIceLite = false;
+  std::vector<std::string> mIceOptions;
+  JsepBundlePolicy mBundlePolicy = kBundleBalanced;
+  std::vector<JsepDtlsFingerprint> mDtlsFingerprints;
+  uint64_t mSessionId = 0;
+  uint64_t mSessionVersion = 0;
+  size_t mMidCounter = 0;
+  std::set<std::string> mUsedMids;
+  size_t mTransportIdCounter = 0;
+  std::vector<JsepExtmapMediaType> mRtpExtensions;
+  std::set<uint16_t> mExtmapEntriesEverUsed;
+  std::map<uint16_t, std::string> mExtmapEntriesEverNegotiated;
+  std::string mDefaultRemoteStreamId;
+  std::string mCNAME;
+  // Used to prevent duplicate local SSRCs. Not used to prevent local/remote or
+  // remote-only duplication, which will be important for EKT but not now.
+  std::set<uint32_t> mSsrcs;
+  std::string mLastError;
+  std::vector<std::pair<size_t, std::string>> mLastSdpParsingErrors;
+  bool mEncodeTrackId = true;
+  SsrcGenerator mSsrcGenerator;
+  // !!!NOT INDEXED BY LEVEL!!! The level mapping is done with
+  // JsepTransceiver::mLevel. The keys are UUIDs.
+  std::vector<JsepTransceiver> mTransceivers;
+  // So we can rollback. Not as simple as just going back to the old, though...
+  std::vector<JsepTransceiver> mOldTransceivers;
+};
+
+class JsepSessionImpl : public JsepSession, public JsepSessionCopyableStuff {
+ public:
+  JsepSessionImpl(const std::string& name, UniquePtr<JsepUuidGenerator> uuidgen)
+      : JsepSession(name),
+        mUuidGen(std::move(uuidgen)),
+        mSdpHelper(&mLastError),
+        mParser(new HybridSdpParser()) {}
+
+  JsepSessionImpl(const JsepSessionImpl& aOrig);
+
+  JsepSession* Clone() const override { return new JsepSessionImpl(*this); }
+
+  // Implement JsepSession methods.
+  virtual nsresult Init() override;
+
+  nsresult SetBundlePolicy(JsepBundlePolicy policy) override;
+
+  virtual bool RemoteIsIceLite() const override { return mRemoteIsIceLite; }
+
+  virtual std::vector<std::string> GetIceOptions() const override {
+    return mIceOptions;
+  }
+
+  virtual nsresult AddDtlsFingerprint(
+      const std::string& algorithm, const std::vector<uint8_t>& value) override;
+
+  virtual nsresult AddRtpExtension(
+      JsepMediaType mediaType, const std::string& extensionName,
+      SdpDirectionAttribute::Direction direction) override;
+  virtual nsresult AddAudioRtpExtension(
+      const std::string& extensionName,
+      SdpDirectionAttribute::Direction direction =
+          SdpDirectionAttribute::Direction::kSendrecv) override;
+
+  virtual nsresult AddVideoRtpExtension(
+      const std::string& extensionName,
+      SdpDirectionAttribute::Direction direction =
+          SdpDirectionAttribute::Direction::kSendrecv) override;
+
+  virtual nsresult AddAudioVideoRtpExtension(
+      const std::string& extensionName,
+      SdpDirectionAttribute::Direction direction =
+          SdpDirectionAttribute::Direction::kSendrecv) override;
+
+  virtual std::vector<UniquePtr<JsepCodecDescription>>& Codecs() override {
+    return mSupportedCodecs;
+  }
+
+  virtual Result CreateOffer(const JsepOfferOptions& options,
+                             std::string* offer) override;
+
+  virtual Result CreateAnswer(const JsepAnswerOptions& options,
+                              std::string* answer) override;
+
+  virtual std::string GetLocalDescription(
+      JsepDescriptionPendingOrCurrent type) const override;
+
+  virtual std::string GetRemoteDescription(
+      JsepDescriptionPendingOrCurrent type) const override;
+
+  virtual Result SetLocalDescription(JsepSdpType type,
+                                     const std::string& sdp) override;
+
+  virtual Result SetRemoteDescription(JsepSdpType type,
+                                      const std::string& sdp) override;
+
+  virtual Result AddRemoteIceCandidate(const std::string& candidate,
+                                       const std::string& mid,
+                                       const Maybe<uint16_t>& level,
+                                       const std::string& ufrag,
+                                       std::string* transportId) override;
+
+  virtual nsresult AddLocalIceCandidate(const std::string& candidate,
+                                        const std::string& transportId,
+                                        const std::string& ufrag,
+                                        uint16_t* level, std::string* mid,
+                                        bool* skipped) override;
+
+  virtual nsresult UpdateDefaultCandidate(
+      const std::string& defaultCandidateAddr, uint16_t defaultCandidatePort,
+      const std::string& defaultRtcpCandidateAddr,
+      uint16_t defaultRtcpCandidatePort,
+      const std::string& transportId) override;
+
+  virtual nsresult Close() override;
+
+  virtual const std::string GetLastError() const override;
+
+  virtual const std::vector<std::pair<size_t, std::string>>&
+  GetLastSdpParsingErrors() const override;
+
+  virtual bool IsIceControlling() const override { return mIceControlling; }
+
+  virtual Maybe<bool> IsPendingOfferer() const override {
+    return mIsPendingOfferer;
+  }
+
+  virtual Maybe<bool> IsCurrentOfferer() const override {
+    return mIsCurrentOfferer;
+  }
+
+  virtual bool IsIceRestarting() const override {
+    return !mOldIceUfrag.empty();
+  }
+
+  virtual std::set<std::pair<std::string, std::string>> GetLocalIceCredentials()
+      const override;
+
+  virtual void AddTransceiver(const JsepTransceiver& transceiver) override;
+
+  virtual bool CheckNegotiationNeeded() const override;
+
+  virtual void SetDefaultCodecs(
+      const std::vector<UniquePtr<JsepCodecDescription>>& aPreferredCodecs)
+      override;
+
+ private:
+  friend class JsepSessionTest;
+  virtual const std::vector<JsepTransceiver>& GetTransceivers() const override {
+    return mTransceivers;
+  }
+
+  virtual std::vector<JsepTransceiver>& GetTransceivers() override {
+    return mTransceivers;
+  }
+
+  // Non-const so it can set mLastError
+  nsresult CreateGenericSDP(UniquePtr<Sdp>* sdp);
+  void AddExtmap(SdpMediaSection* msection);
+  std::vector<SdpExtmapAttributeList::Extmap> GetRtpExtensions(
+      const SdpMediaSection& msection);
+  std::string GetNewMid();
+
+  void AddCommonExtmaps(const SdpMediaSection& remoteMsection,
+                        SdpMediaSection* msection);
+  uint16_t GetNeverUsedExtmapEntry();
+  nsresult SetupIds();
+  void SetState(JsepSignalingState state);
+  // Non-const so it can set mLastError
+  nsresult ParseSdp(const std::string& sdp, UniquePtr<Sdp>* parsedp);
+  nsresult SetLocalDescriptionOffer(UniquePtr<Sdp> offer);
+  nsresult SetLocalDescriptionAnswer(JsepSdpType type, UniquePtr<Sdp> answer);
+  nsresult SetRemoteDescriptionOffer(UniquePtr<Sdp> offer);
+  nsresult SetRemoteDescriptionAnswer(JsepSdpType type, UniquePtr<Sdp> answer);
+  nsresult ValidateLocalDescription(const Sdp& description, JsepSdpType type);
+  nsresult ValidateRemoteDescription(const Sdp& description);
+  nsresult ValidateOffer(const Sdp& offer);
+  nsresult ValidateAnswer(const Sdp& offer, const Sdp& answer);
+  nsresult UpdateTransceiversFromRemoteDescription(const Sdp& remote);
+  Maybe<JsepTransceiver> GetTransceiverForLevel(size_t level) const;
+  Maybe<JsepTransceiver> GetTransceiverForMid(const std::string& mid) const;
+  Maybe<JsepTransceiver> GetTransceiverForLocal(size_t level);
+  Maybe<JsepTransceiver> GetTransceiverForRemote(
+      const SdpMediaSection& msection);
+  Maybe<JsepTransceiver> GetTransceiverWithTransport(
+      const std::string& transportId) const;
+  // The w3c and IETF specs have a lot of "magical" behavior that happens when
+  // addTrack is used. This was a deliberate design choice. Sadface.
+  Maybe<JsepTransceiver> FindUnassociatedTransceiver(
+      SdpMediaSection::MediaType type, bool magic);
+  // Called for rollback of local description
+  void RollbackLocalOffer();
+  // Called for rollback of remote description
+  void RollbackRemoteOffer();
+  nsresult HandleNegotiatedSession(const UniquePtr<Sdp>& local,
+                                   const UniquePtr<Sdp>& remote);
+  nsresult AddTransportAttributes(SdpMediaSection* msection,
+                                  SdpSetupAttribute::Role dtlsRole);
+  nsresult CopyPreviousTransportParams(const Sdp& oldAnswer,
+                                       const Sdp& offerersPreviousSdp,
+                                       const Sdp& newOffer, Sdp* newLocal);
+  void EnsureMsid(Sdp* remote);
+  void SetupBundle(Sdp* sdp) const;
+  nsresult CreateOfferMsection(const JsepOfferOptions& options,
+                               JsepTransceiver& transceiver, Sdp* local);
+  nsresult CreateAnswerMsection(const JsepAnswerOptions& options,
+                                JsepTransceiver& transceiver,
+                                const SdpMediaSection& remoteMsection,
+                                Sdp* sdp);
+  nsresult DetermineAnswererSetupRole(const SdpMediaSection& remoteMsection,
+                                      SdpSetupAttribute::Role* rolep);
+  nsresult MakeNegotiatedTransceiver(const SdpMediaSection& remote,
+                                     const SdpMediaSection& local,
+                                     JsepTransceiver& transceiverOut);
+  void EnsureHasOwnTransport(const SdpMediaSection& msection,
+                             JsepTransceiver& transceiver);
+  void CopyBundleTransports();
+
+  nsresult FinalizeTransport(const SdpAttributeList& remote,
+                             const SdpAttributeList& answer,
+                             JsepTransport* transport) const;
+
+  nsresult GetNegotiatedBundledMids(SdpHelper::BundledMids* bundledMids);
+
+  nsresult EnableOfferMsection(SdpMediaSection* msection);
+
+  mozilla::Sdp* GetParsedLocalDescription(
+      JsepDescriptionPendingOrCurrent type) const;
+  mozilla::Sdp* GetParsedRemoteDescription(
+      JsepDescriptionPendingOrCurrent type) const;
+  const Sdp* GetAnswer() const;
+  void SetIceRestarting(bool restarting);
+
+  void InitTransceiver(JsepTransceiver& aTransceiver);
+
+  UniquePtr<JsepUuidGenerator> mUuidGen;
+  UniquePtr<Sdp> mGeneratedOffer;   // Created but not set.
+  UniquePtr<Sdp> mGeneratedAnswer;  // Created but not set.
+  UniquePtr<Sdp> mCurrentLocalDescription;
+  UniquePtr<Sdp> mCurrentRemoteDescription;
+  UniquePtr<Sdp> mPendingLocalDescription;
+  UniquePtr<Sdp> mPendingRemoteDescription;
+  std::vector<UniquePtr<JsepCodecDescription>> mSupportedCodecs;
+  SdpHelper mSdpHelper;
+  UniquePtr<SdpParser> mParser;
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/jsep/JsepTrack.cpp b/dom/media/webrtc/jsep/JsepTrack.cpp
new file mode 100644
index 0000000000..f60e4ce0d7
--- /dev/null
+++ b/dom/media/webrtc/jsep/JsepTrack.cpp
@@ -0,0 +1,670 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "jsep/JsepTrack.h"
+#include "jsep/JsepCodecDescription.h"
+#include "jsep/JsepTrackEncoding.h"
+
+#include <algorithm>
+
+namespace mozilla {
+void JsepTrack::GetNegotiatedPayloadTypes(
+    std::vector<uint16_t>* payloadTypes) const {
+  if (!mNegotiatedDetails) {
+    return;
+  }
+
+  for (const auto& encoding : mNegotiatedDetails->mEncodings) {
+    GetPayloadTypes(encoding->GetCodecs(), payloadTypes);
+  }
+
+  // Prune out dupes
+  std::sort(payloadTypes->begin(), payloadTypes->end());
+  auto newEnd = std::unique(payloadTypes->begin(), payloadTypes->end());
+  payloadTypes->erase(newEnd, payloadTypes->end());
+}
+
+/* static */
+void JsepTrack::GetPayloadTypes(
+    const std::vector<UniquePtr<JsepCodecDescription>>& codecs,
+    std::vector<uint16_t>* payloadTypes) {
+  for (const auto& codec : codecs) {
+    uint16_t pt;
+    if (!codec->GetPtAsInt(&pt)) {
+      MOZ_ASSERT(false);
+      continue;
+    }
+    payloadTypes->push_back(pt);
+  }
+}
+
+void JsepTrack::EnsureNoDuplicatePayloadTypes(
+    std::vector<UniquePtr<JsepCodecDescription>>* codecs) {
+  std::set<std::string> uniquePayloadTypes;
+  for (auto& codec : *codecs) {
+    codec->EnsureNoDuplicatePayloadTypes(uniquePayloadTypes);
+  }
+}
+
+void JsepTrack::EnsureSsrcs(SsrcGenerator& ssrcGenerator, size_t aNumber) {
+  while (mSsrcs.size() < aNumber) {
+    uint32_t ssrc, rtxSsrc;
+    if (!ssrcGenerator.GenerateSsrc(&ssrc) ||
+        !ssrcGenerator.GenerateSsrc(&rtxSsrc)) {
+      return;
+    }
+    mSsrcs.push_back(ssrc);
+    mSsrcToRtxSsrc[ssrc] = rtxSsrc;
+    MOZ_ASSERT(mSsrcs.size() == mSsrcToRtxSsrc.size());
+  }
+}
+
+void JsepTrack::PopulateCodecs(
+    const std::vector<UniquePtr<JsepCodecDescription>>& prototype) {
+  mPrototypeCodecs.clear();
+  for (const auto& prototypeCodec : prototype) {
+    if (prototypeCodec->Type() == mType) {
+      mPrototypeCodecs.emplace_back(prototypeCodec->Clone());
+      mPrototypeCodecs.back()->mDirection = mDirection;
+    }
+  }
+
+  EnsureNoDuplicatePayloadTypes(&mPrototypeCodecs);
+}
+
+void JsepTrack::AddToOffer(SsrcGenerator& ssrcGenerator,
+                           SdpMediaSection* offer) {
+  AddToMsection(mPrototypeCodecs, offer);
+
+  if (mDirection == sdp::kSend) {
+    std::vector<std::string> rids;
+    if (offer->IsSending()) {
+      rids = mRids;
+    }
+
+    AddToMsection(rids, sdp::kSend, ssrcGenerator,
+                  IsRtxEnabled(mPrototypeCodecs), offer);
+  }
+}
+
+void JsepTrack::AddToAnswer(const SdpMediaSection& offer,
+                            SsrcGenerator& ssrcGenerator,
+                            SdpMediaSection* answer) {
+  // We do not modify mPrototypeCodecs here, since we're only creating an
+  // answer. Once offer/answer concludes, we will update mPrototypeCodecs.
+  std::vector<UniquePtr<JsepCodecDescription>> codecs =
+      NegotiateCodecs(offer, true, Nothing());
+  if (codecs.empty()) {
+    return;
+  }
+
+  AddToMsection(codecs, answer);
+
+  if (mDirection == sdp::kSend) {
+    AddToMsection(mRids, sdp::kSend, ssrcGenerator, IsRtxEnabled(codecs),
+                  answer);
+  }
+}
+
+void JsepTrack::SetRids(const std::vector<std::string>& aRids) {
+  MOZ_ASSERT(!aRids.empty());
+  if (!mRids.empty()) {
+    return;
+  }
+  mRids = aRids;
+}
+
+void JsepTrack::SetMaxEncodings(size_t aMax) {
+  mMaxEncodings = aMax;
+  if (mRids.size() > mMaxEncodings) {
+    mRids.resize(mMaxEncodings);
+  }
+}
+
+void JsepTrack::RecvTrackSetRemote(const Sdp& aSdp,
+                                   const SdpMediaSection& aMsection) {
+  mInHaveRemote = true;
+  MOZ_ASSERT(mDirection == sdp::kRecv);
+  MOZ_ASSERT(aMsection.GetMediaType() !=
+             SdpMediaSection::MediaType::kApplication);
+  std::string error;
+  SdpHelper helper(&error);
+
+  mRemoteSetSendBit = aMsection.IsSending();
+  if (!mRemoteSetSendBit) {
+    mReceptive = false;
+  }
+
+  if (aMsection.IsSending()) {
+    (void)helper.GetIdsFromMsid(aSdp, aMsection, &mStreamIds);
+  } else {
+    mStreamIds.clear();
+  }
+
+  // We do this whether or not the track is active
+  SetCNAME(helper.GetCNAME(aMsection));
+  mSsrcs.clear();
+  if (aMsection.GetAttributeList().HasAttribute(SdpAttribute::kSsrcAttribute)) {
+    for (const auto& ssrcAttr : aMsection.GetAttributeList().GetSsrc().mSsrcs) {
+      mSsrcs.push_back(ssrcAttr.ssrc);
+    }
+  }
+
+  // Use FID ssrc-group to associate rtx ssrcs with "regular" ssrcs. Despite
+  // not being part of RFC 4588, this is how rtx is negotiated by libwebrtc
+  // and jitsi.
+  mSsrcToRtxSsrc.clear();
+  if (aMsection.GetAttributeList().HasAttribute(
+          SdpAttribute::kSsrcGroupAttribute)) {
+    for (const auto& group :
+         aMsection.GetAttributeList().GetSsrcGroup().mSsrcGroups) {
+      if (group.semantics == SdpSsrcGroupAttributeList::kFid &&
+          group.ssrcs.size() == 2) {
+        // Ensure we have a "regular" ssrc for each rtx ssrc.
+        if (std::find(mSsrcs.begin(), mSsrcs.end(), group.ssrcs[0]) !=
+            mSsrcs.end()) {
+          mSsrcToRtxSsrc[group.ssrcs[0]] = group.ssrcs[1];
+
+          // Remove rtx ssrcs from mSsrcs
+          auto res = std::remove_if(
+              mSsrcs.begin(), mSsrcs.end(),
+              [group](uint32_t ssrc) { return ssrc == group.ssrcs[1]; });
+          mSsrcs.erase(res, mSsrcs.end());
+        }
+      }
+    }
+  }
+}
+
+void JsepTrack::RecvTrackSetLocal(const SdpMediaSection& aMsection) {
+  MOZ_ASSERT(mDirection == sdp::kRecv);
+
+  // TODO: Should more stuff live in here? Anything that needs to happen when we
+  // decide we're ready to receive packets should probably go in here.
+  mReceptive = aMsection.IsReceiving();
+}
+
+void JsepTrack::SendTrackSetRemote(SsrcGenerator& aSsrcGenerator,
+                                   const SdpMediaSection& aRemoteMsection) {
+  mInHaveRemote = true;
+  if (mType == SdpMediaSection::kApplication) {
+    return;
+  }
+
+  std::vector<SdpRidAttributeList::Rid> rids;
+
+  // TODO: Current language in webrtc-pc is completely broken, and so I will
+  // not be quoting it here.
+  if ((mType == SdpMediaSection::kVideo) &&
+      aRemoteMsection.GetAttributeList().HasAttribute(
+          SdpAttribute::kSimulcastAttribute)) {
+    // Note: webrtc-pc does not appear to support the full IETF simulcast
+    // spec. In particular, the IETF simulcast spec supports requesting
+    // multiple different sets of encodings. For example, "a=simulcast:send
+    // 1,2;3,4;5,6" means that there are three simulcast streams, the first of
+    // which can use either rid 1 or 2 (but not both), the second of which can
+    // use rid 3 or 4 (but not both), and the third of which can use rid 5 or
+    // 6 (but not both). webrtc-pc does not support this either/or stuff for
+    // rid; each simulcast stream gets exactly one rid.
+    // Also, webrtc-pc does not support the '~' pause syntax at all
+    // See https://github.com/w3c/webrtc-pc/issues/2769
+    GetRids(aRemoteMsection, sdp::kRecv, &rids);
+  }
+
+  if (mRids.empty()) {
+    // Initial configuration
+    for (const auto& ridAttr : rids) {
+      // TODO: Spec might change, making a length > 16 invalid SDP.
+      std::string dummy;
+      if (SdpRidAttributeList::CheckRidValidity(ridAttr.id, &dummy) &&
+          ridAttr.id.size() <= SdpRidAttributeList::kMaxRidLength) {
+        mRids.push_back(ridAttr.id);
+      }
+    }
+    if (mRids.size() > mMaxEncodings) {
+      mRids.resize(mMaxEncodings);
+    }
+  } else {
+    // JSEP is allowed to remove or reorder rids. RTCRtpSender won't pay
+    // attention to reordering.
+    std::vector<std::string> newRids;
+    for (const auto& ridAttr : rids) {
+      for (const auto& oldRid : mRids) {
+        if (oldRid == ridAttr.id) {
+          newRids.push_back(oldRid);
+          break;
+        }
+      }
+    }
+    mRids = std::move(newRids);
+  }
+
+  if (mRids.empty()) {
+    mRids.push_back("");
+  }
+
+  UpdateSsrcs(aSsrcGenerator, mRids.size());
+}
+
+void JsepTrack::AddToMsection(
+    const std::vector<UniquePtr<JsepCodecDescription>>& codecs,
+    SdpMediaSection* msection) const {
+  MOZ_ASSERT(msection->GetMediaType() == mType);
+  MOZ_ASSERT(!codecs.empty());
+
+  for (const auto& codec : codecs) {
+    codec->AddToMediaSection(*msection);
+  }
+
+  if ((mDirection == sdp::kSend) && (mType != SdpMediaSection::kApplication) &&
+      msection->IsSending()) {
+    if (mStreamIds.empty()) {
+      msection->AddMsid("-", mTrackId);
+    } else {
+      for (const std::string& streamId : mStreamIds) {
+        msection->AddMsid(streamId, mTrackId);
+      }
+    }
+  }
+}
+
+void JsepTrack::UpdateSsrcs(SsrcGenerator& ssrcGenerator, size_t encodings) {
+  MOZ_ASSERT(mDirection == sdp::kSend);
+  MOZ_ASSERT(mType != SdpMediaSection::kApplication);
+  size_t numSsrcs = std::max<size_t>(encodings, 1U);
+
+  EnsureSsrcs(ssrcGenerator, numSsrcs);
+  PruneSsrcs(numSsrcs);
+  if (mNegotiatedDetails && mNegotiatedDetails->GetEncodingCount() > numSsrcs) {
+    mNegotiatedDetails->TruncateEncodings(numSsrcs);
+  }
+
+  MOZ_ASSERT(!mSsrcs.empty());
+}
+
+void JsepTrack::PruneSsrcs(size_t aNumSsrcs) {
+  mSsrcs.resize(aNumSsrcs);
+
+  // We might have duplicate entries in mSsrcs, so we need to resize first and
+  // then remove ummatched rtx ssrcs.
+  auto itor = mSsrcToRtxSsrc.begin();
+  while (itor != mSsrcToRtxSsrc.end()) {
+    if (std::find(mSsrcs.begin(), mSsrcs.end(), itor->first) == mSsrcs.end()) {
+      itor = mSsrcToRtxSsrc.erase(itor);
+    } else {
+      ++itor;
+    }
+  }
+}
+
+bool JsepTrack::IsRtxEnabled(
+    const std::vector<UniquePtr<JsepCodecDescription>>& codecs) const {
+  for (const auto& codec : codecs) {
+    if (codec->Type() == SdpMediaSection::kVideo &&
+        static_cast<const JsepVideoCodecDescription*>(codec.get())
+            ->mRtxEnabled) {
+      return true;
+    }
+  }
+
+  return false;
+}
+
+void JsepTrack::AddToMsection(const std::vector<std::string>& aRids,
+                              sdp::Direction direction,
+                              SsrcGenerator& ssrcGenerator, bool rtxEnabled,
+                              SdpMediaSection* msection) {
+  if (aRids.size() > 1) {
+    UniquePtr<SdpSimulcastAttribute> simulcast(new SdpSimulcastAttribute);
+    UniquePtr<SdpRidAttributeList> ridAttrs(new SdpRidAttributeList);
+    for (const std::string& rid : aRids) {
+      SdpRidAttributeList::Rid ridAttr;
+      ridAttr.id = rid;
+      ridAttr.direction = direction;
+      ridAttrs->mRids.push_back(ridAttr);
+
+      SdpSimulcastAttribute::Version version;
+      version.choices.push_back(SdpSimulcastAttribute::Encoding(rid, false));
+      if (direction == sdp::kSend) {
+        simulcast->sendVersions.push_back(version);
+      } else {
+        simulcast->recvVersions.push_back(version);
+      }
+    }
+
+    msection->GetAttributeList().SetAttribute(simulcast.release());
+    msection->GetAttributeList().SetAttribute(ridAttrs.release());
+  }
+
+  bool requireRtxSsrcs = rtxEnabled && msection->IsSending();
+
+  if (mType != SdpMediaSection::kApplication && mDirection == sdp::kSend) {
+    UpdateSsrcs(ssrcGenerator, aRids.size());
+
+    if (requireRtxSsrcs) {
+      MOZ_ASSERT(mSsrcs.size() == mSsrcToRtxSsrc.size());
+      std::vector<uint32_t> allSsrcs;
+      UniquePtr<SdpSsrcGroupAttributeList> group(new SdpSsrcGroupAttributeList);
+      for (const auto& ssrc : mSsrcs) {
+        const auto rtxSsrc = mSsrcToRtxSsrc[ssrc];
+        allSsrcs.push_back(ssrc);
+        allSsrcs.push_back(rtxSsrc);
+        group->PushEntry(SdpSsrcGroupAttributeList::kFid, {ssrc, rtxSsrc});
+      }
+      msection->SetSsrcs(allSsrcs, mCNAME);
+      msection->GetAttributeList().SetAttribute(group.release());
+    } else {
+      msection->SetSsrcs(mSsrcs, mCNAME);
+    }
+  }
+}
+
+void JsepTrack::GetRids(const SdpMediaSection& msection,
+                        sdp::Direction direction,
+                        std::vector<SdpRidAttributeList::Rid>* rids) const {
+  rids->clear();
+  if (!msection.GetAttributeList().HasAttribute(
+          SdpAttribute::kSimulcastAttribute)) {
+    return;
+  }
+
+  const SdpSimulcastAttribute& simulcast(
+      msection.GetAttributeList().GetSimulcast());
+
+  const SdpSimulcastAttribute::Versions* versions = nullptr;
+  switch (direction) {
+    case sdp::kSend:
+      versions = &simulcast.sendVersions;
+      break;
+    case sdp::kRecv:
+      versions = &simulcast.recvVersions;
+      break;
+  }
+
+  if (!versions->IsSet()) {
+    return;
+  }
+
+  // RFC 8853 does not seem to forbid duplicate rids in a simulcast attribute.
+  // So, while this is obviously silly, we should be prepared for it and
+  // ignore those duplicate rids.
+  std::set<std::string> uniqueRids;
+  for (const SdpSimulcastAttribute::Version& version : *versions) {
+    if (!version.choices.empty() && !uniqueRids.count(version.choices[0].rid)) {
+      // We validate that rids are present (and sane) elsewhere.
+      rids->push_back(*msection.FindRid(version.choices[0].rid));
+      uniqueRids.insert(version.choices[0].rid);
+    }
+  }
+}
+
+void JsepTrack::CreateEncodings(
+    const SdpMediaSection& remote,
+    const std::vector<UniquePtr<JsepCodecDescription>>& negotiatedCodecs,
+    JsepTrackNegotiatedDetails* negotiatedDetails) {
+  negotiatedDetails->mTias = remote.GetBandwidth("TIAS");
+
+  webrtc::RtcpMode rtcpMode = webrtc::RtcpMode::kCompound;
+  // rtcp-rsize (video only)
+  if (remote.GetMediaType() == SdpMediaSection::kVideo &&
+      remote.GetAttributeList().HasAttribute(
+          SdpAttribute::kRtcpRsizeAttribute)) {
+    rtcpMode = webrtc::RtcpMode::kReducedSize;
+  }
+  negotiatedDetails->mRtpRtcpConf = RtpRtcpConfig(rtcpMode);
+
+  // TODO add support for b=AS if TIAS is not set (bug 976521)
+
+  if (mRids.empty()) {
+    mRids.push_back("");
+  }
+
+  size_t numEncodings = mRids.size();
+
+  // Drop SSRCs if fewer RIDs were offered than we have encodings
+  if (mSsrcs.size() > numEncodings) {
+    PruneSsrcs(numEncodings);
+  }
+
+  // For each stream make sure we have an encoding, and configure
+  // that encoding appropriately.
+  for (size_t i = 0; i < numEncodings; ++i) {
+    UniquePtr<JsepTrackEncoding> encoding(new JsepTrackEncoding);
+    if (mRids.size() > i) {
+      encoding->mRid = mRids[i];
+    }
+    for (const auto& codec : negotiatedCodecs) {
+      encoding->AddCodec(*codec);
+    }
+    negotiatedDetails->mEncodings.push_back(std::move(encoding));
+  }
+}
+
+std::vector<UniquePtr<JsepCodecDescription>> JsepTrack::GetCodecClones() const {
+  std::vector<UniquePtr<JsepCodecDescription>> clones;
+  for (const auto& codec : mPrototypeCodecs) {
+    clones.emplace_back(codec->Clone());
+  }
+  return clones;
+}
+
+static bool CompareCodec(const UniquePtr<JsepCodecDescription>& lhs,
+                         const UniquePtr<JsepCodecDescription>& rhs) {
+  return lhs->mStronglyPreferred && !rhs->mStronglyPreferred;
+}
+
+std::vector<UniquePtr<JsepCodecDescription>> JsepTrack::NegotiateCodecs(
+    const SdpMediaSection& remote, bool remoteIsOffer,
+    Maybe<const SdpMediaSection&> local) {
+  std::vector<UniquePtr<JsepCodecDescription>> negotiatedCodecs;
+  std::vector<UniquePtr<JsepCodecDescription>> newPrototypeCodecs;
+
+  // Outer loop establishes the remote side's preference
+  for (const std::string& fmt : remote.GetFormats()) {
+    for (auto& codec : mPrototypeCodecs) {
+      if (!codec || !codec->mEnabled || !codec->Matches(fmt, remote)) {
+        continue;
+      }
+
+      // First codec of ours that matches. See if we can negotiate it.
+      UniquePtr<JsepCodecDescription> clone(codec->Clone());
+      if (clone->Negotiate(fmt, remote, remoteIsOffer, local)) {
+        // If negotiation succeeded, remember the payload type the other side
+        // used for reoffers.
+        codec->mDefaultPt = fmt;
+
+        // Remember whether we negotiated rtx and the associated pt for later.
+        if (codec->Type() == SdpMediaSection::kVideo) {
+          JsepVideoCodecDescription* videoCodec =
+              static_cast<JsepVideoCodecDescription*>(codec.get());
+          JsepVideoCodecDescription* cloneVideoCodec =
+              static_cast<JsepVideoCodecDescription*>(clone.get());
+          bool useRtx =
+              mRtxIsAllowed &&
+              Preferences::GetBool("media.peerconnection.video.use_rtx", false);
+          videoCodec->mRtxEnabled = useRtx && cloneVideoCodec->mRtxEnabled;
+          videoCodec->mRtxPayloadType = cloneVideoCodec->mRtxPayloadType;
+        }
+
+        // Moves the codec out of mPrototypeCodecs, leaving an empty
+        // UniquePtr, so we don't use it again. Also causes successfully
+        // negotiated codecs to be placed up front in the future.
+        newPrototypeCodecs.emplace_back(std::move(codec));
+        negotiatedCodecs.emplace_back(std::move(clone));
+        break;
+      }
+    }
+  }
+
+  // newPrototypeCodecs contains just the negotiated stuff so far. Add the rest.
+  for (auto& codec : mPrototypeCodecs) {
+    if (codec) {
+      newPrototypeCodecs.emplace_back(std::move(codec));
+    }
+  }
+
+  // Negotiated stuff is up front, so it will take precedence when ensuring
+  // there are no duplicate payload types.
+  EnsureNoDuplicatePayloadTypes(&newPrototypeCodecs);
+  std::swap(newPrototypeCodecs, mPrototypeCodecs);
+
+  // Find the (potential) red codec and ulpfec codec or telephone-event
+  JsepVideoCodecDescription* red = nullptr;
+  JsepVideoCodecDescription* ulpfec = nullptr;
+  JsepAudioCodecDescription* dtmf = nullptr;
+  // We can safely cast here since JsepTrack has a MediaType and only codecs
+  // that match that MediaType (kAudio or kVideo) are added.
+  for (auto& codec : negotiatedCodecs) {
+    if (codec->mName == "red") {
+      red = static_cast<JsepVideoCodecDescription*>(codec.get());
+    } else if (codec->mName == "ulpfec") {
+      ulpfec = static_cast<JsepVideoCodecDescription*>(codec.get());
+    } else if (codec->mName == "telephone-event") {
+      dtmf = static_cast<JsepAudioCodecDescription*>(codec.get());
+    }
+  }
+  // if we have a red codec remove redundant encodings that don't exist
+  if (red) {
+    // Since we could have an externally specified redundant endcodings
+    // list, we shouldn't simply rebuild the redundant encodings list
+    // based on the current list of codecs.
+    std::vector<uint8_t> unnegotiatedEncodings;
+    std::swap(unnegotiatedEncodings, red->mRedundantEncodings);
+    for (auto redundantPt : unnegotiatedEncodings) {
+      std::string pt = std::to_string(redundantPt);
+      for (const auto& codec : negotiatedCodecs) {
+        if (pt == codec->mDefaultPt) {
+          red->mRedundantEncodings.push_back(redundantPt);
+          break;
+        }
+      }
+    }
+  }
+  // Video FEC is indicated by the existence of the red and ulpfec
+  // codecs and not an attribute on the particular video codec (like in
+  // a rtcpfb attr). If we see both red and ulpfec codecs, we enable FEC
+  // on all the other codecs.
+  if (red && ulpfec) {
+    for (auto& codec : negotiatedCodecs) {
+      if (codec->mName != "red" && codec->mName != "ulpfec") {
+        JsepVideoCodecDescription* videoCodec =
+            static_cast<JsepVideoCodecDescription*>(codec.get());
+        videoCodec->EnableFec(red->mDefaultPt, ulpfec->mDefaultPt);
+      }
+    }
+  }
+
+  // Dtmf support is indicated by the existence of the telephone-event
+  // codec, and not an attribute on the particular audio codec (like in a
+  // rtcpfb attr). If we see the telephone-event codec, we enabled dtmf
+  // support on all the other audio codecs.
+  if (dtmf) {
+    for (auto& codec : negotiatedCodecs) {
+      JsepAudioCodecDescription* audioCodec =
+          static_cast<JsepAudioCodecDescription*>(codec.get());
+      audioCodec->mDtmfEnabled = true;
+    }
+  }
+
+  // Make sure strongly preferred codecs are up front, overriding the remote
+  // side's preference.
+  std::stable_sort(negotiatedCodecs.begin(), negotiatedCodecs.end(),
+                   CompareCodec);
+
+  if (!red) {
+    // No red, remove ulpfec
+    negotiatedCodecs.erase(
+        std::remove_if(negotiatedCodecs.begin(), negotiatedCodecs.end(),
+                       [ulpfec](const UniquePtr<JsepCodecDescription>& codec) {
+                         return codec.get() == ulpfec;
+                       }),
+        negotiatedCodecs.end());
+    // Make sure there's no dangling ptr here
+    ulpfec = nullptr;
+  }
+
+  return negotiatedCodecs;
+}
+
+nsresult JsepTrack::Negotiate(const SdpMediaSection& answer,
+                              const SdpMediaSection& remote,
+                              const SdpMediaSection& local) {
+  std::vector<UniquePtr<JsepCodecDescription>> negotiatedCodecs =
+      NegotiateCodecs(remote, &answer != &remote, SomeRef(local));
+
+  if (negotiatedCodecs.empty()) {
+    return NS_ERROR_FAILURE;
+  }
+
+  UniquePtr<JsepTrackNegotiatedDetails> negotiatedDetails =
+      MakeUnique<JsepTrackNegotiatedDetails>();
+
+  CreateEncodings(remote, negotiatedCodecs, negotiatedDetails.get());
+
+  if (answer.GetAttributeList().HasAttribute(SdpAttribute::kExtmapAttribute)) {
+    for (auto& extmapAttr : answer.GetAttributeList().GetExtmap().mExtmaps) {
+      SdpDirectionAttribute::Direction direction = extmapAttr.direction;
+      if (&remote == &answer) {
+        // Answer is remote, we need to flip this.
+        direction = reverse(direction);
+      }
+
+      if (direction & mDirection) {
+        negotiatedDetails->mExtmap[extmapAttr.extensionname] = extmapAttr;
+      }
+    }
+  }
+
+  mInHaveRemote = false;
+  mNegotiatedDetails = std::move(negotiatedDetails);
+  return NS_OK;
+}
+
+// When doing bundle, if all else fails we can try to figure out which m-line a
+// given RTP packet belongs to by looking at the payload type field. This only
+// works, however, if that payload type appeared in only one m-section.
+// We figure that out here.
+/* static */
+void JsepTrack::SetUniquePayloadTypes(std::vector<JsepTrack*>& tracks) {
+  // Maps to track details if no other track contains the payload type,
+  // otherwise maps to nullptr.
+  std::map<uint16_t, JsepTrackNegotiatedDetails*> payloadTypeToDetailsMap;
+
+  for (JsepTrack* track : tracks) {
+    if (track->GetMediaType() == SdpMediaSection::kApplication) {
+      continue;
+    }
+
+    auto* details = track->GetNegotiatedDetails();
+    if (!details) {
+      // Can happen if negotiation fails on a track
+      continue;
+    }
+
+    std::vector<uint16_t> payloadTypesForTrack;
+    track->GetNegotiatedPayloadTypes(&payloadTypesForTrack);
+
+    for (uint16_t pt : payloadTypesForTrack) {
+      if (payloadTypeToDetailsMap.count(pt)) {
+        // Found in more than one track, not unique
+        payloadTypeToDetailsMap[pt] = nullptr;
+      } else {
+        payloadTypeToDetailsMap[pt] = details;
+      }
+    }
+  }
+
+  for (auto ptAndDetails : payloadTypeToDetailsMap) {
+    uint16_t uniquePt = ptAndDetails.first;
+    MOZ_ASSERT(uniquePt <= UINT8_MAX);
+    auto trackDetails = ptAndDetails.second;
+
+    if (trackDetails) {
+      trackDetails->mUniquePayloadTypes.push_back(
+          static_cast<uint8_t>(uniquePt));
+    }
+  }
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/jsep/JsepTrack.h b/dom/media/webrtc/jsep/JsepTrack.h
new file mode 100644
index 0000000000..70583ccb71
--- /dev/null
+++ b/dom/media/webrtc/jsep/JsepTrack.h
@@ -0,0 +1,305 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _JSEPTRACK_H_
+#define _JSEPTRACK_H_
+
+#include <functional>
+#include <algorithm>
+#include <string>
+#include <map>
+#include <set>
+#include <vector>
+
+#include <mozilla/UniquePtr.h>
+#include "mozilla/Preferences.h"
+#include "nsError.h"
+
+#include "jsep/JsepTrackEncoding.h"
+#include "jsep/SsrcGenerator.h"
+#include "sdp/Sdp.h"
+#include "sdp/SdpAttribute.h"
+#include "sdp/SdpMediaSection.h"
+#include "libwebrtcglue/RtpRtcpConfig.h"
+namespace mozilla {
+
+class JsepTrackNegotiatedDetails {
+ public:
+  JsepTrackNegotiatedDetails()
+      : mTias(0), mRtpRtcpConf(webrtc::RtcpMode::kCompound) {}
+
+  JsepTrackNegotiatedDetails(const JsepTrackNegotiatedDetails& orig)
+      : mExtmap(orig.mExtmap),
+        mUniquePayloadTypes(orig.mUniquePayloadTypes),
+        mTias(orig.mTias),
+        mRtpRtcpConf(orig.mRtpRtcpConf) {
+    for (const auto& encoding : orig.mEncodings) {
+      mEncodings.emplace_back(new JsepTrackEncoding(*encoding));
+    }
+  }
+
+  size_t GetEncodingCount() const { return mEncodings.size(); }
+
+  const JsepTrackEncoding& GetEncoding(size_t index) const {
+    MOZ_RELEASE_ASSERT(index < mEncodings.size());
+    return *mEncodings[index];
+  }
+
+  void TruncateEncodings(size_t aSize) {
+    if (mEncodings.size() < aSize) {
+      MOZ_ASSERT(false);
+      return;
+    }
+    mEncodings.resize(aSize);
+  }
+
+  const SdpExtmapAttributeList::Extmap* GetExt(
+      const std::string& ext_name) const {
+    auto it = mExtmap.find(ext_name);
+    if (it != mExtmap.end()) {
+      return &it->second;
+    }
+    return nullptr;
+  }
+
+  void ForEachRTPHeaderExtension(
+      const std::function<void(const SdpExtmapAttributeList::Extmap& extmap)>&
+          fn) const {
+    for (auto entry : mExtmap) {
+      fn(entry.second);
+    }
+  }
+
+  std::vector<uint8_t> GetUniquePayloadTypes() const {
+    return mUniquePayloadTypes;
+  }
+
+  uint32_t GetTias() const { return mTias; }
+
+  RtpRtcpConfig GetRtpRtcpConfig() const { return mRtpRtcpConf; }
+
+ private:
+  friend class JsepTrack;
+
+  std::map<std::string, SdpExtmapAttributeList::Extmap> mExtmap;
+  std::vector<uint8_t> mUniquePayloadTypes;
+  std::vector<UniquePtr<JsepTrackEncoding>> mEncodings;
+  uint32_t mTias;  // bits per second
+  RtpRtcpConfig mRtpRtcpConf;
+};
+
+class JsepTrack {
+ public:
+  JsepTrack(mozilla::SdpMediaSection::MediaType type, sdp::Direction direction)
+      : mType(type),
+        mDirection(direction),
+        mActive(false),
+        mRemoteSetSendBit(false) {}
+
+  virtual ~JsepTrack() {}
+
+  void UpdateStreamIds(const std::vector<std::string>& streamIds) {
+    mStreamIds = streamIds;
+  }
+
+  void SetTrackId(const std::string& aTrackId) { mTrackId = aTrackId; }
+
+  void ClearStreamIds() { mStreamIds.clear(); }
+
+  void RecvTrackSetRemote(const Sdp& aSdp, const SdpMediaSection& aMsection);
+  void RecvTrackSetLocal(const SdpMediaSection& aMsection);
+
+  // This is called whenever a remote description is set; we do not wait for
+  // offer/answer to complete, since there's nothing to actually negotiate here.
+  void SendTrackSetRemote(SsrcGenerator& aSsrcGenerator,
+                          const SdpMediaSection& aRemoteMsection);
+
+  JsepTrack(const JsepTrack& orig) { *this = orig; }
+
+  JsepTrack(JsepTrack&& orig) = default;
+  JsepTrack& operator=(JsepTrack&& rhs) = default;
+
+  JsepTrack& operator=(const JsepTrack& rhs) {
+    if (this != &rhs) {
+      mType = rhs.mType;
+      mStreamIds = rhs.mStreamIds;
+      mTrackId = rhs.mTrackId;
+      mCNAME = rhs.mCNAME;
+      mDirection = rhs.mDirection;
+      mRids = rhs.mRids;
+      mSsrcs = rhs.mSsrcs;
+      mSsrcToRtxSsrc = rhs.mSsrcToRtxSsrc;
+      mActive = rhs.mActive;
+      mRemoteSetSendBit = rhs.mRemoteSetSendBit;
+      mReceptive = rhs.mReceptive;
+      mMaxEncodings = rhs.mMaxEncodings;
+      mInHaveRemote = rhs.mInHaveRemote;
+      mRtxIsAllowed = rhs.mRtxIsAllowed;
+
+      mPrototypeCodecs.clear();
+      for (const auto& codec : rhs.mPrototypeCodecs) {
+        mPrototypeCodecs.emplace_back(codec->Clone());
+      }
+      if (rhs.mNegotiatedDetails) {
+        mNegotiatedDetails.reset(
+            new JsepTrackNegotiatedDetails(*rhs.mNegotiatedDetails));
+      }
+    }
+    return *this;
+  }
+
+  virtual mozilla::SdpMediaSection::MediaType GetMediaType() const {
+    return mType;
+  }
+
+  virtual const std::vector<std::string>& GetStreamIds() const {
+    return mStreamIds;
+  }
+
+  virtual const std::string& GetCNAME() const { return mCNAME; }
+
+  virtual void SetCNAME(const std::string& cname) { mCNAME = cname; }
+
+  virtual sdp::Direction GetDirection() const { return mDirection; }
+
+  virtual const std::vector<uint32_t>& GetSsrcs() const { return mSsrcs; }
+
+  virtual std::vector<uint32_t> GetRtxSsrcs() const {
+    std::vector<uint32_t> result;
+    if (mRtxIsAllowed &&
+        Preferences::GetBool("media.peerconnection.video.use_rtx", false)) {
+      std::for_each(
+          mSsrcToRtxSsrc.begin(), mSsrcToRtxSsrc.end(),
+          [&result](const auto& pair) { result.push_back(pair.second); });
+    }
+    return result;
+  }
+
+  virtual void EnsureSsrcs(SsrcGenerator& ssrcGenerator, size_t aNumber);
+
+  bool GetActive() const { return mActive; }
+
+  void SetActive(bool active) { mActive = active; }
+
+  bool GetRemoteSetSendBit() const { return mRemoteSetSendBit; }
+  bool GetReceptive() const { return mReceptive; }
+
+  virtual void PopulateCodecs(
+      const std::vector<UniquePtr<JsepCodecDescription>>& prototype);
+
+  template <class UnaryFunction>
+  void ForEachCodec(UnaryFunction func) {
+    std::for_each(mPrototypeCodecs.begin(), mPrototypeCodecs.end(), func);
+  }
+
+  template <class BinaryPredicate>
+  void SortCodecs(BinaryPredicate sorter) {
+    std::stable_sort(mPrototypeCodecs.begin(), mPrototypeCodecs.end(), sorter);
+  }
+
+  // These two are non-const because this is where ssrcs are chosen.
+  virtual void AddToOffer(SsrcGenerator& ssrcGenerator, SdpMediaSection* offer);
+  virtual void AddToAnswer(const SdpMediaSection& offer,
+                           SsrcGenerator& ssrcGenerator,
+                           SdpMediaSection* answer);
+
+  virtual nsresult Negotiate(const SdpMediaSection& answer,
+                             const SdpMediaSection& remote,
+                             const SdpMediaSection& local);
+  static void SetUniquePayloadTypes(std::vector<JsepTrack*>& tracks);
+  virtual void GetNegotiatedPayloadTypes(
+      std::vector<uint16_t>* payloadTypes) const;
+
+  // This will be set when negotiation is carried out.
+  virtual const JsepTrackNegotiatedDetails* GetNegotiatedDetails() const {
+    if (mNegotiatedDetails) {
+      return mNegotiatedDetails.get();
+    }
+    return nullptr;
+  }
+
+  virtual JsepTrackNegotiatedDetails* GetNegotiatedDetails() {
+    if (mNegotiatedDetails) {
+      return mNegotiatedDetails.get();
+    }
+    return nullptr;
+  }
+
+  virtual void ClearNegotiatedDetails() { mNegotiatedDetails.reset(); }
+
+  void SetRids(const std::vector<std::string>& aRids);
+  void ClearRids() { mRids.clear(); }
+  const std::vector<std::string>& GetRids() const { return mRids; }
+
+  void AddToMsection(const std::vector<std::string>& aRids,
+                     sdp::Direction direction, SsrcGenerator& ssrcGenerator,
+                     bool rtxEnabled, SdpMediaSection* msection);
+
+  // See Bug 1642419, this can be removed when all sites are working with RTX.
+  void SetRtxIsAllowed(bool aRtxIsAllowed) { mRtxIsAllowed = aRtxIsAllowed; }
+
+  void SetMaxEncodings(size_t aMax);
+  bool IsInHaveRemote() const { return mInHaveRemote; }
+
+ private:
+  std::vector<UniquePtr<JsepCodecDescription>> GetCodecClones() const;
+  static void EnsureNoDuplicatePayloadTypes(
+      std::vector<UniquePtr<JsepCodecDescription>>* codecs);
+  static void GetPayloadTypes(
+      const std::vector<UniquePtr<JsepCodecDescription>>& codecs,
+      std::vector<uint16_t>* pts);
+  void AddToMsection(const std::vector<UniquePtr<JsepCodecDescription>>& codecs,
+                     SdpMediaSection* msection) const;
+  void GetRids(const SdpMediaSection& msection, sdp::Direction direction,
+               std::vector<SdpRidAttributeList::Rid>* rids) const;
+  void CreateEncodings(
+      const SdpMediaSection& remote,
+      const std::vector<UniquePtr<JsepCodecDescription>>& negotiatedCodecs,
+      JsepTrackNegotiatedDetails* details);
+
+  virtual std::vector<UniquePtr<JsepCodecDescription>> NegotiateCodecs(
+      const SdpMediaSection& remote, bool remoteIsOffer,
+      Maybe<const SdpMediaSection&> local);
+
+  void UpdateSsrcs(SsrcGenerator& ssrcGenerator, size_t encodings);
+  void PruneSsrcs(size_t aNumSsrcs);
+  bool IsRtxEnabled(
+      const std::vector<UniquePtr<JsepCodecDescription>>& codecs) const;
+
+  mozilla::SdpMediaSection::MediaType mType;
+  // These are the ids that everyone outside of JsepSession care about
+  std::vector<std::string> mStreamIds;
+  std::string mTrackId;
+  std::string mCNAME;
+  sdp::Direction mDirection;
+  std::vector<UniquePtr<JsepCodecDescription>> mPrototypeCodecs;
+  // List of rids. May be initially populated from JS, or from a remote SDP.
+  // Can be updated by remote SDP. If no negotiation has taken place at all,
+  // this will be empty. If negotiation has taken place, but no simulcast
+  // attr was negotiated, this will contain the empty string as a single
+  // element. If a simulcast attribute was negotiated, this will contain the
+  // negotiated rids.
+  std::vector<std::string> mRids;
+  UniquePtr<JsepTrackNegotiatedDetails> mNegotiatedDetails;
+  std::vector<uint32_t> mSsrcs;
+  std::map<uint32_t, uint32_t> mSsrcToRtxSsrc;
+  bool mActive;
+  bool mRemoteSetSendBit;
+  // This is used to drive RTCRtpTransceiver.[[Receptive]]. Basically, this
+  // denotes whether we are prepared to receive RTP. When we apply a local
+  // description with the recv bit set, this is set to true, even if we have
+  // not seen the remote description yet. If we apply either a local or remote
+  // description without the recv bit set (from our perspective), this is set
+  // to false.
+  bool mReceptive = false;
+  size_t mMaxEncodings = 3;
+  bool mInHaveRemote = false;
+
+  // See Bug 1642419, this can be removed when all sites are working with RTX.
+  bool mRtxIsAllowed = true;
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/jsep/JsepTrackEncoding.h b/dom/media/webrtc/jsep/JsepTrackEncoding.h
new file mode 100644
index 0000000000..c17dd8534e
--- /dev/null
+++ b/dom/media/webrtc/jsep/JsepTrackEncoding.h
@@ -0,0 +1,62 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _JESPTRACKENCODING_H_
+#define _JESPTRACKENCODING_H_
+
+#include "jsep/JsepCodecDescription.h"
+#include "common/EncodingConstraints.h"
+
+#include <vector>
+
+namespace mozilla {
+// Represents a single encoding of a media track. When simulcast is used, there
+// may be multiple. Each encoding may have some constraints (imposed by JS), and
+// may be able to use any one of multiple codecs (JsepCodecDescription) at any
+// given time.
+class JsepTrackEncoding {
+ public:
+  JsepTrackEncoding() = default;
+  JsepTrackEncoding(const JsepTrackEncoding& orig) { *this = orig; }
+
+  JsepTrackEncoding(JsepTrackEncoding&& aOrig) = default;
+
+  JsepTrackEncoding& operator=(const JsepTrackEncoding& aRhs) {
+    if (this != &aRhs) {
+      mRid = aRhs.mRid;
+      mCodecs.clear();
+      for (const auto& codec : aRhs.mCodecs) {
+        mCodecs.emplace_back(codec->Clone());
+      }
+    }
+    return *this;
+  }
+
+  const std::vector<UniquePtr<JsepCodecDescription>>& GetCodecs() const {
+    return mCodecs;
+  }
+
+  void AddCodec(const JsepCodecDescription& codec) {
+    mCodecs.emplace_back(codec.Clone());
+  }
+
+  bool HasFormat(const std::string& format) const {
+    for (const auto& codec : mCodecs) {
+      if (codec->mDefaultPt == format) {
+        return true;
+      }
+    }
+    return false;
+  }
+
+  std::string mRid;
+
+ private:
+  std::vector<UniquePtr<JsepCodecDescription>> mCodecs;
+};
+}  // namespace mozilla
+
+#endif  // _JESPTRACKENCODING_H_
diff --git a/dom/media/webrtc/jsep/JsepTransceiver.h b/dom/media/webrtc/jsep/JsepTransceiver.h
new file mode 100644
index 0000000000..45c9b85120
--- /dev/null
+++ b/dom/media/webrtc/jsep/JsepTransceiver.h
@@ -0,0 +1,220 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _JSEPTRANSCEIVER_H_
+#define _JSEPTRANSCEIVER_H_
+
+#include <string>
+
+#include "sdp/SdpAttribute.h"
+#include "sdp/SdpMediaSection.h"
+#include "sdp/Sdp.h"
+#include "jsep/JsepTransport.h"
+#include "jsep/JsepTrack.h"
+
+#include "nsError.h"
+
+namespace mozilla {
+
+class JsepUuidGenerator {
+ public:
+  virtual ~JsepUuidGenerator() = default;
+  virtual bool Generate(std::string* id) = 0;
+  virtual JsepUuidGenerator* Clone() const = 0;
+};
+
+class JsepTransceiver {
+ public:
+  explicit JsepTransceiver(SdpMediaSection::MediaType type,
+                           JsepUuidGenerator& aUuidGen,
+                           SdpDirectionAttribute::Direction jsDirection =
+                               SdpDirectionAttribute::kSendrecv)
+      : mJsDirection(jsDirection),
+        mSendTrack(type, sdp::kSend),
+        mRecvTrack(type, sdp::kRecv),
+        mLevel(SIZE_MAX),
+        mBundleLevel(SIZE_MAX),
+        mAddTrackMagic(false),
+        mOnlyExistsBecauseOfSetRemote(false),
+        mStopped(false),
+        mRemoved(false),
+        mNegotiated(false),
+        mCanRecycle(false) {
+    if (!aUuidGen.Generate(&mUuid)) {
+      MOZ_CRASH();
+    }
+  }
+
+  JsepTransceiver(const JsepTransceiver& orig) = default;
+  JsepTransceiver(JsepTransceiver&& orig) = default;
+  JsepTransceiver& operator=(const JsepTransceiver& aRhs) = default;
+  JsepTransceiver& operator=(JsepTransceiver&& aRhs) = default;
+
+  ~JsepTransceiver() = default;
+
+  void Rollback(JsepTransceiver& oldTransceiver, bool aRemote) {
+    MOZ_ASSERT(oldTransceiver.GetMediaType() == GetMediaType());
+    MOZ_ASSERT(!oldTransceiver.IsNegotiated() || !oldTransceiver.HasLevel() ||
+               !HasLevel() || oldTransceiver.GetLevel() == GetLevel());
+    mTransport = oldTransceiver.mTransport;
+    if (aRemote) {
+      mLevel = oldTransceiver.mLevel;
+      mBundleLevel = oldTransceiver.mBundleLevel;
+      mSendTrack = oldTransceiver.mSendTrack;
+    }
+    mRecvTrack = oldTransceiver.mRecvTrack;
+
+    // Don't allow rollback to re-associate a transceiver.
+    if (!oldTransceiver.IsAssociated()) {
+      Disassociate();
+    }
+  }
+
+  bool IsAssociated() const { return !mMid.empty(); }
+
+  const std::string& GetMid() const {
+    MOZ_ASSERT(IsAssociated());
+    return mMid;
+  }
+
+  void Associate(const std::string& mid) {
+    MOZ_ASSERT(HasLevel());
+    mMid = mid;
+  }
+
+  void Disassociate() { mMid.clear(); }
+
+  bool HasLevel() const { return mLevel != SIZE_MAX; }
+
+  void SetLevel(size_t level) {
+    MOZ_ASSERT(level != SIZE_MAX);
+    MOZ_ASSERT(!HasLevel());
+    MOZ_ASSERT(!IsStopped());
+
+    mLevel = level;
+  }
+
+  void ClearLevel() {
+    MOZ_ASSERT(!IsAssociated());
+    mLevel = SIZE_MAX;
+    mBundleLevel = SIZE_MAX;
+  }
+
+  size_t GetLevel() const {
+    MOZ_ASSERT(HasLevel());
+    return mLevel;
+  }
+
+  void Stop() { mStopped = true; }
+
+  bool IsStopped() const { return mStopped; }
+
+  void RestartDatachannelTransceiver() {
+    MOZ_RELEASE_ASSERT(GetMediaType() == SdpMediaSection::kApplication);
+    mStopped = false;
+    mCanRecycle = false;
+  }
+
+  void SetRemoved() { mRemoved = true; }
+
+  bool IsRemoved() const { return mRemoved; }
+
+  bool HasBundleLevel() const { return mBundleLevel != SIZE_MAX; }
+
+  size_t BundleLevel() const {
+    MOZ_ASSERT(HasBundleLevel());
+    return mBundleLevel;
+  }
+
+  void SetBundleLevel(size_t aBundleLevel) {
+    MOZ_ASSERT(aBundleLevel != SIZE_MAX);
+    mBundleLevel = aBundleLevel;
+  }
+
+  void ClearBundleLevel() { mBundleLevel = SIZE_MAX; }
+
+  size_t GetTransportLevel() const {
+    MOZ_ASSERT(HasLevel());
+    if (HasBundleLevel()) {
+      return BundleLevel();
+    }
+    return GetLevel();
+  }
+
+  void SetAddTrackMagic() { mAddTrackMagic = true; }
+
+  bool HasAddTrackMagic() const { return mAddTrackMagic; }
+
+  void SetOnlyExistsBecauseOfSetRemote(bool aValue) {
+    mOnlyExistsBecauseOfSetRemote = aValue;
+  }
+
+  bool OnlyExistsBecauseOfSetRemote() const {
+    return mOnlyExistsBecauseOfSetRemote;
+  }
+
+  void SetNegotiated() {
+    MOZ_ASSERT(IsAssociated());
+    MOZ_ASSERT(HasLevel());
+    mNegotiated = true;
+  }
+
+  bool IsNegotiated() const { return mNegotiated; }
+
+  void SetCanRecycle() { mCanRecycle = true; }
+
+  bool CanRecycle() const { return mCanRecycle; }
+
+  const std::string& GetUuid() const { return mUuid; }
+
+  // Convenience function
+  SdpMediaSection::MediaType GetMediaType() const {
+    MOZ_ASSERT(mRecvTrack.GetMediaType() == mSendTrack.GetMediaType());
+    return mRecvTrack.GetMediaType();
+  }
+
+  bool HasTransport() const { return mTransport.mComponents != 0; }
+
+  bool HasOwnTransport() const {
+    if (HasTransport() &&
+        (!HasBundleLevel() || (GetLevel() == BundleLevel()))) {
+      return true;
+    }
+    return false;
+  }
+
+  // See Bug 1642419, this can be removed when all sites are working with RTX.
+  void SetRtxIsAllowed(bool aRtxIsAllowed) {
+    mSendTrack.SetRtxIsAllowed(aRtxIsAllowed);
+    mRecvTrack.SetRtxIsAllowed(aRtxIsAllowed);
+  }
+
+  // This is the direction JS wants. It might not actually happen.
+  SdpDirectionAttribute::Direction mJsDirection;
+
+  JsepTrack mSendTrack;
+  JsepTrack mRecvTrack;
+  JsepTransport mTransport;
+
+ private:
+  std::string mUuid;
+  // Stuff that is not negotiated
+  std::string mMid;
+  size_t mLevel;  // SIZE_MAX if no level
+  // Is this track pair sharing a transport with another?
+  size_t mBundleLevel;  // SIZE_MAX if no bundle level
+  // The w3c and IETF specs have a lot of "magical" behavior that happens
+  // when addTrack is used to create a transceiver. This was a deliberate
+  // design choice. Sadface.
+  bool mAddTrackMagic;
+  bool mOnlyExistsBecauseOfSetRemote;
+  bool mStopped;
+  bool mRemoved;
+  bool mNegotiated;
+  bool mCanRecycle;
+};
+
+}  // namespace mozilla
+
+#endif  // _JSEPTRANSCEIVER_H_
diff --git a/dom/media/webrtc/jsep/JsepTransport.h b/dom/media/webrtc/jsep/JsepTransport.h
new file mode 100644
index 0000000000..67690b1fd4
--- /dev/null
+++ b/dom/media/webrtc/jsep/JsepTransport.h
@@ -0,0 +1,102 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _JSEPTRANSPORT_H_
+#define _JSEPTRANSPORT_H_
+
+#include <string>
+#include <vector>
+
+#include <mozilla/RefPtr.h>
+#include <mozilla/UniquePtr.h>
+#include "nsISupportsImpl.h"
+
+#include "sdp/SdpAttribute.h"
+
+namespace mozilla {
+
+class JsepDtlsTransport {
+ public:
+  JsepDtlsTransport() : mRole(kJsepDtlsInvalidRole) {}
+
+  virtual ~JsepDtlsTransport() {}
+
+  enum Role { kJsepDtlsClient, kJsepDtlsServer, kJsepDtlsInvalidRole };
+
+  virtual const SdpFingerprintAttributeList& GetFingerprints() const {
+    return mFingerprints;
+  }
+
+  virtual Role GetRole() const { return mRole; }
+
+ private:
+  friend class JsepSessionImpl;
+
+  SdpFingerprintAttributeList mFingerprints;
+  Role mRole;
+};
+
+class JsepIceTransport {
+ public:
+  JsepIceTransport() {}
+
+  virtual ~JsepIceTransport() {}
+
+  const std::string& GetUfrag() const { return mUfrag; }
+  const std::string& GetPassword() const { return mPwd; }
+  const std::vector<std::string>& GetCandidates() const { return mCandidates; }
+
+ private:
+  friend class JsepSessionImpl;
+
+  std::string mUfrag;
+  std::string mPwd;
+  std::vector<std::string> mCandidates;
+};
+
+class JsepTransport {
+ public:
+  JsepTransport() : mComponents(0) {}
+
+  JsepTransport(const JsepTransport& orig) { *this = orig; }
+
+  ~JsepTransport() {}
+
+  JsepTransport& operator=(const JsepTransport& orig) {
+    if (this != &orig) {
+      mIce.reset(orig.mIce ? new JsepIceTransport(*orig.mIce) : nullptr);
+      mDtls.reset(orig.mDtls ? new JsepDtlsTransport(*orig.mDtls) : nullptr);
+      mTransportId = orig.mTransportId;
+      mComponents = orig.mComponents;
+      mLocalUfrag = orig.mLocalUfrag;
+      mLocalPwd = orig.mLocalPwd;
+    }
+    return *this;
+  }
+
+  void Close() {
+    mComponents = 0;
+    mTransportId.clear();
+    mIce.reset();
+    mDtls.reset();
+    mLocalUfrag.clear();
+    mLocalPwd.clear();
+  }
+
+  // Unique identifier for this transport within this call. Group?
+  std::string mTransportId;
+
+  // ICE stuff.
+  UniquePtr<JsepIceTransport> mIce;
+  UniquePtr<JsepDtlsTransport> mDtls;
+
+  // Number of required components.
+  size_t mComponents;
+  std::string mLocalUfrag;
+  std::string mLocalPwd;
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/jsep/SsrcGenerator.cpp b/dom/media/webrtc/jsep/SsrcGenerator.cpp
new file mode 100644
index 0000000000..50025007d1
--- /dev/null
+++ b/dom/media/webrtc/jsep/SsrcGenerator.cpp
@@ -0,0 +1,22 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "jsep/SsrcGenerator.h"
+#include "pk11pub.h"
+
+namespace mozilla {
+
+bool SsrcGenerator::GenerateSsrc(uint32_t* ssrc) {
+  do {
+    SECStatus rv = PK11_GenerateRandom(reinterpret_cast<unsigned char*>(ssrc),
+                                       sizeof(uint32_t));
+    if (rv != SECSuccess) {
+      return false;
+    }
+  } while (mSsrcs.count(*ssrc));
+  mSsrcs.insert(*ssrc);
+
+  return true;
+}
+}  // namespace mozilla
diff --git a/dom/media/webrtc/jsep/SsrcGenerator.h b/dom/media/webrtc/jsep/SsrcGenerator.h
new file mode 100644
index 0000000000..b106358f32
--- /dev/null
+++ b/dom/media/webrtc/jsep/SsrcGenerator.h
@@ -0,0 +1,20 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _SSRCGENERATOR_H_
+#define _SSRCGENERATOR_H_
+
+#include <set>
+
+namespace mozilla {
+class SsrcGenerator {
+ public:
+  bool GenerateSsrc(uint32_t* ssrc);
+
+ private:
+  std::set<uint32_t> mSsrcs;
+};
+}  // namespace mozilla
+
+#endif  // _SSRCGENERATOR_H_
diff --git a/dom/media/webrtc/jsep/moz.build b/dom/media/webrtc/jsep/moz.build
new file mode 100644
index 0000000000..750ac76a30
--- /dev/null
+++ b/dom/media/webrtc/jsep/moz.build
@@ -0,0 +1,18 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+include("/dom/media/webrtc/third_party_build/webrtc.mozbuild")
+
+LOCAL_INCLUDES += [
+    "/dom/media/webrtc",
+    "/media/webrtc",
+    "/third_party/libwebrtc",
+    "/third_party/libwebrtc/third_party/abseil-cpp",
+    "/third_party/sipcc",
+]
+
+UNIFIED_SOURCES += ["JsepSessionImpl.cpp", "JsepTrack.cpp", "SsrcGenerator.cpp"]
+
+FINAL_LIBRARY = "xul"
diff --git a/dom/media/webrtc/libwebrtcglue/AudioConduit.cpp b/dom/media/webrtc/libwebrtcglue/AudioConduit.cpp
new file mode 100644
index 0000000000..d8492f779a
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/AudioConduit.cpp
@@ -0,0 +1,975 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AudioConduit.h"
+
+#include "common/browser_logging/CSFLog.h"
+#include "MediaConduitControl.h"
+#include "mozilla/media/MediaUtils.h"
+#include "mozilla/Telemetry.h"
+#include "transport/runnable_utils.h"
+#include "transport/SrtpFlow.h"  // For SRTP_MAX_EXPANSION
+#include "WebrtcCallWrapper.h"
+
+// libwebrtc includes
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "audio/audio_receive_stream.h"
+#include "media/base/media_constants.h"
+
+// for ntohs
+#ifdef HAVE_NETINET_IN_H
+#  include <netinet/in.h>
+#elif defined XP_WIN
+#  include <winsock2.h>
+#endif
+
+#ifdef MOZ_WIDGET_ANDROID
+#  include "AndroidBridge.h"
+#endif
+
+namespace mozilla {
+
+namespace {
+
+static const char* acLogTag = "WebrtcAudioSessionConduit";
+#ifdef LOGTAG
+#  undef LOGTAG
+#endif
+#define LOGTAG acLogTag
+
+using namespace cricket;
+using LocalDirection = MediaSessionConduitLocalDirection;
+
+const char kCodecParamCbr[] = "cbr";
+
+}  // namespace
+
+/**
+ * Factory Method for AudioConduit
+ */
+RefPtr<AudioSessionConduit> AudioSessionConduit::Create(
+    RefPtr<WebrtcCallWrapper> aCall,
+    nsCOMPtr<nsISerialEventTarget> aStsThread) {
+  CSFLogDebug(LOGTAG, "%s ", __FUNCTION__);
+  MOZ_ASSERT(NS_IsMainThread());
+
+  return MakeRefPtr<WebrtcAudioConduit>(std::move(aCall),
+                                        std::move(aStsThread));
+}
+
+#define INIT_MIRROR(name, val) \
+  name(aCallThread, val, "WebrtcAudioConduit::Control::" #name " (Mirror)")
+WebrtcAudioConduit::Control::Control(const RefPtr<AbstractThread>& aCallThread)
+    : INIT_MIRROR(mReceiving, false),
+      INIT_MIRROR(mTransmitting, false),
+      INIT_MIRROR(mLocalSsrcs, Ssrcs()),
+      INIT_MIRROR(mLocalCname, std::string()),
+      INIT_MIRROR(mMid, std::string()),
+      INIT_MIRROR(mRemoteSsrc, 0),
+      INIT_MIRROR(mSyncGroup, std::string()),
+      INIT_MIRROR(mLocalRecvRtpExtensions, RtpExtList()),
+      INIT_MIRROR(mLocalSendRtpExtensions, RtpExtList()),
+      INIT_MIRROR(mSendCodec, Nothing()),
+      INIT_MIRROR(mRecvCodecs, std::vector<AudioCodecConfig>()) {}
+#undef INIT_MIRROR
+
+RefPtr<GenericPromise> WebrtcAudioConduit::Shutdown() {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  return InvokeAsync(mCallThread, "WebrtcAudioConduit::Shutdown (main thread)",
+                     [this, self = RefPtr<WebrtcAudioConduit>(this)] {
+                       mControl.mReceiving.DisconnectIfConnected();
+                       mControl.mTransmitting.DisconnectIfConnected();
+                       mControl.mLocalSsrcs.DisconnectIfConnected();
+                       mControl.mLocalCname.DisconnectIfConnected();
+                       mControl.mMid.DisconnectIfConnected();
+                       mControl.mRemoteSsrc.DisconnectIfConnected();
+                       mControl.mSyncGroup.DisconnectIfConnected();
+                       mControl.mLocalRecvRtpExtensions.DisconnectIfConnected();
+                       mControl.mLocalSendRtpExtensions.DisconnectIfConnected();
+                       mControl.mSendCodec.DisconnectIfConnected();
+                       mControl.mRecvCodecs.DisconnectIfConnected();
+                       mControl.mOnDtmfEventListener.DisconnectIfExists();
+                       mWatchManager.Shutdown();
+
+                       {
+                         AutoWriteLock lock(mLock);
+                         DeleteSendStream();
+                         DeleteRecvStream();
+                       }
+
+                       return GenericPromise::CreateAndResolve(
+                           true, "WebrtcAudioConduit::Shutdown (call thread)");
+                     });
+}
+
+WebrtcAudioConduit::WebrtcAudioConduit(
+    RefPtr<WebrtcCallWrapper> aCall, nsCOMPtr<nsISerialEventTarget> aStsThread)
+    : mCall(std::move(aCall)),
+      mSendTransport(this),
+      mRecvTransport(this),
+      mRecvStreamConfig(),
+      mRecvStream(nullptr),
+      mSendStreamConfig(&mSendTransport),
+      mSendStream(nullptr),
+      mSendStreamRunning(false),
+      mRecvStreamRunning(false),
+      mDtmfEnabled(false),
+      mLock("WebrtcAudioConduit::mLock"),
+      mCallThread(std::move(mCall->mCallThread)),
+      mStsThread(std::move(aStsThread)),
+      mControl(mCall->mCallThread),
+      mWatchManager(this, mCall->mCallThread) {
+  mRecvStreamConfig.rtcp_send_transport = &mRecvTransport;
+  mRecvStreamConfig.rtp.rtcp_event_observer = this;
+}
+
+/**
+ * Destruction defines for our super-classes
+ */
+WebrtcAudioConduit::~WebrtcAudioConduit() {
+  CSFLogDebug(LOGTAG, "%s ", __FUNCTION__);
+  MOZ_ASSERT(!mSendStream && !mRecvStream,
+             "Call DeleteStreams prior to ~WebrtcAudioConduit.");
+}
+
+#define CONNECT(aCanonical, aMirror)                                          \
+  do {                                                                        \
+    (aMirror).Connect(aCanonical);                                            \
+    mWatchManager.Watch(aMirror, &WebrtcAudioConduit::OnControlConfigChange); \
+  } while (0)
+
+void WebrtcAudioConduit::InitControl(AudioConduitControlInterface* aControl) {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+
+  CONNECT(aControl->CanonicalReceiving(), mControl.mReceiving);
+  CONNECT(aControl->CanonicalTransmitting(), mControl.mTransmitting);
+  CONNECT(aControl->CanonicalLocalSsrcs(), mControl.mLocalSsrcs);
+  CONNECT(aControl->CanonicalLocalCname(), mControl.mLocalCname);
+  CONNECT(aControl->CanonicalMid(), mControl.mMid);
+  CONNECT(aControl->CanonicalRemoteSsrc(), mControl.mRemoteSsrc);
+  CONNECT(aControl->CanonicalSyncGroup(), mControl.mSyncGroup);
+  CONNECT(aControl->CanonicalLocalRecvRtpExtensions(),
+          mControl.mLocalRecvRtpExtensions);
+  CONNECT(aControl->CanonicalLocalSendRtpExtensions(),
+          mControl.mLocalSendRtpExtensions);
+  CONNECT(aControl->CanonicalAudioSendCodec(), mControl.mSendCodec);
+  CONNECT(aControl->CanonicalAudioRecvCodecs(), mControl.mRecvCodecs);
+  mControl.mOnDtmfEventListener = aControl->OnDtmfEvent().Connect(
+      mCall->mCallThread, this, &WebrtcAudioConduit::OnDtmfEvent);
+}
+
+#undef CONNECT
+
+void WebrtcAudioConduit::OnDtmfEvent(const DtmfEvent& aEvent) {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  MOZ_ASSERT(mSendStream);
+  MOZ_ASSERT(mDtmfEnabled);
+  mSendStream->SendTelephoneEvent(aEvent.mPayloadType, aEvent.mPayloadFrequency,
+                                  aEvent.mEventCode, aEvent.mLengthMs);
+}
+
+void WebrtcAudioConduit::OnControlConfigChange() {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+
+  bool recvStreamReconfigureNeeded = false;
+  bool sendStreamReconfigureNeeded = false;
+  bool recvStreamRecreationNeeded = false;
+  bool sendStreamRecreationNeeded = false;
+
+  if (!mControl.mLocalSsrcs.Ref().empty()) {
+    if (mControl.mLocalSsrcs.Ref()[0] != mSendStreamConfig.rtp.ssrc) {
+      sendStreamRecreationNeeded = true;
+
+      // For now...
+      recvStreamRecreationNeeded = true;
+    }
+    mRecvStreamConfig.rtp.local_ssrc = mControl.mLocalSsrcs.Ref()[0];
+    mSendStreamConfig.rtp.ssrc = mControl.mLocalSsrcs.Ref()[0];
+
+    // In the future we can do this instead of recreating the recv stream:
+    // if (mRecvStream) {
+    //   mCall->Call()->OnLocalSsrcUpdated(mRecvStream,
+    //                                     mControl.mLocalSsrcs.Ref()[0]);
+    // }
+  }
+
+  if (mControl.mLocalCname.Ref() != mSendStreamConfig.rtp.c_name) {
+    mSendStreamConfig.rtp.c_name = mControl.mLocalCname.Ref();
+    sendStreamReconfigureNeeded = true;
+  }
+
+  if (mControl.mMid.Ref() != mSendStreamConfig.rtp.mid) {
+    mSendStreamConfig.rtp.mid = mControl.mMid.Ref();
+    sendStreamReconfigureNeeded = true;
+  }
+
+  if (mControl.mRemoteSsrc.Ref() != mControl.mConfiguredRemoteSsrc) {
+    mRecvStreamConfig.rtp.remote_ssrc = mControl.mConfiguredRemoteSsrc =
+        mControl.mRemoteSsrc.Ref();
+    recvStreamRecreationNeeded = true;
+  }
+
+  if (mControl.mSyncGroup.Ref() != mRecvStreamConfig.sync_group) {
+    mRecvStreamConfig.sync_group = mControl.mSyncGroup.Ref();
+    // For now...
+    recvStreamRecreationNeeded = true;
+    // In the future we can do this instead of recreating the recv stream:
+    // if (mRecvStream) {
+    //   mCall->Call()->OnUpdateSyncGroup(mRecvStream,
+    //                                    mRecvStreamConfig.sync_group);
+    // }
+  }
+
+  if (auto filteredExtensions = FilterExtensions(
+          LocalDirection::kRecv, mControl.mLocalRecvRtpExtensions);
+      filteredExtensions != mRecvStreamConfig.rtp.extensions) {
+    mRecvStreamConfig.rtp.extensions = std::move(filteredExtensions);
+    // For now...
+    recvStreamRecreationNeeded = true;
+    // In the future we can do this instead of recreating the recv stream:
+    // if (mRecvStream) {
+    //  mRecvStream->SetRtpExtensions(mRecvStreamConfig.rtp.extensions);
+    //}
+  }
+
+  if (auto filteredExtensions = FilterExtensions(
+          LocalDirection::kSend, mControl.mLocalSendRtpExtensions);
+      filteredExtensions != mSendStreamConfig.rtp.extensions) {
+    // At the very least, we need a reconfigure. Recreation needed if the
+    // extmap for any extension has changed, but not for adding/removing
+    // extensions.
+    sendStreamReconfigureNeeded = true;
+
+    for (const auto& newExt : filteredExtensions) {
+      if (sendStreamRecreationNeeded) {
+        break;
+      }
+      for (const auto& oldExt : mSendStreamConfig.rtp.extensions) {
+        if (newExt.uri == oldExt.uri) {
+          if (newExt.id != oldExt.id) {
+            sendStreamRecreationNeeded = true;
+          }
+          // We're done handling newExt, one way or another
+          break;
+        }
+      }
+    }
+
+    mSendStreamConfig.rtp.extensions = std::move(filteredExtensions);
+  }
+
+  mControl.mSendCodec.Ref().apply([&](const auto& aConfig) {
+    if (mControl.mConfiguredSendCodec != mControl.mSendCodec.Ref()) {
+      mControl.mConfiguredSendCodec = mControl.mSendCodec;
+      if (ValidateCodecConfig(aConfig, true) == kMediaConduitNoError) {
+        mSendStreamConfig.encoder_factory =
+            webrtc::CreateBuiltinAudioEncoderFactory();
+
+        webrtc::AudioSendStream::Config::SendCodecSpec spec(
+            aConfig.mType, CodecConfigToLibwebrtcFormat(aConfig));
+        mSendStreamConfig.send_codec_spec = spec;
+
+        mDtmfEnabled = aConfig.mDtmfEnabled;
+        sendStreamReconfigureNeeded = true;
+      }
+    }
+  });
+
+  if (mControl.mConfiguredRecvCodecs != mControl.mRecvCodecs.Ref()) {
+    mControl.mConfiguredRecvCodecs = mControl.mRecvCodecs;
+    mRecvStreamConfig.decoder_factory = mCall->mAudioDecoderFactory;
+    mRecvStreamConfig.decoder_map.clear();
+
+    for (const auto& codec : mControl.mRecvCodecs.Ref()) {
+      if (ValidateCodecConfig(codec, false) != kMediaConduitNoError) {
+        continue;
+      }
+      mRecvStreamConfig.decoder_map.emplace(
+          codec.mType, CodecConfigToLibwebrtcFormat(codec));
+    }
+
+    recvStreamReconfigureNeeded = true;
+  }
+
+  if (!recvStreamReconfigureNeeded && !sendStreamReconfigureNeeded &&
+      !recvStreamRecreationNeeded && !sendStreamRecreationNeeded &&
+      mControl.mReceiving == mRecvStreamRunning &&
+      mControl.mTransmitting == mSendStreamRunning) {
+    // No changes applied -- no need to lock.
+    return;
+  }
+
+  if (recvStreamRecreationNeeded) {
+    recvStreamReconfigureNeeded = false;
+  }
+  if (sendStreamRecreationNeeded) {
+    sendStreamReconfigureNeeded = false;
+  }
+
+  {
+    AutoWriteLock lock(mLock);
+    // Recreate/Stop/Start streams as needed.
+    if (recvStreamRecreationNeeded) {
+      DeleteRecvStream();
+    }
+    if (mControl.mReceiving) {
+      CreateRecvStream();
+    }
+    if (sendStreamRecreationNeeded) {
+      DeleteSendStream();
+    }
+    if (mControl.mTransmitting) {
+      CreateSendStream();
+    }
+  }
+
+  // We make sure to not hold the lock while stopping/starting/reconfiguring
+  // streams, so as to not cause deadlocks. These methods can cause our platform
+  // codecs to dispatch sync runnables to main, and main may grab the lock.
+
+  if (mRecvStream && recvStreamReconfigureNeeded) {
+    MOZ_ASSERT(!recvStreamRecreationNeeded);
+    mRecvStream->SetDecoderMap(mRecvStreamConfig.decoder_map);
+  }
+
+  if (mSendStream && sendStreamReconfigureNeeded) {
+    MOZ_ASSERT(!sendStreamRecreationNeeded);
+    // TODO: Pass a callback here, so we can react to RTCErrors thrown by
+    // libwebrtc.
+    mSendStream->Reconfigure(mSendStreamConfig, nullptr);
+  }
+
+  if (!mControl.mReceiving) {
+    StopReceiving();
+  }
+  if (!mControl.mTransmitting) {
+    StopTransmitting();
+  }
+
+  if (mControl.mReceiving) {
+    StartReceiving();
+  }
+  if (mControl.mTransmitting) {
+    StartTransmitting();
+  }
+}
+
+std::vector<uint32_t> WebrtcAudioConduit::GetLocalSSRCs() const {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  return std::vector<uint32_t>(1, mRecvStreamConfig.rtp.local_ssrc);
+}
+
+bool WebrtcAudioConduit::OverrideRemoteSSRC(uint32_t aSsrc) {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+
+  if (mRecvStreamConfig.rtp.remote_ssrc == aSsrc) {
+    return true;
+  }
+  mRecvStreamConfig.rtp.remote_ssrc = aSsrc;
+
+  const bool wasReceiving = mRecvStreamRunning;
+  const bool hadRecvStream = mRecvStream;
+
+  StopReceiving();
+
+  if (hadRecvStream) {
+    AutoWriteLock lock(mLock);
+    DeleteRecvStream();
+    CreateRecvStream();
+  }
+
+  if (wasReceiving) {
+    StartReceiving();
+  }
+  return true;
+}
+
+Maybe<Ssrc> WebrtcAudioConduit::GetRemoteSSRC() const {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  // libwebrtc uses 0 to mean a lack of SSRC. That is not to spec.
+  return mRecvStreamConfig.rtp.remote_ssrc == 0
+             ? Nothing()
+             : Some(mRecvStreamConfig.rtp.remote_ssrc);
+}
+
+Maybe<webrtc::AudioReceiveStreamInterface::Stats>
+WebrtcAudioConduit::GetReceiverStats() const {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  if (!mRecvStream) {
+    return Nothing();
+  }
+  return Some(mRecvStream->GetStats());
+}
+
+Maybe<webrtc::AudioSendStream::Stats> WebrtcAudioConduit::GetSenderStats()
+    const {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  if (!mSendStream) {
+    return Nothing();
+  }
+  return Some(mSendStream->GetStats());
+}
+
+Maybe<webrtc::CallBasicStats> WebrtcAudioConduit::GetCallStats() const {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  if (!mCall->Call()) {
+    return Nothing();
+  }
+  return Some(mCall->Call()->GetStats());
+}
+
+void WebrtcAudioConduit::OnRtcpBye() { mRtcpByeEvent.Notify(); }
+
+void WebrtcAudioConduit::OnRtcpTimeout() { mRtcpTimeoutEvent.Notify(); }
+
+void WebrtcAudioConduit::SetTransportActive(bool aActive) {
+  MOZ_ASSERT(mStsThread->IsOnCurrentThread());
+  if (mTransportActive == aActive) {
+    return;
+  }
+
+  // If false, This stops us from sending
+  mTransportActive = aActive;
+
+  // We queue this because there might be notifications to these listeners
+  // pending, and we don't want to drop them by letting this jump ahead of
+  // those notifications. We move the listeners into the lambda in case the
+  // transport comes back up before we disconnect them. (The Connect calls
+  // happen in MediaPipeline)
+  // We retain a strong reference to ourself, because the listeners are holding
+  // a non-refcounted reference to us, and moving them into the lambda could
+  // conceivably allow them to outlive us.
+  if (!aActive) {
+    MOZ_ALWAYS_SUCCEEDS(mCallThread->Dispatch(NS_NewRunnableFunction(
+        __func__,
+        [self = RefPtr<WebrtcAudioConduit>(this),
+         recvRtpListener = std::move(mReceiverRtpEventListener),
+         recvRtcpListener = std::move(mReceiverRtcpEventListener),
+         sendRtcpListener = std::move(mSenderRtcpEventListener)]() mutable {
+          recvRtpListener.DisconnectIfExists();
+          recvRtcpListener.DisconnectIfExists();
+          sendRtcpListener.DisconnectIfExists();
+        })));
+  }
+}
+
+// AudioSessionConduit Implementation
+MediaConduitErrorCode WebrtcAudioConduit::SendAudioFrame(
+    std::unique_ptr<webrtc::AudioFrame> frame) {
+  CSFLogDebug(LOGTAG, "%s ", __FUNCTION__);
+  // Following checks need to be performed
+  // 1. Non null audio buffer pointer, and
+  // 2. Valid sample rate, and
+  // 3. Appropriate Sample Length for 10 ms audio-frame. This represents the
+  //    block size used upstream for processing.
+  //    Ex: for 16000 sample rate , valid block-length is 160.
+  //    Similarly for 32000 sample rate, valid block length is 320.
+
+  if (!frame->data() ||
+      (IsSamplingFreqSupported(frame->sample_rate_hz()) == false) ||
+      ((frame->samples_per_channel() % (frame->sample_rate_hz() / 100) != 0))) {
+    CSFLogError(LOGTAG, "%s Invalid Parameters ", __FUNCTION__);
+    MOZ_ASSERT(PR_FALSE);
+    return kMediaConduitMalformedArgument;
+  }
+
+  // This is the AudioProxyThread, blocking it for a bit is fine.
+  AutoReadLock lock(mLock);
+  if (!mSendStreamRunning) {
+    CSFLogError(LOGTAG, "%s Engine not transmitting ", __FUNCTION__);
+    return kMediaConduitSessionNotInited;
+  }
+
+  mSendStream->SendAudioData(std::move(frame));
+  return kMediaConduitNoError;
+}
+
+MediaConduitErrorCode WebrtcAudioConduit::GetAudioFrame(
+    int32_t samplingFreqHz, webrtc::AudioFrame* frame) {
+  CSFLogDebug(LOGTAG, "%s ", __FUNCTION__);
+
+  // validate params
+  if (!frame) {
+    CSFLogError(LOGTAG, "%s Null Audio Buffer Pointer", __FUNCTION__);
+    MOZ_ASSERT(PR_FALSE);
+    return kMediaConduitMalformedArgument;
+  }
+
+  // Validate sample length
+  if (GetNum10msSamplesForFrequency(samplingFreqHz) == 0) {
+    CSFLogError(LOGTAG, "%s Invalid Sampling Frequency ", __FUNCTION__);
+    MOZ_ASSERT(PR_FALSE);
+    return kMediaConduitMalformedArgument;
+  }
+
+  // If the lock is taken, skip this chunk to avoid blocking the audio thread.
+  AutoTryReadLock tryLock(mLock);
+  if (!tryLock) {
+    CSFLogError(LOGTAG, "%s Conduit going through negotiation ", __FUNCTION__);
+    return kMediaConduitPlayoutError;
+  }
+
+  // Conduit should have reception enabled before we ask for decoded
+  // samples
+  if (!mRecvStreamRunning) {
+    CSFLogError(LOGTAG, "%s Engine not Receiving ", __FUNCTION__);
+    return kMediaConduitSessionNotInited;
+  }
+
+  // Unfortunate to have to cast to an internal class, but that looks like the
+  // only way short of interfacing with a layer above (which mixes all streams,
+  // which we don't want) or a layer below (which we try to avoid because it is
+  // less stable).
+  auto info = static_cast<webrtc::AudioReceiveStreamImpl*>(mRecvStream)
+                  ->GetAudioFrameWithInfo(samplingFreqHz, frame);
+
+  if (info == webrtc::AudioMixer::Source::AudioFrameInfo::kError) {
+    CSFLogError(LOGTAG, "%s Getting audio frame failed", __FUNCTION__);
+    return kMediaConduitPlayoutError;
+  }
+
+  CSFLogDebug(LOGTAG, "%s Got %zu channels of %zu samples", __FUNCTION__,
+              frame->num_channels(), frame->samples_per_channel());
+  return kMediaConduitNoError;
+}
+
+// Transport Layer Callbacks
+void WebrtcAudioConduit::OnRtpReceived(webrtc::RtpPacketReceived&& aPacket,
+                                       webrtc::RTPHeader&& aHeader) {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+
+  if (mAllowSsrcChange && mRecvStreamConfig.rtp.remote_ssrc != aHeader.ssrc) {
+    CSFLogDebug(LOGTAG, "%s: switching from SSRC %u to %u", __FUNCTION__,
+                mRecvStreamConfig.rtp.remote_ssrc, aHeader.ssrc);
+    OverrideRemoteSSRC(aHeader.ssrc);
+  }
+
+  CSFLogVerbose(LOGTAG, "%s: seq# %u, Len %zu, SSRC %u (0x%x) ", __FUNCTION__,
+                aPacket.SequenceNumber(), aPacket.size(), aPacket.Ssrc(),
+                aPacket.Ssrc());
+
+  mRtpPacketEvent.Notify();
+  if (mCall->Call()) {
+    mCall->Call()->Receiver()->DeliverRtpPacket(
+        webrtc::MediaType::AUDIO, std::move(aPacket),
+        [self = RefPtr<WebrtcAudioConduit>(this)](
+            const webrtc::RtpPacketReceived& packet) {
+          CSFLogVerbose(
+              LOGTAG,
+              "AudioConduit %p: failed demuxing packet, ssrc: %u seq: %u",
+              self.get(), packet.Ssrc(), packet.SequenceNumber());
+          return false;
+        });
+  }
+}
+
+void WebrtcAudioConduit::OnRtcpReceived(MediaPacket&& aPacket) {
+  CSFLogDebug(LOGTAG, "%s", __FUNCTION__);
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+
+  if (mCall->Call()) {
+    mCall->Call()->Receiver()->DeliverRtcpPacket(
+        rtc::CopyOnWriteBuffer(aPacket.data(), aPacket.len()));
+  }
+}
+
+Maybe<uint16_t> WebrtcAudioConduit::RtpSendBaseSeqFor(uint32_t aSsrc) const {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  auto it = mRtpSendBaseSeqs.find(aSsrc);
+  if (it == mRtpSendBaseSeqs.end()) {
+    return Nothing();
+  }
+  return Some(it->second);
+}
+
+const dom::RTCStatsTimestampMaker& WebrtcAudioConduit::GetTimestampMaker()
+    const {
+  return mCall->GetTimestampMaker();
+}
+
+void WebrtcAudioConduit::StopTransmitting() {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  MOZ_ASSERT(!mLock.LockedForWritingByCurrentThread());
+
+  if (!mSendStreamRunning) {
+    return;
+  }
+
+  if (mSendStream) {
+    CSFLogDebug(LOGTAG, "%s Stopping send stream", __FUNCTION__);
+    mSendStream->Stop();
+  }
+
+  mSendStreamRunning = false;
+}
+
+void WebrtcAudioConduit::StartTransmitting() {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  MOZ_ASSERT(mSendStream);
+  MOZ_ASSERT(!mLock.LockedForWritingByCurrentThread());
+
+  if (mSendStreamRunning) {
+    return;
+  }
+
+  CSFLogDebug(LOGTAG, "%s Starting send stream", __FUNCTION__);
+
+  mCall->Call()->SignalChannelNetworkState(webrtc::MediaType::AUDIO,
+                                           webrtc::kNetworkUp);
+  mSendStream->Start();
+  mSendStreamRunning = true;
+}
+
+void WebrtcAudioConduit::StopReceiving() {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  MOZ_ASSERT(!mLock.LockedForWritingByCurrentThread());
+
+  if (!mRecvStreamRunning) {
+    return;
+  }
+
+  if (mRecvStream) {
+    CSFLogDebug(LOGTAG, "%s Stopping recv stream", __FUNCTION__);
+    mRecvStream->Stop();
+  }
+
+  mRecvStreamRunning = false;
+}
+
+void WebrtcAudioConduit::StartReceiving() {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  MOZ_ASSERT(mRecvStream);
+  MOZ_ASSERT(!mLock.LockedForWritingByCurrentThread());
+
+  if (mRecvStreamRunning) {
+    return;
+  }
+
+  CSFLogDebug(LOGTAG, "%s Starting receive stream (SSRC %u (0x%x))",
+              __FUNCTION__, mRecvStreamConfig.rtp.remote_ssrc,
+              mRecvStreamConfig.rtp.remote_ssrc);
+
+  mCall->Call()->SignalChannelNetworkState(webrtc::MediaType::AUDIO,
+                                           webrtc::kNetworkUp);
+  mRecvStream->Start();
+  mRecvStreamRunning = true;
+}
+
+bool WebrtcAudioConduit::SendRtp(const uint8_t* aData, size_t aLength,
+                                 const webrtc::PacketOptions& aOptions) {
+  MOZ_ASSERT(aLength >= 12);
+  const uint16_t seqno = ntohs(*((uint16_t*)&aData[2]));
+  const uint32_t ssrc = ntohl(*((uint32_t*)&aData[8]));
+
+  CSFLogVerbose(
+      LOGTAG,
+      "AudioConduit %p: Sending RTP Packet seq# %u, len %zu, SSRC %u (0x%x)",
+      this, seqno, aLength, ssrc, ssrc);
+
+  if (!mTransportActive) {
+    CSFLogError(LOGTAG, "AudioConduit %p: RTP Packet Send Failed ", this);
+    return false;
+  }
+
+  MediaPacket packet;
+  packet.Copy(aData, aLength, aLength + SRTP_MAX_EXPANSION);
+  packet.SetType(MediaPacket::RTP);
+  mSenderRtpSendEvent.Notify(std::move(packet));
+
+  // Parse the sequence number of the first rtp packet as base_seq.
+  const auto inserted = mRtpSendBaseSeqs_n.insert({ssrc, seqno}).second;
+
+  if (inserted || aOptions.packet_id >= 0) {
+    int64_t now_ms = PR_Now() / 1000;
+    MOZ_ALWAYS_SUCCEEDS(mCallThread->Dispatch(NS_NewRunnableFunction(
+        __func__, [this, self = RefPtr<WebrtcAudioConduit>(this),
+                   packet_id = aOptions.packet_id, now_ms, ssrc, seqno] {
+          mRtpSendBaseSeqs.insert({ssrc, seqno});
+          if (packet_id >= 0) {
+            if (mCall->Call()) {
+              // TODO: This notification should ideally happen after the
+              // transport layer has sent the packet on the wire.
+              mCall->Call()->OnSentPacket({packet_id, now_ms});
+            }
+          }
+        })));
+  }
+  return true;
+}
+
+bool WebrtcAudioConduit::SendSenderRtcp(const uint8_t* aData, size_t aLength) {
+  CSFLogVerbose(
+      LOGTAG,
+      "AudioConduit %p: Sending RTCP SR Packet, len %zu, SSRC %u (0x%x)", this,
+      aLength, (uint32_t)ntohl(*((uint32_t*)&aData[4])),
+      (uint32_t)ntohl(*((uint32_t*)&aData[4])));
+
+  if (!mTransportActive) {
+    CSFLogError(LOGTAG, "%s RTCP SR Packet Send Failed ", __FUNCTION__);
+    return false;
+  }
+
+  MediaPacket packet;
+  packet.Copy(aData, aLength, aLength + SRTP_MAX_EXPANSION);
+  packet.SetType(MediaPacket::RTCP);
+  mSenderRtcpSendEvent.Notify(std::move(packet));
+  return true;
+}
+
+bool WebrtcAudioConduit::SendReceiverRtcp(const uint8_t* aData,
+                                          size_t aLength) {
+  CSFLogVerbose(
+      LOGTAG,
+      "AudioConduit %p: Sending RTCP RR Packet, len %zu, SSRC %u (0x%x)", this,
+      aLength, (uint32_t)ntohl(*((uint32_t*)&aData[4])),
+      (uint32_t)ntohl(*((uint32_t*)&aData[4])));
+
+  if (!mTransportActive) {
+    CSFLogError(LOGTAG, "AudioConduit %p: RTCP RR Packet Send Failed", this);
+    return false;
+  }
+
+  MediaPacket packet;
+  packet.Copy(aData, aLength, aLength + SRTP_MAX_EXPANSION);
+  packet.SetType(MediaPacket::RTCP);
+  mReceiverRtcpSendEvent.Notify(std::move(packet));
+  return true;
+}
+
+/**
+ *  Supported Sampling Frequencies.
+ */
+bool WebrtcAudioConduit::IsSamplingFreqSupported(int freq) const {
+  return GetNum10msSamplesForFrequency(freq) != 0;
+}
+
+std::vector<webrtc::RtpSource> WebrtcAudioConduit::GetUpstreamRtpSources()
+    const {
+  MOZ_ASSERT(NS_IsMainThread());
+  std::vector<webrtc::RtpSource> sources;
+  {
+    AutoReadLock lock(mLock);
+    if (mRecvStream) {
+      sources = mRecvStream->GetSources();
+    }
+  }
+  return sources;
+}
+
+/* Return block-length of 10 ms audio frame in number of samples */
+unsigned int WebrtcAudioConduit::GetNum10msSamplesForFrequency(
+    int samplingFreqHz) const {
+  switch (samplingFreqHz) {
+    case 16000:
+      return 160;  // 160 samples
+    case 32000:
+      return 320;  // 320 samples
+    case 44100:
+      return 441;  // 441 samples
+    case 48000:
+      return 480;  // 480 samples
+    default:
+      return 0;  // invalid or unsupported
+  }
+}
+
+/**
+ * Perform validation on the codecConfig to be applied.
+ * Verifies if the codec is already applied.
+ */
+MediaConduitErrorCode WebrtcAudioConduit::ValidateCodecConfig(
+    const AudioCodecConfig& codecInfo, bool send) {
+  if (codecInfo.mName.empty()) {
+    CSFLogError(LOGTAG, "%s Empty Payload Name ", __FUNCTION__);
+    return kMediaConduitMalformedArgument;
+  }
+
+  // Only mono or stereo channels supported
+  if ((codecInfo.mChannels != 1) && (codecInfo.mChannels != 2)) {
+    CSFLogError(LOGTAG, "%s Channel Unsupported ", __FUNCTION__);
+    return kMediaConduitMalformedArgument;
+  }
+
+  return kMediaConduitNoError;
+}
+
+RtpExtList WebrtcAudioConduit::FilterExtensions(LocalDirection aDirection,
+                                                const RtpExtList& aExtensions) {
+  const bool isSend = aDirection == LocalDirection::kSend;
+  RtpExtList filteredExtensions;
+
+  for (const auto& extension : aExtensions) {
+    // ssrc-audio-level RTP header extension
+    if (extension.uri == webrtc::RtpExtension::kAudioLevelUri) {
+      filteredExtensions.push_back(
+          webrtc::RtpExtension(extension.uri, extension.id));
+    }
+
+    // csrc-audio-level RTP header extension
+    if (extension.uri == webrtc::RtpExtension::kCsrcAudioLevelsUri) {
+      if (isSend) {
+        continue;
+      }
+      filteredExtensions.push_back(
+          webrtc::RtpExtension(extension.uri, extension.id));
+    }
+
+    // MID RTP header extension
+    if (extension.uri == webrtc::RtpExtension::kMidUri) {
+      if (!isSend) {
+        // TODO: recv mid support, see also bug 1727211
+        continue;
+      }
+      filteredExtensions.push_back(
+          webrtc::RtpExtension(extension.uri, extension.id));
+    }
+  }
+
+  return filteredExtensions;
+}
+
+webrtc::SdpAudioFormat WebrtcAudioConduit::CodecConfigToLibwebrtcFormat(
+    const AudioCodecConfig& aConfig) {
+  webrtc::SdpAudioFormat::Parameters parameters;
+  if (aConfig.mName == kOpusCodecName) {
+    if (aConfig.mChannels == 2) {
+      parameters[kCodecParamStereo] = kParamValueTrue;
+    }
+    if (aConfig.mFECEnabled) {
+      parameters[kCodecParamUseInbandFec] = kParamValueTrue;
+    }
+    if (aConfig.mDTXEnabled) {
+      parameters[kCodecParamUseDtx] = kParamValueTrue;
+    }
+    if (aConfig.mMaxPlaybackRate) {
+      parameters[kCodecParamMaxPlaybackRate] =
+          std::to_string(aConfig.mMaxPlaybackRate);
+    }
+    if (aConfig.mMaxAverageBitrate) {
+      parameters[kCodecParamMaxAverageBitrate] =
+          std::to_string(aConfig.mMaxAverageBitrate);
+    }
+    if (aConfig.mFrameSizeMs) {
+      parameters[kCodecParamPTime] = std::to_string(aConfig.mFrameSizeMs);
+    }
+    if (aConfig.mMinFrameSizeMs) {
+      parameters[kCodecParamMinPTime] = std::to_string(aConfig.mMinFrameSizeMs);
+    }
+    if (aConfig.mMaxFrameSizeMs) {
+      parameters[kCodecParamMaxPTime] = std::to_string(aConfig.mMaxFrameSizeMs);
+    }
+    if (aConfig.mCbrEnabled) {
+      parameters[kCodecParamCbr] = kParamValueTrue;
+    }
+  }
+
+  return webrtc::SdpAudioFormat(aConfig.mName, aConfig.mFreq, aConfig.mChannels,
+                                parameters);
+}
+
+void WebrtcAudioConduit::DeleteSendStream() {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  MOZ_ASSERT(mLock.LockedForWritingByCurrentThread());
+
+  if (!mSendStream) {
+    return;
+  }
+
+  mCall->Call()->DestroyAudioSendStream(mSendStream);
+  mSendStreamRunning = false;
+  mSendStream = nullptr;
+
+  // Reset base_seqs in case ssrcs get re-used.
+  mRtpSendBaseSeqs.clear();
+}
+
+void WebrtcAudioConduit::CreateSendStream() {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  MOZ_ASSERT(mLock.LockedForWritingByCurrentThread());
+
+  if (mSendStream) {
+    return;
+  }
+
+  mSendStream = mCall->Call()->CreateAudioSendStream(mSendStreamConfig);
+}
+
+void WebrtcAudioConduit::DeleteRecvStream() {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  MOZ_ASSERT(mLock.LockedForWritingByCurrentThread());
+
+  if (!mRecvStream) {
+    return;
+  }
+
+  mCall->Call()->DestroyAudioReceiveStream(mRecvStream);
+  mRecvStreamRunning = false;
+  mRecvStream = nullptr;
+}
+
+void WebrtcAudioConduit::CreateRecvStream() {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  MOZ_ASSERT(mLock.LockedForWritingByCurrentThread());
+
+  if (mRecvStream) {
+    return;
+  }
+
+  mRecvStream = mCall->Call()->CreateAudioReceiveStream(mRecvStreamConfig);
+  // Ensure that we set the jitter buffer target on this stream.
+  mRecvStream->SetBaseMinimumPlayoutDelayMs(mJitterBufferTargetMs);
+}
+
+void WebrtcAudioConduit::SetJitterBufferTarget(DOMHighResTimeStamp aTargetMs) {
+  MOZ_RELEASE_ASSERT(aTargetMs <= std::numeric_limits<uint16_t>::max());
+  MOZ_RELEASE_ASSERT(aTargetMs >= 0);
+
+  MOZ_ALWAYS_SUCCEEDS(mCallThread->Dispatch(NS_NewRunnableFunction(
+      __func__,
+      [this, self = RefPtr<WebrtcAudioConduit>(this), targetMs = aTargetMs] {
+        mJitterBufferTargetMs = static_cast<uint16_t>(targetMs);
+        if (mRecvStream) {
+          mRecvStream->SetBaseMinimumPlayoutDelayMs(targetMs);
+        }
+      })));
+}
+
+void WebrtcAudioConduit::DeliverPacket(rtc::CopyOnWriteBuffer packet,
+                                       PacketType type) {
+  // Currently unused.
+  MOZ_ASSERT(false);
+}
+
+Maybe<int> WebrtcAudioConduit::ActiveSendPayloadType() const {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+
+  auto stats = GetSenderStats();
+  if (!stats) {
+    return Nothing();
+  }
+
+  if (!stats->codec_payload_type) {
+    return Nothing();
+  }
+
+  return Some(*stats->codec_payload_type);
+}
+
+Maybe<int> WebrtcAudioConduit::ActiveRecvPayloadType() const {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+
+  auto stats = GetReceiverStats();
+  if (!stats) {
+    return Nothing();
+  }
+
+  if (!stats->codec_payload_type) {
+    return Nothing();
+  }
+
+  return Some(*stats->codec_payload_type);
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/libwebrtcglue/AudioConduit.h b/dom/media/webrtc/libwebrtcglue/AudioConduit.h
new file mode 100644
index 0000000000..e8de331e12
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/AudioConduit.h
@@ -0,0 +1,303 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef AUDIO_SESSION_H_
+#define AUDIO_SESSION_H_
+
+#include "mozilla/Attributes.h"
+#include "mozilla/ReentrantMonitor.h"
+#include "mozilla/RWLock.h"
+#include "mozilla/StateMirroring.h"
+#include "mozilla/TimeStamp.h"
+
+#include "MediaConduitInterface.h"
+#include "common/MediaEngineWrapper.h"
+
+/**
+ * This file hosts several structures identifying different aspects of a RTP
+ * Session.
+ */
+namespace mozilla {
+
+struct DtmfEvent;
+
+/**
+ * Concrete class for Audio session. Hooks up
+ *  - media-source and target to external transport
+ */
+class WebrtcAudioConduit : public AudioSessionConduit,
+                           public webrtc::RtcpEventObserver {
+ public:
+  Maybe<int> ActiveSendPayloadType() const override;
+  Maybe<int> ActiveRecvPayloadType() const override;
+
+  void OnRtpReceived(webrtc::RtpPacketReceived&& aPacket,
+                     webrtc::RTPHeader&& aHeader);
+  void OnRtcpReceived(MediaPacket&& aPacket);
+
+  void OnRtcpBye() override;
+  void OnRtcpTimeout() override;
+
+  void SetTransportActive(bool aActive) override;
+
+  MediaEventSourceExc<MediaPacket>& SenderRtpSendEvent() override {
+    return mSenderRtpSendEvent;
+  }
+  MediaEventSourceExc<MediaPacket>& SenderRtcpSendEvent() override {
+    return mSenderRtcpSendEvent;
+  }
+  MediaEventSourceExc<MediaPacket>& ReceiverRtcpSendEvent() override {
+    return mReceiverRtcpSendEvent;
+  }
+  void ConnectReceiverRtpEvent(
+      MediaEventSourceExc<webrtc::RtpPacketReceived, webrtc::RTPHeader>& aEvent)
+      override {
+    mReceiverRtpEventListener =
+        aEvent.Connect(mCallThread, this, &WebrtcAudioConduit::OnRtpReceived);
+  }
+  void ConnectReceiverRtcpEvent(
+      MediaEventSourceExc<MediaPacket>& aEvent) override {
+    mReceiverRtcpEventListener =
+        aEvent.Connect(mCallThread, this, &WebrtcAudioConduit::OnRtcpReceived);
+  }
+  void ConnectSenderRtcpEvent(
+      MediaEventSourceExc<MediaPacket>& aEvent) override {
+    mSenderRtcpEventListener =
+        aEvent.Connect(mCallThread, this, &WebrtcAudioConduit::OnRtcpReceived);
+  }
+
+  Maybe<uint16_t> RtpSendBaseSeqFor(uint32_t aSsrc) const override;
+
+  const dom::RTCStatsTimestampMaker& GetTimestampMaker() const override;
+
+  void StopTransmitting();
+  void StartTransmitting();
+  void StopReceiving();
+  void StartReceiving();
+
+  /**
+   * Function to deliver externally captured audio sample for encoding and
+   * transport
+   * @param frame [in]: AudioFrame in upstream's format for forwarding to the
+   *                    send stream. Ownership is passed along.
+   * NOTE: ConfigureSendMediaCodec() SHOULD be called before this function can
+   * be invoked. This ensures the inserted audio-samples can be transmitted by
+   * the conduit.
+   */
+  MediaConduitErrorCode SendAudioFrame(
+      std::unique_ptr<webrtc::AudioFrame> frame) override;
+
+  /**
+   * Function to grab a decoded audio-sample from the media engine for
+   * rendering / playout of length 10 milliseconds.
+   *
+   * @param samplingFreqHz [in]: Frequency of the sampling for playback in
+   *                             Hertz (16000, 32000,..)
+   * @param frame [in/out]: Pointer to an AudioFrame to which audio data will be
+   *                        copied
+   * NOTE: This function should be invoked every 10 milliseconds for the best
+   *       performance
+   * NOTE: ConfigureRecvMediaCodec() SHOULD be called before this function can
+   *       be invoked
+   * This ensures the decoded samples are ready for reading and playout is
+   * enabled.
+   */
+  MediaConduitErrorCode GetAudioFrame(int32_t samplingFreqHz,
+                                      webrtc::AudioFrame* frame) override;
+
+  bool SendRtp(const uint8_t* aData, size_t aLength,
+               const webrtc::PacketOptions& aOptions) override;
+  bool SendSenderRtcp(const uint8_t* aData, size_t aLength) override;
+  bool SendReceiverRtcp(const uint8_t* aData, size_t aLength) override;
+
+  bool HasCodecPluginID(uint64_t aPluginID) const override { return false; }
+
+  void SetJitterBufferTarget(DOMHighResTimeStamp aTargetMs) override;
+
+  void DeliverPacket(rtc::CopyOnWriteBuffer packet, PacketType type) override;
+
+  RefPtr<GenericPromise> Shutdown() override;
+
+  WebrtcAudioConduit(RefPtr<WebrtcCallWrapper> aCall,
+                     nsCOMPtr<nsISerialEventTarget> aStsThread);
+
+  virtual ~WebrtcAudioConduit();
+
+  // Call thread.
+  void InitControl(AudioConduitControlInterface* aControl) override;
+
+  // Handle a DTMF event from mControl.mOnDtmfEventListener.
+  void OnDtmfEvent(const DtmfEvent& aEvent);
+
+  // Called when a parameter in mControl has changed. Call thread.
+  void OnControlConfigChange();
+
+  Ssrcs GetLocalSSRCs() const override;
+  Maybe<Ssrc> GetRemoteSSRC() const override;
+
+  void DisableSsrcChanges() override {
+    MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+    mAllowSsrcChange = false;
+  }
+
+ private:
+  /**
+   * Override the remote ssrc configured on mRecvStreamConfig.
+   *
+   * Recreates and restarts the recv stream if needed. The overriden value is
+   * overwritten the next time the mControl.mRemoteSsrc mirror changes value.
+   *
+   * Call thread only.
+   */
+  bool OverrideRemoteSSRC(uint32_t aSsrc);
+
+ public:
+  void UnsetRemoteSSRC(uint32_t aSsrc) override {}
+
+  Maybe<webrtc::AudioReceiveStreamInterface::Stats> GetReceiverStats()
+      const override;
+  Maybe<webrtc::AudioSendStream::Stats> GetSenderStats() const override;
+  Maybe<webrtc::CallBasicStats> GetCallStats() const override;
+
+  bool IsSamplingFreqSupported(int freq) const override;
+
+  MediaEventSource<void>& RtcpByeEvent() override { return mRtcpByeEvent; }
+  MediaEventSource<void>& RtcpTimeoutEvent() override {
+    return mRtcpTimeoutEvent;
+  }
+  MediaEventSource<void>& RtpPacketEvent() override { return mRtpPacketEvent; }
+
+  std::vector<webrtc::RtpSource> GetUpstreamRtpSources() const override;
+
+ private:
+  WebrtcAudioConduit(const WebrtcAudioConduit& other) = delete;
+  void operator=(const WebrtcAudioConduit& other) = delete;
+
+  // Generate block size in sample length for a given sampling frequency
+  unsigned int GetNum10msSamplesForFrequency(int samplingFreqHz) const;
+
+  // Checks the codec to be applied
+  static MediaConduitErrorCode ValidateCodecConfig(
+      const AudioCodecConfig& codecInfo, bool send);
+  /**
+   * Of all extensions in aExtensions, returns a list of supported extensions.
+   */
+  static RtpExtList FilterExtensions(
+      MediaSessionConduitLocalDirection aDirection,
+      const RtpExtList& aExtensions);
+  static webrtc::SdpAudioFormat CodecConfigToLibwebrtcFormat(
+      const AudioCodecConfig& aConfig);
+
+  void CreateSendStream();
+  void DeleteSendStream();
+  void CreateRecvStream();
+  void DeleteRecvStream();
+
+  // Are SSRC changes without signaling allowed or not.
+  // Call thread only.
+  bool mAllowSsrcChange = true;
+
+  // Const so can be accessed on any thread. Most methods are called on the Call
+  // thread.
+  const RefPtr<WebrtcCallWrapper> mCall;
+
+  // Set up in the ctor and then not touched. Called through by the streams on
+  // any thread.
+  WebrtcSendTransport mSendTransport;
+  WebrtcReceiveTransport mRecvTransport;
+
+  // Accessed only on the Call thread.
+  webrtc::AudioReceiveStreamInterface::Config mRecvStreamConfig;
+
+  // Written only on the Call thread. Guarded by mLock, except for reads on the
+  // Call thread.
+  webrtc::AudioReceiveStreamInterface* mRecvStream;
+
+  // Accessed only on the Call thread.
+  webrtc::AudioSendStream::Config mSendStreamConfig;
+
+  // Written only on the Call thread. Guarded by mLock, except for reads on the
+  // Call thread.
+  webrtc::AudioSendStream* mSendStream;
+
+  // If true => mSendStream started and not stopped
+  // Written only on the Call thread.
+  Atomic<bool> mSendStreamRunning;
+  // If true => mRecvStream started and not stopped
+  // Written only on the Call thread.
+  Atomic<bool> mRecvStreamRunning;
+
+  // Accessed only on the Call thread.
+  bool mDtmfEnabled;
+
+  mutable RWLock mLock MOZ_UNANNOTATED;
+
+  // Call worker thread. All access to mCall->Call() happens here.
+  const RefPtr<AbstractThread> mCallThread;
+
+  // Socket transport service thread. Any thread.
+  const nsCOMPtr<nsISerialEventTarget> mStsThread;
+
+  // Target jitter buffer to be applied to the receive stream in milliseconds.
+  uint16_t mJitterBufferTargetMs = 0;
+
+  struct Control {
+    // Mirrors and events that map to AudioConduitControlInterface for control.
+    // Call thread only.
+    Mirror<bool> mReceiving;
+    Mirror<bool> mTransmitting;
+    Mirror<Ssrcs> mLocalSsrcs;
+    Mirror<std::string> mLocalCname;
+    Mirror<std::string> mMid;
+    Mirror<Ssrc> mRemoteSsrc;
+    Mirror<std::string> mSyncGroup;
+    Mirror<RtpExtList> mLocalRecvRtpExtensions;
+    Mirror<RtpExtList> mLocalSendRtpExtensions;
+    Mirror<Maybe<AudioCodecConfig>> mSendCodec;
+    Mirror<std::vector<AudioCodecConfig>> mRecvCodecs;
+    MediaEventListener mOnDtmfEventListener;
+
+    // For caching mRemoteSsrc, since another caller may change the remote ssrc
+    // in the stream config directly.
+    Ssrc mConfiguredRemoteSsrc = 0;
+    // For tracking changes to mSendCodec.
+    Maybe<AudioCodecConfig> mConfiguredSendCodec;
+    // For tracking changes to mRecvCodecs.
+    std::vector<AudioCodecConfig> mConfiguredRecvCodecs;
+
+    Control() = delete;
+    explicit Control(const RefPtr<AbstractThread>& aCallThread);
+  } mControl;
+
+  // WatchManager allowing Mirrors to trigger functions that will update the
+  // webrtc.org configuration.
+  WatchManager<WebrtcAudioConduit> mWatchManager;
+
+  // Accessed from mStsThread. Last successfully polled RTT
+  Maybe<DOMHighResTimeStamp> mRttSec;
+
+  // Call thread only. ssrc -> base_seq
+  std::map<uint32_t, uint16_t> mRtpSendBaseSeqs;
+  // libwebrtc network thread only. ssrc -> base_seq.
+  // To track changes needed to mRtpSendBaseSeqs.
+  std::map<uint32_t, uint16_t> mRtpSendBaseSeqs_n;
+
+  // Thread safe
+  Atomic<bool> mTransportActive = Atomic<bool>(false);
+  MediaEventProducer<void> mRtcpByeEvent;
+  MediaEventProducer<void> mRtcpTimeoutEvent;
+  MediaEventProducer<void> mRtpPacketEvent;
+  MediaEventProducerExc<MediaPacket> mSenderRtpSendEvent;
+  MediaEventProducerExc<MediaPacket> mSenderRtcpSendEvent;
+  MediaEventProducerExc<MediaPacket> mReceiverRtcpSendEvent;
+
+  // Assigned and revoked on mStsThread. Listeners for receiving packets.
+  MediaEventListener mSenderRtcpEventListener;    // Rtp-transmitting pipeline
+  MediaEventListener mReceiverRtcpEventListener;  // Rtp-receiving pipeline
+  MediaEventListener mReceiverRtpEventListener;   // Rtp-receiving pipeline
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/libwebrtcglue/CallWorkerThread.h b/dom/media/webrtc/libwebrtcglue/CallWorkerThread.h
new file mode 100644
index 0000000000..12d21fbee4
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/CallWorkerThread.h
@@ -0,0 +1,116 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef DOM_MEDIA_WEBRTC_LIBWEBRTCGLUE_CALLWORKERTHREAD_H_
+#define DOM_MEDIA_WEBRTC_LIBWEBRTCGLUE_CALLWORKERTHREAD_H_
+
+#include "mozilla/AbstractThread.h"
+#include "nsIDirectTaskDispatcher.h"
+#include "TaskQueueWrapper.h"
+
+namespace mozilla {
+
+// Implements AbstractThread for running things on the webrtc TaskQueue.
+// Webrtc TaskQueues are not refcounted so cannot implement AbstractThread
+// directly.
+class CallWorkerThread final : public AbstractThread,
+                               public nsIDirectTaskDispatcher {
+ public:
+  NS_DECL_THREADSAFE_ISUPPORTS
+  NS_DECL_NSIDIRECTTASKDISPATCHER
+
+  explicit CallWorkerThread(
+      UniquePtr<TaskQueueWrapper<DeletionPolicy::NonBlocking>> aWebrtcTaskQueue)
+      : AbstractThread(aWebrtcTaskQueue->mTaskQueue->SupportsTailDispatch()),
+        mWebrtcTaskQueue(std::move(aWebrtcTaskQueue)) {}
+
+  // AbstractThread overrides
+  nsresult Dispatch(already_AddRefed<nsIRunnable> aRunnable,
+                    DispatchReason aReason) override;
+  bool IsCurrentThreadIn() const override;
+  TaskDispatcher& TailDispatcher() override;
+  nsIEventTarget* AsEventTarget() override;
+  NS_IMETHOD
+  DelayedDispatch(already_AddRefed<nsIRunnable> aEvent,
+                  uint32_t aDelayMs) override;
+
+  NS_IMETHOD RegisterShutdownTask(nsITargetShutdownTask* aTask) override;
+  NS_IMETHOD UnregisterShutdownTask(nsITargetShutdownTask* aTask) override;
+
+  const UniquePtr<TaskQueueWrapper<DeletionPolicy::NonBlocking>>
+      mWebrtcTaskQueue;
+
+ protected:
+  ~CallWorkerThread() = default;
+};
+
+NS_IMPL_ISUPPORTS(CallWorkerThread, nsIDirectTaskDispatcher,
+                  nsISerialEventTarget, nsIEventTarget);
+
+//-----------------------------------------------------------------------------
+// AbstractThread
+//-----------------------------------------------------------------------------
+
+nsresult CallWorkerThread::Dispatch(already_AddRefed<nsIRunnable> aRunnable,
+                                    DispatchReason aReason) {
+  RefPtr<nsIRunnable> runnable = aRunnable;
+  return mWebrtcTaskQueue->mTaskQueue->Dispatch(
+      mWebrtcTaskQueue->CreateTaskRunner(std::move(runnable)), aReason);
+}
+
+bool CallWorkerThread::IsCurrentThreadIn() const {
+  return mWebrtcTaskQueue->mTaskQueue->IsOnCurrentThreadInfallible() &&
+         mWebrtcTaskQueue->IsCurrent();
+}
+
+TaskDispatcher& CallWorkerThread::TailDispatcher() {
+  return mWebrtcTaskQueue->mTaskQueue->TailDispatcher();
+}
+
+nsIEventTarget* CallWorkerThread::AsEventTarget() {
+  return mWebrtcTaskQueue->mTaskQueue->AsEventTarget();
+}
+
+NS_IMETHODIMP
+CallWorkerThread::DelayedDispatch(already_AddRefed<nsIRunnable> aEvent,
+                                  uint32_t aDelayMs) {
+  RefPtr<nsIRunnable> event = aEvent;
+  return mWebrtcTaskQueue->mTaskQueue->DelayedDispatch(
+      mWebrtcTaskQueue->CreateTaskRunner(std::move(event)), aDelayMs);
+}
+
+NS_IMETHODIMP CallWorkerThread::RegisterShutdownTask(
+    nsITargetShutdownTask* aTask) {
+  return mWebrtcTaskQueue->mTaskQueue->RegisterShutdownTask(aTask);
+}
+
+NS_IMETHODIMP CallWorkerThread::UnregisterShutdownTask(
+    nsITargetShutdownTask* aTask) {
+  return mWebrtcTaskQueue->mTaskQueue->UnregisterShutdownTask(aTask);
+}
+
+//-----------------------------------------------------------------------------
+// nsIDirectTaskDispatcher
+//-----------------------------------------------------------------------------
+
+NS_IMETHODIMP
+CallWorkerThread::DispatchDirectTask(already_AddRefed<nsIRunnable> aEvent) {
+  nsCOMPtr<nsIRunnable> event = aEvent;
+  return mWebrtcTaskQueue->mTaskQueue->DispatchDirectTask(
+      mWebrtcTaskQueue->CreateTaskRunner(std::move(event)));
+}
+
+NS_IMETHODIMP CallWorkerThread::DrainDirectTasks() {
+  return mWebrtcTaskQueue->mTaskQueue->DrainDirectTasks();
+}
+
+NS_IMETHODIMP CallWorkerThread::HaveDirectTasks(bool* aValue) {
+  return mWebrtcTaskQueue->mTaskQueue->HaveDirectTasks(aValue);
+}
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/libwebrtcglue/CodecConfig.h b/dom/media/webrtc/libwebrtcglue/CodecConfig.h
new file mode 100644
index 0000000000..023ea98783
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/CodecConfig.h
@@ -0,0 +1,237 @@
+
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef CODEC_CONFIG_H_
+#define CODEC_CONFIG_H_
+
+#include <string>
+#include <vector>
+
+#include "common/EncodingConstraints.h"
+
+namespace mozilla {
+
+/**
+ * Minimalistic Audio Codec Config Params
+ */
+struct AudioCodecConfig {
+  /*
+   * The data-types for these properties mimic the
+   * corresponding webrtc::CodecInst data-types.
+   */
+  int mType;
+  std::string mName;
+  int mFreq;
+  int mChannels;
+
+  bool mFECEnabled;
+  bool mDtmfEnabled;
+  uint32_t mFrameSizeMs;
+  uint32_t mMaxFrameSizeMs;
+  uint32_t mMinFrameSizeMs;
+
+  // OPUS-specific
+  bool mDTXEnabled;
+  uint32_t mMaxAverageBitrate;
+  int mMaxPlaybackRate;
+  bool mCbrEnabled;
+
+  AudioCodecConfig(int type, std::string name, int freq, int channels,
+                   bool FECEnabled)
+      : mType(type),
+        mName(name),
+        mFreq(freq),
+        mChannels(channels),
+        mFECEnabled(FECEnabled),
+        mDtmfEnabled(false),
+        mFrameSizeMs(0),
+        mMaxFrameSizeMs(0),
+        mMinFrameSizeMs(0),
+        mDTXEnabled(false),
+        mMaxAverageBitrate(0),
+        mMaxPlaybackRate(0),
+        mCbrEnabled(false) {}
+
+  bool operator==(const AudioCodecConfig& aOther) const {
+    return mType == aOther.mType && mName == aOther.mName &&
+           mFreq == aOther.mFreq && mChannels == aOther.mChannels &&
+           mFECEnabled == aOther.mFECEnabled &&
+           mDtmfEnabled == aOther.mDtmfEnabled &&
+           mFrameSizeMs == aOther.mFrameSizeMs &&
+           mMaxFrameSizeMs == aOther.mMaxFrameSizeMs &&
+           mMinFrameSizeMs == aOther.mMinFrameSizeMs &&
+           mDTXEnabled == aOther.mDTXEnabled &&
+           mMaxAverageBitrate == aOther.mMaxAverageBitrate &&
+           mMaxPlaybackRate == aOther.mMaxPlaybackRate &&
+           mCbrEnabled == aOther.mCbrEnabled;
+  }
+};
+
+/*
+ * Minimalistic video codec configuration
+ * More to be added later depending on the use-case
+ */
+
+#define MAX_SPROP_LEN 128
+
+// used for holding SDP negotiation results
+struct VideoCodecConfigH264 {
+  char sprop_parameter_sets[MAX_SPROP_LEN];
+  int packetization_mode;
+  int profile_level_id;
+  int tias_bw;
+
+  bool operator==(const VideoCodecConfigH264& aOther) const {
+    return strncmp(sprop_parameter_sets, aOther.sprop_parameter_sets,
+                   MAX_SPROP_LEN) == 0 &&
+           packetization_mode == aOther.packetization_mode &&
+           profile_level_id == aOther.profile_level_id &&
+           tias_bw == aOther.tias_bw;
+  }
+};
+
+// class so the std::strings can get freed more easily/reliably
+class VideoCodecConfig {
+ public:
+  /*
+   * The data-types for these properties mimic the
+   * corresponding webrtc::VideoCodec data-types.
+   */
+  int mType;  // payload type
+  std::string mName;
+
+  std::vector<std::string> mAckFbTypes;
+  std::vector<std::string> mNackFbTypes;
+  std::vector<std::string> mCcmFbTypes;
+  // Don't pass mOtherFbTypes from JsepVideoCodecDescription because we'd have
+  // to drag SdpRtcpFbAttributeList::Feedback along too.
+  bool mRembFbSet;
+  bool mFECFbSet;
+  bool mTransportCCFbSet;
+
+  int mULPFECPayloadType;
+  int mREDPayloadType;
+  int mREDRTXPayloadType;
+  int mRTXPayloadType;
+
+  uint32_t mTias;
+  EncodingConstraints mEncodingConstraints;
+  struct Encoding {
+    std::string rid;
+    EncodingConstraints constraints;
+    bool active = true;
+    // TODO(bug 1744116): Use = default here
+    bool operator==(const Encoding& aOther) const {
+      return rid == aOther.rid && constraints == aOther.constraints &&
+             active == aOther.active;
+    }
+  };
+  std::vector<Encoding> mEncodings;
+  std::string mSpropParameterSets;
+  uint8_t mProfile;
+  uint8_t mConstraints;
+  uint8_t mLevel;
+  uint8_t mPacketizationMode;
+  // TODO: add external negotiated SPS/PPS
+
+  // TODO(bug 1744116): Use = default here
+  bool operator==(const VideoCodecConfig& aRhs) const {
+    return mType == aRhs.mType && mName == aRhs.mName &&
+           mAckFbTypes == aRhs.mAckFbTypes &&
+           mNackFbTypes == aRhs.mNackFbTypes &&
+           mCcmFbTypes == aRhs.mCcmFbTypes && mRembFbSet == aRhs.mRembFbSet &&
+           mFECFbSet == aRhs.mFECFbSet &&
+           mTransportCCFbSet == aRhs.mTransportCCFbSet &&
+           mULPFECPayloadType == aRhs.mULPFECPayloadType &&
+           mREDPayloadType == aRhs.mREDPayloadType &&
+           mREDRTXPayloadType == aRhs.mREDRTXPayloadType &&
+           mRTXPayloadType == aRhs.mRTXPayloadType && mTias == aRhs.mTias &&
+           mEncodingConstraints == aRhs.mEncodingConstraints &&
+           mEncodings == aRhs.mEncodings &&
+           mSpropParameterSets == aRhs.mSpropParameterSets &&
+           mProfile == aRhs.mProfile && mConstraints == aRhs.mConstraints &&
+           mLevel == aRhs.mLevel &&
+           mPacketizationMode == aRhs.mPacketizationMode;
+  }
+
+  VideoCodecConfig(int type, std::string name,
+                   const EncodingConstraints& constraints,
+                   const struct VideoCodecConfigH264* h264 = nullptr)
+      : mType(type),
+        mName(name),
+        mRembFbSet(false),
+        mFECFbSet(false),
+        mTransportCCFbSet(false),
+        mULPFECPayloadType(-1),
+        mREDPayloadType(-1),
+        mREDRTXPayloadType(-1),
+        mRTXPayloadType(-1),
+        mTias(0),
+        mEncodingConstraints(constraints),
+        mProfile(0x42),
+        mConstraints(0xE0),
+        mLevel(0x0C),
+        mPacketizationMode(1) {
+    if (h264) {
+      mProfile = (h264->profile_level_id & 0x00FF0000) >> 16;
+      mConstraints = (h264->profile_level_id & 0x0000FF00) >> 8;
+      mLevel = (h264->profile_level_id & 0x000000FF);
+      mPacketizationMode = h264->packetization_mode;
+      mSpropParameterSets = h264->sprop_parameter_sets;
+    }
+  }
+
+  bool ResolutionEquals(const VideoCodecConfig& aConfig) const {
+    if (mEncodings.size() != aConfig.mEncodings.size()) {
+      return false;
+    }
+    for (size_t i = 0; i < mEncodings.size(); ++i) {
+      if (!mEncodings[i].constraints.ResolutionEquals(
+              aConfig.mEncodings[i].constraints)) {
+        return false;
+      }
+    }
+    return true;
+  }
+
+  // Nothing seems to use this right now. Do we intend to support this
+  // someday?
+  bool RtcpFbAckIsSet(const std::string& type) const {
+    for (auto i = mAckFbTypes.begin(); i != mAckFbTypes.end(); ++i) {
+      if (*i == type) {
+        return true;
+      }
+    }
+    return false;
+  }
+
+  bool RtcpFbNackIsSet(const std::string& type) const {
+    for (auto i = mNackFbTypes.begin(); i != mNackFbTypes.end(); ++i) {
+      if (*i == type) {
+        return true;
+      }
+    }
+    return false;
+  }
+
+  bool RtcpFbCcmIsSet(const std::string& type) const {
+    for (auto i = mCcmFbTypes.begin(); i != mCcmFbTypes.end(); ++i) {
+      if (*i == type) {
+        return true;
+      }
+    }
+    return false;
+  }
+
+  bool RtcpFbRembIsSet() const { return mRembFbSet; }
+
+  bool RtcpFbFECIsSet() const { return mFECFbSet; }
+
+  bool RtcpFbTransportCCIsSet() const { return mTransportCCFbSet; }
+
+  bool RtxPayloadTypeIsSet() const { return mRTXPayloadType != -1; }
+};
+}  // namespace mozilla
+#endif
diff --git a/dom/media/webrtc/libwebrtcglue/GmpVideoCodec.cpp b/dom/media/webrtc/libwebrtcglue/GmpVideoCodec.cpp
new file mode 100644
index 0000000000..ccadd846e2
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/GmpVideoCodec.cpp
@@ -0,0 +1,22 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "WebrtcGmpVideoCodec.h"
+#include "GmpVideoCodec.h"
+
+namespace mozilla {
+
+WebrtcVideoEncoder* GmpVideoCodec::CreateEncoder(
+    const webrtc::SdpVideoFormat& aFormat, std::string aPCHandle) {
+  return new WebrtcVideoEncoderProxy(
+      new WebrtcGmpVideoEncoder(aFormat, std::move(aPCHandle)));
+}
+
+WebrtcVideoDecoder* GmpVideoCodec::CreateDecoder(std::string aPCHandle,
+                                                 TrackingId aTrackingId) {
+  return new WebrtcVideoDecoderProxy(std::move(aPCHandle),
+                                     std::move(aTrackingId));
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/libwebrtcglue/GmpVideoCodec.h b/dom/media/webrtc/libwebrtcglue/GmpVideoCodec.h
new file mode 100644
index 0000000000..caf125c809
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/GmpVideoCodec.h
@@ -0,0 +1,27 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef GMPVIDEOCODEC_H_
+#define GMPVIDEOCODEC_H_
+
+#include <string>
+
+#include "PerformanceRecorder.h"
+
+namespace mozilla {
+
+class WebrtcVideoDecoder;
+class WebrtcVideoEncoder;
+
+class GmpVideoCodec {
+ public:
+  static WebrtcVideoEncoder* CreateEncoder(
+      const webrtc::SdpVideoFormat& aFormat, std::string aPCHandle);
+  static WebrtcVideoDecoder* CreateDecoder(std::string aPCHandle,
+                                           TrackingId aTrackingId);
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/libwebrtcglue/MediaConduitControl.h b/dom/media/webrtc/libwebrtcglue/MediaConduitControl.h
new file mode 100644
index 0000000000..a860fab146
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/MediaConduitControl.h
@@ -0,0 +1,77 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef DOM_MEDIA_WEBRTC_LIBWEBRTCGLUE_MEDIACONDUITCONTROL_H_
+#define DOM_MEDIA_WEBRTC_LIBWEBRTCGLUE_MEDIACONDUITCONTROL_H_
+
+#include "jsapi/RTCDTMFSender.h"  // For DtmfEvent
+#include "mozilla/StateMirroring.h"
+#include "RtpRtcpConfig.h"
+#include <vector>
+#include <string>
+#include "mozilla/Maybe.h"
+#include "CodecConfig.h"                   // For Audio/VideoCodecConfig
+#include "api/rtp_parameters.h"            // For webrtc::RtpExtension
+#include "api/video_codecs/video_codec.h"  // For webrtc::VideoCodecMode
+
+namespace mozilla {
+
+using RtpExtList = std::vector<webrtc::RtpExtension>;
+using Ssrc = uint32_t;
+using Ssrcs = std::vector<uint32_t>;
+
+/**
+ * These are the interfaces used to control the async conduits. Some parameters
+ * are common, and some are tied to the conduit type. See
+ * MediaSessionConduit::InitConduitControl for how they are used.
+ *
+ * Put simply, the implementer of the interfaces below may set its canonicals on
+ * any thread, and the conduits will react to those changes accordingly, on
+ * their dedicated worker thread. One instance of these interfaces could control
+ * multiple conduits as each canonical can connect to any number of mirrors.
+ */
+
+class MediaConduitControlInterface {
+ public:
+  virtual AbstractCanonical<bool>* CanonicalReceiving() = 0;
+  virtual AbstractCanonical<bool>* CanonicalTransmitting() = 0;
+  virtual AbstractCanonical<Ssrcs>* CanonicalLocalSsrcs() = 0;
+  virtual AbstractCanonical<std::string>* CanonicalLocalCname() = 0;
+  virtual AbstractCanonical<std::string>* CanonicalMid() = 0;
+  virtual AbstractCanonical<Ssrc>* CanonicalRemoteSsrc() = 0;
+  virtual AbstractCanonical<std::string>* CanonicalSyncGroup() = 0;
+  virtual AbstractCanonical<RtpExtList>* CanonicalLocalRecvRtpExtensions() = 0;
+  virtual AbstractCanonical<RtpExtList>* CanonicalLocalSendRtpExtensions() = 0;
+};
+
+class AudioConduitControlInterface : public MediaConduitControlInterface {
+ public:
+  virtual AbstractCanonical<Maybe<AudioCodecConfig>>*
+  CanonicalAudioSendCodec() = 0;
+  virtual AbstractCanonical<std::vector<AudioCodecConfig>>*
+  CanonicalAudioRecvCodecs() = 0;
+  virtual MediaEventSource<DtmfEvent>& OnDtmfEvent() = 0;
+};
+
+class VideoConduitControlInterface : public MediaConduitControlInterface {
+ public:
+  virtual AbstractCanonical<Ssrcs>* CanonicalLocalVideoRtxSsrcs() = 0;
+  virtual AbstractCanonical<Ssrc>* CanonicalRemoteVideoRtxSsrc() = 0;
+  virtual AbstractCanonical<Maybe<VideoCodecConfig>>*
+  CanonicalVideoSendCodec() = 0;
+  virtual AbstractCanonical<Maybe<RtpRtcpConfig>>*
+  CanonicalVideoSendRtpRtcpConfig() = 0;
+  virtual AbstractCanonical<std::vector<VideoCodecConfig>>*
+  CanonicalVideoRecvCodecs() = 0;
+  virtual AbstractCanonical<Maybe<RtpRtcpConfig>>*
+  CanonicalVideoRecvRtpRtcpConfig() = 0;
+  virtual AbstractCanonical<webrtc::VideoCodecMode>*
+  CanonicalVideoCodecMode() = 0;
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/libwebrtcglue/MediaConduitErrors.h b/dom/media/webrtc/libwebrtcglue/MediaConduitErrors.h
new file mode 100644
index 0000000000..34487d77a0
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/MediaConduitErrors.h
@@ -0,0 +1,46 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef MEDIA_SESSION_ERRORS_H_
+#define MEDIA_SESSION_ERRORS_H_
+
+namespace mozilla {
+enum MediaConduitErrorCode {
+  kMediaConduitNoError = 0,  // 0 for Success,greater than 0 imples error
+  kMediaConduitSessionNotInited =
+      10100,                         // Session not initialized.10100 serves as
+                                     // base for the conduit errors
+  kMediaConduitMalformedArgument,    // Malformed input to Conduit API
+  kMediaConduitCaptureError,         // WebRTC capture APIs failed
+  kMediaConduitInvalidSendCodec,     // Wrong Send codec
+  kMediaConduitInvalidReceiveCodec,  // Wrong Recv Codec
+  kMediaConduitCodecInUse,           // Already applied Codec
+  kMediaConduitInvalidRenderer,      // Null or Wrong Renderer object
+  kMediaConduitRendererFail,         // Add Render called multiple times
+  kMediaConduitSendingAlready,       // Engine already trasmitting
+  kMediaConduitReceivingAlready,     // Engine already receiving
+  kMediaConduitTransportRegistrationFail,  // Null or wrong transport interface
+  kMediaConduitInvalidTransport,           // Null or wrong transport interface
+  kMediaConduitChannelError,               // Configuration Error
+  kMediaConduitSocketError,             // Media Engine transport socket error
+  kMediaConduitRTPRTCPModuleError,      // Couldn't start RTP/RTCP processing
+  kMediaConduitRTPProcessingFailed,     // Processing incoming RTP frame failed
+  kMediaConduitUnknownError,            // More information can be found in logs
+  kMediaConduitExternalRecordingError,  // Couldn't start external recording
+  kMediaConduitRecordingError,          // Runtime recording error
+  kMediaConduitExternalPlayoutError,    // Couldn't start external playout
+  kMediaConduitPlayoutError,            // Runtime playout error
+  kMediaConduitMTUError,                // Can't set MTU
+  kMediaConduitRTCPStatusError,         // Can't set RTCP mode
+  kMediaConduitKeyFrameRequestError,    // Can't set KeyFrameRequest mode
+  kMediaConduitNACKStatusError,         // Can't set NACK mode
+  kMediaConduitTMMBRStatusError,        // Can't set TMMBR mode
+  kMediaConduitFECStatusError,          // Can't set FEC mode
+  kMediaConduitHybridNACKFECStatusError,  // Can't set Hybrid NACK / FEC mode
+  kMediaConduitVideoSendStreamError       // WebRTC video send stream failure
+};
+
+}
+
+#endif
diff --git a/dom/media/webrtc/libwebrtcglue/MediaConduitInterface.cpp b/dom/media/webrtc/libwebrtcglue/MediaConduitInterface.cpp
new file mode 100644
index 0000000000..28079b2478
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/MediaConduitInterface.cpp
@@ -0,0 +1,151 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "MediaConduitInterface.h"
+
+#include "nsTArray.h"
+#include "mozilla/Assertions.h"
+#include "MainThreadUtils.h"
+#include "SystemTime.h"
+
+#include "system_wrappers/include/clock.h"
+
+namespace mozilla {
+
+void MediaSessionConduit::GetRtpSources(
+    nsTArray<dom::RTCRtpSourceEntry>& outSources) const {
+  MOZ_ASSERT(NS_IsMainThread());
+  if (mSourcesUpdateNeeded) {
+    UpdateRtpSources(GetUpstreamRtpSources());
+    OnSourcesUpdated();
+  }
+  outSources.Clear();
+  for (auto& [key, entry] : mSourcesCache) {
+    (void)key;
+    outSources.AppendElement(entry);
+  }
+
+  struct TimestampComparator {
+    bool LessThan(const dom::RTCRtpSourceEntry& aLhs,
+                  const dom::RTCRtpSourceEntry& aRhs) const {
+      // Sort descending!
+      return aLhs.mTimestamp > aRhs.mTimestamp;
+    }
+
+    bool Equals(const dom::RTCRtpSourceEntry& aLhs,
+                const dom::RTCRtpSourceEntry& aRhs) const {
+      return aLhs.mTimestamp == aRhs.mTimestamp;
+    }
+  };
+
+  // *sigh* We have to re-sort this by JS timestamp; we can run into cases
+  // where the libwebrtc timestamps are not in exactly the same order as JS
+  // timestamps due to clock differences (wibbly-wobbly, timey-wimey stuff)
+  outSources.Sort(TimestampComparator());
+}
+
+static double rtpToDomAudioLevel(uint8_t aAudioLevel) {
+  if (aAudioLevel == 127) {
+    // Spec indicates that a value of 127 should be set to 0
+    return 0;
+  }
+
+  // All other values are calculated as 10^(-rfc_level/20)
+  return std::pow(10, -aAudioLevel / 20.0);
+}
+
+void MediaSessionConduit::UpdateRtpSources(
+    const std::vector<webrtc::RtpSource>& aSources) const {
+  MOZ_ASSERT(NS_IsMainThread());
+  // Empty out the cache; we'll copy things back as needed
+  auto cache = std::move(mSourcesCache);
+
+  for (const auto& source : aSources) {
+    SourceKey key(source);
+    auto it = cache.find(key);
+    if (it != cache.end()) {
+      // This source entry was already in the cache, and should continue to be
+      // present in exactly the same form as before. This means we do _not_
+      // want to perform the timestamp adjustment again, since it might yield a
+      // slightly different result. This is why we copy this entry from the old
+      // cache instead of simply rebuilding it, and is also why we key the
+      // cache based on timestamp (keying the cache based on timestamp also
+      // gets us the ordering we want, conveniently).
+      mSourcesCache[key] = it->second;
+      continue;
+    }
+
+    // This is something we did not already have in the cache.
+    dom::RTCRtpSourceEntry domEntry;
+    domEntry.mSource = source.source_id();
+    switch (source.source_type()) {
+      case webrtc::RtpSourceType::SSRC:
+        domEntry.mSourceType = dom::RTCRtpSourceEntryType::Synchronization;
+        break;
+      case webrtc::RtpSourceType::CSRC:
+        domEntry.mSourceType = dom::RTCRtpSourceEntryType::Contributing;
+        break;
+      default:
+        MOZ_CRASH("Unexpected RTCRtpSourceEntryType");
+    }
+
+    if (source.audio_level()) {
+      domEntry.mAudioLevel.Construct(rtpToDomAudioLevel(*source.audio_level()));
+    }
+
+    // These timestamps are always **rounded** to milliseconds. That means they
+    // can jump up to half a millisecond into the future. We compensate for that
+    // here so that things seem consistent to js.
+    domEntry.mTimestamp = dom::RTCStatsTimestamp::FromRealtime(
+                              GetTimestampMaker(),
+                              webrtc::Timestamp::Millis(source.timestamp_ms()) -
+                                  webrtc::TimeDelta::Micros(500))
+                              .ToDom();
+    domEntry.mRtpTimestamp = source.rtp_timestamp();
+    mSourcesCache[key] = domEntry;
+  }
+}
+
+void MediaSessionConduit::OnSourcesUpdated() const {
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_ASSERT(mSourcesUpdateNeeded);
+  mSourcesUpdateNeeded = false;
+  // Reset the updateNeeded flag and clear the cache in a direct task, i.e.,
+  // as soon as the current task has finished.
+  AbstractThread::GetCurrent()->TailDispatcher().AddDirectTask(
+      NS_NewRunnableFunction(
+          __func__, [this, self = RefPtr<const MediaSessionConduit>(this)] {
+            mSourcesUpdateNeeded = true;
+            mSourcesCache.clear();
+          }));
+}
+
+void MediaSessionConduit::InsertAudioLevelForContributingSource(
+    const uint32_t aCsrcSource, const int64_t aTimestamp,
+    const uint32_t aRtpTimestamp, const bool aHasAudioLevel,
+    const uint8_t aAudioLevel) {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  if (mSourcesUpdateNeeded) {
+    OnSourcesUpdated();
+  }
+
+  dom::RTCRtpSourceEntry domEntry;
+  domEntry.mSource = aCsrcSource;
+  domEntry.mSourceType = dom::RTCRtpSourceEntryType::Contributing;
+  domEntry.mTimestamp = aTimestamp;
+  domEntry.mRtpTimestamp = aRtpTimestamp;
+  if (aHasAudioLevel) {
+    domEntry.mAudioLevel.Construct(rtpToDomAudioLevel(aAudioLevel));
+  }
+
+  auto now = GetTimestampMaker().GetNow();
+  webrtc::Timestamp convertedTimestamp =
+      now.ToRealtime() - webrtc::TimeDelta::Millis(now.ToDom() - aTimestamp);
+
+  SourceKey key(convertedTimestamp.ms<uint32_t>(), aCsrcSource);
+  mSourcesCache[key] = domEntry;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/libwebrtcglue/MediaConduitInterface.h b/dom/media/webrtc/libwebrtcglue/MediaConduitInterface.h
new file mode 100644
index 0000000000..86e148dc5c
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/MediaConduitInterface.h
@@ -0,0 +1,495 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef MEDIA_CONDUIT_ABSTRACTION_
+#define MEDIA_CONDUIT_ABSTRACTION_
+
+#include <vector>
+#include <functional>
+#include <map>
+
+#include "CodecConfig.h"
+#include "ImageContainer.h"
+#include "jsapi/RTCStatsReport.h"
+#include "MediaConduitErrors.h"
+#include "mozilla/media/MediaUtils.h"
+#include "mozilla/MozPromise.h"
+#include "WebrtcVideoCodecFactory.h"
+#include "nsTArray.h"
+#include "mozilla/dom/RTCRtpSourcesBinding.h"
+#include "PerformanceRecorder.h"
+#include "transport/mediapacket.h"
+#include "MediaConduitControl.h"
+
+// libwebrtc includes
+#include "api/audio/audio_frame.h"
+#include "api/call/transport.h"
+#include "api/rtp_headers.h"
+#include "api/rtp_parameters.h"
+#include "api/transport/rtp/rtp_source.h"
+#include "api/video/video_frame_buffer.h"
+#include "call/audio_receive_stream.h"
+#include "call/audio_send_stream.h"
+#include "call/call_basic_stats.h"
+#include "call/video_receive_stream.h"
+#include "call/video_send_stream.h"
+#include "rtc_base/copy_on_write_buffer.h"
+
+namespace webrtc {
+class RtpPacketReceived;
+class VideoFrame;
+}  // namespace webrtc
+
+namespace mozilla {
+namespace dom {
+struct RTCRtpSourceEntry;
+}
+
+namespace dom {
+struct RTCRtpSourceEntry;
+}
+
+enum class MediaSessionConduitLocalDirection : int { kSend, kRecv };
+
+class VideoSessionConduit;
+class AudioSessionConduit;
+class WebrtcCallWrapper;
+
+/**
+ * 1. Abstract renderer for video data
+ * 2. This class acts as abstract interface between the video-engine and
+ *    video-engine agnostic renderer implementation.
+ * 3. Concrete implementation of this interface is responsible for
+ *    processing and/or rendering the obtained raw video frame to appropriate
+ *    output , say, <video>
+ */
+class VideoRenderer {
+ protected:
+  virtual ~VideoRenderer() {}
+
+ public:
+  /**
+   * Callback Function reportng any change in the video-frame dimensions
+   * @param width:  current width of the video @ decoder
+   * @param height: current height of the video @ decoder
+   */
+  virtual void FrameSizeChange(unsigned int width, unsigned int height) = 0;
+
+  /**
+   * Callback Function reporting decoded frame for processing.
+   * @param buffer: reference to decoded video frame
+   * @param buffer_size: size of the decoded frame
+   * @param time_stamp: Decoder timestamp, typically 90KHz as per RTP
+   * @render_time: Wall-clock time at the decoder for synchronization
+   *                purposes in milliseconds
+   * NOTE: If decoded video frame is passed through buffer , it is the
+   * responsibility of the concrete implementations of this class to own copy
+   * of the frame if needed for time longer than scope of this callback.
+   * Such implementations should be quick in processing the frames and return
+   * immediately.
+   */
+  virtual void RenderVideoFrame(const webrtc::VideoFrameBuffer& buffer,
+                                uint32_t time_stamp, int64_t render_time) = 0;
+
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(VideoRenderer)
+};
+
+/**
+ * Generic Interface for representing Audio/Video Session
+ * MediaSession conduit is identified by 2 main components
+ * 1. Attached Transport Interface (through events) for inbound and outbound RTP
+ *    transport
+ * 2. Attached Renderer Interface for rendering media data off the network
+ * This class hides specifics of Media-Engine implementation from the consumers
+ * of this interface.
+ * Also provides codec configuration API for the media sent and recevied
+ */
+class MediaSessionConduit {
+ protected:
+  virtual ~MediaSessionConduit() {}
+
+ public:
+  enum Type { AUDIO, VIDEO };
+  enum class PacketType { RTP, RTCP };
+
+  static std::string LocalDirectionToString(
+      const MediaSessionConduitLocalDirection aDirection) {
+    return aDirection == MediaSessionConduitLocalDirection::kSend ? "send"
+                                                                  : "receive";
+  }
+
+  virtual Type type() const = 0;
+
+  // Call thread only
+  virtual Maybe<int> ActiveSendPayloadType() const = 0;
+  virtual Maybe<int> ActiveRecvPayloadType() const = 0;
+
+  // Whether transport is currently sending and receiving packets
+  virtual void SetTransportActive(bool aActive) = 0;
+
+  // Sending packets
+  virtual MediaEventSourceExc<MediaPacket>& SenderRtpSendEvent() = 0;
+  virtual MediaEventSourceExc<MediaPacket>& SenderRtcpSendEvent() = 0;
+  virtual MediaEventSourceExc<MediaPacket>& ReceiverRtcpSendEvent() = 0;
+
+  // Receiving packets...
+  // from an rtp-receiving pipeline
+  virtual void ConnectReceiverRtpEvent(
+      MediaEventSourceExc<webrtc::RtpPacketReceived, webrtc::RTPHeader>&
+          aEvent) = 0;
+  // from an rtp-receiving pipeline
+  virtual void ConnectReceiverRtcpEvent(
+      MediaEventSourceExc<MediaPacket>& aEvent) = 0;
+  // from an rtp-transmitting pipeline
+  virtual void ConnectSenderRtcpEvent(
+      MediaEventSourceExc<MediaPacket>& aEvent) = 0;
+
+  // Sts thread only.
+  virtual Maybe<uint16_t> RtpSendBaseSeqFor(uint32_t aSsrc) const = 0;
+
+  // Any thread.
+  virtual const dom::RTCStatsTimestampMaker& GetTimestampMaker() const = 0;
+
+  virtual Ssrcs GetLocalSSRCs() const = 0;
+
+  virtual Maybe<Ssrc> GetRemoteSSRC() const = 0;
+  virtual void UnsetRemoteSSRC(Ssrc aSsrc) = 0;
+
+  virtual void DisableSsrcChanges() = 0;
+
+  virtual bool HasCodecPluginID(uint64_t aPluginID) const = 0;
+
+  // Stuff for driving mute/unmute events
+  virtual MediaEventSource<void>& RtcpByeEvent() = 0;
+  virtual MediaEventSource<void>& RtcpTimeoutEvent() = 0;
+  virtual MediaEventSource<void>& RtpPacketEvent() = 0;
+
+  virtual bool SendRtp(const uint8_t* aData, size_t aLength,
+                       const webrtc::PacketOptions& aOptions) = 0;
+  virtual bool SendSenderRtcp(const uint8_t* aData, size_t aLength) = 0;
+  virtual bool SendReceiverRtcp(const uint8_t* aData, size_t aLength) = 0;
+
+  virtual void DeliverPacket(rtc::CopyOnWriteBuffer packet,
+                             PacketType type) = 0;
+
+  virtual RefPtr<GenericPromise> Shutdown() = 0;
+
+  virtual Maybe<RefPtr<AudioSessionConduit>> AsAudioSessionConduit() = 0;
+  virtual Maybe<RefPtr<VideoSessionConduit>> AsVideoSessionConduit() = 0;
+
+  virtual Maybe<webrtc::CallBasicStats> GetCallStats() const = 0;
+
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(MediaSessionConduit)
+
+  void GetRtpSources(nsTArray<dom::RTCRtpSourceEntry>& outSources) const;
+
+  virtual void SetJitterBufferTarget(DOMHighResTimeStamp aTargetMs) = 0;
+
+  // test-only: inserts fake CSRCs and audio level data.
+  // NB: fake data is only valid during the current main thread task.
+  void InsertAudioLevelForContributingSource(const uint32_t aCsrcSource,
+                                             const int64_t aTimestamp,
+                                             const uint32_t aRtpTimestamp,
+                                             const bool aHasAudioLevel,
+                                             const uint8_t aAudioLevel);
+
+ protected:
+  virtual std::vector<webrtc::RtpSource> GetUpstreamRtpSources() const = 0;
+
+ private:
+  void UpdateRtpSources(const std::vector<webrtc::RtpSource>& aSources) const;
+
+  // Marks the cache as having been updated in the current task, and keeps it
+  // stable until the current task is finished.
+  void OnSourcesUpdated() const;
+
+  // Accessed only on main thread. This exists for a couple of reasons:
+  // 1. The webrtc spec says that source stats are updated using a queued task;
+  //    libwebrtc's internal representation of these stats is updated without
+  //    any task queueing, which means we need a mainthread-only cache.
+  // 2. libwebrtc uses its own clock that is not consistent with the one we
+  // need to use for stats (the so-called JS timestamps), which means we need
+  // to adjust the timestamps.  Since timestamp adjustment is inexact and will
+  // not necessarily yield exactly the same result if performed again later, we
+  // need to avoid performing it more than once for each entry, which means we
+  // need to remember both the JS timestamp (in dom::RTCRtpSourceEntry) and the
+  // libwebrtc timestamp (in SourceKey::mLibwebrtcTimestampMs).
+  class SourceKey {
+   public:
+    explicit SourceKey(const webrtc::RtpSource& aSource)
+        : SourceKey(aSource.timestamp_ms(), aSource.source_id()) {}
+
+    SourceKey(uint32_t aTimestamp, uint32_t aSrc)
+        : mLibwebrtcTimestampMs(aTimestamp), mSrc(aSrc) {}
+
+    // TODO: Once we support = default for this in our toolchain, do so
+    auto operator>(const SourceKey& aRhs) const {
+      if (mLibwebrtcTimestampMs == aRhs.mLibwebrtcTimestampMs) {
+        return mSrc > aRhs.mSrc;
+      }
+      return mLibwebrtcTimestampMs > aRhs.mLibwebrtcTimestampMs;
+    }
+
+   private:
+    uint32_t mLibwebrtcTimestampMs;
+    uint32_t mSrc;
+  };
+  mutable std::map<SourceKey, dom::RTCRtpSourceEntry, std::greater<SourceKey>>
+      mSourcesCache;
+  // Accessed only on main thread. A flag saying whether mSourcesCache needs
+  // updating. Ensures that get*Sources() appear stable from javascript
+  // throughout a main thread task, even though we don't follow the spec to the
+  // letter (dispatch a task to update the sources).
+  mutable bool mSourcesUpdateNeeded = true;
+};
+
+class WebrtcSendTransport : public webrtc::Transport {
+  // WeakRef to the owning conduit
+  MediaSessionConduit* mConduit;
+
+ public:
+  explicit WebrtcSendTransport(MediaSessionConduit* aConduit)
+      : mConduit(aConduit) {}
+  bool SendRtp(const uint8_t* aPacket, size_t aLength,
+               const webrtc::PacketOptions& aOptions) override {
+    return mConduit->SendRtp(aPacket, aLength, aOptions);
+  }
+  bool SendRtcp(const uint8_t* aPacket, size_t aLength) override {
+    return mConduit->SendSenderRtcp(aPacket, aLength);
+  }
+};
+
+class WebrtcReceiveTransport : public webrtc::Transport {
+  // WeakRef to the owning conduit
+  MediaSessionConduit* mConduit;
+
+ public:
+  explicit WebrtcReceiveTransport(MediaSessionConduit* aConduit)
+      : mConduit(aConduit) {}
+  bool SendRtp(const uint8_t* aPacket, size_t aLength,
+               const webrtc::PacketOptions& aOptions) override {
+    MOZ_CRASH("Unexpected RTP packet");
+  }
+  bool SendRtcp(const uint8_t* aPacket, size_t aLength) override {
+    return mConduit->SendReceiverRtcp(aPacket, aLength);
+  }
+};
+
+// Abstract base classes for external encoder/decoder.
+
+// Interface to help signal PluginIDs
+class CodecPluginID {
+ public:
+  virtual MediaEventSource<uint64_t>* InitPluginEvent() { return nullptr; }
+  virtual MediaEventSource<uint64_t>* ReleasePluginEvent() { return nullptr; }
+  virtual ~CodecPluginID() {}
+};
+
+class VideoEncoder : public CodecPluginID {
+ public:
+  virtual ~VideoEncoder() {}
+};
+
+class VideoDecoder : public CodecPluginID {
+ public:
+  virtual ~VideoDecoder() {}
+};
+
+/**
+ * MediaSessionConduit for video
+ * Refer to the comments on MediaSessionConduit above for overall
+ * information
+ */
+class VideoSessionConduit : public MediaSessionConduit {
+ public:
+  struct Options {
+    bool mVideoLatencyTestEnable = false;
+    // All in bps.
+    int mMinBitrate = 0;
+    int mStartBitrate = 0;
+    int mPrefMaxBitrate = 0;
+    int mMinBitrateEstimate = 0;
+    bool mDenoising = false;
+    bool mLockScaling = false;
+    uint8_t mSpatialLayers = 1;
+    uint8_t mTemporalLayers = 1;
+  };
+
+  /**
+   * Factory function to create and initialize a Video Conduit Session
+   * @param  webrtc::Call instance shared by paired audio and video
+   *         media conduits
+   * @param  aOptions are a number of options, typically from prefs, used to
+   *         configure the created VideoConduit.
+   * @param  aPCHandle is a string representing the RTCPeerConnection that is
+   *         creating this VideoConduit. This is used when reporting GMP plugin
+   *         crashes.
+   * @result Concrete VideoSessionConduitObject or nullptr in the case
+   *         of failure
+   */
+  static RefPtr<VideoSessionConduit> Create(
+      RefPtr<WebrtcCallWrapper> aCall,
+      nsCOMPtr<nsISerialEventTarget> aStsThread, Options aOptions,
+      std::string aPCHandle, const TrackingId& aRecvTrackingId);
+
+  enum FrameRequestType {
+    FrameRequestNone,
+    FrameRequestFir,
+    FrameRequestPli,
+    FrameRequestUnknown
+  };
+
+  VideoSessionConduit()
+      : mFrameRequestMethod(FrameRequestNone),
+        mUsingNackBasic(false),
+        mUsingTmmbr(false),
+        mUsingFEC(false) {}
+
+  virtual ~VideoSessionConduit() {}
+
+  Type type() const override { return VIDEO; }
+
+  Maybe<RefPtr<AudioSessionConduit>> AsAudioSessionConduit() override {
+    return Nothing();
+  }
+
+  Maybe<RefPtr<VideoSessionConduit>> AsVideoSessionConduit() override {
+    return Some(RefPtr<VideoSessionConduit>(this));
+  }
+
+  /**
+   * Hooks up mControl Mirrors with aControl Canonicals, and sets up
+   * mWatchManager to react on Mirror changes.
+   */
+  virtual void InitControl(VideoConduitControlInterface* aControl) = 0;
+
+  /**
+   * Function to attach Renderer end-point of the Media-Video conduit.
+   * @param aRenderer : Reference to the concrete Video renderer implementation
+   * Note: Multiple invocations of this API shall remove an existing renderer
+   * and attaches the new to the Conduit.
+   */
+  virtual MediaConduitErrorCode AttachRenderer(
+      RefPtr<mozilla::VideoRenderer> aRenderer) = 0;
+  virtual void DetachRenderer() = 0;
+
+  /**
+   * Function to deliver a capture video frame for encoding and transport.
+   * If the frame's timestamp is 0, it will be automatcally generated.
+   *
+   * NOTE: ConfigureSendMediaCodec() must be called before this function can
+   *       be invoked. This ensures the inserted video-frames can be
+   *       transmitted by the conduit.
+   */
+  virtual MediaConduitErrorCode SendVideoFrame(webrtc::VideoFrame aFrame) = 0;
+
+  /**
+   * These methods allow unit tests to double-check that the
+   * rtcp-fb settings are as expected.
+   */
+  FrameRequestType FrameRequestMethod() const { return mFrameRequestMethod; }
+
+  bool UsingNackBasic() const { return mUsingNackBasic; }
+
+  bool UsingTmmbr() const { return mUsingTmmbr; }
+
+  bool UsingFEC() const { return mUsingFEC; }
+
+  virtual Maybe<webrtc::VideoReceiveStreamInterface::Stats> GetReceiverStats()
+      const = 0;
+  virtual Maybe<webrtc::VideoSendStream::Stats> GetSenderStats() const = 0;
+
+  virtual void CollectTelemetryData() = 0;
+
+  virtual bool AddFrameHistory(
+      dom::Sequence<dom::RTCVideoFrameHistoryInternal>* outHistories) const = 0;
+
+  virtual Maybe<Ssrc> GetAssociatedLocalRtxSSRC(Ssrc aSsrc) const = 0;
+
+ protected:
+  /* RTCP feedback settings, for unit testing purposes */
+  FrameRequestType mFrameRequestMethod;
+  bool mUsingNackBasic;
+  bool mUsingTmmbr;
+  bool mUsingFEC;
+};
+
+/**
+ * MediaSessionConduit for audio
+ * Refer to the comments on MediaSessionConduit above for overall
+ * information
+ */
+class AudioSessionConduit : public MediaSessionConduit {
+ public:
+  /**
+   * Factory function to create and initialize an Audio Conduit Session
+   * @param  webrtc::Call instance shared by paired audio and video
+   *         media conduits
+   * @result Concrete AudioSessionConduitObject or nullptr in the case
+   *         of failure
+   */
+  static RefPtr<AudioSessionConduit> Create(
+      RefPtr<WebrtcCallWrapper> aCall,
+      nsCOMPtr<nsISerialEventTarget> aStsThread);
+
+  virtual ~AudioSessionConduit() {}
+
+  Type type() const override { return AUDIO; }
+
+  Maybe<RefPtr<AudioSessionConduit>> AsAudioSessionConduit() override {
+    return Some(this);
+  }
+
+  Maybe<RefPtr<VideoSessionConduit>> AsVideoSessionConduit() override {
+    return Nothing();
+  }
+
+  /**
+   * Hooks up mControl Mirrors with aControl Canonicals, and sets up
+   * mWatchManager to react on Mirror changes.
+   */
+  virtual void InitControl(AudioConduitControlInterface* aControl) = 0;
+
+  /**
+   * Function to deliver externally captured audio sample for encoding and
+   * transport
+   * @param frame [in]: AudioFrame in upstream's format for forwarding to the
+   *                    send stream. Ownership is passed along.
+   * NOTE: ConfigureSendMediaCodec() SHOULD be called before this function can
+   * be invoked. This ensures the inserted audio-samples can be transmitted by
+   * the conduit.
+   */
+  virtual MediaConduitErrorCode SendAudioFrame(
+      std::unique_ptr<webrtc::AudioFrame> frame) = 0;
+
+  /**
+   * Function to grab a decoded audio-sample from the media engine for
+   * rendering / playout of length 10 milliseconds.
+   *
+   * @param samplingFreqHz [in]: Frequency of the sampling for playback in
+   *                             Hertz (16000, 32000,..)
+   * @param frame [in/out]: Pointer to an AudioFrame to which audio data will be
+   *                        copied
+   * NOTE: This function should be invoked every 10 milliseconds for the best
+   *       performance
+   * NOTE: ConfigureRecvMediaCodec() SHOULD be called before this function can
+   *       be invoked
+   * This ensures the decoded samples are ready for reading and playout is
+   * enabled.
+   */
+  virtual MediaConduitErrorCode GetAudioFrame(int32_t samplingFreqHz,
+                                              webrtc::AudioFrame* frame) = 0;
+
+  /**
+   * Checks if given sampling frequency is supported
+   * @param freq: Sampling rate (in Hz) to check
+   */
+  virtual bool IsSamplingFreqSupported(int freq) const = 0;
+
+  virtual Maybe<webrtc::AudioReceiveStreamInterface::Stats> GetReceiverStats()
+      const = 0;
+  virtual Maybe<webrtc::AudioSendStream::Stats> GetSenderStats() const = 0;
+};
+}  // namespace mozilla
+#endif
diff --git a/dom/media/webrtc/libwebrtcglue/MediaDataCodec.cpp b/dom/media/webrtc/libwebrtcglue/MediaDataCodec.cpp
new file mode 100644
index 0000000000..895fcdc432
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/MediaDataCodec.cpp
@@ -0,0 +1,70 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "MediaDataCodec.h"
+
+#include "PDMFactory.h"
+#include "WebrtcGmpVideoCodec.h"
+#include "WebrtcMediaDataDecoderCodec.h"
+#include "WebrtcMediaDataEncoderCodec.h"
+#include "mozilla/StaticPrefs_media.h"
+
+namespace mozilla {
+
+/* static */
+WebrtcVideoEncoder* MediaDataCodec::CreateEncoder(
+    const webrtc::SdpVideoFormat& aFormat) {
+  if (!StaticPrefs::media_webrtc_platformencoder()) {
+    return nullptr;
+  }
+  if (!WebrtcMediaDataEncoder::CanCreate(
+          webrtc::PayloadStringToCodecType(aFormat.name))) {
+    return nullptr;
+  }
+
+  return new WebrtcVideoEncoderProxy(new WebrtcMediaDataEncoder(aFormat));
+}
+
+/* static */
+WebrtcVideoDecoder* MediaDataCodec::CreateDecoder(
+    webrtc::VideoCodecType aCodecType, TrackingId aTrackingId) {
+  switch (aCodecType) {
+    case webrtc::VideoCodecType::kVideoCodecVP8:
+    case webrtc::VideoCodecType::kVideoCodecVP9:
+      if (!StaticPrefs::media_navigator_mediadatadecoder_vpx_enabled()) {
+        return nullptr;
+      }
+      break;
+    case webrtc::VideoCodecType::kVideoCodecH264:
+      if (!StaticPrefs::media_navigator_mediadatadecoder_h264_enabled()) {
+        return nullptr;
+      }
+      break;
+    default:
+      return nullptr;
+  }
+
+  nsAutoCString codec;
+  switch (aCodecType) {
+    case webrtc::VideoCodecType::kVideoCodecVP8:
+      codec = "video/vp8";
+      break;
+    case webrtc::VideoCodecType::kVideoCodecVP9:
+      codec = "video/vp9";
+      break;
+    case webrtc::VideoCodecType::kVideoCodecH264:
+      codec = "video/avc";
+      break;
+    default:
+      return nullptr;
+  }
+  RefPtr<PDMFactory> pdm = new PDMFactory();
+  if (pdm->SupportsMimeType(codec) == media::DecodeSupport::Unsupported) {
+    return nullptr;
+  }
+
+  return new WebrtcMediaDataDecoder(codec, aTrackingId);
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/libwebrtcglue/MediaDataCodec.h b/dom/media/webrtc/libwebrtcglue/MediaDataCodec.h
new file mode 100644
index 0000000000..b885d6ae0c
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/MediaDataCodec.h
@@ -0,0 +1,32 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef MEDIA_DATA_CODEC_H_
+#define MEDIA_DATA_CODEC_H_
+
+#include "MediaConduitInterface.h"
+
+namespace mozilla {
+
+class WebrtcVideoDecoder;
+class WebrtcVideoEncoder;
+class MediaDataCodec {
+ public:
+  /**
+   * Create encoder object for codec format |aFormat|. Return |nullptr| when
+   * failed.
+   */
+  static WebrtcVideoEncoder* CreateEncoder(
+      const webrtc::SdpVideoFormat& aFormat);
+
+  /**
+   * Create decoder object for codec type |aCodecType|. Return |nullptr| when
+   * failed.
+   */
+  static WebrtcVideoDecoder* CreateDecoder(webrtc::VideoCodecType aCodecType,
+                                           TrackingId aTrackingId);
+};
+}  // namespace mozilla
+
+#endif  // MEDIA_DATA_CODEC_H_
diff --git a/dom/media/webrtc/libwebrtcglue/RtpRtcpConfig.h b/dom/media/webrtc/libwebrtcglue/RtpRtcpConfig.h
new file mode 100644
index 0000000000..03a774ec3b
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/RtpRtcpConfig.h
@@ -0,0 +1,24 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef __RTPRTCP_CONFIG_H__
+#define __RTPRTCP_CONFIG_H__
+#include "api/rtp_headers.h"
+
+namespace mozilla {
+class RtpRtcpConfig {
+ public:
+  RtpRtcpConfig() = delete;
+  explicit RtpRtcpConfig(const webrtc::RtcpMode aMode) : mRtcpMode(aMode) {}
+  webrtc::RtcpMode GetRtcpMode() const { return mRtcpMode; }
+
+  bool operator==(const RtpRtcpConfig& aOther) const {
+    return mRtcpMode == aOther.mRtcpMode;
+  }
+
+ private:
+  webrtc::RtcpMode mRtcpMode;
+};
+}  // namespace mozilla
+#endif
diff --git a/dom/media/webrtc/libwebrtcglue/RunningStat.h b/dom/media/webrtc/libwebrtcglue/RunningStat.h
new file mode 100644
index 0000000000..7a0e88f193
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/RunningStat.h
@@ -0,0 +1,48 @@
+/* Any copyright is dedicated to the Public Domain.
+   http://creativecommons.org/publicdomain/zero/1.0/ */
+/* Adapted from "Accurately computing running variance - John D. Cook"
+   http://www.johndcook.com/standard_deviation.html */
+
+#ifndef RUNNING_STAT_H_
+#define RUNNING_STAT_H_
+#include <math.h>
+
+namespace mozilla {
+
+class RunningStat {
+ public:
+  RunningStat() : mN(0), mOldM(0.0), mNewM(0.0), mOldS(0.0), mNewS(0.0) {}
+
+  void Clear() { mN = 0; }
+
+  void Push(double x) {
+    mN++;
+
+    // See Knuth TAOCP vol 2, 3rd edition, page 232
+    if (mN == 1) {
+      mOldM = mNewM = x;
+      mOldS = 0.0;
+    } else {
+      mNewM = mOldM + (x - mOldM) / mN;
+      mNewS = mOldS + (x - mOldM) * (x - mNewM);
+
+      // set up for next iteration
+      mOldM = mNewM;
+      mOldS = mNewS;
+    }
+  }
+
+  int NumDataValues() const { return mN; }
+
+  double Mean() const { return (mN > 0) ? mNewM : 0.0; }
+
+  double Variance() const { return (mN > 1) ? mNewS / (mN - 1) : 0.0; }
+
+  double StandardDeviation() const { return sqrt(Variance()); }
+
+ private:
+  int mN;
+  double mOldM, mNewM, mOldS, mNewS;
+};
+}  // namespace mozilla
+#endif  // RUNNING_STAT_H_
diff --git a/dom/media/webrtc/libwebrtcglue/SystemTime.cpp b/dom/media/webrtc/libwebrtcglue/SystemTime.cpp
new file mode 100644
index 0000000000..bba7fd788e
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/SystemTime.cpp
@@ -0,0 +1,60 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "SystemTime.h"
+
+#include "TimeUnits.h"
+
+namespace mozilla {
+
+// webrtc::Timestamp may not be negative. `now-base` for the first call to
+// WebrtcSystemTime() is always 0, which makes it impossible for libwebrtc
+// code to calculate a timestamp older than the first one returned. This
+// offset makes sure the clock starts at a value equivalent to roughly 4.5h.
+static constexpr webrtc::TimeDelta kWebrtcTimeOffset =
+    webrtc::TimeDelta::Micros(0x10000000);
+
+RTCStatsTimestampMakerRealtimeClock::RTCStatsTimestampMakerRealtimeClock(
+    const dom::RTCStatsTimestampMaker& aTimestampMaker)
+    : mTimestampMaker(aTimestampMaker) {}
+
+webrtc::Timestamp RTCStatsTimestampMakerRealtimeClock::CurrentTime() {
+  return mTimestampMaker.GetNow().ToRealtime();
+}
+
+webrtc::NtpTime RTCStatsTimestampMakerRealtimeClock::ConvertTimestampToNtpTime(
+    webrtc::Timestamp aRealtime) {
+  return CreateNtp(
+      dom::RTCStatsTimestamp::FromRealtime(mTimestampMaker, aRealtime).ToNtp());
+}
+
+TimeStamp WebrtcSystemTimeBase() {
+  static TimeStamp now = TimeStamp::Now();
+  return now;
+}
+
+webrtc::Timestamp WebrtcSystemTime() {
+  const TimeStamp base = WebrtcSystemTimeBase();
+  const TimeStamp now = TimeStamp::Now();
+  return webrtc::Timestamp::Micros((now - base).ToMicroseconds()) +
+         kWebrtcTimeOffset;
+}
+
+webrtc::NtpTime CreateNtp(webrtc::Timestamp aTime) {
+  const int64_t timeNtpUs = aTime.us();
+  const uint32_t seconds = static_cast<uint32_t>(timeNtpUs / USECS_PER_S);
+
+  constexpr int64_t fractionsPerSec = 1LL << 32;
+  const int64_t fractionsUs = timeNtpUs % USECS_PER_S;
+  const uint32_t fractions = (fractionsUs * fractionsPerSec) / USECS_PER_S;
+
+  return webrtc::NtpTime(seconds, fractions);
+}
+}  // namespace mozilla
+
+namespace rtc {
+int64_t SystemTimeNanos() { return mozilla::WebrtcSystemTime().us() * 1000; }
+}  // namespace rtc
diff --git a/dom/media/webrtc/libwebrtcglue/SystemTime.h b/dom/media/webrtc/libwebrtcglue/SystemTime.h
new file mode 100644
index 0000000000..5c042e689f
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/SystemTime.h
@@ -0,0 +1,44 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef DOM_MEDIA_WEBRTC_LIBWEBRTCGLUE_SYSTEMTIMENANOS_H_
+#define DOM_MEDIA_WEBRTC_LIBWEBRTCGLUE_SYSTEMTIMENANOS_H_
+
+#include "jsapi/RTCStatsReport.h"
+#include "mozilla/TimeStamp.h"
+#include "system_wrappers/include/clock.h"
+
+namespace mozilla {
+class RTCStatsTimestampMakerRealtimeClock : public webrtc::Clock {
+ public:
+  explicit RTCStatsTimestampMakerRealtimeClock(
+      const dom::RTCStatsTimestampMaker& aTimestampMaker);
+
+  webrtc::Timestamp CurrentTime() override;
+
+  // Upstream, this method depend on rtc::TimeUTCMicros for converting the
+  // monotonic system clock to Ntp, if only for the first call when deciding the
+  // Ntp offset.
+  // We override this to be able to use our own clock instead of
+  // rtc::TimeUTCMicros for ntp timestamps.
+  webrtc::NtpTime ConvertTimestampToNtpTime(
+      webrtc::Timestamp aRealtime) override;
+
+  const dom::RTCStatsTimestampMaker mTimestampMaker;
+};
+
+// The time base used for WebrtcSystemTime(). Completely arbitrary. Constant.
+TimeStamp WebrtcSystemTimeBase();
+
+// The returned timestamp denotes the monotonic time passed since
+// WebrtcSystemTimeBase(). Libwebrtc uses this to track how time advances from a
+// specific point in time. It adds an offset to make the timestamps absolute.
+webrtc::Timestamp WebrtcSystemTime();
+
+webrtc::NtpTime CreateNtp(webrtc::Timestamp aTime);
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/libwebrtcglue/TaskQueueWrapper.h b/dom/media/webrtc/libwebrtcglue/TaskQueueWrapper.h
new file mode 100644
index 0000000000..da11bfdf8b
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/TaskQueueWrapper.h
@@ -0,0 +1,182 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef DOM_MEDIA_WEBRTC_LIBWEBRTCGLUE_TASKQUEUEWRAPPER_H_
+#define DOM_MEDIA_WEBRTC_LIBWEBRTCGLUE_TASKQUEUEWRAPPER_H_
+
+#include "api/task_queue/task_queue_factory.h"
+#include "mozilla/DataMutex.h"
+#include "mozilla/RecursiveMutex.h"
+#include "mozilla/ProfilerRunnable.h"
+#include "mozilla/TaskQueue.h"
+#include "VideoUtils.h"
+#include "mozilla/media/MediaUtils.h"  // For media::Await
+
+namespace mozilla {
+
+enum class DeletionPolicy : uint8_t { Blocking, NonBlocking };
+
+/**
+ * A wrapper around Mozilla TaskQueues in the shape of a libwebrtc TaskQueue.
+ *
+ * Allows libwebrtc to use Mozilla threads where tooling, e.g. profiling, is set
+ * up and just works.
+ *
+ * Mozilla APIs like Runnables, MozPromise, etc. can also be used with the
+ * wrapped TaskQueue to run things on the right thread when interacting with
+ * libwebrtc.
+ */
+template <DeletionPolicy Deletion>
+class TaskQueueWrapper : public webrtc::TaskQueueBase {
+ public:
+  TaskQueueWrapper(RefPtr<TaskQueue> aTaskQueue, nsCString aName)
+      : mTaskQueue(std::move(aTaskQueue)), mName(std::move(aName)) {}
+  ~TaskQueueWrapper() = default;
+
+  void Delete() override {
+    {
+      // Scope this to make sure it does not race against the promise chain we
+      // set up below.
+      auto hasShutdown = mHasShutdown.Lock();
+      *hasShutdown = true;
+    }
+
+    MOZ_RELEASE_ASSERT(Deletion == DeletionPolicy::NonBlocking ||
+                       !mTaskQueue->IsOnCurrentThread());
+
+    nsCOMPtr<nsISerialEventTarget> backgroundTaskQueue;
+    NS_CreateBackgroundTaskQueue(__func__, getter_AddRefs(backgroundTaskQueue));
+    if (NS_WARN_IF(!backgroundTaskQueue)) {
+      // Ok... that's pretty broken. Try main instead.
+      MOZ_ASSERT(false);
+      backgroundTaskQueue = GetMainThreadSerialEventTarget();
+    }
+
+    RefPtr<GenericPromise> shutdownPromise = mTaskQueue->BeginShutdown()->Then(
+        backgroundTaskQueue, __func__, [this] {
+          // Wait until shutdown is complete, then delete for real. Although we
+          // prevent queued tasks from executing with mHasShutdown, that is a
+          // member variable, which means we still need to ensure that the
+          // queue is done executing tasks before destroying it.
+          delete this;
+          return GenericPromise::CreateAndResolve(true, __func__);
+        });
+    if constexpr (Deletion == DeletionPolicy::Blocking) {
+      media::Await(backgroundTaskQueue.forget(), shutdownPromise);
+    } else {
+      Unused << shutdownPromise;
+    }
+  }
+
+  already_AddRefed<Runnable> CreateTaskRunner(
+      absl::AnyInvocable<void() &&> aTask) {
+    return NS_NewRunnableFunction(
+        "TaskQueueWrapper::CreateTaskRunner",
+        [this, task = std::move(aTask),
+         name = nsPrintfCString("TQ %s: webrtc::QueuedTask",
+                                mName.get())]() mutable {
+          CurrentTaskQueueSetter current(this);
+          auto hasShutdown = mHasShutdown.Lock();
+          if (*hasShutdown) {
+            return;
+          }
+          AUTO_PROFILE_FOLLOWING_RUNNABLE(name);
+          std::move(task)();
+        });
+  }
+
+  already_AddRefed<Runnable> CreateTaskRunner(nsCOMPtr<nsIRunnable> aRunnable) {
+    return NS_NewRunnableFunction(
+        "TaskQueueWrapper::CreateTaskRunner",
+        [this, runnable = std::move(aRunnable)]() mutable {
+          CurrentTaskQueueSetter current(this);
+          auto hasShutdown = mHasShutdown.Lock();
+          if (*hasShutdown) {
+            return;
+          }
+          AUTO_PROFILE_FOLLOWING_RUNNABLE(runnable);
+          runnable->Run();
+        });
+  }
+
+  void PostTask(absl::AnyInvocable<void() &&> aTask) override {
+    MOZ_ALWAYS_SUCCEEDS(
+        mTaskQueue->Dispatch(CreateTaskRunner(std::move(aTask))));
+  }
+
+  void PostDelayedTask(absl::AnyInvocable<void() &&> aTask,
+                       webrtc::TimeDelta aDelay) override {
+    if (aDelay.ms() == 0) {
+      // AbstractThread::DelayedDispatch doesn't support delay 0
+      PostTask(std::move(aTask));
+      return;
+    }
+    MOZ_ALWAYS_SUCCEEDS(mTaskQueue->DelayedDispatch(
+        CreateTaskRunner(std::move(aTask)), aDelay.ms()));
+  }
+
+  void PostDelayedHighPrecisionTask(absl::AnyInvocable<void() &&> aTask,
+                                    webrtc::TimeDelta aDelay) override {
+    PostDelayedTask(std::move(aTask), aDelay);
+  }
+
+  const RefPtr<TaskQueue> mTaskQueue;
+  const nsCString mName;
+
+  // This is a recursive mutex because a TaskRunner holding this mutex while
+  // running its runnable may end up running other - tail dispatched - runnables
+  // too, and they'll again try to grab the mutex.
+  // The mutex must be held while running the runnable since otherwise there'd
+  // be a race between shutting down the underlying task queue and the runnable
+  // dispatching to that task queue (and we assert it succeeds in e.g.,
+  // PostTask()).
+  DataMutexBase<bool, RecursiveMutex> mHasShutdown{
+      false, "TaskQueueWrapper::mHasShutdown"};
+};
+
+template <DeletionPolicy Deletion>
+class DefaultDelete<TaskQueueWrapper<Deletion>>
+    : public webrtc::TaskQueueDeleter {
+ public:
+  void operator()(TaskQueueWrapper<Deletion>* aPtr) const {
+    webrtc::TaskQueueDeleter::operator()(aPtr);
+  }
+};
+
+class SharedThreadPoolWebRtcTaskQueueFactory : public webrtc::TaskQueueFactory {
+ public:
+  SharedThreadPoolWebRtcTaskQueueFactory() {}
+
+  template <DeletionPolicy Deletion>
+  UniquePtr<TaskQueueWrapper<Deletion>> CreateTaskQueueWrapper(
+      absl::string_view aName, bool aSupportTailDispatch, Priority aPriority,
+      MediaThreadType aThreadType = MediaThreadType::WEBRTC_WORKER) const {
+    // XXX Do something with aPriority
+    nsCString name(aName.data(), aName.size());
+    auto taskQueue = TaskQueue::Create(GetMediaThreadPool(aThreadType),
+                                       name.get(), aSupportTailDispatch);
+    return MakeUnique<TaskQueueWrapper<Deletion>>(std::move(taskQueue),
+                                                  std::move(name));
+  }
+
+  std::unique_ptr<webrtc::TaskQueueBase, webrtc::TaskQueueDeleter>
+  CreateTaskQueue(absl::string_view aName, Priority aPriority) const override {
+    // libwebrtc will dispatch some tasks sync, i.e., block the origin thread
+    // until they've run, and that doesn't play nice with tail dispatching since
+    // there will never be a tail.
+    // DeletionPolicy::Blocking because this is for libwebrtc use and that's
+    // what they expect.
+    constexpr bool supportTailDispatch = false;
+    return std::unique_ptr<webrtc::TaskQueueBase, webrtc::TaskQueueDeleter>(
+        CreateTaskQueueWrapper<DeletionPolicy::Blocking>(
+            std::move(aName), supportTailDispatch, aPriority)
+            .release(),
+        webrtc::TaskQueueDeleter());
+  }
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/libwebrtcglue/VideoConduit.cpp b/dom/media/webrtc/libwebrtcglue/VideoConduit.cpp
new file mode 100644
index 0000000000..f99f818d48
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/VideoConduit.cpp
@@ -0,0 +1,1902 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "VideoConduit.h"
+
+#include <algorithm>
+#include <cinttypes>
+#include <cmath>
+
+#include "common/browser_logging/CSFLog.h"
+#include "common/YuvStamper.h"
+#include "GmpVideoCodec.h"
+#include "MediaConduitControl.h"
+#include "MediaDataCodec.h"
+#include "mozilla/dom/RTCRtpSourcesBinding.h"
+#include "mozilla/media/MediaUtils.h"
+#include "mozilla/StaticPrefs_media.h"
+#include "mozilla/TemplateLib.h"
+#include "nsIGfxInfo.h"
+#include "nsIPrefBranch.h"
+#include "nsIPrefService.h"
+#include "nsServiceManagerUtils.h"
+#include "RtpRtcpConfig.h"
+#include "transport/SrtpFlow.h"  // For SRTP_MAX_EXPANSION
+#include "Tracing.h"
+#include "VideoStreamFactory.h"
+#include "WebrtcCallWrapper.h"
+#include "WebrtcGmpVideoCodec.h"
+
+// libwebrtc includes
+#include "api/transport/bitrate_settings.h"
+#include "api/video_codecs/h264_profile_level_id.h"
+#include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_codec.h"
+#include "media/base/media_constants.h"
+#include "media/engine/encoder_simulcast_proxy.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/video_coding/codecs/vp8/include/vp8.h"
+#include "modules/video_coding/codecs/vp9/include/vp9.h"
+
+#ifdef MOZ_WIDGET_ANDROID
+#  include "VideoEngine.h"
+#endif
+
+// for ntohs
+#ifdef _MSC_VER
+#  include "Winsock2.h"
+#else
+#  include <netinet/in.h>
+#endif
+
+#define INVALID_RTP_PAYLOAD 255  // valid payload types are 0 to 127
+
+namespace mozilla {
+
+namespace {
+
+const char* vcLogTag = "WebrtcVideoSessionConduit";
+#ifdef LOGTAG
+#  undef LOGTAG
+#endif
+#define LOGTAG vcLogTag
+
+using namespace cricket;
+using LocalDirection = MediaSessionConduitLocalDirection;
+
+const int kNullPayloadType = -1;
+const char kRtcpFbCcmParamTmmbr[] = "tmmbr";
+
+// The number of frame buffers WebrtcVideoConduit may create before returning
+// errors.
+// Sometimes these are released synchronously but they can be forwarded all the
+// way to the encoder for asynchronous encoding. With a pool size of 5,
+// we allow 1 buffer for the current conversion, and 4 buffers to be queued at
+// the encoder.
+#define SCALER_BUFFER_POOL_SIZE 5
+
+// The pixel alignment to use for the highest resolution layer when simulcast
+// is active and one or more layers are being scaled.
+#define SIMULCAST_RESOLUTION_ALIGNMENT 16
+
+template <class t>
+void ConstrainPreservingAspectRatioExact(uint32_t max_fs, t* width, t* height) {
+  // We could try to pick a better starting divisor, but it won't make any real
+  // performance difference.
+  for (size_t d = 1; d < std::min(*width, *height); ++d) {
+    if ((*width % d) || (*height % d)) {
+      continue;  // Not divisible
+    }
+
+    if (((*width) * (*height)) / (d * d) <= max_fs) {
+      *width /= d;
+      *height /= d;
+      return;
+    }
+  }
+
+  *width = 0;
+  *height = 0;
+}
+
+/**
+ * Perform validation on the codecConfig to be applied
+ */
+MediaConduitErrorCode ValidateCodecConfig(const VideoCodecConfig& codecInfo) {
+  if (codecInfo.mName.empty()) {
+    CSFLogError(LOGTAG, "%s Empty Payload Name ", __FUNCTION__);
+    return kMediaConduitMalformedArgument;
+  }
+
+  return kMediaConduitNoError;
+}
+
+webrtc::VideoCodecType SupportedCodecType(webrtc::VideoCodecType aType) {
+  switch (aType) {
+    case webrtc::VideoCodecType::kVideoCodecVP8:
+    case webrtc::VideoCodecType::kVideoCodecVP9:
+    case webrtc::VideoCodecType::kVideoCodecH264:
+      return aType;
+    default:
+      return webrtc::VideoCodecType::kVideoCodecGeneric;
+  }
+  // NOTREACHED
+}
+
+// Call thread only.
+rtc::scoped_refptr<webrtc::VideoEncoderConfig::EncoderSpecificSettings>
+ConfigureVideoEncoderSettings(const VideoCodecConfig& aConfig,
+                              const WebrtcVideoConduit* aConduit,
+                              webrtc::SdpVideoFormat::Parameters& aParameters) {
+  bool is_screencast =
+      aConduit->CodecMode() == webrtc::VideoCodecMode::kScreensharing;
+  // No automatic resizing when using simulcast or screencast.
+  bool automatic_resize = !is_screencast && aConfig.mEncodings.size() <= 1;
+  bool denoising;
+  bool codec_default_denoising = false;
+  if (is_screencast) {
+    denoising = false;
+  } else {
+    // Use codec default if video_noise_reduction is unset.
+    denoising = aConduit->Denoising();
+    codec_default_denoising = !denoising;
+  }
+
+  if (aConfig.mName == kH264CodecName) {
+    aParameters[kH264FmtpPacketizationMode] =
+        std::to_string(aConfig.mPacketizationMode);
+    {
+      std::stringstream ss;
+      ss << std::hex << std::setfill('0');
+      ss << std::setw(2) << static_cast<uint32_t>(aConfig.mProfile);
+      ss << std::setw(2) << static_cast<uint32_t>(aConfig.mConstraints);
+      ss << std::setw(2) << static_cast<uint32_t>(aConfig.mLevel);
+      std::string profileLevelId = ss.str();
+      auto parsedProfileLevelId =
+          webrtc::ParseH264ProfileLevelId(profileLevelId.c_str());
+      MOZ_DIAGNOSTIC_ASSERT(parsedProfileLevelId);
+      if (parsedProfileLevelId) {
+        aParameters[kH264FmtpProfileLevelId] = profileLevelId;
+      }
+    }
+    aParameters[kH264FmtpSpropParameterSets] = aConfig.mSpropParameterSets;
+  }
+  if (aConfig.mName == kVp8CodecName) {
+    webrtc::VideoCodecVP8 vp8_settings =
+        webrtc::VideoEncoder::GetDefaultVp8Settings();
+    vp8_settings.automaticResizeOn = automatic_resize;
+    // VP8 denoising is enabled by default.
+    vp8_settings.denoisingOn = codec_default_denoising ? true : denoising;
+    return rtc::scoped_refptr<
+        webrtc::VideoEncoderConfig::EncoderSpecificSettings>(
+        new rtc::RefCountedObject<
+            webrtc::VideoEncoderConfig::Vp8EncoderSpecificSettings>(
+            vp8_settings));
+  }
+  if (aConfig.mName == kVp9CodecName) {
+    webrtc::VideoCodecVP9 vp9_settings =
+        webrtc::VideoEncoder::GetDefaultVp9Settings();
+    if (!is_screencast) {
+      // Always configure only 1 spatial layer for screencapture as libwebrtc
+      // has some special requirements when SVC is active. For non-screencapture
+      // the spatial layers are experimentally configurable via a pref.
+      vp9_settings.numberOfSpatialLayers = aConduit->SpatialLayers();
+    }
+    // VP9 denoising is disabled by default.
+    vp9_settings.denoisingOn = codec_default_denoising ? false : denoising;
+    return rtc::scoped_refptr<
+        webrtc::VideoEncoderConfig::EncoderSpecificSettings>(
+        new rtc::RefCountedObject<
+            webrtc::VideoEncoderConfig::Vp9EncoderSpecificSettings>(
+            vp9_settings));
+  }
+  return nullptr;
+}
+
+uint32_t GenerateRandomSSRC() {
+  uint32_t ssrc;
+  do {
+    SECStatus rv = PK11_GenerateRandom(reinterpret_cast<unsigned char*>(&ssrc),
+                                       sizeof(ssrc));
+    MOZ_RELEASE_ASSERT(rv == SECSuccess);
+  } while (ssrc == 0);  // webrtc.org code has fits if you select an SSRC of 0
+
+  return ssrc;
+}
+
+// TODO: Make this a defaulted operator when we have c++20 (bug 1731036).
+bool operator==(const rtc::VideoSinkWants& aThis,
+                const rtc::VideoSinkWants& aOther) {
+  // This would have to be expanded should we make use of more members of
+  // rtc::VideoSinkWants.
+  return aThis.max_pixel_count == aOther.max_pixel_count &&
+         aThis.max_framerate_fps == aOther.max_framerate_fps &&
+         aThis.resolution_alignment == aOther.resolution_alignment;
+}
+
+// TODO: Make this a defaulted operator when we have c++20 (bug 1731036).
+bool operator!=(const rtc::VideoSinkWants& aThis,
+                const rtc::VideoSinkWants& aOther) {
+  return !(aThis == aOther);
+}
+
+// TODO: Make this a defaulted operator when we have c++20 (bug 1731036).
+bool operator!=(
+    const webrtc::VideoReceiveStreamInterface::Config::Rtp& aThis,
+    const webrtc::VideoReceiveStreamInterface::Config::Rtp& aOther) {
+  return aThis.remote_ssrc != aOther.remote_ssrc ||
+         aThis.local_ssrc != aOther.local_ssrc ||
+         aThis.rtcp_mode != aOther.rtcp_mode ||
+         aThis.rtcp_xr.receiver_reference_time_report !=
+             aOther.rtcp_xr.receiver_reference_time_report ||
+         aThis.remb != aOther.remb || aThis.tmmbr != aOther.tmmbr ||
+         aThis.keyframe_method != aOther.keyframe_method ||
+         aThis.lntf.enabled != aOther.lntf.enabled ||
+         aThis.nack.rtp_history_ms != aOther.nack.rtp_history_ms ||
+         aThis.ulpfec_payload_type != aOther.ulpfec_payload_type ||
+         aThis.red_payload_type != aOther.red_payload_type ||
+         aThis.rtx_ssrc != aOther.rtx_ssrc ||
+         aThis.protected_by_flexfec != aOther.protected_by_flexfec ||
+         aThis.rtx_associated_payload_types !=
+             aOther.rtx_associated_payload_types ||
+         aThis.raw_payload_types != aOther.raw_payload_types ||
+         aThis.extensions != aOther.extensions;
+}
+
+#ifdef DEBUG
+// TODO: Make this a defaulted operator when we have c++20 (bug 1731036).
+bool operator==(
+    const webrtc::VideoReceiveStreamInterface::Config::Rtp& aThis,
+    const webrtc::VideoReceiveStreamInterface::Config::Rtp& aOther) {
+  return !(aThis != aOther);
+}
+#endif
+
+// TODO: Make this a defaulted operator when we have c++20 (bug 1731036).
+bool operator!=(const webrtc::RtpConfig& aThis,
+                const webrtc::RtpConfig& aOther) {
+  return aThis.ssrcs != aOther.ssrcs || aThis.rids != aOther.rids ||
+         aThis.mid != aOther.mid || aThis.rtcp_mode != aOther.rtcp_mode ||
+         aThis.max_packet_size != aOther.max_packet_size ||
+         aThis.extmap_allow_mixed != aOther.extmap_allow_mixed ||
+         aThis.extensions != aOther.extensions ||
+         aThis.payload_name != aOther.payload_name ||
+         aThis.payload_type != aOther.payload_type ||
+         aThis.raw_payload != aOther.raw_payload ||
+         aThis.lntf.enabled != aOther.lntf.enabled ||
+         aThis.nack.rtp_history_ms != aOther.nack.rtp_history_ms ||
+         !(aThis.ulpfec == aOther.ulpfec) ||
+         aThis.flexfec.payload_type != aOther.flexfec.payload_type ||
+         aThis.flexfec.ssrc != aOther.flexfec.ssrc ||
+         aThis.flexfec.protected_media_ssrcs !=
+             aOther.flexfec.protected_media_ssrcs ||
+         aThis.rtx.ssrcs != aOther.rtx.ssrcs ||
+         aThis.rtx.payload_type != aOther.rtx.payload_type ||
+         aThis.c_name != aOther.c_name;
+}
+
+#ifdef DEBUG
+// TODO: Make this a defaulted operator when we have c++20 (bug 1731036).
+bool operator==(const webrtc::RtpConfig& aThis,
+                const webrtc::RtpConfig& aOther) {
+  return !(aThis != aOther);
+}
+#endif
+
+}  // namespace
+
+/**
+ * Factory Method for VideoConduit
+ */
+RefPtr<VideoSessionConduit> VideoSessionConduit::Create(
+    RefPtr<WebrtcCallWrapper> aCall, nsCOMPtr<nsISerialEventTarget> aStsThread,
+    Options aOptions, std::string aPCHandle,
+    const TrackingId& aRecvTrackingId) {
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_ASSERT(aCall, "missing required parameter: aCall");
+  CSFLogVerbose(LOGTAG, "%s", __FUNCTION__);
+
+  if (!aCall) {
+    return nullptr;
+  }
+
+  auto obj = MakeRefPtr<WebrtcVideoConduit>(
+      std::move(aCall), std::move(aStsThread), std::move(aOptions),
+      std::move(aPCHandle), aRecvTrackingId);
+  if (obj->Init() != kMediaConduitNoError) {
+    CSFLogError(LOGTAG, "%s VideoConduit Init Failed ", __FUNCTION__);
+    return nullptr;
+  }
+  CSFLogVerbose(LOGTAG, "%s Successfully created VideoConduit ", __FUNCTION__);
+  return obj.forget();
+}
+
+#define INIT_MIRROR(name, val) \
+  name(aCallThread, val, "WebrtcVideoConduit::Control::" #name " (Mirror)")
+WebrtcVideoConduit::Control::Control(const RefPtr<AbstractThread>& aCallThread)
+    : INIT_MIRROR(mReceiving, false),
+      INIT_MIRROR(mTransmitting, false),
+      INIT_MIRROR(mLocalSsrcs, Ssrcs()),
+      INIT_MIRROR(mLocalRtxSsrcs, Ssrcs()),
+      INIT_MIRROR(mLocalCname, std::string()),
+      INIT_MIRROR(mMid, std::string()),
+      INIT_MIRROR(mRemoteSsrc, 0),
+      INIT_MIRROR(mRemoteRtxSsrc, 0),
+      INIT_MIRROR(mSyncGroup, std::string()),
+      INIT_MIRROR(mLocalRecvRtpExtensions, RtpExtList()),
+      INIT_MIRROR(mLocalSendRtpExtensions, RtpExtList()),
+      INIT_MIRROR(mSendCodec, Nothing()),
+      INIT_MIRROR(mSendRtpRtcpConfig, Nothing()),
+      INIT_MIRROR(mRecvCodecs, std::vector<VideoCodecConfig>()),
+      INIT_MIRROR(mRecvRtpRtcpConfig, Nothing()),
+      INIT_MIRROR(mCodecMode, webrtc::VideoCodecMode::kRealtimeVideo) {}
+#undef INIT_MIRROR
+
+WebrtcVideoConduit::WebrtcVideoConduit(
+    RefPtr<WebrtcCallWrapper> aCall, nsCOMPtr<nsISerialEventTarget> aStsThread,
+    Options aOptions, std::string aPCHandle, const TrackingId& aRecvTrackingId)
+    : mRendererMonitor("WebrtcVideoConduit::mRendererMonitor"),
+      mCallThread(aCall->mCallThread),
+      mStsThread(std::move(aStsThread)),
+      mControl(aCall->mCallThread),
+      mWatchManager(this, aCall->mCallThread),
+      mMutex("WebrtcVideoConduit::mMutex"),
+      mDecoderFactory(MakeUnique<WebrtcVideoDecoderFactory>(
+          mCallThread.get(), aPCHandle, aRecvTrackingId)),
+      mEncoderFactory(MakeUnique<WebrtcVideoEncoderFactory>(
+          mCallThread.get(), std::move(aPCHandle))),
+      mBufferPool(false, SCALER_BUFFER_POOL_SIZE),
+      mEngineTransmitting(false),
+      mEngineReceiving(false),
+      mVideoLatencyTestEnable(aOptions.mVideoLatencyTestEnable),
+      mMinBitrate(aOptions.mMinBitrate),
+      mStartBitrate(aOptions.mStartBitrate),
+      mPrefMaxBitrate(aOptions.mPrefMaxBitrate),
+      mMinBitrateEstimate(aOptions.mMinBitrateEstimate),
+      mDenoising(aOptions.mDenoising),
+      mLockScaling(aOptions.mLockScaling),
+      mSpatialLayers(aOptions.mSpatialLayers),
+      mTemporalLayers(aOptions.mTemporalLayers),
+      mCall(std::move(aCall)),
+      mSendTransport(this),
+      mRecvTransport(this),
+      mSendStreamConfig(&mSendTransport),
+      mVideoStreamFactory("WebrtcVideoConduit::mVideoStreamFactory"),
+      mRecvStreamConfig(&mRecvTransport) {
+  mRecvStreamConfig.rtp.rtcp_event_observer = this;
+}
+
+WebrtcVideoConduit::~WebrtcVideoConduit() {
+  CSFLogDebug(LOGTAG, "%s ", __FUNCTION__);
+
+  MOZ_ASSERT(!mSendStream && !mRecvStream,
+             "Call DeleteStreams prior to ~WebrtcVideoConduit.");
+}
+
+#define CONNECT(aCanonical, aMirror)                                          \
+  do {                                                                        \
+    (aMirror).Connect(aCanonical);                                            \
+    mWatchManager.Watch(aMirror, &WebrtcVideoConduit::OnControlConfigChange); \
+  } while (0)
+
+void WebrtcVideoConduit::InitControl(VideoConduitControlInterface* aControl) {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+
+  CONNECT(aControl->CanonicalReceiving(), mControl.mReceiving);
+  CONNECT(aControl->CanonicalTransmitting(), mControl.mTransmitting);
+  CONNECT(aControl->CanonicalLocalSsrcs(), mControl.mLocalSsrcs);
+  CONNECT(aControl->CanonicalLocalVideoRtxSsrcs(), mControl.mLocalRtxSsrcs);
+  CONNECT(aControl->CanonicalLocalCname(), mControl.mLocalCname);
+  CONNECT(aControl->CanonicalMid(), mControl.mMid);
+  CONNECT(aControl->CanonicalRemoteSsrc(), mControl.mRemoteSsrc);
+  CONNECT(aControl->CanonicalRemoteVideoRtxSsrc(), mControl.mRemoteRtxSsrc);
+  CONNECT(aControl->CanonicalSyncGroup(), mControl.mSyncGroup);
+  CONNECT(aControl->CanonicalLocalRecvRtpExtensions(),
+          mControl.mLocalRecvRtpExtensions);
+  CONNECT(aControl->CanonicalLocalSendRtpExtensions(),
+          mControl.mLocalSendRtpExtensions);
+  CONNECT(aControl->CanonicalVideoSendCodec(), mControl.mSendCodec);
+  CONNECT(aControl->CanonicalVideoSendRtpRtcpConfig(),
+          mControl.mSendRtpRtcpConfig);
+  CONNECT(aControl->CanonicalVideoRecvCodecs(), mControl.mRecvCodecs);
+  CONNECT(aControl->CanonicalVideoRecvRtpRtcpConfig(),
+          mControl.mRecvRtpRtcpConfig);
+  CONNECT(aControl->CanonicalVideoCodecMode(), mControl.mCodecMode);
+}
+
+#undef CONNECT
+
+void WebrtcVideoConduit::OnControlConfigChange() {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+
+  bool encoderReconfigureNeeded = false;
+  bool remoteSsrcUpdateNeeded = false;
+  bool sendStreamRecreationNeeded = false;
+
+  if (mControl.mRemoteSsrc.Ref() != mControl.mConfiguredRemoteSsrc) {
+    mControl.mConfiguredRemoteSsrc = mControl.mRemoteSsrc;
+    remoteSsrcUpdateNeeded = true;
+  }
+
+  if (mControl.mRemoteRtxSsrc.Ref() != mControl.mConfiguredRemoteRtxSsrc) {
+    mControl.mConfiguredRemoteRtxSsrc = mControl.mRemoteRtxSsrc;
+    remoteSsrcUpdateNeeded = true;
+  }
+
+  if (mControl.mSyncGroup.Ref() != mRecvStreamConfig.sync_group) {
+    mRecvStreamConfig.sync_group = mControl.mSyncGroup;
+  }
+
+  if (mControl.mLocalRecvRtpExtensions.Ref() !=
+      mRecvStreamConfig.rtp.extensions) {
+    mRecvStreamConfig.rtp.extensions = mControl.mLocalRecvRtpExtensions;
+  }
+
+  if (const auto [codecConfigList, rtpRtcpConfig] = std::make_pair(
+          mControl.mRecvCodecs.Ref(), mControl.mRecvRtpRtcpConfig.Ref());
+      !codecConfigList.empty() && rtpRtcpConfig.isSome() &&
+      (codecConfigList != mControl.mConfiguredRecvCodecs ||
+       rtpRtcpConfig != mControl.mConfiguredRecvRtpRtcpConfig)) {
+    mControl.mConfiguredRecvCodecs = codecConfigList;
+    mControl.mConfiguredRecvRtpRtcpConfig = rtpRtcpConfig;
+
+    webrtc::VideoReceiveStreamInterface::Config::Rtp newRtp(
+        mRecvStreamConfig.rtp);
+    MOZ_ASSERT(newRtp == mRecvStreamConfig.rtp);
+    newRtp.rtx_associated_payload_types.clear();
+    newRtp.rtcp_mode = rtpRtcpConfig->GetRtcpMode();
+    newRtp.nack.rtp_history_ms = 0;
+    newRtp.remb = false;
+    newRtp.tmmbr = false;
+    newRtp.keyframe_method = webrtc::KeyFrameReqMethod::kNone;
+    newRtp.ulpfec_payload_type = kNullPayloadType;
+    newRtp.red_payload_type = kNullPayloadType;
+    bool use_fec = false;
+    bool configuredH264 = false;
+    std::vector<webrtc::VideoReceiveStreamInterface::Decoder> recv_codecs;
+
+    // Try Applying the codecs in the list
+    // we treat as success if at least one codec was applied and reception was
+    // started successfully.
+    for (const auto& codec_config : codecConfigList) {
+      if (auto condError = ValidateCodecConfig(codec_config);
+          condError != kMediaConduitNoError) {
+        CSFLogError(LOGTAG, "Invalid recv codec config for %s decoder: %i",
+                    codec_config.mName.c_str(), condError);
+        continue;
+      }
+
+      if (codec_config.mName == kH264CodecName) {
+        // TODO(bug 1200768): We can only handle configuring one recv H264 codec
+        if (configuredH264) {
+          continue;
+        }
+        configuredH264 = true;
+      }
+
+      if (codec_config.mName == kUlpfecCodecName) {
+        newRtp.ulpfec_payload_type = codec_config.mType;
+        continue;
+      }
+
+      if (codec_config.mName == kRedCodecName) {
+        newRtp.red_payload_type = codec_config.mType;
+        continue;
+      }
+
+      if (SupportedCodecType(
+              webrtc::PayloadStringToCodecType(codec_config.mName)) ==
+          webrtc::VideoCodecType::kVideoCodecGeneric) {
+        CSFLogError(LOGTAG, "%s Unknown decoder type: %s", __FUNCTION__,
+                    codec_config.mName.c_str());
+        continue;
+      }
+
+      // Check for the keyframe request type: PLI is preferred over FIR, and FIR
+      // is preferred over none.
+      if (codec_config.RtcpFbNackIsSet(kRtcpFbNackParamPli)) {
+        newRtp.keyframe_method = webrtc::KeyFrameReqMethod::kPliRtcp;
+      } else if (newRtp.keyframe_method !=
+                     webrtc::KeyFrameReqMethod::kPliRtcp &&
+                 codec_config.RtcpFbCcmIsSet(kRtcpFbCcmParamFir)) {
+        newRtp.keyframe_method = webrtc::KeyFrameReqMethod::kFirRtcp;
+      }
+
+      // What if codec A has Nack and REMB, and codec B has TMMBR, and codec C
+      // has none? In practice, that's not a useful configuration, and
+      // VideoReceiveStream::Config can't represent that, so simply union the
+      // (boolean) settings
+      if (codec_config.RtcpFbNackIsSet(kParamValueEmpty)) {
+        newRtp.nack.rtp_history_ms = 1000;
+      }
+      newRtp.tmmbr |= codec_config.RtcpFbCcmIsSet(kRtcpFbCcmParamTmmbr);
+      newRtp.remb |= codec_config.RtcpFbRembIsSet();
+      use_fec |= codec_config.RtcpFbFECIsSet();
+
+      if (codec_config.RtxPayloadTypeIsSet()) {
+        newRtp.rtx_associated_payload_types[codec_config.mRTXPayloadType] =
+            codec_config.mType;
+      }
+
+      auto& decoder = recv_codecs.emplace_back();
+      decoder.video_format = webrtc::SdpVideoFormat(codec_config.mName);
+      decoder.payload_type = codec_config.mType;
+    }
+
+    if (!use_fec) {
+      // Reset to defaults
+      newRtp.ulpfec_payload_type = kNullPayloadType;
+      newRtp.red_payload_type = kNullPayloadType;
+    }
+
+    // TODO: This would be simpler, but for some reason gives
+    //       "error: invalid operands to binary expression
+    //       ('webrtc::VideoReceiveStreamInterface::Decoder' and
+    //       'webrtc::VideoReceiveStreamInterface::Decoder')"
+    // if (recv_codecs != mRecvStreamConfig.decoders) {
+    if (!std::equal(recv_codecs.begin(), recv_codecs.end(),
+                    mRecvStreamConfig.decoders.begin(),
+                    mRecvStreamConfig.decoders.end(),
+                    [](const auto& aLeft, const auto& aRight) {
+                      return aLeft == aRight;
+                    })) {
+      if (recv_codecs.empty()) {
+        CSFLogError(LOGTAG, "%s Found no valid receive codecs", __FUNCTION__);
+      }
+      mRecvStreamConfig.decoders = std::move(recv_codecs);
+    }
+
+    if (mRecvStreamConfig.rtp != newRtp) {
+      mRecvStreamConfig.rtp = newRtp;
+    }
+  }
+
+  {
+    // mSendStreamConfig and other members need the lock
+    MutexAutoLock lock(mMutex);
+    if (mControl.mLocalSsrcs.Ref() != mSendStreamConfig.rtp.ssrcs) {
+      mSendStreamConfig.rtp.ssrcs = mControl.mLocalSsrcs;
+      sendStreamRecreationNeeded = true;
+
+      const uint32_t localSsrc = mSendStreamConfig.rtp.ssrcs.empty()
+                                     ? 0
+                                     : mSendStreamConfig.rtp.ssrcs.front();
+      if (localSsrc != mRecvStreamConfig.rtp.local_ssrc) {
+        mRecvStreamConfig.rtp.local_ssrc = localSsrc;
+      }
+    }
+
+    {
+      Ssrcs localRtxSsrcs = mControl.mLocalRtxSsrcs.Ref();
+      if (!mControl.mSendCodec.Ref()
+               .map([](const auto& aCodec) {
+                 return aCodec.RtxPayloadTypeIsSet();
+               })
+               .valueOr(false)) {
+        localRtxSsrcs.clear();
+      }
+      if (localRtxSsrcs != mSendStreamConfig.rtp.rtx.ssrcs) {
+        mSendStreamConfig.rtp.rtx.ssrcs = localRtxSsrcs;
+        sendStreamRecreationNeeded = true;
+      }
+    }
+
+    if (mControl.mLocalCname.Ref() != mSendStreamConfig.rtp.c_name) {
+      mSendStreamConfig.rtp.c_name = mControl.mLocalCname;
+      sendStreamRecreationNeeded = true;
+    }
+
+    if (mControl.mMid.Ref() != mSendStreamConfig.rtp.mid) {
+      mSendStreamConfig.rtp.mid = mControl.mMid;
+      sendStreamRecreationNeeded = true;
+    }
+
+    if (mControl.mLocalSendRtpExtensions.Ref() !=
+        mSendStreamConfig.rtp.extensions) {
+      mSendStreamConfig.rtp.extensions = mControl.mLocalSendRtpExtensions;
+      sendStreamRecreationNeeded = true;
+    }
+
+    if (const auto [codecConfig, rtpRtcpConfig] = std::make_pair(
+            mControl.mSendCodec.Ref(), mControl.mSendRtpRtcpConfig.Ref());
+        codecConfig.isSome() && rtpRtcpConfig.isSome() &&
+        (codecConfig != mControl.mConfiguredSendCodec ||
+         rtpRtcpConfig != mControl.mConfiguredSendRtpRtcpConfig)) {
+      CSFLogDebug(LOGTAG, "Configuring codec %s", codecConfig->mName.c_str());
+      mControl.mConfiguredSendCodec = codecConfig;
+      mControl.mConfiguredSendRtpRtcpConfig = rtpRtcpConfig;
+
+      if (ValidateCodecConfig(*codecConfig) == kMediaConduitNoError) {
+        encoderReconfigureNeeded = true;
+
+        mCurSendCodecConfig = codecConfig;
+
+        size_t streamCount = std::min(codecConfig->mEncodings.size(),
+                                      (size_t)webrtc::kMaxSimulcastStreams);
+        size_t highestResolutionIndex = 0;
+        for (size_t i = 1; i < streamCount; ++i) {
+          if (codecConfig->mEncodings[i].constraints.scaleDownBy <
+              codecConfig->mEncodings[highestResolutionIndex]
+                  .constraints.scaleDownBy) {
+            highestResolutionIndex = i;
+          }
+        }
+        MOZ_RELEASE_ASSERT(streamCount >= 1,
+                           "streamCount should be at least one");
+
+        CSFLogDebug(LOGTAG,
+                    "Updating send codec for VideoConduit:%p stream count:%zu",
+                    this, streamCount);
+
+        // So we can comply with b=TIAS/b=AS/maxbr=X when input resolution
+        // changes
+        MOZ_ASSERT(codecConfig->mTias < INT_MAX);
+        mNegotiatedMaxBitrate = static_cast<int>(codecConfig->mTias);
+
+        if (mLastWidth == 0 && mMinBitrateEstimate != 0) {
+          // Only do this at the start; use "have we sent a frame" as a
+          // reasonable stand-in. min <= start <= max (but all three parameters
+          // are optional)
+          webrtc::BitrateSettings settings;
+          settings.min_bitrate_bps = mMinBitrateEstimate;
+          settings.start_bitrate_bps = mMinBitrateEstimate;
+          mCall->Call()->SetClientBitratePreferences(settings);
+        }
+
+        // XXX parse the encoded SPS/PPS data and set
+        // spsData/spsLen/ppsData/ppsLen
+        mEncoderConfig.video_format =
+            webrtc::SdpVideoFormat(codecConfig->mName);
+        mEncoderConfig.encoder_specific_settings =
+            ConfigureVideoEncoderSettings(
+                *codecConfig, this, mEncoderConfig.video_format.parameters);
+
+        mEncoderConfig.codec_type = SupportedCodecType(
+            webrtc::PayloadStringToCodecType(codecConfig->mName));
+        MOZ_RELEASE_ASSERT(mEncoderConfig.codec_type !=
+                           webrtc::VideoCodecType::kVideoCodecGeneric);
+
+        mEncoderConfig.content_type =
+            mControl.mCodecMode.Ref() == webrtc::VideoCodecMode::kRealtimeVideo
+                ? webrtc::VideoEncoderConfig::ContentType::kRealtimeVideo
+                : webrtc::VideoEncoderConfig::ContentType::kScreen;
+
+        mEncoderConfig.frame_drop_enabled =
+            mControl.mCodecMode.Ref() != webrtc::VideoCodecMode::kScreensharing;
+
+        mEncoderConfig.min_transmit_bitrate_bps = mMinBitrate;
+
+        // Set the max bitrate, defaulting to 10Mbps, checking:
+        // - pref
+        // - b=TIAS
+        // - codec constraints
+        // - encoding parameter if there's a single stream
+        int maxBps = KBPS(10000);
+        maxBps = MinIgnoreZero(maxBps, mPrefMaxBitrate);
+        maxBps = MinIgnoreZero(maxBps, mNegotiatedMaxBitrate);
+        maxBps = MinIgnoreZero(
+            maxBps, static_cast<int>(codecConfig->mEncodingConstraints.maxBr));
+        if (codecConfig->mEncodings.size() == 1) {
+          maxBps = MinIgnoreZero(
+              maxBps,
+              static_cast<int>(codecConfig->mEncodings[0].constraints.maxBr));
+        }
+        mEncoderConfig.max_bitrate_bps = maxBps;
+
+        // TODO this is for webrtc-priority, but needs plumbing bits
+        mEncoderConfig.bitrate_priority = 1.0;
+
+        // Expected max number of encodings
+        mEncoderConfig.number_of_streams = streamCount;
+
+        // libwebrtc disables this by default.
+        mSendStreamConfig.suspend_below_min_bitrate = false;
+
+        webrtc::RtpConfig newRtp = mSendStreamConfig.rtp;
+        MOZ_ASSERT(newRtp == mSendStreamConfig.rtp);
+        newRtp.payload_name = codecConfig->mName;
+        newRtp.payload_type = codecConfig->mType;
+        newRtp.rtcp_mode = rtpRtcpConfig->GetRtcpMode();
+        newRtp.max_packet_size = kVideoMtu;
+        newRtp.rtx.payload_type = codecConfig->RtxPayloadTypeIsSet()
+                                      ? codecConfig->mRTXPayloadType
+                                      : kNullPayloadType;
+
+        {
+          // See Bug 1297058, enabling FEC when basic NACK is to be enabled in
+          // H.264 is problematic
+          const bool useFECDefaults =
+              !codecConfig->RtcpFbFECIsSet() ||
+              (codecConfig->mName == kH264CodecName &&
+               codecConfig->RtcpFbNackIsSet(kParamValueEmpty));
+          newRtp.ulpfec.ulpfec_payload_type =
+              useFECDefaults ? kNullPayloadType
+                             : codecConfig->mULPFECPayloadType;
+          newRtp.ulpfec.red_payload_type =
+              useFECDefaults ? kNullPayloadType : codecConfig->mREDPayloadType;
+          newRtp.ulpfec.red_rtx_payload_type =
+              useFECDefaults ? kNullPayloadType
+                             : codecConfig->mREDRTXPayloadType;
+        }
+
+        newRtp.nack.rtp_history_ms =
+            codecConfig->RtcpFbNackIsSet(kParamValueEmpty) ? 1000 : 0;
+
+        {
+          newRtp.rids.clear();
+          bool has_rid = false;
+          for (size_t idx = 0; idx < streamCount; idx++) {
+            const auto& encoding = codecConfig->mEncodings[idx];
+            if (encoding.rid[0]) {
+              has_rid = true;
+              break;
+            }
+          }
+          if (has_rid) {
+            for (size_t idx = streamCount; idx > 0; idx--) {
+              const auto& encoding = codecConfig->mEncodings[idx - 1];
+              newRtp.rids.push_back(encoding.rid);
+            }
+          }
+        }
+        if (mSendStreamConfig.rtp != newRtp) {
+          mSendStreamConfig.rtp = newRtp;
+          sendStreamRecreationNeeded = true;
+        }
+
+        mEncoderConfig.video_stream_factory = CreateVideoStreamFactory();
+      }
+    }
+
+    {
+      const auto& mode = mControl.mCodecMode.Ref();
+      MOZ_ASSERT(mode == webrtc::VideoCodecMode::kRealtimeVideo ||
+                 mode == webrtc::VideoCodecMode::kScreensharing);
+
+      auto contentType =
+          mode == webrtc::VideoCodecMode::kRealtimeVideo
+              ? webrtc::VideoEncoderConfig::ContentType::kRealtimeVideo
+              : webrtc::VideoEncoderConfig::ContentType::kScreen;
+
+      if (contentType != mEncoderConfig.content_type) {
+        mEncoderConfig.video_stream_factory = CreateVideoStreamFactory();
+        encoderReconfigureNeeded = true;
+      }
+    }
+
+    if (remoteSsrcUpdateNeeded) {
+      SetRemoteSSRCConfig(mControl.mConfiguredRemoteSsrc,
+                          mControl.mConfiguredRemoteRtxSsrc);
+    }
+
+    // Handle un-signalled SSRCs by creating random ones and then when they
+    // actually get set, we'll destroy and recreate.
+    if (mControl.mReceiving || mControl.mTransmitting) {
+      const auto remoteSsrc = mRecvStreamConfig.rtp.remote_ssrc;
+      const auto localSsrc = mRecvStreamConfig.rtp.local_ssrc;
+      const auto localSsrcs = mSendStreamConfig.rtp.ssrcs;
+      EnsureLocalSSRC();
+      if (mControl.mReceiving) {
+        EnsureRemoteSSRC();
+      }
+      if (localSsrc != mRecvStreamConfig.rtp.local_ssrc ||
+          remoteSsrc != mRecvStreamConfig.rtp.remote_ssrc) {
+      }
+      if (localSsrcs != mSendStreamConfig.rtp.ssrcs) {
+        sendStreamRecreationNeeded = true;
+      }
+    }
+
+    // Recreate receiving streams
+    if (mControl.mReceiving) {
+      DeleteRecvStream();
+      CreateRecvStream();
+    }
+    if (sendStreamRecreationNeeded) {
+      DeleteSendStream();
+    }
+    if (mControl.mTransmitting) {
+      CreateSendStream();
+    }
+  }
+
+  // We make sure to not hold the lock while stopping/starting/reconfiguring
+  // streams, so as to not cause deadlocks. These methods can cause our platform
+  // codecs to dispatch sync runnables to main, and main may grab the lock.
+
+  if (mSendStream && encoderReconfigureNeeded) {
+    MOZ_DIAGNOSTIC_ASSERT(
+        mSendStreamConfig.rtp.ssrcs.size() == mEncoderConfig.number_of_streams,
+        "Each video substream must have a corresponding ssrc.");
+    mSendStream->ReconfigureVideoEncoder(mEncoderConfig.Copy());
+  }
+
+  if (!mControl.mReceiving) {
+    StopReceiving();
+  }
+  if (!mControl.mTransmitting) {
+    StopTransmitting();
+  }
+
+  if (mControl.mReceiving) {
+    StartReceiving();
+  }
+  if (mControl.mTransmitting) {
+    StartTransmitting();
+  }
+}
+
+std::vector<unsigned int> WebrtcVideoConduit::GetLocalSSRCs() const {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  return mSendStreamConfig.rtp.ssrcs;
+}
+
+Maybe<Ssrc> WebrtcVideoConduit::GetAssociatedLocalRtxSSRC(Ssrc aSsrc) const {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  for (size_t i = 0; i < mSendStreamConfig.rtp.ssrcs.size() &&
+                     i < mSendStreamConfig.rtp.rtx.ssrcs.size();
+       ++i) {
+    if (mSendStreamConfig.rtp.ssrcs[i] == aSsrc) {
+      return Some(mSendStreamConfig.rtp.rtx.ssrcs[i]);
+    }
+  }
+  return Nothing();
+}
+
+void WebrtcVideoConduit::DeleteSendStream() {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  mMutex.AssertCurrentThreadOwns();
+
+  if (!mSendStream) {
+    return;
+  }
+
+  mCall->Call()->DestroyVideoSendStream(mSendStream);
+  mEngineTransmitting = false;
+  mSendStream = nullptr;
+
+  // Reset base_seqs in case ssrcs get re-used.
+  mRtpSendBaseSeqs.clear();
+}
+
+void WebrtcVideoConduit::CreateSendStream() {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  mMutex.AssertCurrentThreadOwns();
+
+  if (mSendStream) {
+    return;
+  }
+
+  nsAutoString codecName;
+  codecName.AssignASCII(mSendStreamConfig.rtp.payload_name.c_str());
+  Telemetry::ScalarAdd(Telemetry::ScalarID::WEBRTC_VIDEO_SEND_CODEC_USED,
+                       codecName, 1);
+
+  mSendStreamConfig.encoder_settings.encoder_factory = mEncoderFactory.get();
+  mSendStreamConfig.encoder_settings.bitrate_allocator_factory =
+      mCall->mVideoBitrateAllocatorFactory.get();
+
+  MOZ_DIAGNOSTIC_ASSERT(
+      mSendStreamConfig.rtp.ssrcs.size() == mEncoderConfig.number_of_streams,
+      "Each video substream must have a corresponding ssrc.");
+
+  mSendStream = mCall->Call()->CreateVideoSendStream(mSendStreamConfig.Copy(),
+                                                     mEncoderConfig.Copy());
+
+  mSendStream->SetSource(this, webrtc::DegradationPreference::BALANCED);
+}
+
+void WebrtcVideoConduit::DeleteRecvStream() {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  mMutex.AssertCurrentThreadOwns();
+
+  if (!mRecvStream) {
+    return;
+  }
+
+  mCall->Call()->DestroyVideoReceiveStream(mRecvStream);
+  mEngineReceiving = false;
+  mRecvStream = nullptr;
+}
+
+void WebrtcVideoConduit::CreateRecvStream() {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  mMutex.AssertCurrentThreadOwns();
+
+  if (mRecvStream) {
+    return;
+  }
+
+  mRecvStreamConfig.renderer = this;
+
+  for (auto& decoder : mRecvStreamConfig.decoders) {
+    nsAutoString codecName;
+    codecName.AssignASCII(decoder.video_format.name.c_str());
+    Telemetry::ScalarAdd(Telemetry::ScalarID::WEBRTC_VIDEO_RECV_CODEC_USED,
+                         codecName, 1);
+  }
+
+  mRecvStreamConfig.decoder_factory = mDecoderFactory.get();
+
+  mRecvStream =
+      mCall->Call()->CreateVideoReceiveStream(mRecvStreamConfig.Copy());
+  // Ensure that we set the jitter buffer target on this stream.
+  mRecvStream->SetBaseMinimumPlayoutDelayMs(mJitterBufferTargetMs);
+
+  CSFLogDebug(LOGTAG, "Created VideoReceiveStream %p for SSRC %u (0x%x)",
+              mRecvStream, mRecvStreamConfig.rtp.remote_ssrc,
+              mRecvStreamConfig.rtp.remote_ssrc);
+}
+
+void WebrtcVideoConduit::NotifyUnsetCurrentRemoteSSRC() {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  CSFLogDebug(LOGTAG, "%s (%p): Unsetting SSRC %u in other conduits",
+              __FUNCTION__, this, mRecvStreamConfig.rtp.remote_ssrc);
+  mCall->UnregisterConduit(this);
+  mCall->UnsetRemoteSSRC(mRecvStreamConfig.rtp.remote_ssrc);
+  mCall->RegisterConduit(this);
+}
+
+void WebrtcVideoConduit::SetRemoteSSRCConfig(uint32_t aSsrc,
+                                             uint32_t aRtxSsrc) {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+
+  CSFLogDebug(LOGTAG, "%s: SSRC %u (0x%x)", __FUNCTION__, aSsrc, aSsrc);
+
+  if (mRecvStreamConfig.rtp.remote_ssrc != aSsrc) {
+    nsCOMPtr<nsIDirectTaskDispatcher> dtd = do_QueryInterface(mCallThread);
+    MOZ_ALWAYS_SUCCEEDS(dtd->DispatchDirectTask(NewRunnableMethod(
+        "WebrtcVideoConduit::NotifyUnsetCurrentRemoteSSRC", this,
+        &WebrtcVideoConduit::NotifyUnsetCurrentRemoteSSRC)));
+  }
+
+  mRecvSSRC = mRecvStreamConfig.rtp.remote_ssrc = aSsrc;
+  mRecvStreamConfig.rtp.rtx_ssrc = aRtxSsrc;
+}
+
+void WebrtcVideoConduit::SetRemoteSSRCAndRestartAsNeeded(uint32_t aSsrc,
+                                                         uint32_t aRtxSsrc) {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+
+  if (mRecvStreamConfig.rtp.remote_ssrc == aSsrc &&
+      mRecvStreamConfig.rtp.rtx_ssrc == aRtxSsrc) {
+    return;
+  }
+
+  SetRemoteSSRCConfig(aSsrc, aRtxSsrc);
+
+  const bool wasReceiving = mEngineReceiving;
+  const bool hadRecvStream = mRecvStream;
+
+  StopReceiving();
+
+  if (hadRecvStream) {
+    MutexAutoLock lock(mMutex);
+    DeleteRecvStream();
+    CreateRecvStream();
+  }
+
+  if (wasReceiving) {
+    StartReceiving();
+  }
+}
+
+void WebrtcVideoConduit::EnsureRemoteSSRC() {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  mMutex.AssertCurrentThreadOwns();
+
+  const auto& ssrcs = mSendStreamConfig.rtp.ssrcs;
+  if (mRecvStreamConfig.rtp.remote_ssrc != 0 &&
+      std::find(ssrcs.begin(), ssrcs.end(),
+                mRecvStreamConfig.rtp.remote_ssrc) == ssrcs.end()) {
+    return;
+  }
+
+  uint32_t ssrc;
+  do {
+    ssrc = GenerateRandomSSRC();
+  } while (
+      NS_WARN_IF(std::find(ssrcs.begin(), ssrcs.end(), ssrc) != ssrcs.end()));
+  CSFLogDebug(LOGTAG, "VideoConduit %p: Generated remote SSRC %u", this, ssrc);
+  SetRemoteSSRCConfig(ssrc, 0);
+}
+
+void WebrtcVideoConduit::EnsureLocalSSRC() {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  mMutex.AssertCurrentThreadOwns();
+
+  auto& ssrcs = mSendStreamConfig.rtp.ssrcs;
+  if (ssrcs.empty()) {
+    ssrcs.push_back(GenerateRandomSSRC());
+  }
+
+  // Reverse-iterating here so that the first dupe in `ssrcs` always wins.
+  for (auto& ssrc : Reversed(ssrcs)) {
+    if (ssrc != 0 && ssrc != mRecvStreamConfig.rtp.remote_ssrc &&
+        std::count(ssrcs.begin(), ssrcs.end(), ssrc) == 1) {
+      continue;
+    }
+    do {
+      ssrc = GenerateRandomSSRC();
+    } while (NS_WARN_IF(ssrc == mRecvStreamConfig.rtp.remote_ssrc) ||
+             NS_WARN_IF(std::count(ssrcs.begin(), ssrcs.end(), ssrc) > 1));
+    CSFLogDebug(LOGTAG, "%s (%p): Generated local SSRC %u", __FUNCTION__, this,
+                ssrc);
+  }
+  mRecvStreamConfig.rtp.local_ssrc = ssrcs[0];
+}
+
+void WebrtcVideoConduit::UnsetRemoteSSRC(uint32_t aSsrc) {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  mMutex.AssertNotCurrentThreadOwns();
+
+  if (mRecvStreamConfig.rtp.remote_ssrc != aSsrc &&
+      mRecvStreamConfig.rtp.rtx_ssrc != aSsrc) {
+    return;
+  }
+
+  const auto& ssrcs = mSendStreamConfig.rtp.ssrcs;
+  uint32_t our_ssrc = 0;
+  do {
+    our_ssrc = GenerateRandomSSRC();
+  } while (NS_WARN_IF(our_ssrc == aSsrc) ||
+           NS_WARN_IF(std::find(ssrcs.begin(), ssrcs.end(), our_ssrc) !=
+                      ssrcs.end()));
+
+  CSFLogDebug(LOGTAG, "%s (%p): Generated remote SSRC %u", __FUNCTION__, this,
+              our_ssrc);
+
+  // There is a (tiny) chance that this new random ssrc will collide with some
+  // other conduit's remote ssrc, in which case that conduit will choose a new
+  // one.
+  SetRemoteSSRCAndRestartAsNeeded(our_ssrc, 0);
+}
+
+/*static*/
+unsigned WebrtcVideoConduit::ToLibwebrtcMaxFramerate(
+    const Maybe<double>& aMaxFramerate) {
+  Maybe<unsigned> negotiatedMaxFps;
+  if (aMaxFramerate.isSome()) {
+    // libwebrtc does not handle non-integer max framerate.
+    unsigned integerMaxFps = static_cast<unsigned>(std::round(*aMaxFramerate));
+    // libwebrtc crashes with a max framerate of 0, even though the
+    // spec says this is valid. For now, we treat this as no limit.
+    if (integerMaxFps) {
+      negotiatedMaxFps = Some(integerMaxFps);
+    }
+  }
+  // We do not use DEFAULT_VIDEO_MAX_FRAMERATE here; that is used at the very
+  // end in VideoStreamFactory, once codec-wide and per-encoding limits are
+  // known.
+  return negotiatedMaxFps.refOr(std::numeric_limits<unsigned int>::max());
+}
+
+Maybe<Ssrc> WebrtcVideoConduit::GetRemoteSSRC() const {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  // libwebrtc uses 0 to mean a lack of SSRC. That is not to spec.
+  return mRecvStreamConfig.rtp.remote_ssrc == 0
+             ? Nothing()
+             : Some(mRecvStreamConfig.rtp.remote_ssrc);
+}
+
+Maybe<webrtc::VideoReceiveStreamInterface::Stats>
+WebrtcVideoConduit::GetReceiverStats() const {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  if (!mRecvStream) {
+    return Nothing();
+  }
+  return Some(mRecvStream->GetStats());
+}
+
+Maybe<webrtc::VideoSendStream::Stats> WebrtcVideoConduit::GetSenderStats()
+    const {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  if (!mSendStream) {
+    return Nothing();
+  }
+  return Some(mSendStream->GetStats());
+}
+
+Maybe<webrtc::Call::Stats> WebrtcVideoConduit::GetCallStats() const {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  if (!mCall->Call()) {
+    return Nothing();
+  }
+  return Some(mCall->Call()->GetStats());
+}
+
+MediaConduitErrorCode WebrtcVideoConduit::Init() {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  CSFLogDebug(LOGTAG, "%s this=%p", __FUNCTION__, this);
+
+#ifdef MOZ_WIDGET_ANDROID
+  if (mozilla::camera::VideoEngine::SetAndroidObjects() != 0) {
+    CSFLogError(LOGTAG, "%s: could not set Android objects", __FUNCTION__);
+    return kMediaConduitSessionNotInited;
+  }
+#endif  // MOZ_WIDGET_ANDROID
+
+  mSendPluginCreated = mEncoderFactory->CreatedGmpPluginEvent().Connect(
+      GetMainThreadSerialEventTarget(),
+      [self = detail::RawPtr(this)](uint64_t aPluginID) {
+        self.get()->mSendCodecPluginIDs.AppendElement(aPluginID);
+      });
+  mSendPluginReleased = mEncoderFactory->ReleasedGmpPluginEvent().Connect(
+      GetMainThreadSerialEventTarget(),
+      [self = detail::RawPtr(this)](uint64_t aPluginID) {
+        self.get()->mSendCodecPluginIDs.RemoveElement(aPluginID);
+      });
+  mRecvPluginCreated = mDecoderFactory->CreatedGmpPluginEvent().Connect(
+      GetMainThreadSerialEventTarget(),
+      [self = detail::RawPtr(this)](uint64_t aPluginID) {
+        self.get()->mRecvCodecPluginIDs.AppendElement(aPluginID);
+      });
+  mRecvPluginReleased = mDecoderFactory->ReleasedGmpPluginEvent().Connect(
+      GetMainThreadSerialEventTarget(),
+      [self = detail::RawPtr(this)](uint64_t aPluginID) {
+        self.get()->mRecvCodecPluginIDs.RemoveElement(aPluginID);
+      });
+
+  MOZ_ALWAYS_SUCCEEDS(mCallThread->Dispatch(NS_NewRunnableFunction(
+      __func__, [this, self = RefPtr<WebrtcVideoConduit>(this)] {
+        mCall->RegisterConduit(this);
+      })));
+
+  CSFLogDebug(LOGTAG, "%s Initialization Done", __FUNCTION__);
+  return kMediaConduitNoError;
+}
+
+RefPtr<GenericPromise> WebrtcVideoConduit::Shutdown() {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  mSendPluginCreated.DisconnectIfExists();
+  mSendPluginReleased.DisconnectIfExists();
+  mRecvPluginCreated.DisconnectIfExists();
+  mRecvPluginReleased.DisconnectIfExists();
+
+  return InvokeAsync(
+      mCallThread, __func__, [this, self = RefPtr<WebrtcVideoConduit>(this)] {
+        using namespace Telemetry;
+        if (mSendBitrate.NumDataValues() > 0) {
+          Accumulate(WEBRTC_VIDEO_ENCODER_BITRATE_AVG_PER_CALL_KBPS,
+                     static_cast<unsigned>(mSendBitrate.Mean() / 1000));
+          Accumulate(
+              WEBRTC_VIDEO_ENCODER_BITRATE_STD_DEV_PER_CALL_KBPS,
+              static_cast<unsigned>(mSendBitrate.StandardDeviation() / 1000));
+          mSendBitrate.Clear();
+        }
+        if (mSendFramerate.NumDataValues() > 0) {
+          Accumulate(WEBRTC_VIDEO_ENCODER_FRAMERATE_AVG_PER_CALL,
+                     static_cast<unsigned>(mSendFramerate.Mean()));
+          Accumulate(
+              WEBRTC_VIDEO_ENCODER_FRAMERATE_10X_STD_DEV_PER_CALL,
+              static_cast<unsigned>(mSendFramerate.StandardDeviation() * 10));
+          mSendFramerate.Clear();
+        }
+
+        if (mRecvBitrate.NumDataValues() > 0) {
+          Accumulate(WEBRTC_VIDEO_DECODER_BITRATE_AVG_PER_CALL_KBPS,
+                     static_cast<unsigned>(mRecvBitrate.Mean() / 1000));
+          Accumulate(
+              WEBRTC_VIDEO_DECODER_BITRATE_STD_DEV_PER_CALL_KBPS,
+              static_cast<unsigned>(mRecvBitrate.StandardDeviation() / 1000));
+          mRecvBitrate.Clear();
+        }
+        if (mRecvFramerate.NumDataValues() > 0) {
+          Accumulate(WEBRTC_VIDEO_DECODER_FRAMERATE_AVG_PER_CALL,
+                     static_cast<unsigned>(mRecvFramerate.Mean()));
+          Accumulate(
+              WEBRTC_VIDEO_DECODER_FRAMERATE_10X_STD_DEV_PER_CALL,
+              static_cast<unsigned>(mRecvFramerate.StandardDeviation() * 10));
+          mRecvFramerate.Clear();
+        }
+
+        mControl.mReceiving.DisconnectIfConnected();
+        mControl.mTransmitting.DisconnectIfConnected();
+        mControl.mLocalSsrcs.DisconnectIfConnected();
+        mControl.mLocalRtxSsrcs.DisconnectIfConnected();
+        mControl.mLocalCname.DisconnectIfConnected();
+        mControl.mMid.DisconnectIfConnected();
+        mControl.mRemoteSsrc.DisconnectIfConnected();
+        mControl.mRemoteRtxSsrc.DisconnectIfConnected();
+        mControl.mSyncGroup.DisconnectIfConnected();
+        mControl.mLocalRecvRtpExtensions.DisconnectIfConnected();
+        mControl.mLocalSendRtpExtensions.DisconnectIfConnected();
+        mControl.mSendCodec.DisconnectIfConnected();
+        mControl.mSendRtpRtcpConfig.DisconnectIfConnected();
+        mControl.mRecvCodecs.DisconnectIfConnected();
+        mControl.mRecvRtpRtcpConfig.DisconnectIfConnected();
+        mControl.mCodecMode.DisconnectIfConnected();
+        mWatchManager.Shutdown();
+
+        mCall->UnregisterConduit(this);
+        mDecoderFactory->DisconnectAll();
+        mEncoderFactory->DisconnectAll();
+        {
+          MutexAutoLock lock(mMutex);
+          DeleteSendStream();
+          DeleteRecvStream();
+        }
+
+        return GenericPromise::CreateAndResolve(true, __func__);
+      });
+}
+
+webrtc::VideoCodecMode WebrtcVideoConduit::CodecMode() const {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  return mControl.mCodecMode;
+}
+
+MediaConduitErrorCode WebrtcVideoConduit::AttachRenderer(
+    RefPtr<mozilla::VideoRenderer> aVideoRenderer) {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  CSFLogDebug(LOGTAG, "%s", __FUNCTION__);
+
+  // null renderer
+  if (!aVideoRenderer) {
+    CSFLogError(LOGTAG, "%s NULL Renderer", __FUNCTION__);
+    MOZ_ASSERT(false);
+    return kMediaConduitInvalidRenderer;
+  }
+
+  // This function is called only from main, so we only need to protect against
+  // modifying mRenderer while any webrtc.org code is trying to use it.
+  {
+    ReentrantMonitorAutoEnter enter(mRendererMonitor);
+    mRenderer = aVideoRenderer;
+    // Make sure the renderer knows the resolution
+    mRenderer->FrameSizeChange(mReceivingWidth, mReceivingHeight);
+  }
+
+  return kMediaConduitNoError;
+}
+
+void WebrtcVideoConduit::DetachRenderer() {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  ReentrantMonitorAutoEnter enter(mRendererMonitor);
+  if (mRenderer) {
+    mRenderer = nullptr;
+  }
+}
+
+rtc::RefCountedObject<mozilla::VideoStreamFactory>*
+WebrtcVideoConduit::CreateVideoStreamFactory() {
+  auto videoStreamFactory = mVideoStreamFactory.Lock();
+  *videoStreamFactory = new rtc::RefCountedObject<VideoStreamFactory>(
+      *mCurSendCodecConfig, mControl.mCodecMode, mMinBitrate, mStartBitrate,
+      mPrefMaxBitrate, mNegotiatedMaxBitrate, mVideoBroadcaster.wants(),
+      mLockScaling);
+  return videoStreamFactory->get();
+}
+
+void WebrtcVideoConduit::AddOrUpdateSink(
+    rtc::VideoSinkInterface<webrtc::VideoFrame>* sink,
+    const rtc::VideoSinkWants& wants) {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  if (!mRegisteredSinks.Contains(sink)) {
+    mRegisteredSinks.AppendElement(sink);
+  }
+  auto oldWants = mVideoBroadcaster.wants();
+  mVideoBroadcaster.AddOrUpdateSink(sink, wants);
+  if (oldWants != mVideoBroadcaster.wants()) {
+    mEncoderConfig.video_stream_factory = CreateVideoStreamFactory();
+    mSendStream->ReconfigureVideoEncoder(mEncoderConfig.Copy());
+  }
+}
+
+void WebrtcVideoConduit::RemoveSink(
+    rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+
+  mRegisteredSinks.RemoveElement(sink);
+  auto oldWants = mVideoBroadcaster.wants();
+  mVideoBroadcaster.RemoveSink(sink);
+  if (oldWants != mVideoBroadcaster.wants()) {
+    mEncoderConfig.video_stream_factory = CreateVideoStreamFactory();
+    mSendStream->ReconfigureVideoEncoder(mEncoderConfig.Copy());
+  }
+}
+
+MediaConduitErrorCode WebrtcVideoConduit::SendVideoFrame(
+    webrtc::VideoFrame aFrame) {
+  // XXX Google uses a "timestamp_aligner" to translate timestamps from the
+  // camera via TranslateTimestamp(); we should look at doing the same.  This
+  // avoids sampling error when capturing frames, but google had to deal with
+  // some broken cameras, include Logitech c920's IIRC.
+
+  {
+    MutexAutoLock lock(mMutex);
+    if (mSendStreamConfig.rtp.ssrcs.empty()) {
+      CSFLogVerbose(LOGTAG, "WebrtcVideoConduit %p %s No SSRC set", this,
+                    __FUNCTION__);
+      return kMediaConduitNoError;
+    }
+    if (!mCurSendCodecConfig) {
+      CSFLogVerbose(LOGTAG, "WebrtcVideoConduit %p %s No send codec set", this,
+                    __FUNCTION__);
+      return kMediaConduitNoError;
+    }
+
+    // Workaround for bug in libwebrtc where all encodings are transmitted
+    // if they are all inactive.
+    bool anyActive = false;
+    for (const auto& encoding : mCurSendCodecConfig->mEncodings) {
+      if (encoding.active) {
+        anyActive = true;
+        break;
+      }
+    }
+    if (!anyActive) {
+      CSFLogVerbose(LOGTAG, "WebrtcVideoConduit %p %s No active encodings",
+                    this, __FUNCTION__);
+      return kMediaConduitNoError;
+    }
+
+    CSFLogVerbose(LOGTAG, "WebrtcVideoConduit %p %s (send SSRC %u (0x%x))",
+                  this, __FUNCTION__, mSendStreamConfig.rtp.ssrcs.front(),
+                  mSendStreamConfig.rtp.ssrcs.front());
+
+    if (aFrame.width() != mLastWidth || aFrame.height() != mLastHeight) {
+      // See if we need to recalculate what we're sending.
+      CSFLogVerbose(LOGTAG, "%s: call SelectSendResolution with %ux%u",
+                    __FUNCTION__, aFrame.width(), aFrame.height());
+      MOZ_ASSERT(aFrame.width() != 0 && aFrame.height() != 0);
+      // Note coverity will flag this since it thinks they can be 0
+      MOZ_ASSERT(mCurSendCodecConfig);
+
+      mLastWidth = aFrame.width();
+      mLastHeight = aFrame.height();
+    }
+
+    // adapt input video to wants of sink
+    if (!mVideoBroadcaster.frame_wanted()) {
+      return kMediaConduitNoError;
+    }
+
+    // Check if we need to drop this frame to meet a requested FPS
+    auto videoStreamFactory = mVideoStreamFactory.Lock();
+    auto& videoStreamFactoryRef = videoStreamFactory.ref();
+    if (videoStreamFactoryRef->ShouldDropFrame(aFrame)) {
+      return kMediaConduitNoError;
+    }
+  }
+
+  // If we have zero width or height, drop the frame here. Attempting to send
+  // it will cause all sorts of problems in the webrtc.org code.
+  if (aFrame.width() == 0 || aFrame.height() == 0) {
+    return kMediaConduitNoError;
+  }
+
+  rtc::scoped_refptr<webrtc::VideoFrameBuffer> buffer =
+      aFrame.video_frame_buffer();
+
+  MOZ_ASSERT(!aFrame.color_space(), "Unexpected use of color space");
+  MOZ_ASSERT(!aFrame.has_update_rect(), "Unexpected use of update rect");
+
+#ifdef MOZ_REAL_TIME_TRACING
+  if (profiler_is_active()) {
+    MutexAutoLock lock(mMutex);
+    nsAutoCStringN<256> ssrcsCommaSeparated;
+    bool first = true;
+    for (auto ssrc : mSendStreamConfig.rtp.ssrcs) {
+      if (!first) {
+        ssrcsCommaSeparated.AppendASCII(", ");
+      } else {
+        first = false;
+      }
+      ssrcsCommaSeparated.AppendInt(ssrc);
+    }
+    // The first frame has a delta of zero.
+    uint64_t timestampDelta =
+        mLastTimestampSendUs.isSome()
+            ? aFrame.timestamp_us() - mLastTimestampSendUs.value()
+            : 0;
+    mLastTimestampSendUs = Some(aFrame.timestamp_us());
+    TRACE_COMMENT("VideoConduit::SendVideoFrame", "t-delta=%.1fms, ssrcs=%s",
+                  timestampDelta / 1000.f, ssrcsCommaSeparated.get());
+  }
+#endif
+
+  mVideoBroadcaster.OnFrame(aFrame);
+
+  return kMediaConduitNoError;
+}
+
+// Transport Layer Callbacks
+
+void WebrtcVideoConduit::DeliverPacket(rtc::CopyOnWriteBuffer packet,
+                                       PacketType type) {
+  // Currently unused.
+  MOZ_ASSERT(false);
+}
+
+void WebrtcVideoConduit::OnRtpReceived(webrtc::RtpPacketReceived&& aPacket,
+                                       webrtc::RTPHeader&& aHeader) {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+
+  mRemoteSendSSRC = aHeader.ssrc;
+
+  if (mAllowSsrcChange || mRecvStreamConfig.rtp.remote_ssrc == 0) {
+    bool switchRequired = mRecvStreamConfig.rtp.remote_ssrc != aHeader.ssrc;
+    if (switchRequired) {
+      // Handle the unknown ssrc (and ssrc-not-signaled case).
+
+      // We need to check that the newly received ssrc is not already
+      // associated with ulpfec or rtx. This is how webrtc.org handles
+      // things, see https://codereview.webrtc.org/1226093002.
+      const webrtc::VideoReceiveStreamInterface::Config::Rtp& rtp =
+          mRecvStreamConfig.rtp;
+      switchRequired =
+          rtp.rtx_associated_payload_types.find(aHeader.payloadType) ==
+              rtp.rtx_associated_payload_types.end() &&
+          rtp.ulpfec_payload_type != aHeader.payloadType;
+    }
+
+    if (switchRequired) {
+      CSFLogInfo(LOGTAG, "VideoConduit %p: Switching remote SSRC from %u to %u",
+                 this, mRecvStreamConfig.rtp.remote_ssrc, aHeader.ssrc);
+      SetRemoteSSRCAndRestartAsNeeded(aHeader.ssrc, 0);
+    }
+  }
+
+  CSFLogVerbose(LOGTAG, "%s: seq# %u, Len %zu, SSRC %u (0x%x) ", __FUNCTION__,
+                aPacket.SequenceNumber(), aPacket.size(), aPacket.Ssrc(),
+                aPacket.Ssrc());
+
+  mRtpPacketEvent.Notify();
+  if (mCall->Call()) {
+    mCall->Call()->Receiver()->DeliverRtpPacket(
+        webrtc::MediaType::VIDEO, std::move(aPacket),
+        [self = RefPtr<WebrtcVideoConduit>(this)](
+            const webrtc::RtpPacketReceived& packet) {
+          CSFLogVerbose(
+              LOGTAG,
+              "VideoConduit %p: failed demuxing packet, ssrc: %u seq: %u",
+              self.get(), packet.Ssrc(), packet.SequenceNumber());
+          return false;
+        });
+  }
+}
+
+void WebrtcVideoConduit::OnRtcpReceived(MediaPacket&& aPacket) {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+
+  CSFLogVerbose(LOGTAG, "VideoConduit %p: Received RTCP Packet, len %zu ", this,
+                aPacket.len());
+
+  if (mCall->Call()) {
+    mCall->Call()->Receiver()->DeliverRtcpPacket(
+        rtc::CopyOnWriteBuffer(aPacket.data(), aPacket.len()));
+  }
+}
+
+Maybe<uint16_t> WebrtcVideoConduit::RtpSendBaseSeqFor(uint32_t aSsrc) const {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  auto it = mRtpSendBaseSeqs.find(aSsrc);
+  if (it == mRtpSendBaseSeqs.end()) {
+    return Nothing();
+  }
+  return Some(it->second);
+}
+
+const dom::RTCStatsTimestampMaker& WebrtcVideoConduit::GetTimestampMaker()
+    const {
+  return mCall->GetTimestampMaker();
+}
+
+void WebrtcVideoConduit::StopTransmitting() {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  mMutex.AssertNotCurrentThreadOwns();
+
+  if (!mEngineTransmitting) {
+    return;
+  }
+
+  if (mSendStream) {
+    CSFLogDebug(LOGTAG, "%s Stopping send stream", __FUNCTION__);
+    mSendStream->Stop();
+  }
+
+  mEngineTransmitting = false;
+}
+
+void WebrtcVideoConduit::StartTransmitting() {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  MOZ_ASSERT(mSendStream);
+  mMutex.AssertNotCurrentThreadOwns();
+
+  if (mEngineTransmitting) {
+    return;
+  }
+
+  CSFLogDebug(LOGTAG, "%s Starting send stream", __FUNCTION__);
+
+  mSendStream->Start();
+  // XXX File a bug to consider hooking this up to the state of mtransport
+  mCall->Call()->SignalChannelNetworkState(webrtc::MediaType::VIDEO,
+                                           webrtc::kNetworkUp);
+  mEngineTransmitting = true;
+}
+
+void WebrtcVideoConduit::StopReceiving() {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  mMutex.AssertNotCurrentThreadOwns();
+
+  // Are we receiving already? If so, stop receiving and playout
+  // since we can't apply new recv codec when the engine is playing.
+  if (!mEngineReceiving) {
+    return;
+  }
+
+  if (mRecvStream) {
+    CSFLogDebug(LOGTAG, "%s Stopping receive stream", __FUNCTION__);
+    mRecvStream->Stop();
+  }
+
+  mEngineReceiving = false;
+}
+
+void WebrtcVideoConduit::StartReceiving() {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  MOZ_ASSERT(mRecvStream);
+  mMutex.AssertNotCurrentThreadOwns();
+
+  if (mEngineReceiving) {
+    return;
+  }
+
+  CSFLogDebug(LOGTAG, "%s Starting receive stream (SSRC %u (0x%x))",
+              __FUNCTION__, mRecvStreamConfig.rtp.remote_ssrc,
+              mRecvStreamConfig.rtp.remote_ssrc);
+  // Start Receiving on the video engine
+  mRecvStream->Start();
+
+  // XXX File a bug to consider hooking this up to the state of mtransport
+  mCall->Call()->SignalChannelNetworkState(webrtc::MediaType::VIDEO,
+                                           webrtc::kNetworkUp);
+  mEngineReceiving = true;
+}
+
+bool WebrtcVideoConduit::SendRtp(const uint8_t* aData, size_t aLength,
+                                 const webrtc::PacketOptions& aOptions) {
+  MOZ_ASSERT(aLength >= 12);
+  const uint16_t seqno = ntohs(*((uint16_t*)&aData[2]));
+  const uint32_t ssrc = ntohl(*((uint32_t*)&aData[8]));
+
+  CSFLogVerbose(
+      LOGTAG,
+      "VideoConduit %p: Sending RTP Packet seq# %u, len %zu, SSRC %u (0x%x)",
+      this, seqno, aLength, ssrc, ssrc);
+
+  if (!mTransportActive) {
+    CSFLogError(LOGTAG, "VideoConduit %p: RTP Packet Send Failed", this);
+    return false;
+  }
+
+  MediaPacket packet;
+  packet.Copy(aData, aLength, aLength + SRTP_MAX_EXPANSION);
+  packet.SetType(MediaPacket::RTP);
+  mSenderRtpSendEvent.Notify(std::move(packet));
+
+  // Parse the sequence number of the first rtp packet as base_seq.
+  const auto inserted = mRtpSendBaseSeqs_n.insert({ssrc, seqno}).second;
+
+  if (inserted || aOptions.packet_id >= 0) {
+    int64_t now_ms = PR_Now() / 1000;
+    MOZ_ALWAYS_SUCCEEDS(mCallThread->Dispatch(NS_NewRunnableFunction(
+        __func__, [this, self = RefPtr<WebrtcVideoConduit>(this),
+                   packet_id = aOptions.packet_id, now_ms, ssrc, seqno] {
+          mRtpSendBaseSeqs.insert({ssrc, seqno});
+          if (packet_id >= 0) {
+            if (mCall->Call()) {
+              // TODO: This notification should ideally happen after the
+              // transport layer has sent the packet on the wire.
+              mCall->Call()->OnSentPacket({packet_id, now_ms});
+            }
+          }
+        })));
+  }
+  return true;
+}
+
+bool WebrtcVideoConduit::SendSenderRtcp(const uint8_t* aData, size_t aLength) {
+  CSFLogVerbose(
+      LOGTAG,
+      "VideoConduit %p: Sending RTCP SR Packet, len %zu, SSRC %u (0x%x)", this,
+      aLength, (uint32_t)ntohl(*((uint32_t*)&aData[4])),
+      (uint32_t)ntohl(*((uint32_t*)&aData[4])));
+
+  if (!mTransportActive) {
+    CSFLogError(LOGTAG, "VideoConduit %p: RTCP SR Packet Send Failed", this);
+    return false;
+  }
+
+  MediaPacket packet;
+  packet.Copy(aData, aLength, aLength + SRTP_MAX_EXPANSION);
+  packet.SetType(MediaPacket::RTCP);
+  mSenderRtcpSendEvent.Notify(std::move(packet));
+  return true;
+}
+
+bool WebrtcVideoConduit::SendReceiverRtcp(const uint8_t* aData,
+                                          size_t aLength) {
+  CSFLogVerbose(
+      LOGTAG,
+      "VideoConduit %p: Sending RTCP RR Packet, len %zu, SSRC %u (0x%x)", this,
+      aLength, (uint32_t)ntohl(*((uint32_t*)&aData[4])),
+      (uint32_t)ntohl(*((uint32_t*)&aData[4])));
+
+  if (!mTransportActive) {
+    CSFLogError(LOGTAG, "VideoConduit %p: RTCP RR Packet Send Failed", this);
+    return false;
+  }
+
+  MediaPacket packet;
+  packet.Copy(aData, aLength, aLength + SRTP_MAX_EXPANSION);
+  packet.SetType(MediaPacket::RTCP);
+  mReceiverRtcpSendEvent.Notify(std::move(packet));
+  return true;
+}
+
+void WebrtcVideoConduit::OnFrame(const webrtc::VideoFrame& video_frame) {
+  const uint32_t localRecvSsrc = mRecvSSRC;
+  const uint32_t remoteSendSsrc = mRemoteSendSSRC;
+
+  CSFLogVerbose(
+      LOGTAG,
+      "VideoConduit %p: Rendering frame, Remote SSRC %u (0x%x), size %ux%u",
+      this, static_cast<uint32_t>(remoteSendSsrc),
+      static_cast<uint32_t>(remoteSendSsrc), video_frame.width(),
+      video_frame.height());
+  ReentrantMonitorAutoEnter enter(mRendererMonitor);
+
+  if (!mRenderer) {
+    CSFLogError(LOGTAG, "VideoConduit %p: Cannot render frame, no renderer",
+                this);
+    return;
+  }
+
+  bool needsNewHistoryElement = mReceivedFrameHistory.mEntries.IsEmpty();
+
+  if (mReceivingWidth != video_frame.width() ||
+      mReceivingHeight != video_frame.height()) {
+    mReceivingWidth = video_frame.width();
+    mReceivingHeight = video_frame.height();
+    mRenderer->FrameSizeChange(mReceivingWidth, mReceivingHeight);
+    needsNewHistoryElement = true;
+  }
+
+  if (!needsNewHistoryElement) {
+    auto& currentEntry = mReceivedFrameHistory.mEntries.LastElement();
+    needsNewHistoryElement =
+        currentEntry.mRotationAngle !=
+            static_cast<unsigned long>(video_frame.rotation()) ||
+        currentEntry.mLocalSsrc != localRecvSsrc ||
+        currentEntry.mRemoteSsrc != remoteSendSsrc;
+  }
+
+  // Record frame history
+  const auto historyNow = mCall->GetTimestampMaker().GetNow().ToDom();
+  if (needsNewHistoryElement) {
+    dom::RTCVideoFrameHistoryEntryInternal frameHistoryElement;
+    frameHistoryElement.mConsecutiveFrames = 0;
+    frameHistoryElement.mWidth = video_frame.width();
+    frameHistoryElement.mHeight = video_frame.height();
+    frameHistoryElement.mRotationAngle =
+        static_cast<unsigned long>(video_frame.rotation());
+    frameHistoryElement.mFirstFrameTimestamp = historyNow;
+    frameHistoryElement.mLocalSsrc = localRecvSsrc;
+    frameHistoryElement.mRemoteSsrc = remoteSendSsrc;
+    if (!mReceivedFrameHistory.mEntries.AppendElement(frameHistoryElement,
+                                                      fallible)) {
+      mozalloc_handle_oom(0);
+    }
+  }
+  auto& currentEntry = mReceivedFrameHistory.mEntries.LastElement();
+
+  currentEntry.mConsecutiveFrames++;
+  currentEntry.mLastFrameTimestamp = historyNow;
+  // Attempt to retrieve an timestamp encoded in the image pixels if enabled.
+  if (mVideoLatencyTestEnable && mReceivingWidth && mReceivingHeight) {
+    uint64_t now = PR_Now();
+    uint64_t timestamp = 0;
+    uint8_t* data = const_cast<uint8_t*>(
+        video_frame.video_frame_buffer()->GetI420()->DataY());
+    bool ok = YuvStamper::Decode(
+        mReceivingWidth, mReceivingHeight, mReceivingWidth, data,
+        reinterpret_cast<unsigned char*>(&timestamp), sizeof(timestamp), 0, 0);
+    if (ok) {
+      VideoLatencyUpdate(now - timestamp);
+    }
+  }
+#ifdef MOZ_REAL_TIME_TRACING
+  if (profiler_is_active()) {
+    MutexAutoLock lock(mMutex);
+    // The first frame has a delta of zero.
+    uint32_t rtpTimestamp = video_frame.timestamp();
+    uint32_t timestampDelta =
+        mLastRTPTimestampReceive.isSome()
+            ? rtpTimestamp - mLastRTPTimestampReceive.value()
+            : 0;
+    mLastRTPTimestampReceive = Some(rtpTimestamp);
+    TRACE_COMMENT("VideoConduit::OnFrame", "t-delta=%.1fms, ssrc=%u",
+                  timestampDelta * 1000.f / webrtc::kVideoPayloadTypeFrequency,
+                  localRecvSsrc);
+  }
+#endif
+
+  mRenderer->RenderVideoFrame(*video_frame.video_frame_buffer(),
+                              video_frame.timestamp(),
+                              video_frame.render_time_ms());
+}
+
+bool WebrtcVideoConduit::AddFrameHistory(
+    dom::Sequence<dom::RTCVideoFrameHistoryInternal>* outHistories) const {
+  ReentrantMonitorAutoEnter enter(mRendererMonitor);
+  if (!outHistories->AppendElement(mReceivedFrameHistory, fallible)) {
+    mozalloc_handle_oom(0);
+    return false;
+  }
+  return true;
+}
+
+void WebrtcVideoConduit::SetJitterBufferTarget(DOMHighResTimeStamp aTargetMs) {
+  MOZ_RELEASE_ASSERT(aTargetMs <= std::numeric_limits<uint16_t>::max());
+  MOZ_RELEASE_ASSERT(aTargetMs >= 0);
+
+  MOZ_ALWAYS_SUCCEEDS(mCallThread->Dispatch(NS_NewRunnableFunction(
+      __func__,
+      [this, self = RefPtr<WebrtcVideoConduit>(this), targetMs = aTargetMs] {
+        mJitterBufferTargetMs = static_cast<uint16_t>(targetMs);
+        if (mRecvStream) {
+          mRecvStream->SetBaseMinimumPlayoutDelayMs(targetMs);
+        }
+      })));
+}
+
+void WebrtcVideoConduit::DumpCodecDB() const {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+
+  for (const auto& entry : mControl.mConfiguredRecvCodecs) {
+    CSFLogDebug(LOGTAG, "Payload Name: %s", entry.mName.c_str());
+    CSFLogDebug(LOGTAG, "Payload Type: %d", entry.mType);
+    CSFLogDebug(LOGTAG, "Payload Max Frame Size: %d",
+                entry.mEncodingConstraints.maxFs);
+    if (entry.mEncodingConstraints.maxFps.isSome()) {
+      CSFLogDebug(LOGTAG, "Payload Max Frame Rate: %f",
+                  *entry.mEncodingConstraints.maxFps);
+    }
+  }
+}
+
+void WebrtcVideoConduit::VideoLatencyUpdate(uint64_t aNewSample) {
+  mRendererMonitor.AssertCurrentThreadIn();
+
+  mVideoLatencyAvg =
+      (sRoundingPadding * aNewSample + sAlphaNum * mVideoLatencyAvg) /
+      sAlphaDen;
+}
+
+uint64_t WebrtcVideoConduit::MozVideoLatencyAvg() {
+  mRendererMonitor.AssertCurrentThreadIn();
+
+  return mVideoLatencyAvg / sRoundingPadding;
+}
+
+void WebrtcVideoConduit::CollectTelemetryData() {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+
+  if (mEngineTransmitting) {
+    webrtc::VideoSendStream::Stats stats = mSendStream->GetStats();
+    mSendBitrate.Push(stats.media_bitrate_bps);
+    mSendFramerate.Push(stats.encode_frame_rate);
+  }
+  if (mEngineReceiving) {
+    webrtc::VideoReceiveStreamInterface::Stats stats = mRecvStream->GetStats();
+    mRecvBitrate.Push(stats.total_bitrate_bps);
+    mRecvFramerate.Push(stats.decode_frame_rate);
+  }
+}
+
+void WebrtcVideoConduit::OnRtcpBye() { mRtcpByeEvent.Notify(); }
+
+void WebrtcVideoConduit::OnRtcpTimeout() { mRtcpTimeoutEvent.Notify(); }
+
+void WebrtcVideoConduit::SetTransportActive(bool aActive) {
+  MOZ_ASSERT(mStsThread->IsOnCurrentThread());
+  if (mTransportActive == aActive) {
+    return;
+  }
+
+  // If false, This stops us from sending
+  mTransportActive = aActive;
+
+  // We queue this because there might be notifications to these listeners
+  // pending, and we don't want to drop them by letting this jump ahead of
+  // those notifications. We move the listeners into the lambda in case the
+  // transport comes back up before we disconnect them. (The Connect calls
+  // happen in MediaPipeline)
+  // We retain a strong reference to ourself, because the listeners are holding
+  // a non-refcounted reference to us, and moving them into the lambda could
+  // conceivably allow them to outlive us.
+  if (!aActive) {
+    MOZ_ALWAYS_SUCCEEDS(mCallThread->Dispatch(NS_NewRunnableFunction(
+        __func__,
+        [self = RefPtr<WebrtcVideoConduit>(this),
+         recvRtpListener = std::move(mReceiverRtpEventListener),
+         recvRtcpListener = std::move(mReceiverRtcpEventListener),
+         sendRtcpListener = std::move(mSenderRtcpEventListener)]() mutable {
+          recvRtpListener.DisconnectIfExists();
+          recvRtcpListener.DisconnectIfExists();
+          sendRtcpListener.DisconnectIfExists();
+        })));
+  }
+}
+
+std::vector<webrtc::RtpSource> WebrtcVideoConduit::GetUpstreamRtpSources()
+    const {
+  MOZ_ASSERT(NS_IsMainThread());
+  std::vector<webrtc::RtpSource> sources;
+  {
+    MutexAutoLock lock(mMutex);
+    if (mRecvStream) {
+      sources = mRecvStream->GetSources();
+    }
+  }
+  return sources;
+}
+
+bool WebrtcVideoConduit::HasCodecPluginID(uint64_t aPluginID) const {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  return mSendCodecPluginIDs.Contains(aPluginID) ||
+         mRecvCodecPluginIDs.Contains(aPluginID);
+}
+
+bool WebrtcVideoConduit::HasH264Hardware() {
+  nsCOMPtr<nsIGfxInfo> gfxInfo = do_GetService("@mozilla.org/gfx/info;1");
+  if (!gfxInfo) {
+    return false;
+  }
+  int32_t status;
+  nsCString discardFailureId;
+  return NS_SUCCEEDED(gfxInfo->GetFeatureStatus(
+             nsIGfxInfo::FEATURE_WEBRTC_HW_ACCELERATION_H264, discardFailureId,
+             &status)) &&
+         status == nsIGfxInfo::FEATURE_STATUS_OK;
+}
+
+Maybe<int> WebrtcVideoConduit::ActiveSendPayloadType() const {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+
+  if (!mSendStream) {
+    return Nothing();
+  }
+
+  if (mSendStreamConfig.rtp.payload_type == -1) {
+    return Nothing();
+  }
+
+  return Some(mSendStreamConfig.rtp.payload_type);
+}
+
+Maybe<int> WebrtcVideoConduit::ActiveRecvPayloadType() const {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+
+  auto stats = GetReceiverStats();
+  if (!stats) {
+    return Nothing();
+  }
+
+  if (stats->current_payload_type == -1) {
+    return Nothing();
+  }
+
+  return Some(stats->current_payload_type);
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/libwebrtcglue/VideoConduit.h b/dom/media/webrtc/libwebrtcglue/VideoConduit.h
new file mode 100644
index 0000000000..f46f1f962c
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/VideoConduit.h
@@ -0,0 +1,494 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef VIDEO_SESSION_H_
+#define VIDEO_SESSION_H_
+
+#include "mozilla/Atomics.h"
+#include "mozilla/Attributes.h"
+#include "mozilla/DataMutex.h"
+#include "mozilla/ReentrantMonitor.h"
+#include "mozilla/SharedThreadPool.h"
+#include "mozilla/StateMirroring.h"
+#include "mozilla/UniquePtr.h"
+#include "nsITimer.h"
+
+#include "MediaConduitInterface.h"
+#include "common/MediaEngineWrapper.h"
+#include "RtpRtcpConfig.h"
+#include "RunningStat.h"
+#include "transport/runnable_utils.h"
+
+// conflicts with #include of scoped_ptr.h
+#undef FF
+// Video Engine Includes
+#include "api/video_codecs/video_decoder.h"
+#include "api/video_codecs/video_encoder.h"
+#include "api/video_codecs/sdp_video_format.h"
+#include "call/call_basic_stats.h"
+#include "common_video/include/video_frame_buffer_pool.h"
+#include "media/base/video_broadcaster.h"
+#include <functional>
+#include <memory>
+/** This file hosts several structures identifying different aspects
+ * of a RTP Session.
+ */
+
+namespace mozilla {
+
+// Convert (SI) kilobits/sec to (SI) bits/sec
+#define KBPS(kbps) kbps * 1000
+
+const int kViEMinCodecBitrate_bps = KBPS(30);
+const unsigned int kVideoMtu = 1200;
+const int kQpMax = 56;
+
+template <typename T>
+T MinIgnoreZero(const T& a, const T& b) {
+  return std::min(a ? a : b, b ? b : a);
+}
+
+class VideoStreamFactory;
+class WebrtcAudioConduit;
+
+// Interface of external video encoder for WebRTC.
+class WebrtcVideoEncoder : public VideoEncoder, public webrtc::VideoEncoder {};
+
+// Interface of external video decoder for WebRTC.
+class WebrtcVideoDecoder : public VideoDecoder, public webrtc::VideoDecoder {};
+
+/**
+ * Concrete class for Video session. Hooks up
+ *  - media-source and target to external transport
+ */
+class WebrtcVideoConduit
+    : public VideoSessionConduit,
+      public webrtc::RtcpEventObserver,
+      public rtc::VideoSinkInterface<webrtc::VideoFrame>,
+      public rtc::VideoSourceInterface<webrtc::VideoFrame> {
+ public:
+  // Returns true when both encoder and decoder are HW accelerated.
+  static bool HasH264Hardware();
+
+  Maybe<int> ActiveSendPayloadType() const override;
+  Maybe<int> ActiveRecvPayloadType() const override;
+
+  /**
+   * Function to attach Renderer end-point for the Media-Video conduit.
+   * @param aRenderer : Reference to the concrete mozilla Video renderer
+   * implementation Note: Multiple invocations of this API shall remove an
+   * existing renderer and attaches the new to the Conduit.
+   */
+  MediaConduitErrorCode AttachRenderer(
+      RefPtr<mozilla::VideoRenderer> aVideoRenderer) override;
+  void DetachRenderer() override;
+
+  Maybe<uint16_t> RtpSendBaseSeqFor(uint32_t aSsrc) const override;
+
+  const dom::RTCStatsTimestampMaker& GetTimestampMaker() const override;
+
+  void StopTransmitting();
+  void StartTransmitting();
+  void StopReceiving();
+  void StartReceiving();
+
+  /**
+   * Function to deliver a capture video frame for encoding and transport.
+   * If the frame's timestamp is 0, it will be automatically generated.
+   *
+   * NOTE: ConfigureSendMediaCodec() must be called before this function can
+   *       be invoked. This ensures the inserted video-frames can be
+   *       transmitted by the conduit.
+   */
+  MediaConduitErrorCode SendVideoFrame(webrtc::VideoFrame aFrame) override;
+
+  bool SendRtp(const uint8_t* aData, size_t aLength,
+               const webrtc::PacketOptions& aOptions) override;
+  bool SendSenderRtcp(const uint8_t* aData, size_t aLength) override;
+  bool SendReceiverRtcp(const uint8_t* aData, size_t aLength) override;
+
+  /*
+   * webrtc:VideoSinkInterface implementation
+   * -------------------------------
+   */
+  void OnFrame(const webrtc::VideoFrame& frame) override;
+
+  /*
+   * webrtc:VideoSourceInterface implementation
+   * -------------------------------
+   */
+  void AddOrUpdateSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink,
+                       const rtc::VideoSinkWants& wants) override;
+  void RemoveSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
+
+  bool HasCodecPluginID(uint64_t aPluginID) const override;
+
+  RefPtr<GenericPromise> Shutdown() override;
+
+  bool Denoising() const { return mDenoising; }
+
+  uint8_t SpatialLayers() const { return mSpatialLayers; }
+
+  uint8_t TemporalLayers() const { return mTemporalLayers; }
+
+  webrtc::VideoCodecMode CodecMode() const;
+
+  WebrtcVideoConduit(RefPtr<WebrtcCallWrapper> aCall,
+                     nsCOMPtr<nsISerialEventTarget> aStsThread,
+                     Options aOptions, std::string aPCHandle,
+                     const TrackingId& aRecvTrackingId);
+  virtual ~WebrtcVideoConduit();
+
+  // Call thread.
+  void InitControl(VideoConduitControlInterface* aControl) override;
+
+  // Called when a parameter in mControl has changed. Call thread.
+  void OnControlConfigChange();
+
+  // Necessary Init steps on main thread.
+  MediaConduitErrorCode Init();
+
+  Ssrcs GetLocalSSRCs() const override;
+  Maybe<Ssrc> GetAssociatedLocalRtxSSRC(Ssrc aSsrc) const override;
+  Maybe<Ssrc> GetRemoteSSRC() const override;
+
+  // Call thread.
+  void UnsetRemoteSSRC(uint32_t aSsrc) override;
+
+  static unsigned ToLibwebrtcMaxFramerate(const Maybe<double>& aMaxFramerate);
+
+ private:
+  void NotifyUnsetCurrentRemoteSSRC();
+  void SetRemoteSSRCConfig(uint32_t aSsrc, uint32_t aRtxSsrc);
+  void SetRemoteSSRCAndRestartAsNeeded(uint32_t aSsrc, uint32_t aRtxSsrc);
+  rtc::RefCountedObject<mozilla::VideoStreamFactory>*
+  CreateVideoStreamFactory();
+
+ public:
+  // Creating a recv stream or a send stream requires a local ssrc to be
+  // configured. This method will generate one if needed.
+  void EnsureLocalSSRC();
+  // Creating a recv stream requires a remote ssrc to be configured. This method
+  // will generate one if needed.
+  void EnsureRemoteSSRC();
+
+  Maybe<webrtc::VideoReceiveStreamInterface::Stats> GetReceiverStats()
+      const override;
+  Maybe<webrtc::VideoSendStream::Stats> GetSenderStats() const override;
+  Maybe<webrtc::CallBasicStats> GetCallStats() const override;
+
+  bool AddFrameHistory(dom::Sequence<dom::RTCVideoFrameHistoryInternal>*
+                           outHistories) const override;
+
+  void SetJitterBufferTarget(DOMHighResTimeStamp aTargetMs) override;
+
+  uint64_t MozVideoLatencyAvg();
+
+  void DisableSsrcChanges() override {
+    MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+    mAllowSsrcChange = false;
+  }
+
+  void CollectTelemetryData() override;
+
+  void OnRtpReceived(webrtc::RtpPacketReceived&& aPacket,
+                     webrtc::RTPHeader&& aHeader);
+  void OnRtcpReceived(MediaPacket&& aPacket);
+
+  void OnRtcpBye() override;
+  void OnRtcpTimeout() override;
+
+  void SetTransportActive(bool aActive) override;
+
+  MediaEventSourceExc<MediaPacket>& SenderRtpSendEvent() override {
+    return mSenderRtpSendEvent;
+  }
+  MediaEventSourceExc<MediaPacket>& SenderRtcpSendEvent() override {
+    return mSenderRtcpSendEvent;
+  }
+  MediaEventSourceExc<MediaPacket>& ReceiverRtcpSendEvent() override {
+    return mReceiverRtcpSendEvent;
+  }
+  void ConnectReceiverRtpEvent(
+      MediaEventSourceExc<webrtc::RtpPacketReceived, webrtc::RTPHeader>& aEvent)
+      override {
+    mReceiverRtpEventListener =
+        aEvent.Connect(mCallThread, this, &WebrtcVideoConduit::OnRtpReceived);
+  }
+  void ConnectReceiverRtcpEvent(
+      MediaEventSourceExc<MediaPacket>& aEvent) override {
+    mReceiverRtcpEventListener =
+        aEvent.Connect(mCallThread, this, &WebrtcVideoConduit::OnRtcpReceived);
+  }
+  void ConnectSenderRtcpEvent(
+      MediaEventSourceExc<MediaPacket>& aEvent) override {
+    mSenderRtcpEventListener =
+        aEvent.Connect(mCallThread, this, &WebrtcVideoConduit::OnRtcpReceived);
+  }
+
+  std::vector<webrtc::RtpSource> GetUpstreamRtpSources() const override;
+
+ private:
+  // Don't allow copying/assigning.
+  WebrtcVideoConduit(const WebrtcVideoConduit&) = delete;
+  void operator=(const WebrtcVideoConduit&) = delete;
+
+  // Utility function to dump recv codec database
+  void DumpCodecDB() const;
+
+  // Video Latency Test averaging filter
+  void VideoLatencyUpdate(uint64_t aNewSample);
+
+  void CreateSendStream();
+  void DeleteSendStream();
+  void CreateRecvStream();
+  void DeleteRecvStream();
+
+  void DeliverPacket(rtc::CopyOnWriteBuffer packet, PacketType type) override;
+
+  MediaEventSource<void>& RtcpByeEvent() override { return mRtcpByeEvent; }
+  MediaEventSource<void>& RtcpTimeoutEvent() override {
+    return mRtcpTimeoutEvent;
+  }
+  MediaEventSource<void>& RtpPacketEvent() override { return mRtpPacketEvent; }
+
+  bool RequiresNewSendStream(const VideoCodecConfig& newConfig) const;
+
+  mutable mozilla::ReentrantMonitor mRendererMonitor MOZ_UNANNOTATED;
+
+  // Accessed on any thread under mRendererMonitor.
+  RefPtr<mozilla::VideoRenderer> mRenderer;
+
+  // Accessed on any thread under mRendererMonitor.
+  unsigned short mReceivingWidth = 0;
+
+  // Accessed on any thread under mRendererMonitor.
+  unsigned short mReceivingHeight = 0;
+
+  // Call worker thread. All access to mCall->Call() happens here.
+  const nsCOMPtr<nsISerialEventTarget> mCallThread;
+
+  // Socket transport service thread that runs stats queries against us. Any
+  // thread.
+  const nsCOMPtr<nsISerialEventTarget> mStsThread;
+
+  // Thread on which we are fed video frames. Set lazily on first call to
+  // SendVideoFrame().
+  nsCOMPtr<nsISerialEventTarget> mFrameSendingThread;
+
+  struct Control {
+    // Mirrors that map to VideoConduitControlInterface for control. Call thread
+    // only.
+    Mirror<bool> mReceiving;
+    Mirror<bool> mTransmitting;
+    Mirror<Ssrcs> mLocalSsrcs;
+    Mirror<Ssrcs> mLocalRtxSsrcs;
+    Mirror<std::string> mLocalCname;
+    Mirror<std::string> mMid;
+    Mirror<Ssrc> mRemoteSsrc;
+    Mirror<Ssrc> mRemoteRtxSsrc;
+    Mirror<std::string> mSyncGroup;
+    Mirror<RtpExtList> mLocalRecvRtpExtensions;
+    Mirror<RtpExtList> mLocalSendRtpExtensions;
+    Mirror<Maybe<VideoCodecConfig>> mSendCodec;
+    Mirror<Maybe<RtpRtcpConfig>> mSendRtpRtcpConfig;
+    Mirror<std::vector<VideoCodecConfig>> mRecvCodecs;
+    Mirror<Maybe<RtpRtcpConfig>> mRecvRtpRtcpConfig;
+    Mirror<webrtc::VideoCodecMode> mCodecMode;
+
+    // For caching mRemoteSsrc and mRemoteRtxSsrc, since another caller may
+    // change the remote ssrc in the stream config directly.
+    Ssrc mConfiguredRemoteSsrc = 0;
+    Ssrc mConfiguredRemoteRtxSsrc = 0;
+    // For tracking changes to mSendCodec and mSendRtpRtcpConfig.
+    Maybe<VideoCodecConfig> mConfiguredSendCodec;
+    Maybe<RtpRtcpConfig> mConfiguredSendRtpRtcpConfig;
+    // For tracking changes to mRecvCodecs and mRecvRtpRtcpConfig.
+    std::vector<VideoCodecConfig> mConfiguredRecvCodecs;
+    Maybe<RtpRtcpConfig> mConfiguredRecvRtpRtcpConfig;
+
+    Control() = delete;
+    explicit Control(const RefPtr<AbstractThread>& aCallThread);
+  } mControl;
+
+  // WatchManager allowing Mirrors and other watch targets to trigger functions
+  // that will update the webrtc.org configuration.
+  WatchManager<WebrtcVideoConduit> mWatchManager;
+
+  mutable Mutex mMutex MOZ_UNANNOTATED;
+
+  // Decoder factory used by mRecvStream when it needs new decoders. This is
+  // not shared broader like some state in the WebrtcCallWrapper because it
+  // handles CodecPluginID plumbing tied to this VideoConduit.
+  const UniquePtr<WebrtcVideoDecoderFactory> mDecoderFactory;
+
+  // Encoder factory used by mSendStream when it needs new encoders. This is
+  // not shared broader like some state in the WebrtcCallWrapper because it
+  // handles CodecPluginID plumbing tied to this VideoConduit.
+  const UniquePtr<WebrtcVideoEncoderFactory> mEncoderFactory;
+
+  // Our own record of the sinks added to mVideoBroadcaster so we can support
+  // dispatching updates to sinks from off-Call-thread. Call thread only.
+  AutoTArray<rtc::VideoSinkInterface<webrtc::VideoFrame>*, 1> mRegisteredSinks;
+
+  // Broadcaster that distributes our frames to all registered sinks.
+  // Threadsafe.
+  rtc::VideoBroadcaster mVideoBroadcaster;
+
+  // Buffer pool used for scaling frames.
+  // Accessed on the frame-feeding thread only.
+  webrtc::VideoFrameBufferPool mBufferPool;
+
+  // Engine state we are concerned with. Written on the Call thread and read
+  // anywhere.
+  mozilla::Atomic<bool>
+      mEngineTransmitting;  // If true ==> Transmit Subsystem is up and running
+  mozilla::Atomic<bool>
+      mEngineReceiving;  // if true ==> Receive Subsystem up and running
+
+  // Written only on the Call thread. Guarded by mMutex, except for reads on the
+  // Call thread.
+  Maybe<VideoCodecConfig> mCurSendCodecConfig;
+
+  // Bookkeeping of stats for telemetry. Call thread only.
+  RunningStat mSendFramerate;
+  RunningStat mSendBitrate;
+  RunningStat mRecvFramerate;
+  RunningStat mRecvBitrate;
+
+  // Must call webrtc::Call::DestroyVideoReceive/SendStream to delete this.
+  // Written only on the Call thread. Guarded by mMutex, except for reads on the
+  // Call thread.
+  webrtc::VideoReceiveStreamInterface* mRecvStream = nullptr;
+
+  // Must call webrtc::Call::DestroyVideoReceive/SendStream to delete this.
+  webrtc::VideoSendStream* mSendStream = nullptr;
+
+  // Written on the frame feeding thread.
+  // Guarded by mMutex, except for reads on the frame feeding thread.
+  unsigned short mLastWidth = 0;
+
+  // Written on the frame feeding thread.
+  // Guarded by mMutex, except for reads on the frame feeding thread.
+  unsigned short mLastHeight = 0;
+
+  // Written on the frame feeding thread, the timestamp of the last frame on the
+  // send side, in microseconds. This is a local timestamp using the system
+  // clock with a unspecified epoch (Like mozilla::TimeStamp).
+  // Guarded by mMutex.
+  Maybe<uint64_t> mLastTimestampSendUs;
+
+  // Written on the frame receive thread, the rtp timestamp of the last frame
+  // on the receive side, in 90kHz base. This comes from the RTP packet.
+  // Guarded by mMutex.
+  Maybe<uint32_t> mLastRTPTimestampReceive;
+
+  // Accessed from any thread under mRendererMonitor.
+  uint64_t mVideoLatencyAvg = 0;
+
+  const bool mVideoLatencyTestEnable;
+
+  // All in bps.
+  const int mMinBitrate;
+  const int mStartBitrate;
+  const int mPrefMaxBitrate;
+  const int mMinBitrateEstimate;
+
+  // Max bitrate in bps as provided by negotiation. Call thread only.
+  int mNegotiatedMaxBitrate = 0;
+
+  // Set to true to force denoising on.
+  const bool mDenoising;
+
+  // Set to true to ignore sink wants (scaling due to bwe and cpu usage).
+  const bool mLockScaling;
+
+  const uint8_t mSpatialLayers;
+  const uint8_t mTemporalLayers;
+
+  static const unsigned int sAlphaNum = 7;
+  static const unsigned int sAlphaDen = 8;
+  static const unsigned int sRoundingPadding = 1024;
+
+  // Target jitter buffer to be applied to the receive stream in milliseconds.
+  uint16_t mJitterBufferTargetMs = 0;
+
+  // WEBRTC.ORG Call API
+  // Const so can be accessed on any thread. All methods are called on the Call
+  // thread.
+  const RefPtr<WebrtcCallWrapper> mCall;
+
+  // Set up in the ctor and then not touched. Called through by the streams on
+  // any thread. Safe since we own and control the lifetime of the streams.
+  WebrtcSendTransport mSendTransport;
+  WebrtcReceiveTransport mRecvTransport;
+
+  // Written only on the Call thread. Guarded by mMutex, except for reads on the
+  // Call thread. Typical non-Call thread access is on the frame delivery
+  // thread.
+  webrtc::VideoSendStream::Config mSendStreamConfig;
+
+  // Call thread only.
+  webrtc::VideoEncoderConfig mEncoderConfig;
+
+  // Written only on the Call thread. Guarded by mMutex, except for reads on the
+  // Call thread. Calls can happen under mMutex on any thread.
+  DataMutex<RefPtr<rtc::RefCountedObject<VideoStreamFactory>>>
+      mVideoStreamFactory;
+
+  // Call thread only.
+  webrtc::VideoReceiveStreamInterface::Config mRecvStreamConfig;
+
+  // Are SSRC changes without signaling allowed or not.
+  // Call thread only.
+  bool mAllowSsrcChange = true;
+
+  // Accessed during configuration/signaling (Call thread), and on the frame
+  // delivery thread for frame history tracking. Set only on the Call thread.
+  Atomic<uint32_t> mRecvSSRC =
+      Atomic<uint32_t>(0);  // this can change during a stream!
+
+  // Accessed from both the STS and frame delivery thread for frame history
+  // tracking. Set when receiving packets.
+  Atomic<uint32_t> mRemoteSendSSRC =
+      Atomic<uint32_t>(0);  // this can change during a stream!
+
+  // Main thread only
+  nsTArray<uint64_t> mSendCodecPluginIDs;
+  // Main thread only
+  nsTArray<uint64_t> mRecvCodecPluginIDs;
+
+  // Main thread only
+  MediaEventListener mSendPluginCreated;
+  MediaEventListener mSendPluginReleased;
+  MediaEventListener mRecvPluginCreated;
+  MediaEventListener mRecvPluginReleased;
+
+  // Call thread only. ssrc -> base_seq
+  std::map<uint32_t, uint16_t> mRtpSendBaseSeqs;
+  // libwebrtc network thread only. ssrc -> base_seq.
+  // To track changes needed to mRtpSendBaseSeqs.
+  std::map<uint32_t, uint16_t> mRtpSendBaseSeqs_n;
+
+  // Tracking the attributes of received frames over time
+  // Protected by mRendererMonitor
+  dom::RTCVideoFrameHistoryInternal mReceivedFrameHistory;
+
+  // Thread safe
+  Atomic<bool> mTransportActive = Atomic<bool>(false);
+  MediaEventProducer<void> mRtcpByeEvent;
+  MediaEventProducer<void> mRtcpTimeoutEvent;
+  MediaEventProducer<void> mRtpPacketEvent;
+  MediaEventProducerExc<MediaPacket> mSenderRtpSendEvent;
+  MediaEventProducerExc<MediaPacket> mSenderRtcpSendEvent;
+  MediaEventProducerExc<MediaPacket> mReceiverRtcpSendEvent;
+
+  // Assigned and revoked on mStsThread. Listeners for receiving packets.
+  MediaEventListener mSenderRtcpEventListener;    // Rtp-transmitting pipeline
+  MediaEventListener mReceiverRtcpEventListener;  // Rtp-receiving pipeline
+  MediaEventListener mReceiverRtpEventListener;   // Rtp-receiving pipeline
+};
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/libwebrtcglue/VideoStreamFactory.cpp b/dom/media/webrtc/libwebrtcglue/VideoStreamFactory.cpp
new file mode 100644
index 0000000000..9782f8760b
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/VideoStreamFactory.cpp
@@ -0,0 +1,387 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/. */
+
+#include "VideoStreamFactory.h"
+
+#include "common/browser_logging/CSFLog.h"
+#include "nsThreadUtils.h"
+#include "VideoConduit.h"
+
+template <class t>
+void ConstrainPreservingAspectRatio(uint16_t aMaxWidth, uint16_t aMaxHeight,
+                                    t* aWidth, t* aHeight) {
+  if (((*aWidth) <= aMaxWidth) && ((*aHeight) <= aMaxHeight)) {
+    return;
+  }
+
+  if ((*aWidth) * aMaxHeight > aMaxWidth * (*aHeight)) {
+    (*aHeight) = aMaxWidth * (*aHeight) / (*aWidth);
+    (*aWidth) = aMaxWidth;
+  } else {
+    (*aWidth) = aMaxHeight * (*aWidth) / (*aHeight);
+    (*aHeight) = aMaxHeight;
+  }
+}
+
+namespace mozilla {
+
+#ifdef LOGTAG
+#  undef LOGTAG
+#endif
+#define LOGTAG "WebrtcVideoSessionConduit"
+
+#define DEFAULT_VIDEO_MAX_FRAMERATE 30u
+
+#define MB_OF(w, h) \
+  ((unsigned int)((((w + 15) >> 4)) * ((unsigned int)((h + 15) >> 4))))
+// For now, try to set the max rates well above the knee in the curve.
+// Chosen somewhat arbitrarily; it's hard to find good data oriented for
+// realtime interactive/talking-head recording.  These rates assume
+// 30fps.
+
+// XXX Populate this based on a pref (which we should consider sorting because
+// people won't assume they need to).
+static VideoStreamFactory::ResolutionAndBitrateLimits
+    kResolutionAndBitrateLimits[] = {
+        // clang-format off
+  {MB_OF(1920, 1200), KBPS(1500), KBPS(2000), KBPS(10000)}, // >HD (3K, 4K, etc)
+  {MB_OF(1280, 720), KBPS(1200), KBPS(1500), KBPS(5000)}, // HD ~1080-1200
+  {MB_OF(800, 480), KBPS(200), KBPS(800), KBPS(2500)}, // HD ~720
+  {MB_OF(480, 270), KBPS(150), KBPS(500), KBPS(2000)}, // WVGA
+  {tl::Max<MB_OF(400, 240), MB_OF(352, 288)>::value, KBPS(125), KBPS(300), KBPS(1300)}, // VGA
+  {MB_OF(176, 144), KBPS(100), KBPS(150), KBPS(500)}, // WQVGA, CIF
+  {0 , KBPS(40), KBPS(80), KBPS(250)} // QCIF and below
+        // clang-format on
+};
+
+auto VideoStreamFactory::GetLimitsFor(unsigned int aWidth, unsigned int aHeight,
+                                      int aCapBps /* = 0 */)
+    -> ResolutionAndBitrateLimits {
+  // max bandwidth should be proportional (not linearly!) to resolution, and
+  // proportional (perhaps linearly, or close) to current frame rate.
+  int fs = MB_OF(aWidth, aHeight);
+
+  for (const auto& resAndLimits : kResolutionAndBitrateLimits) {
+    if (fs > resAndLimits.resolution_in_mb &&
+        // pick the highest range where at least start rate is within cap
+        // (or if we're at the end of the array).
+        (aCapBps == 0 || resAndLimits.start_bitrate_bps <= aCapBps ||
+         resAndLimits.resolution_in_mb == 0)) {
+      return resAndLimits;
+    }
+  }
+
+  MOZ_CRASH("Loop should have handled fallback");
+}
+
+/**
+ * Function to set the encoding bitrate limits based on incoming frame size and
+ * rate
+ * @param width, height: dimensions of the frame
+ * @param min: minimum bitrate in bps
+ * @param start: bitrate in bps that the encoder should start with
+ * @param cap: user-enforced max bitrate, or 0
+ * @param pref_cap: cap enforced by prefs
+ * @param negotiated_cap: cap negotiated through SDP
+ * @param aVideoStream stream to apply bitrates to
+ */
+static void SelectBitrates(unsigned short width, unsigned short height, int min,
+                           int start, int cap, int pref_cap, int negotiated_cap,
+                           webrtc::VideoStream& aVideoStream) {
+  int& out_min = aVideoStream.min_bitrate_bps;
+  int& out_start = aVideoStream.target_bitrate_bps;
+  int& out_max = aVideoStream.max_bitrate_bps;
+
+  VideoStreamFactory::ResolutionAndBitrateLimits resAndLimits =
+      VideoStreamFactory::GetLimitsFor(width, height);
+  out_min = MinIgnoreZero(resAndLimits.min_bitrate_bps, cap);
+  out_start = MinIgnoreZero(resAndLimits.start_bitrate_bps, cap);
+  out_max = MinIgnoreZero(resAndLimits.max_bitrate_bps, cap);
+
+  // Note: negotiated_cap is the max transport bitrate - it applies to
+  // a single codec encoding, but should also apply to the sum of all
+  // simulcast layers in this encoding! So sum(layers.maxBitrate) <=
+  // negotiated_cap
+  // Note that out_max already has had pref_cap applied to it
+  out_max = MinIgnoreZero(negotiated_cap, out_max);
+  out_min = std::min(out_min, out_max);
+  out_start = std::min(out_start, out_max);
+
+  if (min && min > out_min) {
+    out_min = min;
+  }
+  // If we try to set a minimum bitrate that is too low, ViE will reject it.
+  out_min = std::max(kViEMinCodecBitrate_bps, out_min);
+  out_max = std::max(kViEMinCodecBitrate_bps, out_max);
+  if (start && start > out_start) {
+    out_start = start;
+  }
+
+  // Ensure that min <= start <= max
+  if (out_min > out_max) {
+    out_min = out_max;
+  }
+  out_start = std::min(out_max, std::max(out_start, out_min));
+
+  MOZ_ASSERT(pref_cap == 0 || out_max <= pref_cap);
+}
+
+std::vector<webrtc::VideoStream> VideoStreamFactory::CreateEncoderStreams(
+    int aWidth, int aHeight, const webrtc::VideoEncoderConfig& aConfig) {
+  // We only allow one layer when screensharing
+  const size_t streamCount =
+      mCodecMode == webrtc::VideoCodecMode::kScreensharing
+          ? 1
+          : aConfig.number_of_streams;
+
+  MOZ_RELEASE_ASSERT(streamCount >= 1, "Should request at least one stream");
+
+  std::vector<webrtc::VideoStream> streams;
+  streams.reserve(streamCount);
+
+  {
+    auto frameRateController = mFramerateController.Lock();
+    frameRateController->Reset();
+  }
+
+  for (int idx = streamCount - 1; idx >= 0; --idx) {
+    webrtc::VideoStream video_stream;
+    auto& encoding = mCodecConfig.mEncodings[idx];
+    video_stream.active = encoding.active;
+    MOZ_ASSERT(encoding.constraints.scaleDownBy >= 1.0);
+
+    gfx::IntSize newSize(0, 0);
+
+    if (aWidth && aHeight) {
+      auto maxPixelCount = mLockScaling ? 0U : mWants.max_pixel_count;
+      newSize = CalculateScaledResolution(
+          aWidth, aHeight, encoding.constraints.scaleDownBy, maxPixelCount);
+    }
+
+    if (newSize.width == 0 || newSize.height == 0) {
+      CSFLogInfo(LOGTAG,
+                 "%s Stream with RID %s ignored because of no resolution.",
+                 __FUNCTION__, encoding.rid.c_str());
+      continue;
+    }
+
+    uint16_t max_width = mCodecConfig.mEncodingConstraints.maxWidth;
+    uint16_t max_height = mCodecConfig.mEncodingConstraints.maxHeight;
+    if (max_width || max_height) {
+      max_width = max_width ? max_width : UINT16_MAX;
+      max_height = max_height ? max_height : UINT16_MAX;
+      ConstrainPreservingAspectRatio(max_width, max_height, &newSize.width,
+                                     &newSize.height);
+    }
+
+    MOZ_ASSERT(newSize.width > 0);
+    MOZ_ASSERT(newSize.height > 0);
+    video_stream.width = newSize.width;
+    video_stream.height = newSize.height;
+    SelectMaxFramerateForAllStreams(newSize.width, newSize.height);
+
+    CSFLogInfo(LOGTAG, "%s Input frame %ux%u, RID %s scaling to %zux%zu",
+               __FUNCTION__, aWidth, aHeight, encoding.rid.c_str(),
+               video_stream.width, video_stream.height);
+
+    // mMaxFramerateForAllStreams is based on codec-wide stuff like fmtp, and
+    // hard-coded limits based on the source resolution.
+    // mCodecConfig.mEncodingConstraints.maxFps does not take the hard-coded
+    // limits into account, so we have mMaxFramerateForAllStreams which
+    // incorporates those. Per-encoding max framerate is based on parameters
+    // from JS, and maybe rid
+    unsigned int max_framerate = SelectFrameRate(
+        mMaxFramerateForAllStreams, video_stream.width, video_stream.height);
+    max_framerate = std::min(WebrtcVideoConduit::ToLibwebrtcMaxFramerate(
+                                 encoding.constraints.maxFps),
+                             max_framerate);
+    if (max_framerate >= std::numeric_limits<int>::max()) {
+      // If nothing has specified any kind of limit (uncommon), pick something
+      // reasonable.
+      max_framerate = DEFAULT_VIDEO_MAX_FRAMERATE;
+    }
+    video_stream.max_framerate = static_cast<int>(max_framerate);
+    CSFLogInfo(LOGTAG, "%s Stream with RID %s maxFps=%d (global max fps = %u)",
+               __FUNCTION__, encoding.rid.c_str(), video_stream.max_framerate,
+               (unsigned)mMaxFramerateForAllStreams);
+
+    SelectBitrates(video_stream.width, video_stream.height, mMinBitrate,
+                   mStartBitrate, encoding.constraints.maxBr, mPrefMaxBitrate,
+                   mNegotiatedMaxBitrate, video_stream);
+
+    video_stream.bitrate_priority = aConfig.bitrate_priority;
+    video_stream.max_qp = kQpMax;
+
+    if (streamCount > 1) {
+      if (mCodecMode == webrtc::VideoCodecMode::kScreensharing) {
+        video_stream.num_temporal_layers = 1;
+      } else {
+        video_stream.num_temporal_layers = 2;
+      }
+      // XXX Bug 1390215 investigate using more of
+      // simulcast.cc:GetSimulcastConfig() or our own algorithm to replace it
+    }
+
+    if (mCodecConfig.mName == "H264") {
+      if (mCodecConfig.mEncodingConstraints.maxMbps > 0) {
+        // Not supported yet!
+        CSFLogError(LOGTAG, "%s H.264 max_mbps not supported yet",
+                    __FUNCTION__);
+      }
+    }
+    streams.push_back(video_stream);
+  }
+
+  MOZ_RELEASE_ASSERT(streams.size(), "Should configure at least one stream");
+  return streams;
+}
+
+gfx::IntSize VideoStreamFactory::CalculateScaledResolution(
+    int aWidth, int aHeight, double aScaleDownByResolution,
+    unsigned int aMaxPixelCount) {
+  // If any adjustments like scaleResolutionDownBy or maxFS are being given
+  // we want to choose a height and width here to provide for more variety
+  // in possible resolutions.
+  int width = aWidth;
+  int height = aHeight;
+
+  if (aScaleDownByResolution > 1) {
+    width = static_cast<int>(aWidth / aScaleDownByResolution);
+    height = static_cast<int>(aHeight / aScaleDownByResolution);
+  }
+
+  // Check if we still need to adjust resolution down more due to other
+  // constraints.
+  if (mCodecConfig.mEncodingConstraints.maxFs > 0 || aMaxPixelCount > 0) {
+    auto currentFs = static_cast<unsigned int>(width * height);
+    auto maxFs =
+        (mCodecConfig.mEncodingConstraints.maxFs > 0 && aMaxPixelCount > 0)
+            ? std::min((mCodecConfig.mEncodingConstraints.maxFs * 16 * 16),
+                       aMaxPixelCount)
+            : std::max((mCodecConfig.mEncodingConstraints.maxFs * 16 * 16),
+                       aMaxPixelCount);
+
+    // If our currentFs is greater than maxFs we calculate a width and height
+    // that will get as close as possible to maxFs and try to maintain aspect
+    // ratio.
+    if (currentFs > maxFs) {
+      if (aWidth > aHeight) {  // Landscape
+        auto aspectRatio = static_cast<double>(aWidth) / aHeight;
+
+        height = static_cast<int>(std::sqrt(maxFs / aspectRatio));
+        width = static_cast<int>(height * aspectRatio);
+      } else {  // Portrait
+        auto aspectRatio = static_cast<double>(aHeight) / aWidth;
+
+        width = static_cast<int>(std::sqrt(maxFs / aspectRatio));
+        height = static_cast<int>(width * aspectRatio);
+      }
+    }
+  }
+
+  // Simplest possible adaptation to resolution alignment.
+  width -= width % mWants.resolution_alignment;
+  height -= height % mWants.resolution_alignment;
+
+  // Dont scale below our minimum value to prevent problems.
+  const int minSize = 1;
+  if (width < minSize || height < minSize) {
+    width = minSize;
+    height = minSize;
+  }
+
+  return gfx::IntSize(width, height);
+}
+
+void VideoStreamFactory::SelectMaxFramerateForAllStreams(
+    unsigned short aWidth, unsigned short aHeight) {
+  int max_fs = std::numeric_limits<int>::max();
+  if (!mLockScaling) {
+    max_fs = mWants.max_pixel_count;
+  }
+  // Limit resolution to max-fs
+  if (mCodecConfig.mEncodingConstraints.maxFs) {
+    // max-fs is in macroblocks, convert to pixels
+    max_fs = std::min(
+        max_fs,
+        static_cast<int>(mCodecConfig.mEncodingConstraints.maxFs * (16 * 16)));
+  }
+
+  unsigned int framerate_all_streams =
+      SelectFrameRate(mMaxFramerateForAllStreams, aWidth, aHeight);
+  unsigned int maxFrameRate = mMaxFramerateForAllStreams;
+  if (mMaxFramerateForAllStreams != framerate_all_streams) {
+    CSFLogDebug(LOGTAG, "%s: framerate changing to %u (from %u)", __FUNCTION__,
+                framerate_all_streams, maxFrameRate);
+    mMaxFramerateForAllStreams = framerate_all_streams;
+  }
+
+  int framerate_with_wants;
+  if (framerate_all_streams > std::numeric_limits<int>::max()) {
+    framerate_with_wants = std::numeric_limits<int>::max();
+  } else {
+    framerate_with_wants = static_cast<int>(framerate_all_streams);
+  }
+
+  framerate_with_wants =
+      std::min(framerate_with_wants, mWants.max_framerate_fps);
+  CSFLogDebug(LOGTAG,
+              "%s: Calling OnOutputFormatRequest, max_fs=%d, max_fps=%d",
+              __FUNCTION__, max_fs, framerate_with_wants);
+  auto frameRateController = mFramerateController.Lock();
+  frameRateController->SetMaxFramerate(framerate_with_wants);
+}
+
+unsigned int VideoStreamFactory::SelectFrameRate(
+    unsigned int aOldFramerate, unsigned short aSendingWidth,
+    unsigned short aSendingHeight) {
+  unsigned int new_framerate = aOldFramerate;
+
+  // Limit frame rate based on max-mbps
+  if (mCodecConfig.mEncodingConstraints.maxMbps) {
+    unsigned int cur_fs, mb_width, mb_height;
+
+    mb_width = (aSendingWidth + 15) >> 4;
+    mb_height = (aSendingHeight + 15) >> 4;
+
+    cur_fs = mb_width * mb_height;
+    if (cur_fs > 0) {  // in case no frames have been sent
+      new_framerate = mCodecConfig.mEncodingConstraints.maxMbps / cur_fs;
+    }
+  }
+
+  new_framerate =
+      std::min(new_framerate, WebrtcVideoConduit::ToLibwebrtcMaxFramerate(
+                                  mCodecConfig.mEncodingConstraints.maxFps));
+  return new_framerate;
+}
+
+bool VideoStreamFactory::ShouldDropFrame(const webrtc::VideoFrame& aFrame) {
+  bool hasNonZeroLayer = false;
+  {
+    const size_t streamCount =
+        mCodecMode == webrtc::VideoCodecMode::kScreensharing
+            ? 1
+            : mCodecConfig.mEncodings.size();
+    for (int idx = streamCount - 1; idx >= 0; --idx) {
+      const auto& encoding = mCodecConfig.mEncodings[idx];
+      if (aFrame.width() / encoding.constraints.scaleDownBy >= 1.0 &&
+          aFrame.height() / encoding.constraints.scaleDownBy >= 1.0) {
+        hasNonZeroLayer = true;
+        break;
+      }
+    }
+  }
+  if (!hasNonZeroLayer) {
+    return true;
+  }
+
+  auto frameRateController = mFramerateController.Lock();
+  return frameRateController->ShouldDropFrame(aFrame.timestamp_us() *
+                                              rtc::kNumNanosecsPerMicrosec);
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/libwebrtcglue/VideoStreamFactory.h b/dom/media/webrtc/libwebrtcglue/VideoStreamFactory.h
new file mode 100644
index 0000000000..832d1e9399
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/VideoStreamFactory.h
@@ -0,0 +1,132 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at https://mozilla.org/MPL/2.0/. */
+
+#ifndef VideoStreamFactory_h
+#define VideoStreamFactory_h
+
+#include "CodecConfig.h"
+#include "mozilla/Atomics.h"
+#include "mozilla/DataMutex.h"
+#include "mozilla/gfx/Point.h"
+#include "mozilla/UniquePtr.h"
+#include "api/video/video_source_interface.h"
+#include "common_video/framerate_controller.h"
+#include "rtc_base/time_utils.h"
+#include "video/config/video_encoder_config.h"
+
+namespace webrtc {
+class VideoFrame;
+}
+
+namespace mozilla {
+
+// Factory class for VideoStreams... vie_encoder.cc will call this to
+// reconfigure.
+class VideoStreamFactory
+    : public webrtc::VideoEncoderConfig::VideoStreamFactoryInterface {
+ public:
+  struct ResolutionAndBitrateLimits {
+    int resolution_in_mb;
+    int min_bitrate_bps;
+    int start_bitrate_bps;
+    int max_bitrate_bps;
+  };
+
+  static ResolutionAndBitrateLimits GetLimitsFor(unsigned int aWidth,
+                                                 unsigned int aHeight,
+                                                 int aCapBps = 0);
+
+  VideoStreamFactory(VideoCodecConfig aConfig,
+                     webrtc::VideoCodecMode aCodecMode, int aMinBitrate,
+                     int aStartBitrate, int aPrefMaxBitrate,
+                     int aNegotiatedMaxBitrate,
+                     const rtc::VideoSinkWants& aWants, bool aLockScaling)
+      : mCodecMode(aCodecMode),
+        mMaxFramerateForAllStreams(std::numeric_limits<unsigned int>::max()),
+        mCodecConfig(std::forward<VideoCodecConfig>(aConfig)),
+        mMinBitrate(aMinBitrate),
+        mStartBitrate(aStartBitrate),
+        mPrefMaxBitrate(aPrefMaxBitrate),
+        mNegotiatedMaxBitrate(aNegotiatedMaxBitrate),
+        mFramerateController("VideoStreamFactory::mFramerateController"),
+        mWants(aWants),
+        mLockScaling(aLockScaling) {}
+
+  // This gets called off-main thread and may hold internal webrtc.org
+  // locks. May *NOT* lock the conduit's mutex, to avoid deadlocks.
+  std::vector<webrtc::VideoStream> CreateEncoderStreams(
+      int aWidth, int aHeight,
+      const webrtc::VideoEncoderConfig& aConfig) override;
+  /**
+   * Function to select and change the encoding resolution based on incoming
+   * frame size and current available bandwidth.
+   * @param width, height: dimensions of the frame
+   */
+  void SelectMaxFramerateForAllStreams(unsigned short aWidth,
+                                       unsigned short aHeight);
+
+  /**
+   * Function to determine if the frame should be dropped based on the given
+   * frame's resolution (combined with the factory's scaleResolutionDownBy) or
+   * timestamp.
+   * @param aFrame frame to be evaluated.
+   * @return true if frame should be dropped, false otehrwise.
+   */
+  bool ShouldDropFrame(const webrtc::VideoFrame& aFrame);
+
+ private:
+  /**
+   * Function to calculate a scaled down width and height based on
+   * scaleDownByResolution, maxFS, and max pixel count settings.
+   * @param aWidth current frame width
+   * @param aHeight current frame height
+   * @param aScaleDownByResolution value to scale width and height down by.
+   * @param aMaxPixelCount maximum number of pixels wanted in a frame.
+   * @return a gfx:IntSize containing  width and height to use. These may match
+   * the aWidth and aHeight passed in if no scaling was needed.
+   */
+  gfx::IntSize CalculateScaledResolution(int aWidth, int aHeight,
+                                         double aScaleDownByResolution,
+                                         unsigned int aMaxPixelCount);
+
+  /**
+   * Function to select and change the encoding frame rate based on incoming
+   * frame rate, current frame size and max-mbps setting.
+   * @param aOldFramerate current framerate
+   * @param aSendingWidth width of frames being sent
+   * @param aSendingHeight height of frames being sent
+   * @return new framerate meeting max-mbps requriements based on frame size
+   */
+  unsigned int SelectFrameRate(unsigned int aOldFramerate,
+                               unsigned short aSendingWidth,
+                               unsigned short aSendingHeight);
+
+  // Used to limit number of streams for screensharing.
+  Atomic<webrtc::VideoCodecMode> mCodecMode;
+
+  // The framerate we're currently sending at.
+  Atomic<unsigned int> mMaxFramerateForAllStreams;
+
+  // The current send codec config, containing simulcast layer configs.
+  const VideoCodecConfig mCodecConfig;
+
+  // Bitrate limits in bps.
+  const int mMinBitrate = 0;
+  const int mStartBitrate = 0;
+  const int mPrefMaxBitrate = 0;
+  const int mNegotiatedMaxBitrate = 0;
+
+  // DatamMutex used as object is mutated from a libwebrtc thread and
+  // a seperate thread used to pass video frames to libwebrtc.
+  DataMutex<webrtc::FramerateController> mFramerateController;
+
+  const rtc::VideoSinkWants mWants;
+  const bool mLockScaling;
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/libwebrtcglue/WebrtcCallWrapper.cpp b/dom/media/webrtc/libwebrtcglue/WebrtcCallWrapper.cpp
new file mode 100644
index 0000000000..7d4547ddb6
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/WebrtcCallWrapper.cpp
@@ -0,0 +1,105 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "WebrtcCallWrapper.h"
+
+#include "jsapi/PeerConnectionCtx.h"
+#include "MediaConduitInterface.h"
+#include "TaskQueueWrapper.h"
+
+// libwebrtc includes
+#include "call/rtp_transport_controller_send_factory.h"
+
+namespace mozilla {
+
+/* static */ RefPtr<WebrtcCallWrapper> WebrtcCallWrapper::Create(
+    const dom::RTCStatsTimestampMaker& aTimestampMaker,
+    UniquePtr<media::ShutdownBlockingTicket> aShutdownTicket,
+    const RefPtr<SharedWebrtcState>& aSharedState) {
+  auto eventLog = MakeUnique<webrtc::RtcEventLogNull>();
+  auto taskQueueFactory = MakeUnique<SharedThreadPoolWebRtcTaskQueueFactory>();
+  auto videoBitrateAllocatorFactory =
+      WrapUnique(webrtc::CreateBuiltinVideoBitrateAllocatorFactory().release());
+  RefPtr<WebrtcCallWrapper> wrapper = new WebrtcCallWrapper(
+      aSharedState, std::move(videoBitrateAllocatorFactory),
+      std::move(eventLog), std::move(taskQueueFactory), aTimestampMaker,
+      std::move(aShutdownTicket));
+
+  wrapper->mCallThread->Dispatch(
+      NS_NewRunnableFunction(__func__, [wrapper, aSharedState] {
+        webrtc::Call::Config config(wrapper->mEventLog.get());
+        config.audio_state =
+            webrtc::AudioState::Create(aSharedState->mAudioStateConfig);
+        config.task_queue_factory = wrapper->mTaskQueueFactory.get();
+        config.trials = aSharedState->mTrials.get();
+        wrapper->SetCall(WrapUnique(webrtc::Call::Create(
+            config, &wrapper->mClock,
+            webrtc::RtpTransportControllerSendFactory().Create(
+                config.ExtractTransportConfig(), &wrapper->mClock))));
+      }));
+
+  return wrapper;
+}
+
+void WebrtcCallWrapper::SetCall(UniquePtr<webrtc::Call> aCall) {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  MOZ_ASSERT(!mCall);
+  mCall = std::move(aCall);
+}
+
+webrtc::Call* WebrtcCallWrapper::Call() const {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  return mCall.get();
+}
+
+void WebrtcCallWrapper::UnsetRemoteSSRC(uint32_t aSsrc) {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  for (const auto& conduit : mConduits) {
+    conduit->UnsetRemoteSSRC(aSsrc);
+  }
+}
+
+void WebrtcCallWrapper::RegisterConduit(MediaSessionConduit* conduit) {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  mConduits.insert(conduit);
+}
+
+void WebrtcCallWrapper::UnregisterConduit(MediaSessionConduit* conduit) {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  mConduits.erase(conduit);
+}
+
+void WebrtcCallWrapper::Destroy() {
+  MOZ_ASSERT(mCallThread->IsOnCurrentThread());
+  mCall = nullptr;
+  mShutdownTicket = nullptr;
+}
+
+const dom::RTCStatsTimestampMaker& WebrtcCallWrapper::GetTimestampMaker()
+    const {
+  return mClock.mTimestampMaker;
+}
+
+WebrtcCallWrapper::~WebrtcCallWrapper() { MOZ_ASSERT(!mCall); }
+
+WebrtcCallWrapper::WebrtcCallWrapper(
+    RefPtr<SharedWebrtcState> aSharedState,
+    UniquePtr<webrtc::VideoBitrateAllocatorFactory>
+        aVideoBitrateAllocatorFactory,
+    UniquePtr<webrtc::RtcEventLog> aEventLog,
+    UniquePtr<webrtc::TaskQueueFactory> aTaskQueueFactory,
+    const dom::RTCStatsTimestampMaker& aTimestampMaker,
+    UniquePtr<media::ShutdownBlockingTicket> aShutdownTicket)
+    : mSharedState(std::move(aSharedState)),
+      mClock(aTimestampMaker),
+      mShutdownTicket(std::move(aShutdownTicket)),
+      mCallThread(mSharedState->mCallWorkerThread),
+      mAudioDecoderFactory(mSharedState->mAudioDecoderFactory),
+      mVideoBitrateAllocatorFactory(std::move(aVideoBitrateAllocatorFactory)),
+      mEventLog(std::move(aEventLog)),
+      mTaskQueueFactory(std::move(aTaskQueueFactory)) {}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/libwebrtcglue/WebrtcCallWrapper.h b/dom/media/webrtc/libwebrtcglue/WebrtcCallWrapper.h
new file mode 100644
index 0000000000..17054c5f75
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/WebrtcCallWrapper.h
@@ -0,0 +1,114 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef DOM_MEDIA_WEBRTC_LIBWEBRTCGLUE_WEBRTCCALLWRAPPER_H_
+#define DOM_MEDIA_WEBRTC_LIBWEBRTCGLUE_WEBRTCCALLWRAPPER_H_
+
+#include <set>
+
+#include "domstubs.h"
+#include "jsapi/RTCStatsReport.h"
+#include "nsISupportsImpl.h"
+#include "SystemTime.h"
+
+// libwebrtc includes
+#include "api/video/builtin_video_bitrate_allocator_factory.h"
+#include "call/call.h"
+#include "call/call_config.h"
+
+namespace mozilla {
+class AbstractThread;
+class MediaSessionConduit;
+class SharedWebrtcState;
+
+namespace media {
+class ShutdownBlockingTicket;
+}
+
+// Wrap the webrtc.org Call class adding mozilla add/ref support.
+class WebrtcCallWrapper {
+ public:
+  typedef webrtc::CallConfig Config;
+
+  static RefPtr<WebrtcCallWrapper> Create(
+      const dom::RTCStatsTimestampMaker& aTimestampMaker,
+      UniquePtr<media::ShutdownBlockingTicket> aShutdownTicket,
+      const RefPtr<SharedWebrtcState>& aSharedState);
+
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(WebrtcCallWrapper)
+
+  // Don't allow copying/assigning.
+  WebrtcCallWrapper(const WebrtcCallWrapper&) = delete;
+  void operator=(const WebrtcCallWrapper&) = delete;
+
+  void SetCall(UniquePtr<webrtc::Call> aCall);
+
+  webrtc::Call* Call() const;
+
+  void UnsetRemoteSSRC(uint32_t aSsrc);
+
+  // Idempotent.
+  void RegisterConduit(MediaSessionConduit* conduit);
+
+  // Idempotent.
+  void UnregisterConduit(MediaSessionConduit* conduit);
+
+  // Allow destroying the Call instance on the Call worker thread.
+  //
+  // Note that shutdown is blocked until the Call instance is destroyed.
+  //
+  // This CallWrapper is designed to be sharable, and is held by several objects
+  // that are cycle-collectable. TaskQueueWrapper that the Call instances use
+  // for worker threads are based off SharedThreadPools, and will block
+  // xpcom-shutdown-threads until destroyed. The Call instance however will hold
+  // on to its worker threads until destruction.
+  //
+  // If the last ref to this CallWrapper is held to cycle collector shutdown we
+  // end up in a deadlock where cycle collector shutdown is required to destroy
+  // the SharedThreadPool that is blocking xpcom-shutdown-threads from finishing
+  // and triggering cycle collector shutdown.
+  //
+  // It would be nice to have the invariant where this class is immutable to the
+  // degree that mCall is const, but given the above that is not possible.
+  void Destroy();
+
+  const dom::RTCStatsTimestampMaker& GetTimestampMaker() const;
+
+ protected:
+  virtual ~WebrtcCallWrapper();
+
+  WebrtcCallWrapper(RefPtr<SharedWebrtcState> aSharedState,
+                    UniquePtr<webrtc::VideoBitrateAllocatorFactory>
+                        aVideoBitrateAllocatorFactory,
+                    UniquePtr<webrtc::RtcEventLog> aEventLog,
+                    UniquePtr<webrtc::TaskQueueFactory> aTaskQueueFactory,
+                    const dom::RTCStatsTimestampMaker& aTimestampMaker,
+                    UniquePtr<media::ShutdownBlockingTicket> aShutdownTicket);
+
+  const RefPtr<SharedWebrtcState> mSharedState;
+
+  // Allows conduits to know about one another, to avoid remote SSRC
+  // collisions.
+  std::set<RefPtr<MediaSessionConduit>> mConduits;
+  RTCStatsTimestampMakerRealtimeClock mClock;
+  UniquePtr<media::ShutdownBlockingTicket> mShutdownTicket;
+
+ public:
+  const RefPtr<AbstractThread> mCallThread;
+  const RefPtr<webrtc::AudioDecoderFactory> mAudioDecoderFactory;
+  const UniquePtr<webrtc::VideoBitrateAllocatorFactory>
+      mVideoBitrateAllocatorFactory;
+  const UniquePtr<webrtc::RtcEventLog> mEventLog;
+  const UniquePtr<webrtc::TaskQueueFactory> mTaskQueueFactory;
+
+ protected:
+  // Call worker thread only.
+  UniquePtr<webrtc::Call> mCall;
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/libwebrtcglue/WebrtcGmpVideoCodec.cpp b/dom/media/webrtc/libwebrtcglue/WebrtcGmpVideoCodec.cpp
new file mode 100644
index 0000000000..50e39b22d0
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/WebrtcGmpVideoCodec.cpp
@@ -0,0 +1,1043 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "WebrtcGmpVideoCodec.h"
+
+#include <utility>
+#include <vector>
+
+#include "GMPLog.h"
+#include "MainThreadUtils.h"
+#include "VideoConduit.h"
+#include "gmp-video-frame-encoded.h"
+#include "gmp-video-frame-i420.h"
+#include "mozilla/CheckedInt.h"
+#include "mozilla/EndianUtils.h"
+#include "mozilla/IntegerPrintfMacros.h"
+#include "mozilla/SyncRunnable.h"
+#include "nsServiceManagerUtils.h"
+#include "transport/runnable_utils.h"
+#include "api/video/video_frame_type.h"
+#include "common_video/include/video_frame_buffer.h"
+#include "media/base/media_constants.h"
+// #include "rtc_base/bind.h"
+
+namespace mozilla {
+
+// QP scaling thresholds.
+static const int kLowH264QpThreshold = 24;
+static const int kHighH264QpThreshold = 37;
+
+// Encoder.
+WebrtcGmpVideoEncoder::WebrtcGmpVideoEncoder(
+    const webrtc::SdpVideoFormat& aFormat, std::string aPCHandle)
+    : mGMP(nullptr),
+      mInitting(false),
+      mHost(nullptr),
+      mMaxPayloadSize(0),
+      mFormatParams(aFormat.parameters),
+      mCallbackMutex("WebrtcGmpVideoEncoder encoded callback mutex"),
+      mCallback(nullptr),
+      mPCHandle(std::move(aPCHandle)),
+      mInputImageMap("WebrtcGmpVideoEncoder::mInputImageMap") {
+  mCodecParams.mGMPApiVersion = 0;
+  mCodecParams.mCodecType = kGMPVideoCodecInvalid;
+  mCodecParams.mPLType = 0;
+  mCodecParams.mWidth = 0;
+  mCodecParams.mHeight = 0;
+  mCodecParams.mStartBitrate = 0;
+  mCodecParams.mMaxBitrate = 0;
+  mCodecParams.mMinBitrate = 0;
+  mCodecParams.mMaxFramerate = 0;
+  mCodecParams.mFrameDroppingOn = false;
+  mCodecParams.mKeyFrameInterval = 0;
+  mCodecParams.mQPMax = 0;
+  mCodecParams.mNumberOfSimulcastStreams = 0;
+  mCodecParams.mMode = kGMPCodecModeInvalid;
+  mCodecParams.mLogLevel = GetGMPLibraryLogLevel();
+  MOZ_ASSERT(!mPCHandle.empty());
+}
+
+WebrtcGmpVideoEncoder::~WebrtcGmpVideoEncoder() {
+  // We should not have been destroyed if we never closed our GMP
+  MOZ_ASSERT(!mGMP);
+}
+
+static int WebrtcFrameTypeToGmpFrameType(webrtc::VideoFrameType aIn,
+                                         GMPVideoFrameType* aOut) {
+  MOZ_ASSERT(aOut);
+  switch (aIn) {
+    case webrtc::VideoFrameType::kVideoFrameKey:
+      *aOut = kGMPKeyFrame;
+      break;
+    case webrtc::VideoFrameType::kVideoFrameDelta:
+      *aOut = kGMPDeltaFrame;
+      break;
+    case webrtc::VideoFrameType::kEmptyFrame:
+      *aOut = kGMPSkipFrame;
+      break;
+    default:
+      MOZ_CRASH("Unexpected webrtc::FrameType");
+  }
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+static int GmpFrameTypeToWebrtcFrameType(GMPVideoFrameType aIn,
+                                         webrtc::VideoFrameType* aOut) {
+  MOZ_ASSERT(aOut);
+  switch (aIn) {
+    case kGMPKeyFrame:
+      *aOut = webrtc::VideoFrameType::kVideoFrameKey;
+      break;
+    case kGMPDeltaFrame:
+      *aOut = webrtc::VideoFrameType::kVideoFrameDelta;
+      break;
+    case kGMPSkipFrame:
+      *aOut = webrtc::VideoFrameType::kEmptyFrame;
+      break;
+    default:
+      MOZ_CRASH("Unexpected GMPVideoFrameType");
+  }
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+static int SizeNumBytes(GMPBufferType aBufferType) {
+  switch (aBufferType) {
+    case GMP_BufferSingle:
+      return 0;
+    case GMP_BufferLength8:
+      return 1;
+    case GMP_BufferLength16:
+      return 2;
+    case GMP_BufferLength24:
+      return 3;
+    case GMP_BufferLength32:
+      return 4;
+    default:
+      MOZ_CRASH("Unexpected buffer type");
+  }
+}
+
+int32_t WebrtcGmpVideoEncoder::InitEncode(
+    const webrtc::VideoCodec* aCodecSettings,
+    const webrtc::VideoEncoder::Settings& aSettings) {
+  if (!mMPS) {
+    mMPS = do_GetService("@mozilla.org/gecko-media-plugin-service;1");
+  }
+  MOZ_ASSERT(mMPS);
+
+  if (!mGMPThread) {
+    if (NS_WARN_IF(NS_FAILED(mMPS->GetThread(getter_AddRefs(mGMPThread))))) {
+      return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+  }
+
+  MOZ_ASSERT(aCodecSettings->numberOfSimulcastStreams == 1,
+             "Simulcast not implemented for GMP-H264");
+
+  // Bug XXXXXX: transfer settings from codecSettings to codec.
+  GMPVideoCodec codecParams;
+  memset(&codecParams, 0, sizeof(codecParams));
+
+  codecParams.mGMPApiVersion = kGMPVersion34;
+  codecParams.mLogLevel = GetGMPLibraryLogLevel();
+  codecParams.mStartBitrate = aCodecSettings->startBitrate;
+  codecParams.mMinBitrate = aCodecSettings->minBitrate;
+  codecParams.mMaxBitrate = aCodecSettings->maxBitrate;
+  codecParams.mMaxFramerate = aCodecSettings->maxFramerate;
+
+  memset(&mCodecSpecificInfo.codecSpecific, 0,
+         sizeof(mCodecSpecificInfo.codecSpecific));
+  mCodecSpecificInfo.codecType = webrtc::kVideoCodecH264;
+  mCodecSpecificInfo.codecSpecific.H264.packetization_mode =
+      mFormatParams.count(cricket::kH264FmtpPacketizationMode) == 1 &&
+              mFormatParams.at(cricket::kH264FmtpPacketizationMode) == "1"
+          ? webrtc::H264PacketizationMode::NonInterleaved
+          : webrtc::H264PacketizationMode::SingleNalUnit;
+
+  uint32_t maxPayloadSize = aSettings.max_payload_size;
+  if (mCodecSpecificInfo.codecSpecific.H264.packetization_mode ==
+      webrtc::H264PacketizationMode::NonInterleaved) {
+    maxPayloadSize = 0;  // No limit, use FUAs
+  }
+
+  if (aCodecSettings->mode == webrtc::VideoCodecMode::kScreensharing) {
+    codecParams.mMode = kGMPScreensharing;
+  } else {
+    codecParams.mMode = kGMPRealtimeVideo;
+  }
+
+  codecParams.mWidth = aCodecSettings->width;
+  codecParams.mHeight = aCodecSettings->height;
+
+  RefPtr<GmpInitDoneRunnable> initDone(new GmpInitDoneRunnable(mPCHandle));
+  mGMPThread->Dispatch(
+      WrapRunnableNM(WebrtcGmpVideoEncoder::InitEncode_g,
+                     RefPtr<WebrtcGmpVideoEncoder>(this), codecParams,
+                     aSettings.number_of_cores, maxPayloadSize, initDone),
+      NS_DISPATCH_NORMAL);
+
+  // Since init of the GMP encoder is a multi-step async dispatch (including
+  // dispatches to main), and since this function is invoked on main, there's
+  // no safe way to block until this init is done. If an error occurs, we'll
+  // handle it later.
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+/* static */
+void WebrtcGmpVideoEncoder::InitEncode_g(
+    const RefPtr<WebrtcGmpVideoEncoder>& aThis,
+    const GMPVideoCodec& aCodecParams, int32_t aNumberOfCores,
+    uint32_t aMaxPayloadSize, const RefPtr<GmpInitDoneRunnable>& aInitDone) {
+  nsTArray<nsCString> tags;
+  tags.AppendElement("h264"_ns);
+  UniquePtr<GetGMPVideoEncoderCallback> callback(
+      new InitDoneCallback(aThis, aInitDone, aCodecParams));
+  aThis->mInitting = true;
+  aThis->mMaxPayloadSize = aMaxPayloadSize;
+  nsresult rv = aThis->mMPS->GetGMPVideoEncoder(nullptr, &tags, ""_ns,
+                                                std::move(callback));
+  if (NS_WARN_IF(NS_FAILED(rv))) {
+    GMP_LOG_DEBUG("GMP Encode: GetGMPVideoEncoder failed");
+    aThis->Close_g();
+    aInitDone->Dispatch(WEBRTC_VIDEO_CODEC_ERROR,
+                        "GMP Encode: GetGMPVideoEncoder failed");
+  }
+}
+
+int32_t WebrtcGmpVideoEncoder::GmpInitDone(GMPVideoEncoderProxy* aGMP,
+                                           GMPVideoHost* aHost,
+                                           std::string* aErrorOut) {
+  if (!mInitting || !aGMP || !aHost) {
+    *aErrorOut =
+        "GMP Encode: Either init was aborted, "
+        "or init failed to supply either a GMP Encoder or GMP host.";
+    if (aGMP) {
+      // This could destroy us, since aGMP may be the last thing holding a ref
+      // Return immediately.
+      aGMP->Close();
+    }
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  mInitting = false;
+
+  if (mGMP && mGMP != aGMP) {
+    Close_g();
+  }
+
+  mGMP = aGMP;
+  mHost = aHost;
+  mCachedPluginId = Some(mGMP->GetPluginId());
+  mInitPluginEvent.Notify(*mCachedPluginId);
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t WebrtcGmpVideoEncoder::GmpInitDone(GMPVideoEncoderProxy* aGMP,
+                                           GMPVideoHost* aHost,
+                                           const GMPVideoCodec& aCodecParams,
+                                           std::string* aErrorOut) {
+  int32_t r = GmpInitDone(aGMP, aHost, aErrorOut);
+  if (r != WEBRTC_VIDEO_CODEC_OK) {
+    // We might have been destroyed if GmpInitDone failed.
+    // Return immediately.
+    return r;
+  }
+  mCodecParams = aCodecParams;
+  return InitEncoderForSize(aCodecParams.mWidth, aCodecParams.mHeight,
+                            aErrorOut);
+}
+
+void WebrtcGmpVideoEncoder::Close_g() {
+  GMPVideoEncoderProxy* gmp(mGMP);
+  mGMP = nullptr;
+  mHost = nullptr;
+  mInitting = false;
+
+  if (mCachedPluginId) {
+    mReleasePluginEvent.Notify(*mCachedPluginId);
+  }
+  mCachedPluginId = Nothing();
+
+  if (gmp) {
+    // Do this last, since this could cause us to be destroyed
+    gmp->Close();
+  }
+}
+
+int32_t WebrtcGmpVideoEncoder::InitEncoderForSize(unsigned short aWidth,
+                                                  unsigned short aHeight,
+                                                  std::string* aErrorOut) {
+  mCodecParams.mWidth = aWidth;
+  mCodecParams.mHeight = aHeight;
+  // Pass dummy codecSpecific data for now...
+  nsTArray<uint8_t> codecSpecific;
+
+  GMPErr err =
+      mGMP->InitEncode(mCodecParams, codecSpecific, this, 1, mMaxPayloadSize);
+  if (err != GMPNoErr) {
+    *aErrorOut = "GMP Encode: InitEncode failed";
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t WebrtcGmpVideoEncoder::Encode(
+    const webrtc::VideoFrame& aInputImage,
+    const std::vector<webrtc::VideoFrameType>* aFrameTypes) {
+  MOZ_ASSERT(aInputImage.width() >= 0 && aInputImage.height() >= 0);
+  if (!aFrameTypes) {
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  // It is safe to copy aInputImage here because the frame buffer is held by
+  // a refptr.
+  mGMPThread->Dispatch(WrapRunnableNM(&WebrtcGmpVideoEncoder::Encode_g,
+                                      RefPtr<WebrtcGmpVideoEncoder>(this),
+                                      aInputImage, *aFrameTypes),
+                       NS_DISPATCH_NORMAL);
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+void WebrtcGmpVideoEncoder::RegetEncoderForResolutionChange(
+    uint32_t aWidth, uint32_t aHeight,
+    const RefPtr<GmpInitDoneRunnable>& aInitDone) {
+  Close_g();
+
+  UniquePtr<GetGMPVideoEncoderCallback> callback(
+      new InitDoneForResolutionChangeCallback(this, aInitDone, aWidth,
+                                              aHeight));
+
+  // OpenH264 codec (at least) can't handle dynamic input resolution changes
+  // re-init the plugin when the resolution changes
+  // XXX allow codec to indicate it doesn't need re-init!
+  nsTArray<nsCString> tags;
+  tags.AppendElement("h264"_ns);
+  mInitting = true;
+  if (NS_WARN_IF(NS_FAILED(mMPS->GetGMPVideoEncoder(nullptr, &tags, ""_ns,
+                                                    std::move(callback))))) {
+    aInitDone->Dispatch(WEBRTC_VIDEO_CODEC_ERROR,
+                        "GMP Encode: GetGMPVideoEncoder failed");
+  }
+}
+
+void WebrtcGmpVideoEncoder::Encode_g(
+    const RefPtr<WebrtcGmpVideoEncoder>& aEncoder,
+    webrtc::VideoFrame aInputImage,
+    std::vector<webrtc::VideoFrameType> aFrameTypes) {
+  if (!aEncoder->mGMP) {
+    // destroyed via Terminate(), failed to init, or just not initted yet
+    GMP_LOG_DEBUG("GMP Encode: not initted yet");
+    return;
+  }
+  MOZ_ASSERT(aEncoder->mHost);
+
+  if (static_cast<uint32_t>(aInputImage.width()) !=
+          aEncoder->mCodecParams.mWidth ||
+      static_cast<uint32_t>(aInputImage.height()) !=
+          aEncoder->mCodecParams.mHeight) {
+    GMP_LOG_DEBUG("GMP Encode: resolution change from %ux%u to %dx%d",
+                  aEncoder->mCodecParams.mWidth, aEncoder->mCodecParams.mHeight,
+                  aInputImage.width(), aInputImage.height());
+
+    RefPtr<GmpInitDoneRunnable> initDone(
+        new GmpInitDoneRunnable(aEncoder->mPCHandle));
+    aEncoder->RegetEncoderForResolutionChange(aInputImage.width(),
+                                              aInputImage.height(), initDone);
+    if (!aEncoder->mGMP) {
+      // We needed to go async to re-get the encoder. Bail.
+      return;
+    }
+  }
+
+  GMPVideoFrame* ftmp = nullptr;
+  GMPErr err = aEncoder->mHost->CreateFrame(kGMPI420VideoFrame, &ftmp);
+  if (err != GMPNoErr) {
+    GMP_LOG_DEBUG("GMP Encode: failed to create frame on host");
+    return;
+  }
+  GMPUniquePtr<GMPVideoi420Frame> frame(static_cast<GMPVideoi420Frame*>(ftmp));
+  const webrtc::I420BufferInterface* input_image =
+      aInputImage.video_frame_buffer()->GetI420();
+  // check for overflow of stride * height
+  CheckedInt32 ysize =
+      CheckedInt32(input_image->StrideY()) * input_image->height();
+  MOZ_RELEASE_ASSERT(ysize.isValid());
+  // I will assume that if that doesn't overflow, the others case - YUV
+  // 4:2:0 has U/V widths <= Y, even with alignment issues.
+  err = frame->CreateFrame(
+      ysize.value(), input_image->DataY(),
+      input_image->StrideU() * ((input_image->height() + 1) / 2),
+      input_image->DataU(),
+      input_image->StrideV() * ((input_image->height() + 1) / 2),
+      input_image->DataV(), input_image->width(), input_image->height(),
+      input_image->StrideY(), input_image->StrideU(), input_image->StrideV());
+  if (err != GMPNoErr) {
+    GMP_LOG_DEBUG("GMP Encode: failed to create frame");
+    return;
+  }
+  frame->SetTimestamp((aInputImage.timestamp() * 1000ll) /
+                      90);  // note: rounds down!
+  // frame->SetDuration(1000000ll/30); // XXX base duration on measured current
+  // FPS - or don't bother
+
+  // Bug XXXXXX: Set codecSpecific info
+  GMPCodecSpecificInfo info;
+  memset(&info, 0, sizeof(info));
+  info.mCodecType = kGMPVideoCodecH264;
+  nsTArray<uint8_t> codecSpecificInfo;
+  codecSpecificInfo.AppendElements((uint8_t*)&info,
+                                   sizeof(GMPCodecSpecificInfo));
+
+  nsTArray<GMPVideoFrameType> gmp_frame_types;
+  for (auto it = aFrameTypes.begin(); it != aFrameTypes.end(); ++it) {
+    GMPVideoFrameType ft;
+
+    int32_t ret = WebrtcFrameTypeToGmpFrameType(*it, &ft);
+    if (ret != WEBRTC_VIDEO_CODEC_OK) {
+      GMP_LOG_DEBUG(
+          "GMP Encode: failed to map webrtc frame type to gmp frame type");
+      return;
+    }
+
+    gmp_frame_types.AppendElement(ft);
+  }
+
+  {
+    auto inputImageMap = aEncoder->mInputImageMap.Lock();
+    DebugOnly<bool> inserted = false;
+    std::tie(std::ignore, inserted) = inputImageMap->insert(
+        {frame->Timestamp(), {aInputImage.timestamp_us()}});
+    MOZ_ASSERT(inserted, "Duplicate timestamp");
+  }
+
+  GMP_LOG_DEBUG("GMP Encode: %" PRIu64, (frame->Timestamp()));
+  err = aEncoder->mGMP->Encode(std::move(frame), codecSpecificInfo,
+                               gmp_frame_types);
+  if (err != GMPNoErr) {
+    GMP_LOG_DEBUG("GMP Encode: failed to encode frame");
+  }
+}
+
+int32_t WebrtcGmpVideoEncoder::RegisterEncodeCompleteCallback(
+    webrtc::EncodedImageCallback* aCallback) {
+  MutexAutoLock lock(mCallbackMutex);
+  mCallback = aCallback;
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+/* static */
+void WebrtcGmpVideoEncoder::ReleaseGmp_g(
+    const RefPtr<WebrtcGmpVideoEncoder>& aEncoder) {
+  aEncoder->Close_g();
+}
+
+int32_t WebrtcGmpVideoEncoder::Shutdown() {
+  GMP_LOG_DEBUG("GMP Released:");
+  RegisterEncodeCompleteCallback(nullptr);
+  if (mGMPThread) {
+    mGMPThread->Dispatch(WrapRunnableNM(&WebrtcGmpVideoEncoder::ReleaseGmp_g,
+                                        RefPtr<WebrtcGmpVideoEncoder>(this)),
+                         NS_DISPATCH_NORMAL);
+  }
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t WebrtcGmpVideoEncoder::SetRates(
+    const webrtc::VideoEncoder::RateControlParameters& aParameters) {
+  MOZ_ASSERT(mGMPThread);
+  MOZ_ASSERT(aParameters.bitrate.IsSpatialLayerUsed(0));
+  MOZ_ASSERT(!aParameters.bitrate.HasBitrate(0, 1),
+             "No simulcast support for H264");
+  MOZ_ASSERT(!aParameters.bitrate.IsSpatialLayerUsed(1),
+             "No simulcast support for H264");
+  mGMPThread->Dispatch(
+      WrapRunnableNM(&WebrtcGmpVideoEncoder::SetRates_g,
+                     RefPtr<WebrtcGmpVideoEncoder>(this),
+                     aParameters.bitrate.GetBitrate(0, 0) / 1000,
+                     aParameters.framerate_fps > 0.0
+                         ? Some(aParameters.framerate_fps)
+                         : Nothing()),
+      NS_DISPATCH_NORMAL);
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+WebrtcVideoEncoder::EncoderInfo WebrtcGmpVideoEncoder::GetEncoderInfo() const {
+  WebrtcVideoEncoder::EncoderInfo info;
+  info.supports_native_handle = false;
+  info.implementation_name = "GMPOpenH264";
+  info.scaling_settings = WebrtcVideoEncoder::ScalingSettings(
+      kLowH264QpThreshold, kHighH264QpThreshold);
+  info.is_hardware_accelerated = false;
+  info.supports_simulcast = false;
+  return info;
+}
+
+/* static */
+int32_t WebrtcGmpVideoEncoder::SetRates_g(RefPtr<WebrtcGmpVideoEncoder> aThis,
+                                          uint32_t aNewBitRateKbps,
+                                          Maybe<double> aFrameRate) {
+  if (!aThis->mGMP) {
+    // destroyed via Terminate()
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  GMPErr err = aThis->mGMP->SetRates(
+      aNewBitRateKbps, aFrameRate
+                           .map([](double aFr) {
+                             // Avoid rounding to 0
+                             return std::max(1U, static_cast<uint32_t>(aFr));
+                           })
+                           .valueOr(aThis->mCodecParams.mMaxFramerate));
+  if (err != GMPNoErr) {
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+// GMPVideoEncoderCallback virtual functions.
+void WebrtcGmpVideoEncoder::Terminated() {
+  GMP_LOG_DEBUG("GMP Encoder Terminated: %p", (void*)this);
+
+  GMPVideoEncoderProxy* gmp(mGMP);
+  mGMP = nullptr;
+  mHost = nullptr;
+  mInitting = false;
+
+  if (gmp) {
+    // Do this last, since this could cause us to be destroyed
+    gmp->Close();
+  }
+
+  // Could now notify that it's dead
+}
+
+void WebrtcGmpVideoEncoder::Encoded(
+    GMPVideoEncodedFrame* aEncodedFrame,
+    const nsTArray<uint8_t>& aCodecSpecificInfo) {
+  webrtc::Timestamp capture_time = webrtc::Timestamp::Micros(0);
+  {
+    auto inputImageMap = mInputImageMap.Lock();
+    auto handle = inputImageMap->extract(aEncodedFrame->TimeStamp());
+    MOZ_ASSERT(handle);
+    if (handle) {
+      capture_time = webrtc::Timestamp::Micros(handle.mapped().timestamp_us);
+    }
+  }
+
+  MutexAutoLock lock(mCallbackMutex);
+  if (!mCallback) {
+    return;
+  }
+
+  webrtc::VideoFrameType ft;
+  GmpFrameTypeToWebrtcFrameType(aEncodedFrame->FrameType(), &ft);
+  uint32_t timestamp = (aEncodedFrame->TimeStamp() * 90ll + 999) / 1000;
+
+  GMP_LOG_DEBUG("GMP Encoded: %" PRIu64 ", type %d, len %d",
+                aEncodedFrame->TimeStamp(), aEncodedFrame->BufferType(),
+                aEncodedFrame->Size());
+
+  if (!aEncodedFrame->Buffer()) {
+    GMP_LOG_ERROR("GMP plugin returned null buffer");
+    return;
+  }
+
+  // Libwebrtc's RtpPacketizerH264 expects a 3- or 4-byte NALU start sequence
+  // before the start of the NALU payload. {0,0,1} or {0,0,0,1}. We set this
+  // in-place. Any other length of the length field we reject.
+
+  const int sizeNumBytes = SizeNumBytes(aEncodedFrame->BufferType());
+  uint32_t unitOffset = 0;
+  uint32_t unitSize = 0;
+  // Make sure we don't read past the end of the buffer getting the size
+  while (unitOffset + sizeNumBytes < aEncodedFrame->Size()) {
+    uint8_t* unitBuffer = aEncodedFrame->Buffer() + unitOffset;
+    switch (aEncodedFrame->BufferType()) {
+      case GMP_BufferLength24: {
+#if MOZ_LITTLE_ENDIAN()
+        unitSize = (static_cast<uint32_t>(*unitBuffer)) |
+                   (static_cast<uint32_t>(*(unitBuffer + 1)) << 8) |
+                   (static_cast<uint32_t>(*(unitBuffer + 2)) << 16);
+#else
+        unitSize = (static_cast<uint32_t>(*unitBuffer) << 16) |
+                   (static_cast<uint32_t>(*(unitBuffer + 1)) << 8) |
+                   (static_cast<uint32_t>(*(unitBuffer + 2)));
+#endif
+        const uint8_t startSequence[] = {0, 0, 1};
+        if (memcmp(unitBuffer, startSequence, 3) == 0) {
+          // This is a bug in OpenH264 where it misses to convert the NALU start
+          // sequence to the NALU size per the GMP protocol. We mitigate this by
+          // letting it through as this is what libwebrtc already expects and
+          // scans for.
+          unitSize = aEncodedFrame->Size() - 3;
+          break;
+        }
+        memcpy(unitBuffer, startSequence, 3);
+        break;
+      }
+      case GMP_BufferLength32: {
+#if MOZ_LITTLE_ENDIAN()
+        unitSize = LittleEndian::readUint32(unitBuffer);
+#else
+        unitSize = BigEndian::readUint32(unitBuffer);
+#endif
+        const uint8_t startSequence[] = {0, 0, 0, 1};
+        if (memcmp(unitBuffer, startSequence, 4) == 0) {
+          // This is a bug in OpenH264 where it misses to convert the NALU start
+          // sequence to the NALU size per the GMP protocol. We mitigate this by
+          // letting it through as this is what libwebrtc already expects and
+          // scans for.
+          unitSize = aEncodedFrame->Size() - 4;
+          break;
+        }
+        memcpy(unitBuffer, startSequence, 4);
+        break;
+      }
+      default:
+        GMP_LOG_ERROR("GMP plugin returned type we cannot handle (%d)",
+                      aEncodedFrame->BufferType());
+        return;
+    }
+
+    MOZ_ASSERT(unitSize != 0);
+    MOZ_ASSERT(unitOffset + sizeNumBytes + unitSize <= aEncodedFrame->Size());
+    if (unitSize == 0 ||
+        unitOffset + sizeNumBytes + unitSize > aEncodedFrame->Size()) {
+      // XXX Should we kill the plugin for returning extra bytes? Probably
+      GMP_LOG_ERROR(
+          "GMP plugin returned badly formatted encoded data: "
+          "unitOffset=%u, sizeNumBytes=%d, unitSize=%u, size=%u",
+          unitOffset, sizeNumBytes, unitSize, aEncodedFrame->Size());
+      return;
+    }
+
+    unitOffset += sizeNumBytes + unitSize;
+  }
+
+  if (unitOffset != aEncodedFrame->Size()) {
+    // At most 3 bytes can be left over, depending on buffertype
+    GMP_LOG_DEBUG("GMP plugin returned %u extra bytes",
+                  aEncodedFrame->Size() - unitOffset);
+  }
+
+  webrtc::EncodedImage unit;
+  unit.SetEncodedData(webrtc::EncodedImageBuffer::Create(
+      aEncodedFrame->Buffer(), aEncodedFrame->Size()));
+  unit._frameType = ft;
+  unit.SetTimestamp(timestamp);
+  unit.capture_time_ms_ = capture_time.ms();
+  unit._encodedWidth = aEncodedFrame->EncodedWidth();
+  unit._encodedHeight = aEncodedFrame->EncodedHeight();
+
+  // Parse QP.
+  mH264BitstreamParser.ParseBitstream(unit);
+  unit.qp_ = mH264BitstreamParser.GetLastSliceQp().value_or(-1);
+
+  // TODO: Currently the OpenH264 codec does not preserve any codec
+  //       specific info passed into it and just returns default values.
+  //       If this changes in the future, it would be nice to get rid of
+  //       mCodecSpecificInfo.
+  mCallback->OnEncodedImage(unit, &mCodecSpecificInfo);
+}
+
+// Decoder.
+WebrtcGmpVideoDecoder::WebrtcGmpVideoDecoder(std::string aPCHandle,
+                                             TrackingId aTrackingId)
+    : mGMP(nullptr),
+      mInitting(false),
+      mHost(nullptr),
+      mCallbackMutex("WebrtcGmpVideoDecoder decoded callback mutex"),
+      mCallback(nullptr),
+      mDecoderStatus(GMPNoErr),
+      mPCHandle(std::move(aPCHandle)),
+      mTrackingId(std::move(aTrackingId)) {
+  MOZ_ASSERT(!mPCHandle.empty());
+}
+
+WebrtcGmpVideoDecoder::~WebrtcGmpVideoDecoder() {
+  // We should not have been destroyed if we never closed our GMP
+  MOZ_ASSERT(!mGMP);
+}
+
+bool WebrtcGmpVideoDecoder::Configure(
+    const webrtc::VideoDecoder::Settings& settings) {
+  if (!mMPS) {
+    mMPS = do_GetService("@mozilla.org/gecko-media-plugin-service;1");
+  }
+  MOZ_ASSERT(mMPS);
+
+  if (!mGMPThread) {
+    if (NS_WARN_IF(NS_FAILED(mMPS->GetThread(getter_AddRefs(mGMPThread))))) {
+      return false;
+    }
+  }
+
+  RefPtr<GmpInitDoneRunnable> initDone(new GmpInitDoneRunnable(mPCHandle));
+  mGMPThread->Dispatch(
+      WrapRunnableNM(&WebrtcGmpVideoDecoder::Configure_g,
+                     RefPtr<WebrtcGmpVideoDecoder>(this), settings, initDone),
+      NS_DISPATCH_NORMAL);
+
+  return true;
+}
+
+/* static */
+void WebrtcGmpVideoDecoder::Configure_g(
+    const RefPtr<WebrtcGmpVideoDecoder>& aThis,
+    const webrtc::VideoDecoder::Settings& settings,  // unused
+    const RefPtr<GmpInitDoneRunnable>& aInitDone) {
+  nsTArray<nsCString> tags;
+  tags.AppendElement("h264"_ns);
+  UniquePtr<GetGMPVideoDecoderCallback> callback(
+      new InitDoneCallback(aThis, aInitDone));
+  aThis->mInitting = true;
+  nsresult rv = aThis->mMPS->GetGMPVideoDecoder(nullptr, &tags, ""_ns,
+                                                std::move(callback));
+  if (NS_WARN_IF(NS_FAILED(rv))) {
+    GMP_LOG_DEBUG("GMP Decode: GetGMPVideoDecoder failed");
+    aThis->Close_g();
+    aInitDone->Dispatch(WEBRTC_VIDEO_CODEC_ERROR,
+                        "GMP Decode: GetGMPVideoDecoder failed.");
+  }
+}
+
+int32_t WebrtcGmpVideoDecoder::GmpInitDone(GMPVideoDecoderProxy* aGMP,
+                                           GMPVideoHost* aHost,
+                                           std::string* aErrorOut) {
+  if (!mInitting || !aGMP || !aHost) {
+    *aErrorOut =
+        "GMP Decode: Either init was aborted, "
+        "or init failed to supply either a GMP decoder or GMP host.";
+    if (aGMP) {
+      // This could destroy us, since aGMP may be the last thing holding a ref
+      // Return immediately.
+      aGMP->Close();
+    }
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  mInitting = false;
+
+  if (mGMP && mGMP != aGMP) {
+    Close_g();
+  }
+
+  mGMP = aGMP;
+  mHost = aHost;
+  mCachedPluginId = Some(mGMP->GetPluginId());
+  mInitPluginEvent.Notify(*mCachedPluginId);
+  // Bug XXXXXX: transfer settings from codecSettings to codec.
+  GMPVideoCodec codec;
+  memset(&codec, 0, sizeof(codec));
+  codec.mGMPApiVersion = kGMPVersion34;
+  codec.mLogLevel = GetGMPLibraryLogLevel();
+
+  // XXX this is currently a hack
+  // GMPVideoCodecUnion codecSpecific;
+  // memset(&codecSpecific, 0, sizeof(codecSpecific));
+  nsTArray<uint8_t> codecSpecific;
+  nsresult rv = mGMP->InitDecode(codec, codecSpecific, this, 1);
+  if (NS_FAILED(rv)) {
+    *aErrorOut = "GMP Decode: InitDecode failed";
+    mQueuedFrames.Clear();
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  // now release any frames that got queued waiting for InitDone
+  if (!mQueuedFrames.IsEmpty()) {
+    // So we're safe to call Decode_g(), which asserts it's empty
+    nsTArray<UniquePtr<GMPDecodeData>> temp = std::move(mQueuedFrames);
+    for (auto& queued : temp) {
+      Decode_g(RefPtr<WebrtcGmpVideoDecoder>(this), std::move(queued));
+    }
+  }
+
+  // This is an ugly solution to asynchronous decoding errors
+  // from Decode_g() not being returned to the synchronous Decode() method.
+  // If we don't return an error code at this point, our caller ultimately won't
+  // know to request a PLI and the video stream will remain frozen unless an IDR
+  // happens to arrive for other reasons. Bug 1492852 tracks implementing a
+  // proper solution.
+  if (mDecoderStatus != GMPNoErr) {
+    GMP_LOG_ERROR("%s: Decoder status is bad (%u)!", __PRETTY_FUNCTION__,
+                  static_cast<unsigned>(mDecoderStatus));
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+void WebrtcGmpVideoDecoder::Close_g() {
+  GMPVideoDecoderProxy* gmp(mGMP);
+  mGMP = nullptr;
+  mHost = nullptr;
+  mInitting = false;
+
+  if (mCachedPluginId) {
+    mReleasePluginEvent.Notify(*mCachedPluginId);
+  }
+  mCachedPluginId = Nothing();
+
+  if (gmp) {
+    // Do this last, since this could cause us to be destroyed
+    gmp->Close();
+  }
+}
+
+int32_t WebrtcGmpVideoDecoder::Decode(const webrtc::EncodedImage& aInputImage,
+                                      bool aMissingFrames,
+                                      int64_t aRenderTimeMs) {
+  MOZ_ASSERT(mGMPThread);
+  MOZ_ASSERT(!NS_IsMainThread());
+  if (!aInputImage.size()) {
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  MediaInfoFlag flag = MediaInfoFlag::None;
+  flag |= (aInputImage._frameType == webrtc::VideoFrameType::kVideoFrameKey
+               ? MediaInfoFlag::KeyFrame
+               : MediaInfoFlag::NonKeyFrame);
+  flag |= MediaInfoFlag::SoftwareDecoding;
+  flag |= MediaInfoFlag::VIDEO_H264;
+  mPerformanceRecorder.Start((aInputImage.Timestamp() * 1000ll) / 90,
+                             "WebrtcGmpVideoDecoder"_ns, mTrackingId, flag);
+
+  // This is an ugly solution to asynchronous decoding errors
+  // from Decode_g() not being returned to the synchronous Decode() method.
+  // If we don't return an error code at this point, our caller ultimately won't
+  // know to request a PLI and the video stream will remain frozen unless an IDR
+  // happens to arrive for other reasons. Bug 1492852 tracks implementing a
+  // proper solution.
+  auto decodeData =
+      MakeUnique<GMPDecodeData>(aInputImage, aMissingFrames, aRenderTimeMs);
+
+  mGMPThread->Dispatch(WrapRunnableNM(&WebrtcGmpVideoDecoder::Decode_g,
+                                      RefPtr<WebrtcGmpVideoDecoder>(this),
+                                      std::move(decodeData)),
+                       NS_DISPATCH_NORMAL);
+
+  if (mDecoderStatus != GMPNoErr) {
+    GMP_LOG_ERROR("%s: Decoder status is bad (%u)!", __PRETTY_FUNCTION__,
+                  static_cast<unsigned>(mDecoderStatus));
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+/* static */
+void WebrtcGmpVideoDecoder::Decode_g(const RefPtr<WebrtcGmpVideoDecoder>& aThis,
+                                     UniquePtr<GMPDecodeData>&& aDecodeData) {
+  if (!aThis->mGMP) {
+    if (aThis->mInitting) {
+      // InitDone hasn't been called yet (race)
+      aThis->mQueuedFrames.AppendElement(std::move(aDecodeData));
+      return;
+    }
+    // destroyed via Terminate(), failed to init, or just not initted yet
+    GMP_LOG_DEBUG("GMP Decode: not initted yet");
+
+    aThis->mDecoderStatus = GMPDecodeErr;
+    return;
+  }
+
+  MOZ_ASSERT(aThis->mQueuedFrames.IsEmpty());
+  MOZ_ASSERT(aThis->mHost);
+
+  GMPVideoFrame* ftmp = nullptr;
+  GMPErr err = aThis->mHost->CreateFrame(kGMPEncodedVideoFrame, &ftmp);
+  if (err != GMPNoErr) {
+    GMP_LOG_ERROR("%s: CreateFrame failed (%u)!", __PRETTY_FUNCTION__,
+                  static_cast<unsigned>(err));
+    aThis->mDecoderStatus = err;
+    return;
+  }
+
+  GMPUniquePtr<GMPVideoEncodedFrame> frame(
+      static_cast<GMPVideoEncodedFrame*>(ftmp));
+  err = frame->CreateEmptyFrame(aDecodeData->mImage.size());
+  if (err != GMPNoErr) {
+    GMP_LOG_ERROR("%s: CreateEmptyFrame failed (%u)!", __PRETTY_FUNCTION__,
+                  static_cast<unsigned>(err));
+    aThis->mDecoderStatus = err;
+    return;
+  }
+
+  // XXX At this point, we only will get mode1 data (a single length and a
+  // buffer) Session_info.cc/etc code needs to change to support mode 0.
+  *(reinterpret_cast<uint32_t*>(frame->Buffer())) = frame->Size();
+
+  // XXX It'd be wonderful not to have to memcpy the encoded data!
+  memcpy(frame->Buffer() + 4, aDecodeData->mImage.data() + 4,
+         frame->Size() - 4);
+
+  frame->SetEncodedWidth(aDecodeData->mImage._encodedWidth);
+  frame->SetEncodedHeight(aDecodeData->mImage._encodedHeight);
+  frame->SetTimeStamp((aDecodeData->mImage.Timestamp() * 1000ll) /
+                      90);  // rounds down
+  frame->SetCompleteFrame(
+      true);  // upstream no longer deals with incomplete frames
+  frame->SetBufferType(GMP_BufferLength32);
+
+  GMPVideoFrameType ft;
+  int32_t ret =
+      WebrtcFrameTypeToGmpFrameType(aDecodeData->mImage._frameType, &ft);
+  if (ret != WEBRTC_VIDEO_CODEC_OK) {
+    GMP_LOG_ERROR("%s: WebrtcFrameTypeToGmpFrameType failed (%u)!",
+                  __PRETTY_FUNCTION__, static_cast<unsigned>(ret));
+    aThis->mDecoderStatus = GMPDecodeErr;
+    return;
+  }
+
+  // Bug XXXXXX: Set codecSpecific info
+  GMPCodecSpecificInfo info;
+  memset(&info, 0, sizeof(info));
+  info.mCodecType = kGMPVideoCodecH264;
+  info.mCodecSpecific.mH264.mSimulcastIdx = 0;
+  nsTArray<uint8_t> codecSpecificInfo;
+  codecSpecificInfo.AppendElements((uint8_t*)&info,
+                                   sizeof(GMPCodecSpecificInfo));
+
+  GMP_LOG_DEBUG("GMP Decode: %" PRIu64 ", len %zu%s", frame->TimeStamp(),
+                aDecodeData->mImage.size(),
+                ft == kGMPKeyFrame ? ", KeyFrame" : "");
+
+  nsresult rv =
+      aThis->mGMP->Decode(std::move(frame), aDecodeData->mMissingFrames,
+                          codecSpecificInfo, aDecodeData->mRenderTimeMs);
+  if (NS_FAILED(rv)) {
+    GMP_LOG_ERROR("%s: Decode failed (rv=%u)!", __PRETTY_FUNCTION__,
+                  static_cast<unsigned>(rv));
+    aThis->mDecoderStatus = GMPDecodeErr;
+    return;
+  }
+
+  aThis->mDecoderStatus = GMPNoErr;
+}
+
+int32_t WebrtcGmpVideoDecoder::RegisterDecodeCompleteCallback(
+    webrtc::DecodedImageCallback* aCallback) {
+  MutexAutoLock lock(mCallbackMutex);
+  mCallback = aCallback;
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+/* static */
+void WebrtcGmpVideoDecoder::ReleaseGmp_g(
+    const RefPtr<WebrtcGmpVideoDecoder>& aDecoder) {
+  aDecoder->Close_g();
+}
+
+int32_t WebrtcGmpVideoDecoder::ReleaseGmp() {
+  GMP_LOG_DEBUG("GMP Released:");
+  RegisterDecodeCompleteCallback(nullptr);
+
+  if (mGMPThread) {
+    mGMPThread->Dispatch(WrapRunnableNM(&WebrtcGmpVideoDecoder::ReleaseGmp_g,
+                                        RefPtr<WebrtcGmpVideoDecoder>(this)),
+                         NS_DISPATCH_NORMAL);
+  }
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+void WebrtcGmpVideoDecoder::Terminated() {
+  GMP_LOG_DEBUG("GMP Decoder Terminated: %p", (void*)this);
+
+  GMPVideoDecoderProxy* gmp(mGMP);
+  mGMP = nullptr;
+  mHost = nullptr;
+  mInitting = false;
+
+  if (gmp) {
+    // Do this last, since this could cause us to be destroyed
+    gmp->Close();
+  }
+
+  // Could now notify that it's dead
+}
+
+void WebrtcGmpVideoDecoder::Decoded(GMPVideoi420Frame* aDecodedFrame) {
+  // we have two choices here: wrap the frame with a callback that frees
+  // the data later (risking running out of shmems), or copy the data out
+  // always.  Also, we can only Destroy() the frame on the gmp thread, so
+  // copying is simplest if expensive.
+  // I420 size including rounding...
+  CheckedInt32 length =
+      (CheckedInt32(aDecodedFrame->Stride(kGMPYPlane)) *
+       aDecodedFrame->Height()) +
+      (aDecodedFrame->Stride(kGMPVPlane) + aDecodedFrame->Stride(kGMPUPlane)) *
+          ((aDecodedFrame->Height() + 1) / 2);
+  int32_t size = length.value();
+  MOZ_RELEASE_ASSERT(length.isValid() && size > 0);
+
+  // Don't use MakeUniqueFallible here, because UniquePtr isn't copyable, and
+  // the closure below in WrapI420Buffer uses std::function which _is_ copyable.
+  // We'll alloc the buffer here, so we preserve the "fallible" nature, and
+  // then hand a shared_ptr, which is copyable, to WrapI420Buffer.
+  auto falliblebuffer = new (std::nothrow) uint8_t[size];
+  if (falliblebuffer) {
+    auto buffer = std::shared_ptr<uint8_t>(falliblebuffer);
+
+    // This is 3 separate buffers currently anyways, no use in trying to
+    // see if we can use a single memcpy.
+    uint8_t* buffer_y = buffer.get();
+    memcpy(buffer_y, aDecodedFrame->Buffer(kGMPYPlane),
+           aDecodedFrame->Stride(kGMPYPlane) * aDecodedFrame->Height());
+    // Should this be aligned, making it non-contiguous?  Assume no, this is
+    // already factored into the strides.
+    uint8_t* buffer_u =
+        buffer_y + aDecodedFrame->Stride(kGMPYPlane) * aDecodedFrame->Height();
+    memcpy(buffer_u, aDecodedFrame->Buffer(kGMPUPlane),
+           aDecodedFrame->Stride(kGMPUPlane) *
+               ((aDecodedFrame->Height() + 1) / 2));
+    uint8_t* buffer_v = buffer_u + aDecodedFrame->Stride(kGMPUPlane) *
+                                       ((aDecodedFrame->Height() + 1) / 2);
+    memcpy(buffer_v, aDecodedFrame->Buffer(kGMPVPlane),
+           aDecodedFrame->Stride(kGMPVPlane) *
+               ((aDecodedFrame->Height() + 1) / 2));
+
+    MutexAutoLock lock(mCallbackMutex);
+    if (mCallback) {
+      // Note: the last parameter to WrapI420Buffer is named no_longer_used,
+      // but is currently called in the destructor of WrappedYuvBuffer when
+      // the buffer is "no_longer_used".
+      rtc::scoped_refptr<webrtc::I420BufferInterface> video_frame_buffer =
+          webrtc::WrapI420Buffer(
+              aDecodedFrame->Width(), aDecodedFrame->Height(), buffer_y,
+              aDecodedFrame->Stride(kGMPYPlane), buffer_u,
+              aDecodedFrame->Stride(kGMPUPlane), buffer_v,
+              aDecodedFrame->Stride(kGMPVPlane), [buffer] {});
+
+      GMP_LOG_DEBUG("GMP Decoded: %" PRIu64, aDecodedFrame->Timestamp());
+      auto videoFrame =
+          webrtc::VideoFrame::Builder()
+              .set_video_frame_buffer(video_frame_buffer)
+              .set_timestamp_rtp(
+                  // round up
+                  (aDecodedFrame->UpdatedTimestamp() * 90ll + 999) / 1000)
+              .build();
+      mPerformanceRecorder.Record(
+          static_cast<int64_t>(aDecodedFrame->Timestamp()),
+          [&](DecodeStage& aStage) {
+            aStage.SetImageFormat(DecodeStage::YUV420P);
+            aStage.SetResolution(aDecodedFrame->Width(),
+                                 aDecodedFrame->Height());
+            aStage.SetColorDepth(gfx::ColorDepth::COLOR_8);
+          });
+      mCallback->Decoded(videoFrame);
+    }
+  }
+  aDecodedFrame->Destroy();
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/libwebrtcglue/WebrtcGmpVideoCodec.h b/dom/media/webrtc/libwebrtcglue/WebrtcGmpVideoCodec.h
new file mode 100644
index 0000000000..ef3e1bd5af
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/WebrtcGmpVideoCodec.h
@@ -0,0 +1,507 @@
+/*
+ *  Copyright (c) 2012, The WebRTC project authors. All rights reserved.
+ *
+ *  Redistribution and use in source and binary forms, with or without
+ *  modification, are permitted provided that the following conditions are
+ *  met:
+ *
+ *    * Redistributions of source code must retain the above copyright
+ *      notice, this list of conditions and the following disclaimer.
+ *
+ *    * Redistributions in binary form must reproduce the above copyright
+ *      notice, this list of conditions and the following disclaimer in
+ *      the documentation and/or other materials provided with the
+ *      distribution.
+ *
+ *    * Neither the name of Google nor the names of its contributors may
+ *      be used to endorse or promote products derived from this software
+ *      without specific prior written permission.
+ *
+ *  THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ *  "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ *  LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ *  A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+ *  HOLDER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ *  SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+ *  LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+ *  DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+ *  THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ *  (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ *  OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef WEBRTCGMPVIDEOCODEC_H_
+#define WEBRTCGMPVIDEOCODEC_H_
+
+#include <queue>
+#include <string>
+
+#include "nsThreadUtils.h"
+#include "mozilla/Monitor.h"
+#include "mozilla/Mutex.h"
+#include "mozilla/Telemetry.h"
+
+#include "mozIGeckoMediaPluginService.h"
+#include "MediaConduitInterface.h"
+#include "AudioConduit.h"
+#include "PerformanceRecorder.h"
+#include "VideoConduit.h"
+#include "api/video/video_frame_type.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "common_video/h264/h264_bitstream_parser.h"
+
+#include "gmp-video-host.h"
+#include "GMPVideoDecoderProxy.h"
+#include "GMPVideoEncoderProxy.h"
+
+#include "jsapi/PeerConnectionImpl.h"
+
+namespace mozilla {
+
+class GmpInitDoneRunnable : public Runnable {
+ public:
+  explicit GmpInitDoneRunnable(std::string aPCHandle)
+      : Runnable("GmpInitDoneRunnable"),
+        mResult(WEBRTC_VIDEO_CODEC_OK),
+        mPCHandle(std::move(aPCHandle)) {}
+
+  NS_IMETHOD Run() override {
+    Telemetry::Accumulate(Telemetry::WEBRTC_GMP_INIT_SUCCESS,
+                          mResult == WEBRTC_VIDEO_CODEC_OK);
+    if (mResult == WEBRTC_VIDEO_CODEC_OK) {
+      // Might be useful to notify the PeerConnection about successful init
+      // someday.
+      return NS_OK;
+    }
+
+    PeerConnectionWrapper wrapper(mPCHandle);
+    if (wrapper.impl()) {
+      wrapper.impl()->OnMediaError(mError);
+    }
+    return NS_OK;
+  }
+
+  void Dispatch(int32_t aResult, const std::string& aError = "") {
+    mResult = aResult;
+    mError = aError;
+    nsCOMPtr<nsIThread> mainThread(do_GetMainThread());
+    if (mainThread) {
+      // For some reason, the compiler on CI is treating |this| as a const
+      // pointer, despite the fact that we're in a non-const function. And,
+      // interestingly enough, correcting this doesn't require a const_cast.
+      mainThread->Dispatch(do_AddRef(static_cast<nsIRunnable*>(this)),
+                           NS_DISPATCH_NORMAL);
+    }
+  }
+
+  int32_t Result() { return mResult; }
+
+ private:
+  int32_t mResult;
+  const std::string mPCHandle;
+  std::string mError;
+};
+
+// Hold a frame for later decode
+class GMPDecodeData {
+ public:
+  GMPDecodeData(const webrtc::EncodedImage& aInputImage, bool aMissingFrames,
+                int64_t aRenderTimeMs)
+      : mImage(aInputImage),
+        mMissingFrames(aMissingFrames),
+        mRenderTimeMs(aRenderTimeMs) {
+    // We want to use this for queuing, and the calling code recycles the
+    // buffer on return from Decode()
+    MOZ_RELEASE_ASSERT(aInputImage.size() <
+                       (std::numeric_limits<size_t>::max() >> 1));
+  }
+
+  ~GMPDecodeData() = default;
+
+  const webrtc::EncodedImage mImage;
+  const bool mMissingFrames;
+  const int64_t mRenderTimeMs;
+};
+
+class RefCountedWebrtcVideoEncoder {
+ public:
+  NS_INLINE_DECL_PURE_VIRTUAL_REFCOUNTING
+
+  // Implement sort of WebrtcVideoEncoder interface and support refcounting.
+  // (We cannot use |Release|, since that's needed for nsRefPtr)
+  virtual int32_t InitEncode(
+      const webrtc::VideoCodec* aCodecSettings,
+      const webrtc::VideoEncoder::Settings& aSettings) = 0;
+
+  virtual int32_t Encode(
+      const webrtc::VideoFrame& aInputImage,
+      const std::vector<webrtc::VideoFrameType>* aFrameTypes) = 0;
+
+  virtual int32_t RegisterEncodeCompleteCallback(
+      webrtc::EncodedImageCallback* aCallback) = 0;
+
+  virtual int32_t Shutdown() = 0;
+
+  virtual int32_t SetRates(
+      const webrtc::VideoEncoder::RateControlParameters& aParameters) = 0;
+
+  virtual MediaEventSource<uint64_t>* InitPluginEvent() = 0;
+
+  virtual MediaEventSource<uint64_t>* ReleasePluginEvent() = 0;
+
+  virtual WebrtcVideoEncoder::EncoderInfo GetEncoderInfo() const = 0;
+
+ protected:
+  virtual ~RefCountedWebrtcVideoEncoder() = default;
+};
+
+class WebrtcGmpVideoEncoder : public GMPVideoEncoderCallbackProxy,
+                              public RefCountedWebrtcVideoEncoder {
+ public:
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(WebrtcGmpVideoEncoder, final);
+
+  WebrtcGmpVideoEncoder(const webrtc::SdpVideoFormat& aFormat,
+                        std::string aPCHandle);
+
+  // Implement VideoEncoder interface, sort of.
+  // (We cannot use |Release|, since that's needed for nsRefPtr)
+  int32_t InitEncode(const webrtc::VideoCodec* aCodecSettings,
+                     const webrtc::VideoEncoder::Settings& aSettings) override;
+
+  int32_t Encode(
+      const webrtc::VideoFrame& aInputImage,
+      const std::vector<webrtc::VideoFrameType>* aFrameTypes) override;
+
+  int32_t RegisterEncodeCompleteCallback(
+      webrtc::EncodedImageCallback* aCallback) override;
+
+  int32_t Shutdown() override;
+
+  int32_t SetRates(
+      const webrtc::VideoEncoder::RateControlParameters& aParameters) override;
+
+  WebrtcVideoEncoder::EncoderInfo GetEncoderInfo() const override;
+
+  MediaEventSource<uint64_t>* InitPluginEvent() override {
+    return &mInitPluginEvent;
+  }
+
+  MediaEventSource<uint64_t>* ReleasePluginEvent() override {
+    return &mReleasePluginEvent;
+  }
+
+  // GMPVideoEncoderCallback virtual functions.
+  virtual void Terminated() override;
+
+  virtual void Encoded(GMPVideoEncodedFrame* aEncodedFrame,
+                       const nsTArray<uint8_t>& aCodecSpecificInfo) override;
+
+  virtual void Error(GMPErr aError) override {}
+
+ private:
+  virtual ~WebrtcGmpVideoEncoder();
+
+  static void InitEncode_g(const RefPtr<WebrtcGmpVideoEncoder>& aThis,
+                           const GMPVideoCodec& aCodecParams,
+                           int32_t aNumberOfCores, uint32_t aMaxPayloadSize,
+                           const RefPtr<GmpInitDoneRunnable>& aInitDone);
+  int32_t GmpInitDone(GMPVideoEncoderProxy* aGMP, GMPVideoHost* aHost,
+                      const GMPVideoCodec& aCodecParams,
+                      std::string* aErrorOut);
+  int32_t GmpInitDone(GMPVideoEncoderProxy* aGMP, GMPVideoHost* aHost,
+                      std::string* aErrorOut);
+  int32_t InitEncoderForSize(unsigned short aWidth, unsigned short aHeight,
+                             std::string* aErrorOut);
+  static void ReleaseGmp_g(const RefPtr<WebrtcGmpVideoEncoder>& aEncoder);
+  void Close_g();
+
+  class InitDoneCallback : public GetGMPVideoEncoderCallback {
+   public:
+    InitDoneCallback(const RefPtr<WebrtcGmpVideoEncoder>& aEncoder,
+                     const RefPtr<GmpInitDoneRunnable>& aInitDone,
+                     const GMPVideoCodec& aCodecParams)
+        : mEncoder(aEncoder),
+          mInitDone(aInitDone),
+          mCodecParams(aCodecParams) {}
+
+    virtual void Done(GMPVideoEncoderProxy* aGMP,
+                      GMPVideoHost* aHost) override {
+      std::string errorOut;
+      int32_t result =
+          mEncoder->GmpInitDone(aGMP, aHost, mCodecParams, &errorOut);
+
+      mInitDone->Dispatch(result, errorOut);
+    }
+
+   private:
+    const RefPtr<WebrtcGmpVideoEncoder> mEncoder;
+    const RefPtr<GmpInitDoneRunnable> mInitDone;
+    const GMPVideoCodec mCodecParams;
+  };
+
+  static void Encode_g(const RefPtr<WebrtcGmpVideoEncoder>& aEncoder,
+                       webrtc::VideoFrame aInputImage,
+                       std::vector<webrtc::VideoFrameType> aFrameTypes);
+  void RegetEncoderForResolutionChange(
+      uint32_t aWidth, uint32_t aHeight,
+      const RefPtr<GmpInitDoneRunnable>& aInitDone);
+
+  class InitDoneForResolutionChangeCallback
+      : public GetGMPVideoEncoderCallback {
+   public:
+    InitDoneForResolutionChangeCallback(
+        const RefPtr<WebrtcGmpVideoEncoder>& aEncoder,
+        const RefPtr<GmpInitDoneRunnable>& aInitDone, uint32_t aWidth,
+        uint32_t aHeight)
+        : mEncoder(aEncoder),
+          mInitDone(aInitDone),
+          mWidth(aWidth),
+          mHeight(aHeight) {}
+
+    virtual void Done(GMPVideoEncoderProxy* aGMP,
+                      GMPVideoHost* aHost) override {
+      std::string errorOut;
+      int32_t result = mEncoder->GmpInitDone(aGMP, aHost, &errorOut);
+      if (result != WEBRTC_VIDEO_CODEC_OK) {
+        mInitDone->Dispatch(result, errorOut);
+        return;
+      }
+
+      result = mEncoder->InitEncoderForSize(mWidth, mHeight, &errorOut);
+      mInitDone->Dispatch(result, errorOut);
+    }
+
+   private:
+    const RefPtr<WebrtcGmpVideoEncoder> mEncoder;
+    const RefPtr<GmpInitDoneRunnable> mInitDone;
+    const uint32_t mWidth;
+    const uint32_t mHeight;
+  };
+
+  static int32_t SetRates_g(RefPtr<WebrtcGmpVideoEncoder> aThis,
+                            uint32_t aNewBitRateKbps, Maybe<double> aFrameRate);
+
+  nsCOMPtr<mozIGeckoMediaPluginService> mMPS;
+  nsCOMPtr<nsIThread> mGMPThread;
+  GMPVideoEncoderProxy* mGMP;
+  // Used to handle a race where Release() is called while init is in progress
+  bool mInitting;
+  GMPVideoHost* mHost;
+  GMPVideoCodec mCodecParams;
+  uint32_t mMaxPayloadSize;
+  const webrtc::SdpVideoFormat::Parameters mFormatParams;
+  webrtc::CodecSpecificInfo mCodecSpecificInfo;
+  webrtc::H264BitstreamParser mH264BitstreamParser;
+  // Protects mCallback
+  Mutex mCallbackMutex MOZ_UNANNOTATED;
+  webrtc::EncodedImageCallback* mCallback;
+  Maybe<uint64_t> mCachedPluginId;
+  const std::string mPCHandle;
+
+  struct InputImageData {
+    int64_t timestamp_us;
+  };
+  // Map rtp time -> input image data
+  DataMutex<std::map<uint32_t, InputImageData>> mInputImageMap;
+
+  MediaEventProducer<uint64_t> mInitPluginEvent;
+  MediaEventProducer<uint64_t> mReleasePluginEvent;
+};
+
+// Basically a strong ref to a RefCountedWebrtcVideoEncoder, that also
+// translates from Release() to RefCountedWebrtcVideoEncoder::Shutdown(),
+// since we need RefCountedWebrtcVideoEncoder::Release() for managing the
+// refcount. The webrtc.org code gets one of these, so it doesn't unilaterally
+// delete the "real" encoder.
+class WebrtcVideoEncoderProxy : public WebrtcVideoEncoder {
+ public:
+  explicit WebrtcVideoEncoderProxy(
+      RefPtr<RefCountedWebrtcVideoEncoder> aEncoder)
+      : mEncoderImpl(std::move(aEncoder)) {}
+
+  virtual ~WebrtcVideoEncoderProxy() {
+    RegisterEncodeCompleteCallback(nullptr);
+  }
+
+  MediaEventSource<uint64_t>* InitPluginEvent() override {
+    return mEncoderImpl->InitPluginEvent();
+  }
+
+  MediaEventSource<uint64_t>* ReleasePluginEvent() override {
+    return mEncoderImpl->ReleasePluginEvent();
+  }
+
+  int32_t InitEncode(const webrtc::VideoCodec* aCodecSettings,
+                     const WebrtcVideoEncoder::Settings& aSettings) override {
+    return mEncoderImpl->InitEncode(aCodecSettings, aSettings);
+  }
+
+  int32_t Encode(
+      const webrtc::VideoFrame& aInputImage,
+      const std::vector<webrtc::VideoFrameType>* aFrameTypes) override {
+    return mEncoderImpl->Encode(aInputImage, aFrameTypes);
+  }
+
+  int32_t RegisterEncodeCompleteCallback(
+      webrtc::EncodedImageCallback* aCallback) override {
+    return mEncoderImpl->RegisterEncodeCompleteCallback(aCallback);
+  }
+
+  int32_t Release() override { return mEncoderImpl->Shutdown(); }
+
+  void SetRates(const RateControlParameters& aParameters) override {
+    mEncoderImpl->SetRates(aParameters);
+  }
+
+  EncoderInfo GetEncoderInfo() const override {
+    return mEncoderImpl->GetEncoderInfo();
+  }
+
+ private:
+  const RefPtr<RefCountedWebrtcVideoEncoder> mEncoderImpl;
+};
+
+class WebrtcGmpVideoDecoder : public GMPVideoDecoderCallbackProxy {
+ public:
+  WebrtcGmpVideoDecoder(std::string aPCHandle, TrackingId aTrackingId);
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(WebrtcGmpVideoDecoder, final);
+
+  // Implement VideoEncoder interface, sort of.
+  // (We cannot use |Release|, since that's needed for nsRefPtr)
+  virtual bool Configure(const webrtc::VideoDecoder::Settings& settings);
+  virtual int32_t Decode(const webrtc::EncodedImage& aInputImage,
+                         bool aMissingFrames, int64_t aRenderTimeMs);
+  virtual int32_t RegisterDecodeCompleteCallback(
+      webrtc::DecodedImageCallback* aCallback);
+
+  virtual int32_t ReleaseGmp();
+
+  MediaEventSource<uint64_t>* InitPluginEvent() { return &mInitPluginEvent; }
+
+  MediaEventSource<uint64_t>* ReleasePluginEvent() {
+    return &mReleasePluginEvent;
+  }
+
+  // GMPVideoDecoderCallbackProxy
+  virtual void Terminated() override;
+
+  virtual void Decoded(GMPVideoi420Frame* aDecodedFrame) override;
+
+  virtual void ReceivedDecodedReferenceFrame(
+      const uint64_t aPictureId) override {
+    MOZ_CRASH();
+  }
+
+  virtual void ReceivedDecodedFrame(const uint64_t aPictureId) override {
+    MOZ_CRASH();
+  }
+
+  virtual void InputDataExhausted() override {}
+
+  virtual void DrainComplete() override {}
+
+  virtual void ResetComplete() override {}
+
+  virtual void Error(GMPErr aError) override { mDecoderStatus = aError; }
+
+ private:
+  virtual ~WebrtcGmpVideoDecoder();
+
+  static void Configure_g(const RefPtr<WebrtcGmpVideoDecoder>& aThis,
+                          const webrtc::VideoDecoder::Settings& settings,
+                          const RefPtr<GmpInitDoneRunnable>& aInitDone);
+  int32_t GmpInitDone(GMPVideoDecoderProxy* aGMP, GMPVideoHost* aHost,
+                      std::string* aErrorOut);
+  static void ReleaseGmp_g(const RefPtr<WebrtcGmpVideoDecoder>& aDecoder);
+  void Close_g();
+
+  class InitDoneCallback : public GetGMPVideoDecoderCallback {
+   public:
+    explicit InitDoneCallback(const RefPtr<WebrtcGmpVideoDecoder>& aDecoder,
+                              const RefPtr<GmpInitDoneRunnable>& aInitDone)
+        : mDecoder(aDecoder), mInitDone(aInitDone) {}
+
+    virtual void Done(GMPVideoDecoderProxy* aGMP,
+                      GMPVideoHost* aHost) override {
+      std::string errorOut;
+      int32_t result = mDecoder->GmpInitDone(aGMP, aHost, &errorOut);
+
+      mInitDone->Dispatch(result, errorOut);
+    }
+
+   private:
+    const RefPtr<WebrtcGmpVideoDecoder> mDecoder;
+    const RefPtr<GmpInitDoneRunnable> mInitDone;
+  };
+
+  static void Decode_g(const RefPtr<WebrtcGmpVideoDecoder>& aThis,
+                       UniquePtr<GMPDecodeData>&& aDecodeData);
+
+  nsCOMPtr<mozIGeckoMediaPluginService> mMPS;
+  nsCOMPtr<nsIThread> mGMPThread;
+  GMPVideoDecoderProxy* mGMP;  // Addref is held for us
+  // Used to handle a race where Release() is called while init is in progress
+  bool mInitting;
+  // Frames queued for decode while mInitting is true
+  nsTArray<UniquePtr<GMPDecodeData>> mQueuedFrames;
+  GMPVideoHost* mHost;
+  // Protects mCallback
+  Mutex mCallbackMutex MOZ_UNANNOTATED;
+  webrtc::DecodedImageCallback* mCallback;
+  Maybe<uint64_t> mCachedPluginId;
+  Atomic<GMPErr, ReleaseAcquire> mDecoderStatus;
+  const std::string mPCHandle;
+  const TrackingId mTrackingId;
+  PerformanceRecorderMulti<DecodeStage> mPerformanceRecorder;
+
+  MediaEventProducer<uint64_t> mInitPluginEvent;
+  MediaEventProducer<uint64_t> mReleasePluginEvent;
+};
+
+// Basically a strong ref to a WebrtcGmpVideoDecoder, that also translates
+// from Release() to WebrtcGmpVideoDecoder::ReleaseGmp(), since we need
+// WebrtcGmpVideoDecoder::Release() for managing the refcount.
+// The webrtc.org code gets one of these, so it doesn't unilaterally delete
+// the "real" encoder.
+class WebrtcVideoDecoderProxy : public WebrtcVideoDecoder {
+ public:
+  explicit WebrtcVideoDecoderProxy(std::string aPCHandle,
+                                   TrackingId aTrackingId)
+      : mDecoderImpl(new WebrtcGmpVideoDecoder(std::move(aPCHandle),
+                                               std::move(aTrackingId))) {}
+
+  virtual ~WebrtcVideoDecoderProxy() {
+    RegisterDecodeCompleteCallback(nullptr);
+  }
+
+  MediaEventSource<uint64_t>* InitPluginEvent() override {
+    return mDecoderImpl->InitPluginEvent();
+  }
+
+  MediaEventSource<uint64_t>* ReleasePluginEvent() override {
+    return mDecoderImpl->ReleasePluginEvent();
+  }
+
+  bool Configure(const Settings& settings) override {
+    return mDecoderImpl->Configure(settings);
+  }
+
+  int32_t Decode(const webrtc::EncodedImage& aInputImage, bool aMissingFrames,
+                 int64_t aRenderTimeMs) override {
+    return mDecoderImpl->Decode(aInputImage, aMissingFrames, aRenderTimeMs);
+  }
+
+  int32_t RegisterDecodeCompleteCallback(
+      webrtc::DecodedImageCallback* aCallback) override {
+    return mDecoderImpl->RegisterDecodeCompleteCallback(aCallback);
+  }
+
+  int32_t Release() override { return mDecoderImpl->ReleaseGmp(); }
+
+ private:
+  const RefPtr<WebrtcGmpVideoDecoder> mDecoderImpl;
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/libwebrtcglue/WebrtcImageBuffer.h b/dom/media/webrtc/libwebrtcglue/WebrtcImageBuffer.h
new file mode 100644
index 0000000000..305f4df577
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/WebrtcImageBuffer.h
@@ -0,0 +1,53 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef WebrtcImageBuffer_h__
+#define WebrtcImageBuffer_h__
+
+#include "common_video/include/video_frame_buffer.h"
+
+namespace mozilla {
+namespace layers {
+class Image;
+}
+
+class ImageBuffer : public webrtc::VideoFrameBuffer {
+ public:
+  explicit ImageBuffer(RefPtr<layers::Image>&& aImage)
+      : mImage(std::move(aImage)) {}
+
+  rtc::scoped_refptr<webrtc::I420BufferInterface> ToI420() override {
+    RefPtr<layers::PlanarYCbCrImage> image = mImage->AsPlanarYCbCrImage();
+    MOZ_ASSERT(image);
+    if (!image) {
+      // TODO. YUV420 ReadBack, Image only provides a RGB readback.
+      return nullptr;
+    }
+    const layers::PlanarYCbCrData* data = image->GetData();
+    rtc::scoped_refptr<webrtc::I420BufferInterface> buf =
+        webrtc::WrapI420Buffer(
+            data->mPictureRect.width, data->mPictureRect.height,
+            data->mYChannel, data->mYStride, data->mCbChannel,
+            data->mCbCrStride, data->mCrChannel, data->mCbCrStride,
+            [image] { /* keep reference alive*/ });
+    return buf;
+  }
+
+  Type type() const override { return Type::kNative; }
+
+  int width() const override { return mImage->GetSize().width; }
+
+  int height() const override { return mImage->GetSize().height; }
+
+  RefPtr<layers::Image> GetNativeImage() const { return mImage; }
+
+ private:
+  const RefPtr<layers::Image> mImage;
+};
+
+}  // namespace mozilla
+
+#endif  // WebrtcImageBuffer_h__
diff --git a/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataDecoderCodec.cpp b/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataDecoderCodec.cpp
new file mode 100644
index 0000000000..f132e6a124
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataDecoderCodec.cpp
@@ -0,0 +1,209 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "WebrtcMediaDataDecoderCodec.h"
+
+#include "ImageContainer.h"
+#include "MediaDataDecoderProxy.h"
+#include "PDMFactory.h"
+#include "VideoUtils.h"
+#include "mozilla/layers/ImageBridgeChild.h"
+#include "mozilla/media/MediaUtils.h"
+#include "mozilla/StaticPrefs_media.h"
+
+namespace mozilla {
+
+WebrtcMediaDataDecoder::WebrtcMediaDataDecoder(nsACString& aCodecMimeType,
+                                               TrackingId aTrackingId)
+    : mThreadPool(GetMediaThreadPool(MediaThreadType::SUPERVISOR)),
+      mTaskQueue(TaskQueue::Create(do_AddRef(mThreadPool),
+                                   "WebrtcMediaDataDecoder::mTaskQueue")),
+      mImageContainer(MakeAndAddRef<layers::ImageContainer>(
+          layers::ImageContainer::ASYNCHRONOUS)),
+      mFactory(new PDMFactory()),
+      mTrackType(TrackInfo::kUndefinedTrack),
+      mCodecType(aCodecMimeType),
+      mTrackingId(std::move(aTrackingId)) {}
+
+WebrtcMediaDataDecoder::~WebrtcMediaDataDecoder() {}
+
+bool WebrtcMediaDataDecoder::Configure(
+    const webrtc::VideoDecoder::Settings& settings) {
+  nsCString codec;
+  mTrackType = TrackInfo::kVideoTrack;
+  mInfo = VideoInfo(settings.max_render_resolution().Width(),
+                    settings.max_render_resolution().Height());
+  mInfo.mMimeType = mCodecType;
+
+#ifdef MOZ_WIDGET_GTK
+  if (mInfo.mMimeType.EqualsLiteral("video/vp8") &&
+      !StaticPrefs::media_navigator_mediadatadecoder_vp8_hardware_enabled()) {
+    mDisabledHardwareAcceleration = true;
+  }
+#endif
+
+  return WEBRTC_VIDEO_CODEC_OK == CreateDecoder();
+}
+
+int32_t WebrtcMediaDataDecoder::Decode(const webrtc::EncodedImage& aInputImage,
+                                       bool aMissingFrames,
+                                       int64_t aRenderTimeMs) {
+  if (!mCallback || !mDecoder) {
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+
+  if (!aInputImage.data() || !aInputImage.size()) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+
+  // Always start with a complete key frame.
+  if (mNeedKeyframe) {
+    if (aInputImage._frameType != webrtc::VideoFrameType::kVideoFrameKey)
+      return WEBRTC_VIDEO_CODEC_ERROR;
+    // We have a key frame - is it complete?
+    mNeedKeyframe = false;
+  }
+
+  auto disabledHardwareAcceleration =
+      MakeScopeExit([&] { mDisabledHardwareAcceleration = true; });
+
+  RefPtr<MediaRawData> compressedFrame =
+      new MediaRawData(aInputImage.data(), aInputImage.size());
+  if (!compressedFrame->Data()) {
+    return WEBRTC_VIDEO_CODEC_MEMORY;
+  }
+
+  compressedFrame->mTime =
+      media::TimeUnit::FromMicroseconds(aInputImage.Timestamp());
+  compressedFrame->mTimecode =
+      media::TimeUnit::FromMicroseconds(aRenderTimeMs * 1000);
+  compressedFrame->mKeyframe =
+      aInputImage._frameType == webrtc::VideoFrameType::kVideoFrameKey;
+  {
+    media::Await(
+        do_AddRef(mThreadPool), mDecoder->Decode(compressedFrame),
+        [&](const MediaDataDecoder::DecodedData& aResults) {
+          mResults = aResults.Clone();
+          mError = NS_OK;
+        },
+        [&](const MediaResult& aError) { mError = aError; });
+
+    for (auto& frame : mResults) {
+      MOZ_ASSERT(frame->mType == MediaData::Type::VIDEO_DATA);
+      RefPtr<VideoData> video = frame->As<VideoData>();
+      MOZ_ASSERT(video);
+      if (!video->mImage) {
+        // Nothing to display.
+        continue;
+      }
+      rtc::scoped_refptr<ImageBuffer> image(
+          new rtc::RefCountedObject<ImageBuffer>(std::move(video->mImage)));
+
+      auto videoFrame = webrtc::VideoFrame::Builder()
+                            .set_video_frame_buffer(image)
+                            .set_timestamp_rtp(aInputImage.Timestamp())
+                            .set_rotation(aInputImage.rotation_)
+                            .build();
+      mCallback->Decoded(videoFrame);
+    }
+    mResults.Clear();
+  }
+
+  if (NS_FAILED(mError) && mError != NS_ERROR_DOM_MEDIA_CANCELED) {
+    CreateDecoder();
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  if (NS_FAILED(mError)) {
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  disabledHardwareAcceleration.release();
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t WebrtcMediaDataDecoder::RegisterDecodeCompleteCallback(
+    webrtc::DecodedImageCallback* aCallback) {
+  mCallback = aCallback;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t WebrtcMediaDataDecoder::Release() {
+  if (mDecoder) {
+    RefPtr<MediaDataDecoder> decoder = std::move(mDecoder);
+    decoder->Flush()->Then(mTaskQueue, __func__,
+                           [decoder]() { decoder->Shutdown(); });
+  }
+
+  mNeedKeyframe = true;
+  mError = NS_OK;
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+bool WebrtcMediaDataDecoder::OnTaskQueue() const {
+  return mTaskQueue->IsOnCurrentThread();
+}
+
+int32_t WebrtcMediaDataDecoder::CreateDecoder() {
+  RefPtr<layers::KnowsCompositor> knowsCompositor =
+      layers::ImageBridgeChild::GetSingleton();
+
+  if (mDecoder) {
+    Release();
+  }
+
+  RefPtr<TaskQueue> tq =
+      TaskQueue::Create(GetMediaThreadPool(MediaThreadType::PLATFORM_DECODER),
+                        "webrtc decode TaskQueue");
+  RefPtr<MediaDataDecoder> decoder;
+
+  media::Await(do_AddRef(mThreadPool), InvokeAsync(tq, __func__, [&] {
+                 RefPtr<GenericPromise> p =
+                     mFactory
+                         ->CreateDecoder(
+                             {mInfo,
+                              CreateDecoderParams::OptionSet(
+                                  CreateDecoderParams::Option::LowLatency,
+                                  CreateDecoderParams::Option::FullH264Parsing,
+                                  CreateDecoderParams::Option::
+                                      ErrorIfNoInitializationData,
+                                  mDisabledHardwareAcceleration
+                                      ? CreateDecoderParams::Option::
+                                            HardwareDecoderNotAllowed
+                                      : CreateDecoderParams::Option::Default),
+                              mTrackType, mImageContainer, knowsCompositor,
+                              Some(mTrackingId)})
+                         ->Then(
+                             tq, __func__,
+                             [&](RefPtr<MediaDataDecoder>&& aDecoder) {
+                               decoder = std::move(aDecoder);
+                               return GenericPromise::CreateAndResolve(
+                                   true, __func__);
+                             },
+                             [](const MediaResult& aResult) {
+                               return GenericPromise::CreateAndReject(
+                                   NS_ERROR_DOM_MEDIA_FATAL_ERR, __func__);
+                             });
+                 return p;
+               }));
+
+  if (!decoder) {
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  // We need to wrap our decoder in a MediaDataDecoderProxy so that it always
+  // run on an nsISerialEventTarget (which the webrtc code doesn't do)
+  mDecoder = new MediaDataDecoderProxy(decoder.forget(), tq.forget());
+
+  media::Await(
+      do_AddRef(mThreadPool), mDecoder->Init(),
+      [&](TrackInfo::TrackType) { mError = NS_OK; },
+      [&](const MediaResult& aError) { mError = aError; });
+
+  return NS_SUCCEEDED(mError) ? WEBRTC_VIDEO_CODEC_OK
+                              : WEBRTC_VIDEO_CODEC_ERROR;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataDecoderCodec.h b/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataDecoderCodec.h
new file mode 100644
index 0000000000..ccb54c692b
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataDecoderCodec.h
@@ -0,0 +1,70 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef WebrtcMediaDataDecoderCodec_h__
+#define WebrtcMediaDataDecoderCodec_h__
+
+#include "MediaConduitInterface.h"
+#include "MediaInfo.h"
+#include "MediaResult.h"
+#include "PlatformDecoderModule.h"
+#include "VideoConduit.h"
+#include "WebrtcImageBuffer.h"
+#include "common_video/include/video_frame_buffer.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+
+namespace webrtc {
+class DecodedImageCallback;
+}
+namespace mozilla {
+namespace layers {
+class Image;
+class ImageContainer;
+}  // namespace layers
+
+class PDMFactory;
+class SharedThreadPool;
+class TaskQueue;
+
+class WebrtcMediaDataDecoder : public WebrtcVideoDecoder {
+ public:
+  WebrtcMediaDataDecoder(nsACString& aCodecMimeType, TrackingId aTrackingId);
+
+  bool Configure(const webrtc::VideoDecoder::Settings& settings) override;
+
+  int32_t Decode(const webrtc::EncodedImage& inputImage, bool missingFrames,
+                 int64_t renderTimeMs = -1) override;
+
+  int32_t RegisterDecodeCompleteCallback(
+      webrtc::DecodedImageCallback* callback) override;
+
+  int32_t Release() override;
+
+ private:
+  ~WebrtcMediaDataDecoder();
+  void QueueFrame(MediaRawData* aFrame);
+  bool OnTaskQueue() const;
+  int32_t CreateDecoder();
+
+  const RefPtr<SharedThreadPool> mThreadPool;
+  const RefPtr<TaskQueue> mTaskQueue;
+  const RefPtr<layers::ImageContainer> mImageContainer;
+  const RefPtr<PDMFactory> mFactory;
+  RefPtr<MediaDataDecoder> mDecoder;
+  webrtc::DecodedImageCallback* mCallback = nullptr;
+  VideoInfo mInfo;
+  TrackInfo::TrackType mTrackType;
+  bool mNeedKeyframe = true;
+  MozPromiseRequestHolder<MediaDataDecoder::DecodePromise> mDecodeRequest;
+
+  MediaResult mError = NS_OK;
+  MediaDataDecoder::DecodedData mResults;
+  const nsCString mCodecType;
+  bool mDisabledHardwareAcceleration = false;
+  const TrackingId mTrackingId;
+};
+
+}  // namespace mozilla
+
+#endif  // WebrtcMediaDataDecoderCodec_h__
diff --git a/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataEncoderCodec.cpp b/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataEncoderCodec.cpp
new file mode 100644
index 0000000000..332d49b4e5
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataEncoderCodec.cpp
@@ -0,0 +1,518 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "WebrtcMediaDataEncoderCodec.h"
+
+#include "AnnexB.h"
+#include "api/video_codecs/h264_profile_level_id.h"
+#include "ImageContainer.h"
+#include "media/base/media_constants.h"
+#include "MediaData.h"
+#include "modules/video_coding/utility/vp8_header_parser.h"
+#include "modules/video_coding/utility/vp9_uncompressed_header_parser.h"
+#include "mozilla/Maybe.h"
+#include "mozilla/Span.h"
+#include "mozilla/gfx/Point.h"
+#include "mozilla/media/MediaUtils.h"
+#include "mozilla/StaticPrefs_media.h"
+#include "PEMFactory.h"
+#include "system_wrappers/include/clock.h"
+#include "VideoUtils.h"
+
+namespace mozilla {
+
+extern LazyLogModule sPEMLog;
+
+#undef LOG
+#define LOG(msg, ...)               \
+  MOZ_LOG(sPEMLog, LogLevel::Debug, \
+          ("WebrtcMediaDataEncoder=%p, " msg, this, ##__VA_ARGS__))
+
+#undef LOG_V
+#define LOG_V(msg, ...)               \
+  MOZ_LOG(sPEMLog, LogLevel::Verbose, \
+          ("WebrtcMediaDataEncoder=%p, " msg, this, ##__VA_ARGS__))
+
+using namespace media;
+using namespace layers;
+using MimeTypeResult = Maybe<nsLiteralCString>;
+
+static MimeTypeResult ConvertWebrtcCodecTypeToMimeType(
+    const webrtc::VideoCodecType& aType) {
+  switch (aType) {
+    case webrtc::VideoCodecType::kVideoCodecVP8:
+      return Some("video/vp8"_ns);
+    case webrtc::VideoCodecType::kVideoCodecVP9:
+      return Some("video/vp9"_ns);
+    case webrtc::VideoCodecType::kVideoCodecH264:
+      return Some("video/avc"_ns);
+    default:
+      MOZ_MAKE_COMPILER_ASSUME_IS_UNREACHABLE("Unsupported codec type");
+  }
+  return Nothing();
+}
+
+bool WebrtcMediaDataEncoder::CanCreate(
+    const webrtc::VideoCodecType aCodecType) {
+  auto factory = MakeRefPtr<PEMFactory>();
+  MimeTypeResult mimeType = ConvertWebrtcCodecTypeToMimeType(aCodecType);
+  return mimeType ? factory->SupportsMimeType(mimeType.ref()) : false;
+}
+
+static const char* PacketModeStr(const webrtc::CodecSpecificInfo& aInfo) {
+  MOZ_ASSERT(aInfo.codecType != webrtc::VideoCodecType::kVideoCodecGeneric);
+
+  if (aInfo.codecType != webrtc::VideoCodecType::kVideoCodecH264) {
+    return "N/A";
+  }
+  switch (aInfo.codecSpecific.H264.packetization_mode) {
+    case webrtc::H264PacketizationMode::SingleNalUnit:
+      return "SingleNalUnit";
+    case webrtc::H264PacketizationMode::NonInterleaved:
+      return "NonInterleaved";
+    default:
+      return "Unknown";
+  }
+}
+
+static MediaDataEncoder::H264Specific::ProfileLevel ConvertProfileLevel(
+    const webrtc::SdpVideoFormat::Parameters& aParameters) {
+  const absl::optional<webrtc::H264ProfileLevelId> profileLevel =
+      webrtc::ParseSdpForH264ProfileLevelId(aParameters);
+  if (profileLevel &&
+      (profileLevel->profile == webrtc::H264Profile::kProfileBaseline ||
+       profileLevel->profile ==
+           webrtc::H264Profile::kProfileConstrainedBaseline)) {
+    return MediaDataEncoder::H264Specific::ProfileLevel::BaselineAutoLevel;
+  }
+  return MediaDataEncoder::H264Specific::ProfileLevel::MainAutoLevel;
+}
+
+static MediaDataEncoder::VPXSpecific::Complexity MapComplexity(
+    webrtc::VideoCodecComplexity aComplexity) {
+  switch (aComplexity) {
+    case webrtc::VideoCodecComplexity::kComplexityNormal:
+      return MediaDataEncoder::VPXSpecific::Complexity::Normal;
+    case webrtc::VideoCodecComplexity::kComplexityHigh:
+      return MediaDataEncoder::VPXSpecific::Complexity::High;
+    case webrtc::VideoCodecComplexity::kComplexityHigher:
+      return MediaDataEncoder::VPXSpecific::Complexity::Higher;
+    case webrtc::VideoCodecComplexity::kComplexityMax:
+      return MediaDataEncoder::VPXSpecific::Complexity::Max;
+    default:
+      MOZ_MAKE_COMPILER_ASSUME_IS_UNREACHABLE("Bad complexity value");
+  }
+}
+
+WebrtcMediaDataEncoder::WebrtcMediaDataEncoder(
+    const webrtc::SdpVideoFormat& aFormat)
+    : mTaskQueue(
+          TaskQueue::Create(GetMediaThreadPool(MediaThreadType::SUPERVISOR),
+                            "WebrtcMediaDataEncoder::mTaskQueue")),
+      mFactory(new PEMFactory()),
+      mCallbackMutex("WebrtcMediaDataEncoderCodec encoded callback mutex"),
+      mFormatParams(aFormat.parameters),
+      // Use the same lower and upper bound as h264_video_toolbox_encoder which
+      // is an encoder from webrtc's upstream codebase.
+      // 0.5 is set as a mininum to prevent overcompensating for large temporary
+      // overshoots. We don't want to degrade video quality too badly.
+      // 0.95 is set to prevent oscillations. When a lower bitrate is set on the
+      // encoder than previously set, its output seems to have a brief period of
+      // drastically reduced bitrate, so we want to avoid that. In steady state
+      // conditions, 0.95 seems to give us better overall bitrate over long
+      // periods of time.
+      mBitrateAdjuster(0.5, 0.95) {
+  PodZero(&mCodecSpecific.codecSpecific);
+}
+
+WebrtcMediaDataEncoder::~WebrtcMediaDataEncoder() = default;
+
+static void InitCodecSpecficInfo(
+    webrtc::CodecSpecificInfo& aInfo, const webrtc::VideoCodec* aCodecSettings,
+    const webrtc::SdpVideoFormat::Parameters& aParameters) {
+  MOZ_ASSERT(aCodecSettings);
+
+  aInfo.codecType = aCodecSettings->codecType;
+  switch (aCodecSettings->codecType) {
+    case webrtc::VideoCodecType::kVideoCodecH264: {
+      aInfo.codecSpecific.H264.packetization_mode =
+          aParameters.count(cricket::kH264FmtpPacketizationMode) == 1 &&
+                  aParameters.at(cricket::kH264FmtpPacketizationMode) == "1"
+              ? webrtc::H264PacketizationMode::NonInterleaved
+              : webrtc::H264PacketizationMode::SingleNalUnit;
+      break;
+    }
+    case webrtc::VideoCodecType::kVideoCodecVP9: {
+      MOZ_ASSERT(aCodecSettings->VP9().numberOfSpatialLayers == 1);
+      aInfo.codecSpecific.VP9.flexible_mode =
+          aCodecSettings->VP9().flexibleMode;
+      aInfo.codecSpecific.VP9.first_frame_in_picture = true;
+      break;
+    }
+    default:
+      break;
+  }
+}
+
+int32_t WebrtcMediaDataEncoder::InitEncode(
+    const webrtc::VideoCodec* aCodecSettings,
+    const webrtc::VideoEncoder::Settings& aSettings) {
+  MOZ_ASSERT(aCodecSettings);
+
+  if (aCodecSettings->numberOfSimulcastStreams > 1) {
+    LOG("Only one stream is supported. Falling back to simulcast adaptor");
+    return WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED;
+  }
+
+  // TODO: enable max output size setting when supported.
+  if (aCodecSettings->codecType == webrtc::VideoCodecType::kVideoCodecH264 &&
+      !(mFormatParams.count(cricket::kH264FmtpPacketizationMode) == 1 &&
+        mFormatParams.at(cricket::kH264FmtpPacketizationMode) == "1")) {
+    LOG("Some platform encoders don't support setting max output size."
+        " Falling back to SW");
+    return WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
+  }
+
+  if (mEncoder) {
+    // Clean existing encoder.
+    Shutdown();
+  }
+
+  RefPtr<MediaDataEncoder> encoder = CreateEncoder(aCodecSettings);
+  if (!encoder) {
+    LOG("Fail to create encoder. Falling back to SW");
+    return WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
+  }
+
+  InitCodecSpecficInfo(mCodecSpecific, aCodecSettings, mFormatParams);
+  LOG("Init encode, mimeType %s, mode %s", mInfo.mMimeType.get(),
+      PacketModeStr(mCodecSpecific));
+  if (!media::Await(do_AddRef(mTaskQueue), encoder->Init()).IsResolve()) {
+    LOG("Fail to init encoder. Falling back to SW");
+    return WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
+  }
+  mEncoder = std::move(encoder);
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+bool WebrtcMediaDataEncoder::SetupConfig(
+    const webrtc::VideoCodec* aCodecSettings) {
+  MimeTypeResult mimeType =
+      ConvertWebrtcCodecTypeToMimeType(aCodecSettings->codecType);
+  if (!mimeType) {
+    LOG("Get incorrect mime type");
+    return false;
+  }
+  mInfo = VideoInfo(aCodecSettings->width, aCodecSettings->height);
+  mInfo.mMimeType = mimeType.extract();
+  mMaxFrameRate = aCodecSettings->maxFramerate;
+  // Those bitrates in codec setting are all kbps, so we have to covert them to
+  // bps.
+  mMaxBitrateBps = aCodecSettings->maxBitrate * 1000;
+  mMinBitrateBps = aCodecSettings->minBitrate * 1000;
+  mBitrateAdjuster.SetTargetBitrateBps(aCodecSettings->startBitrate * 1000);
+  return true;
+}
+
+already_AddRefed<MediaDataEncoder> WebrtcMediaDataEncoder::CreateEncoder(
+    const webrtc::VideoCodec* aCodecSettings) {
+  if (!SetupConfig(aCodecSettings)) {
+    return nullptr;
+  }
+  const bool swOnly = StaticPrefs::media_webrtc_platformencoder_sw_only();
+  LOG("Request platform encoder for %s, bitRate=%u bps, frameRate=%u"
+      ", sw-only=%d",
+      mInfo.mMimeType.get(), mBitrateAdjuster.GetTargetBitrateBps(),
+      aCodecSettings->maxFramerate, swOnly);
+
+  size_t keyframeInterval = 1;
+  switch (aCodecSettings->codecType) {
+    case webrtc::VideoCodecType::kVideoCodecH264: {
+      keyframeInterval = aCodecSettings->H264().keyFrameInterval;
+      break;
+    }
+    case webrtc::VideoCodecType::kVideoCodecVP8: {
+      keyframeInterval = aCodecSettings->VP8().keyFrameInterval;
+      break;
+    }
+    case webrtc::VideoCodecType::kVideoCodecVP9: {
+      keyframeInterval = aCodecSettings->VP9().keyFrameInterval;
+      break;
+    }
+    default:
+      MOZ_ASSERT_UNREACHABLE("Unsupported codec type");
+      return nullptr;
+  }
+  CreateEncoderParams params(
+      mInfo, MediaDataEncoder::Usage::Realtime,
+      TaskQueue::Create(GetMediaThreadPool(MediaThreadType::PLATFORM_ENCODER),
+                        "WebrtcMediaDataEncoder::mEncoder"),
+      MediaDataEncoder::PixelFormat::YUV420P, aCodecSettings->maxFramerate,
+      keyframeInterval, mBitrateAdjuster.GetTargetBitrateBps());
+  switch (aCodecSettings->codecType) {
+    case webrtc::VideoCodecType::kVideoCodecH264: {
+      params.SetCodecSpecific(
+          MediaDataEncoder::H264Specific(ConvertProfileLevel(mFormatParams)));
+      break;
+    }
+    case webrtc::VideoCodecType::kVideoCodecVP8: {
+      const webrtc::VideoCodecVP8& vp8 = aCodecSettings->VP8();
+      const webrtc::VideoCodecComplexity complexity =
+          aCodecSettings->GetVideoEncoderComplexity();
+      const bool frameDropEnabled = aCodecSettings->GetFrameDropEnabled();
+      params.SetCodecSpecific(MediaDataEncoder::VPXSpecific::VP8(
+          MapComplexity(complexity), false, vp8.numberOfTemporalLayers,
+          vp8.denoisingOn, vp8.automaticResizeOn, frameDropEnabled));
+      break;
+    }
+    case webrtc::VideoCodecType::kVideoCodecVP9: {
+      const webrtc::VideoCodecVP9& vp9 = aCodecSettings->VP9();
+      const webrtc::VideoCodecComplexity complexity =
+          aCodecSettings->GetVideoEncoderComplexity();
+      const bool frameDropEnabled = aCodecSettings->GetFrameDropEnabled();
+      params.SetCodecSpecific(MediaDataEncoder::VPXSpecific::VP9(
+          MapComplexity(complexity), false, vp9.numberOfTemporalLayers,
+          vp9.denoisingOn, vp9.automaticResizeOn, frameDropEnabled,
+          vp9.adaptiveQpMode, vp9.numberOfSpatialLayers, vp9.flexibleMode));
+      break;
+    }
+    default:
+      MOZ_MAKE_COMPILER_ASSUME_IS_UNREACHABLE("Unsupported codec type");
+  }
+  return mFactory->CreateEncoder(params, swOnly);
+}
+
+WebrtcVideoEncoder::EncoderInfo WebrtcMediaDataEncoder::GetEncoderInfo() const {
+  WebrtcVideoEncoder::EncoderInfo info;
+  info.supports_native_handle = false;
+  info.implementation_name = "MediaDataEncoder";
+  info.is_hardware_accelerated = false;
+  info.supports_simulcast = false;
+
+#ifdef MOZ_WIDGET_ANDROID
+  // Assume MediaDataEncoder is used mainly for hardware encoding. 16-alignment
+  // seems required on Android. This could be improved by querying the
+  // underlying encoder.
+  info.requested_resolution_alignment = 16;
+  info.apply_alignment_to_all_simulcast_layers = true;
+#endif
+  return info;
+}
+
+int32_t WebrtcMediaDataEncoder::RegisterEncodeCompleteCallback(
+    webrtc::EncodedImageCallback* aCallback) {
+  MutexAutoLock lock(mCallbackMutex);
+  mCallback = aCallback;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t WebrtcMediaDataEncoder::Shutdown() {
+  LOG("Release encoder");
+  {
+    MutexAutoLock lock(mCallbackMutex);
+    mCallback = nullptr;
+    mError = NS_OK;
+  }
+  if (mEncoder) {
+    media::Await(do_AddRef(mTaskQueue), mEncoder->Shutdown());
+    mEncoder = nullptr;
+  }
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+static already_AddRefed<VideoData> CreateVideoDataFromWebrtcVideoFrame(
+    const webrtc::VideoFrame& aFrame, const bool aIsKeyFrame,
+    const TimeUnit aDuration) {
+  MOZ_ASSERT(aFrame.video_frame_buffer()->type() ==
+                 webrtc::VideoFrameBuffer::Type::kI420,
+             "Only support YUV420!");
+  const webrtc::I420BufferInterface* i420 =
+      aFrame.video_frame_buffer()->GetI420();
+
+  PlanarYCbCrData yCbCrData;
+  yCbCrData.mYChannel = const_cast<uint8_t*>(i420->DataY());
+  yCbCrData.mYStride = i420->StrideY();
+  yCbCrData.mCbChannel = const_cast<uint8_t*>(i420->DataU());
+  yCbCrData.mCrChannel = const_cast<uint8_t*>(i420->DataV());
+  MOZ_ASSERT(i420->StrideU() == i420->StrideV());
+  yCbCrData.mCbCrStride = i420->StrideU();
+  yCbCrData.mPictureRect = gfx::IntRect(0, 0, i420->width(), i420->height());
+  yCbCrData.mChromaSubsampling = gfx::ChromaSubsampling::HALF_WIDTH_AND_HEIGHT;
+
+  RefPtr<PlanarYCbCrImage> image =
+      new RecyclingPlanarYCbCrImage(new BufferRecycleBin());
+  image->CopyData(yCbCrData);
+
+  // Although webrtc::VideoFrame::timestamp_rtp_ will likely be deprecated,
+  // webrtc::EncodedImage and the VPx encoders still use it in the imported
+  // version of libwebrtc. Not using the same timestamp values generates
+  // discontinuous time and confuses the video receiver when switching from
+  // platform to libwebrtc encoder.
+  TimeUnit timestamp =
+      media::TimeUnit(aFrame.timestamp(), cricket::kVideoCodecClockrate);
+  return VideoData::CreateFromImage(image->GetSize(), 0, timestamp, aDuration,
+                                    image, aIsKeyFrame, timestamp);
+}
+
+static void UpdateCodecSpecificInfo(webrtc::CodecSpecificInfo& aInfo,
+                                    const gfx::IntSize& aSize,
+                                    const bool aIsKeyframe) {
+  switch (aInfo.codecType) {
+    case webrtc::VideoCodecType::kVideoCodecVP8: {
+      // See webrtc::VP8EncoderImpl::PopulateCodecSpecific().
+      webrtc::CodecSpecificInfoVP8& vp8 = aInfo.codecSpecific.VP8;
+      vp8.keyIdx = webrtc::kNoKeyIdx;
+      // Cannot be 100% sure unless parsing significant portion of the
+      // bitstream. Treat all frames as referenced just to be safe.
+      vp8.nonReference = false;
+      // One temporal layer only.
+      vp8.temporalIdx = webrtc::kNoTemporalIdx;
+      vp8.layerSync = false;
+      break;
+    }
+    case webrtc::VideoCodecType::kVideoCodecVP9: {
+      // See webrtc::VP9EncoderImpl::PopulateCodecSpecific().
+      webrtc::CodecSpecificInfoVP9& vp9 = aInfo.codecSpecific.VP9;
+      vp9.inter_pic_predicted = !aIsKeyframe;
+      vp9.ss_data_available = aIsKeyframe && !vp9.flexible_mode;
+      // One temporal & spatial layer only.
+      vp9.temporal_idx = webrtc::kNoTemporalIdx;
+      vp9.temporal_up_switch = false;
+      vp9.num_spatial_layers = 1;
+      vp9.end_of_picture = true;
+      vp9.gof_idx = webrtc::kNoGofIdx;
+      vp9.width[0] = aSize.width;
+      vp9.height[0] = aSize.height;
+      break;
+    }
+    default:
+      break;
+  }
+}
+
+static void GetVPXQp(const webrtc::VideoCodecType aType,
+                     webrtc::EncodedImage& aImage) {
+  switch (aType) {
+    case webrtc::VideoCodecType::kVideoCodecVP8:
+      webrtc::vp8::GetQp(aImage.data(), aImage.size(), &(aImage.qp_));
+      break;
+    case webrtc::VideoCodecType::kVideoCodecVP9:
+      webrtc::vp9::GetQp(aImage.data(), aImage.size(), &(aImage.qp_));
+      break;
+    default:
+      break;
+  }
+}
+
+int32_t WebrtcMediaDataEncoder::Encode(
+    const webrtc::VideoFrame& aInputFrame,
+    const std::vector<webrtc::VideoFrameType>* aFrameTypes) {
+  if (!aInputFrame.size() || !aInputFrame.video_frame_buffer() ||
+      aFrameTypes->empty()) {
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+
+  if (!mEncoder) {
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+  {
+    MutexAutoLock lock(mCallbackMutex);
+    if (!mCallback) {
+      return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+    }
+    if (NS_FAILED(mError)) {
+      return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+  }
+
+  LOG_V("Encode frame, type %d size %u", static_cast<int>((*aFrameTypes)[0]),
+        aInputFrame.size());
+  MOZ_ASSERT(aInputFrame.video_frame_buffer()->type() ==
+             webrtc::VideoFrameBuffer::Type::kI420);
+  RefPtr<VideoData> data = CreateVideoDataFromWebrtcVideoFrame(
+      aInputFrame, (*aFrameTypes)[0] == webrtc::VideoFrameType::kVideoFrameKey,
+      TimeUnit::FromSeconds(1.0 / mMaxFrameRate));
+  const gfx::IntSize displaySize = data->mDisplay;
+
+  mEncoder->Encode(data)->Then(
+      mTaskQueue, __func__,
+      [self = RefPtr<WebrtcMediaDataEncoder>(this), this,
+       displaySize](MediaDataEncoder::EncodedData aFrames) {
+        LOG_V("Received encoded frame, nums %zu width %d height %d",
+              aFrames.Length(), displaySize.width, displaySize.height);
+        for (auto& frame : aFrames) {
+          MutexAutoLock lock(mCallbackMutex);
+          if (!mCallback) {
+            break;
+          }
+          webrtc::EncodedImage image;
+          image.SetEncodedData(
+              webrtc::EncodedImageBuffer::Create(frame->Data(), frame->Size()));
+          image._encodedWidth = displaySize.width;
+          image._encodedHeight = displaySize.height;
+          CheckedInt64 time =
+              TimeUnitToFrames(frame->mTime, cricket::kVideoCodecClockrate);
+          if (!time.isValid()) {
+            self->mError = MediaResult(NS_ERROR_DOM_MEDIA_FATAL_ERR,
+                                       "invalid timestamp from encoder");
+            break;
+          }
+          image.SetTimestamp(time.value());
+          image._frameType = frame->mKeyframe
+                                 ? webrtc::VideoFrameType::kVideoFrameKey
+                                 : webrtc::VideoFrameType::kVideoFrameDelta;
+          GetVPXQp(mCodecSpecific.codecType, image);
+          UpdateCodecSpecificInfo(mCodecSpecific, displaySize,
+                                  frame->mKeyframe);
+
+          LOG_V("Send encoded image");
+          self->mCallback->OnEncodedImage(image, &mCodecSpecific);
+          self->mBitrateAdjuster.Update(image.size());
+        }
+      },
+      [self = RefPtr<WebrtcMediaDataEncoder>(this)](const MediaResult aError) {
+        self->mError = aError;
+      });
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t WebrtcMediaDataEncoder::SetRates(
+    const webrtc::VideoEncoder::RateControlParameters& aParameters) {
+  if (!aParameters.bitrate.HasBitrate(0, 0)) {
+    LOG("%s: no bitrate value to set.", __func__);
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+  MOZ_ASSERT(aParameters.bitrate.IsSpatialLayerUsed(0));
+  MOZ_ASSERT(!aParameters.bitrate.IsSpatialLayerUsed(1),
+             "No simulcast support for platform encoder");
+
+  const uint32_t newBitrateBps = aParameters.bitrate.GetBitrate(0, 0);
+  if (newBitrateBps < mMinBitrateBps || newBitrateBps > mMaxBitrateBps) {
+    LOG("%s: bitrate value out of range.", __func__);
+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
+  }
+
+  // We have already been in this bitrate.
+  if (mBitrateAdjuster.GetAdjustedBitrateBps() == newBitrateBps) {
+    return WEBRTC_VIDEO_CODEC_OK;
+  }
+
+  if (!mEncoder) {
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+  {
+    MutexAutoLock lock(mCallbackMutex);
+    if (NS_FAILED(mError)) {
+      return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+  }
+  mBitrateAdjuster.SetTargetBitrateBps(newBitrateBps);
+  LOG("Set bitrate %u bps, minBitrate %u bps, maxBitrate %u bps", newBitrateBps,
+      mMinBitrateBps, mMaxBitrateBps);
+  auto rv =
+      media::Await(do_AddRef(mTaskQueue), mEncoder->SetBitrate(newBitrateBps));
+  return rv.IsResolve() ? WEBRTC_VIDEO_CODEC_OK : WEBRTC_VIDEO_CODEC_ERROR;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataEncoderCodec.h b/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataEncoderCodec.h
new file mode 100644
index 0000000000..bc462cc382
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/WebrtcMediaDataEncoderCodec.h
@@ -0,0 +1,78 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef WebrtcMediaDataEncoderCodec_h__
+#define WebrtcMediaDataEncoderCodec_h__
+
+#include "MediaConduitInterface.h"
+#include "MediaInfo.h"
+#include "MediaResult.h"
+#include "PlatformEncoderModule.h"
+#include "WebrtcGmpVideoCodec.h"
+#include "common_video/include/bitrate_adjuster.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+
+namespace mozilla {
+
+class MediaData;
+class PEMFactory;
+class SharedThreadPool;
+class TaskQueue;
+
+class WebrtcMediaDataEncoder : public RefCountedWebrtcVideoEncoder {
+ public:
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(WebrtcMediaDataEncoder, final);
+
+  static bool CanCreate(const webrtc::VideoCodecType aCodecType);
+
+  explicit WebrtcMediaDataEncoder(const webrtc::SdpVideoFormat& aFormat);
+
+  int32_t InitEncode(const webrtc::VideoCodec* aCodecSettings,
+                     const webrtc::VideoEncoder::Settings& aSettings) override;
+
+  int32_t RegisterEncodeCompleteCallback(
+      webrtc::EncodedImageCallback* aCallback) override;
+
+  int32_t Shutdown() override;
+
+  int32_t Encode(
+      const webrtc::VideoFrame& aFrame,
+      const std::vector<webrtc::VideoFrameType>* aFrameTypes) override;
+
+  int32_t SetRates(
+      const webrtc::VideoEncoder::RateControlParameters& aParameters) override;
+
+  WebrtcVideoEncoder::EncoderInfo GetEncoderInfo() const override;
+  MediaEventSource<uint64_t>* InitPluginEvent() override { return nullptr; }
+
+  MediaEventSource<uint64_t>* ReleasePluginEvent() override { return nullptr; }
+
+ private:
+  virtual ~WebrtcMediaDataEncoder();
+
+  bool SetupConfig(const webrtc::VideoCodec* aCodecSettings);
+  already_AddRefed<MediaDataEncoder> CreateEncoder(
+      const webrtc::VideoCodec* aCodecSettings);
+  bool InitEncoder();
+
+  const RefPtr<TaskQueue> mTaskQueue;
+  const RefPtr<PEMFactory> mFactory;
+  RefPtr<MediaDataEncoder> mEncoder;
+
+  Mutex mCallbackMutex MOZ_UNANNOTATED;  // Protects mCallback and mError.
+  webrtc::EncodedImageCallback* mCallback = nullptr;
+  MediaResult mError = NS_OK;
+
+  VideoInfo mInfo;
+  webrtc::SdpVideoFormat::Parameters mFormatParams;
+  webrtc::CodecSpecificInfo mCodecSpecific;
+  webrtc::BitrateAdjuster mBitrateAdjuster;
+  uint32_t mMaxFrameRate;
+  uint32_t mMinBitrateBps;
+  uint32_t mMaxBitrateBps;
+};
+
+}  // namespace mozilla
+
+#endif  // WebrtcMediaDataEncoderCodec_h__
diff --git a/dom/media/webrtc/libwebrtcglue/WebrtcVideoCodecFactory.cpp b/dom/media/webrtc/libwebrtcglue/WebrtcVideoCodecFactory.cpp
new file mode 100644
index 0000000000..6acec07ea3
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/WebrtcVideoCodecFactory.cpp
@@ -0,0 +1,139 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "WebrtcVideoCodecFactory.h"
+
+#include "GmpVideoCodec.h"
+#include "MediaDataCodec.h"
+#include "VideoConduit.h"
+#include "mozilla/StaticPrefs_media.h"
+
+// libwebrtc includes
+#include "api/rtp_headers.h"
+#include "api/video_codecs/video_codec.h"
+#include "api/video_codecs/video_encoder_software_fallback_wrapper.h"
+#include "media/engine/encoder_simulcast_proxy.h"
+#include "modules/video_coding/codecs/vp8/include/vp8.h"
+#include "modules/video_coding/codecs/vp9/include/vp9.h"
+
+namespace mozilla {
+
+std::unique_ptr<webrtc::VideoDecoder>
+WebrtcVideoDecoderFactory::CreateVideoDecoder(
+    const webrtc::SdpVideoFormat& aFormat) {
+  std::unique_ptr<webrtc::VideoDecoder> decoder;
+  auto type = webrtc::PayloadStringToCodecType(aFormat.name);
+
+  // Attempt to create a decoder using MediaDataDecoder.
+  decoder.reset(MediaDataCodec::CreateDecoder(type, mTrackingId));
+  if (decoder) {
+    return decoder;
+  }
+
+  switch (type) {
+    case webrtc::VideoCodecType::kVideoCodecH264: {
+      // Get an external decoder
+      auto gmpDecoder =
+          WrapUnique(GmpVideoCodec::CreateDecoder(mPCHandle, mTrackingId));
+      mCreatedGmpPluginEvent.Forward(*gmpDecoder->InitPluginEvent());
+      mReleasedGmpPluginEvent.Forward(*gmpDecoder->ReleasePluginEvent());
+      decoder.reset(gmpDecoder.release());
+      break;
+    }
+
+    // Use libvpx decoders as fallbacks.
+    case webrtc::VideoCodecType::kVideoCodecVP8:
+      if (!decoder) {
+        decoder = webrtc::VP8Decoder::Create();
+      }
+      break;
+    case webrtc::VideoCodecType::kVideoCodecVP9:
+      decoder = webrtc::VP9Decoder::Create();
+      break;
+
+    default:
+      break;
+  }
+
+  return decoder;
+}
+
+std::unique_ptr<webrtc::VideoEncoder>
+WebrtcVideoEncoderFactory::CreateVideoEncoder(
+    const webrtc::SdpVideoFormat& aFormat) {
+  if (!mInternalFactory->Supports(aFormat)) {
+    return nullptr;
+  }
+  auto type = webrtc::PayloadStringToCodecType(aFormat.name);
+  switch (type) {
+    case webrtc::VideoCodecType::kVideoCodecVP8:
+      // XXX We might be able to use the simulcast proxy for more codecs, but
+      // that requires testing.
+      return std::make_unique<webrtc::EncoderSimulcastProxy>(
+          mInternalFactory.get(), aFormat);
+    default:
+      return mInternalFactory->CreateVideoEncoder(aFormat);
+  }
+}
+
+bool WebrtcVideoEncoderFactory::InternalFactory::Supports(
+    const webrtc::SdpVideoFormat& aFormat) {
+  switch (webrtc::PayloadStringToCodecType(aFormat.name)) {
+    case webrtc::VideoCodecType::kVideoCodecVP8:
+    case webrtc::VideoCodecType::kVideoCodecVP9:
+    case webrtc::VideoCodecType::kVideoCodecH264:
+      return true;
+    default:
+      return false;
+  }
+}
+
+std::unique_ptr<webrtc::VideoEncoder>
+WebrtcVideoEncoderFactory::InternalFactory::CreateVideoEncoder(
+    const webrtc::SdpVideoFormat& aFormat) {
+  MOZ_ASSERT(Supports(aFormat));
+
+  std::unique_ptr<webrtc::VideoEncoder> platformEncoder;
+  platformEncoder.reset(MediaDataCodec::CreateEncoder(aFormat));
+  const bool fallback = StaticPrefs::media_webrtc_software_encoder_fallback();
+  if (!fallback && platformEncoder) {
+    return platformEncoder;
+  }
+
+  std::unique_ptr<webrtc::VideoEncoder> encoder;
+  switch (webrtc::PayloadStringToCodecType(aFormat.name)) {
+    case webrtc::VideoCodecType::kVideoCodecH264: {
+      // get an external encoder
+      auto gmpEncoder =
+          WrapUnique(GmpVideoCodec::CreateEncoder(aFormat, mPCHandle));
+      mCreatedGmpPluginEvent.Forward(*gmpEncoder->InitPluginEvent());
+      mReleasedGmpPluginEvent.Forward(*gmpEncoder->ReleasePluginEvent());
+      encoder.reset(gmpEncoder.release());
+      break;
+    }
+    // libvpx fallbacks.
+    case webrtc::VideoCodecType::kVideoCodecVP8:
+      if (!encoder) {
+        encoder = webrtc::VP8Encoder::Create();
+      }
+      break;
+    case webrtc::VideoCodecType::kVideoCodecVP9:
+      encoder = webrtc::VP9Encoder::Create();
+      break;
+
+    default:
+      break;
+  }
+  if (fallback && encoder && platformEncoder) {
+    return webrtc::CreateVideoEncoderSoftwareFallbackWrapper(
+        std::move(encoder), std::move(platformEncoder), false);
+  }
+  if (platformEncoder) {
+    return platformEncoder;
+  }
+  return encoder;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/libwebrtcglue/WebrtcVideoCodecFactory.h b/dom/media/webrtc/libwebrtcglue/WebrtcVideoCodecFactory.h
new file mode 100644
index 0000000000..ef5765043f
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/WebrtcVideoCodecFactory.h
@@ -0,0 +1,124 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef DOM_MEDIA_WEBRTC_LIBWEBRTCGLUE_WEBRTCVIDEOCODECFACTORY_H_
+#define DOM_MEDIA_WEBRTC_LIBWEBRTCGLUE_WEBRTCVIDEOCODECFACTORY_H_
+
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "MediaEventSource.h"
+#include "PerformanceRecorder.h"
+
+namespace mozilla {
+class GmpPluginNotifierInterface {
+  virtual void DisconnectAll() = 0;
+  virtual MediaEventSource<uint64_t>& CreatedGmpPluginEvent() = 0;
+  virtual MediaEventSource<uint64_t>& ReleasedGmpPluginEvent() = 0;
+};
+
+class GmpPluginNotifier : public GmpPluginNotifierInterface {
+ public:
+  explicit GmpPluginNotifier(nsCOMPtr<nsISerialEventTarget> aOwningThread)
+      : mOwningThread(std::move(aOwningThread)),
+        mCreatedGmpPluginEvent(mOwningThread),
+        mReleasedGmpPluginEvent(mOwningThread) {}
+
+  ~GmpPluginNotifier() = default;
+
+  void DisconnectAll() override {
+    MOZ_ASSERT(mOwningThread->IsOnCurrentThread());
+    mCreatedGmpPluginEvent.DisconnectAll();
+    mReleasedGmpPluginEvent.DisconnectAll();
+  }
+
+  MediaEventSource<uint64_t>& CreatedGmpPluginEvent() override {
+    return mCreatedGmpPluginEvent;
+  }
+
+  MediaEventSource<uint64_t>& ReleasedGmpPluginEvent() override {
+    return mReleasedGmpPluginEvent;
+  }
+
+ protected:
+  const nsCOMPtr<nsISerialEventTarget> mOwningThread;
+  MediaEventForwarder<uint64_t> mCreatedGmpPluginEvent;
+  MediaEventForwarder<uint64_t> mReleasedGmpPluginEvent;
+};
+
+class WebrtcVideoDecoderFactory : public GmpPluginNotifier,
+                                  public webrtc::VideoDecoderFactory {
+ public:
+  WebrtcVideoDecoderFactory(nsCOMPtr<nsISerialEventTarget> aOwningThread,
+                            std::string aPCHandle, TrackingId aTrackingId)
+      : GmpPluginNotifier(std::move(aOwningThread)),
+        mPCHandle(std::move(aPCHandle)),
+        mTrackingId(std::move(aTrackingId)) {}
+
+  std::vector<webrtc::SdpVideoFormat> GetSupportedFormats() const override {
+    MOZ_CRASH("Unexpected call");
+    return std::vector<webrtc::SdpVideoFormat>();
+  }
+
+  std::unique_ptr<webrtc::VideoDecoder> CreateVideoDecoder(
+      const webrtc::SdpVideoFormat& aFormat) override;
+
+ private:
+  const std::string mPCHandle;
+  const TrackingId mTrackingId;
+};
+
+class WebrtcVideoEncoderFactory : public GmpPluginNotifierInterface,
+                                  public webrtc::VideoEncoderFactory {
+  class InternalFactory : public GmpPluginNotifier,
+                          public webrtc::VideoEncoderFactory {
+   public:
+    InternalFactory(nsCOMPtr<nsISerialEventTarget> aOwningThread,
+                    std::string aPCHandle)
+        : GmpPluginNotifier(std::move(aOwningThread)),
+          mPCHandle(std::move(aPCHandle)) {}
+
+    std::vector<webrtc::SdpVideoFormat> GetSupportedFormats() const override {
+      MOZ_CRASH("Unexpected call");
+      return std::vector<webrtc::SdpVideoFormat>();
+    }
+
+    std::unique_ptr<webrtc::VideoEncoder> CreateVideoEncoder(
+        const webrtc::SdpVideoFormat& aFormat) override;
+
+    bool Supports(const webrtc::SdpVideoFormat& aFormat);
+
+   private:
+    const std::string mPCHandle;
+  };
+
+ public:
+  explicit WebrtcVideoEncoderFactory(
+      nsCOMPtr<nsISerialEventTarget> aOwningThread, std::string aPCHandle)
+      : mInternalFactory(MakeUnique<InternalFactory>(std::move(aOwningThread),
+                                                     std::move(aPCHandle))) {}
+
+  std::vector<webrtc::SdpVideoFormat> GetSupportedFormats() const override {
+    MOZ_CRASH("Unexpected call");
+    return std::vector<webrtc::SdpVideoFormat>();
+  }
+
+  std::unique_ptr<webrtc::VideoEncoder> CreateVideoEncoder(
+      const webrtc::SdpVideoFormat& aFormat) override;
+
+  void DisconnectAll() override { mInternalFactory->DisconnectAll(); }
+
+  MediaEventSource<uint64_t>& CreatedGmpPluginEvent() override {
+    return mInternalFactory->CreatedGmpPluginEvent();
+  }
+  MediaEventSource<uint64_t>& ReleasedGmpPluginEvent() override {
+    return mInternalFactory->ReleasedGmpPluginEvent();
+  }
+
+ private:
+  const UniquePtr<InternalFactory> mInternalFactory;
+};
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/libwebrtcglue/moz.build b/dom/media/webrtc/libwebrtcglue/moz.build
new file mode 100644
index 0000000000..62bac2ffe6
--- /dev/null
+++ b/dom/media/webrtc/libwebrtcglue/moz.build
@@ -0,0 +1,35 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+include("/dom/media/webrtc/third_party_build/webrtc.mozbuild")
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "/dom/media/gmp",  # for GMPLog.h,
+    "/dom/media/webrtc",
+    "/ipc/chromium/src",
+    "/media/libyuv/libyuv/include",
+    "/media/webrtc",
+    "/third_party/libsrtp/src/include",
+    "/third_party/libwebrtc",
+    "/third_party/libwebrtc/third_party/abseil-cpp",
+]
+
+UNIFIED_SOURCES += [
+    "AudioConduit.cpp",
+    "GmpVideoCodec.cpp",
+    "MediaConduitInterface.cpp",
+    "MediaDataCodec.cpp",
+    "SystemTime.cpp",
+    "VideoConduit.cpp",
+    "VideoStreamFactory.cpp",
+    "WebrtcCallWrapper.cpp",
+    "WebrtcGmpVideoCodec.cpp",
+    "WebrtcMediaDataDecoderCodec.cpp",
+    "WebrtcMediaDataEncoderCodec.cpp",
+    "WebrtcVideoCodecFactory.cpp",
+]
+
+FINAL_LIBRARY = "xul"
diff --git a/dom/media/webrtc/metrics.yaml b/dom/media/webrtc/metrics.yaml
new file mode 100644
index 0000000000..77d9112543
--- /dev/null
+++ b/dom/media/webrtc/metrics.yaml
@@ -0,0 +1,358 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+# Adding a new metric? We have docs for that!
+# https://firefox-source-docs.mozilla.org/toolkit/components/glean/user/new_definitions_file.html
+
+---
+$schema: moz://mozilla.org/schemas/glean/metrics/2-0-0
+$tags:
+  - 'Core :: WebRTC'
+
+rtcrtpsender:
+  count:
+    type: counter
+    description: >
+      The number of RTCRtpSenders created.
+    bugs:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_reviews:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_sensitivity:
+      - technical
+    notification_emails:
+      - webrtc-telemetry-alerts@mozilla.com
+    expires: 120
+
+  count_setparameters_compat:
+    type: counter
+    description: >
+      The number of RTCRtpSenders created that use the compatibility mode for
+      setParameters.
+    bugs:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_reviews:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_sensitivity:
+      - technical
+    notification_emails:
+      - webrtc-telemetry-alerts@mozilla.com
+    expires: 120
+
+  used_sendencodings:
+    type: rate
+    description: >
+      The proportion of RTCRtpSenders that were created by an addTransceivers
+      call that was passed a sendEncodings.
+    denominator_metric: rtcrtpsender.count
+    bugs:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_reviews:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_sensitivity:
+      - technical
+    notification_emails:
+      - webrtc-telemetry-alerts@mozilla.com
+    expires: 120
+
+rtcrtpsender.setparameters:
+  warn_no_getparameters:
+    type: rate
+    description: >
+      The proportion of RTCRtpSenders configured with the setParameters compat
+      mode that have warned at least once about a setParameters call because
+      [[LastReturnedParameters]] was not set.  (ie; there was not a recent
+      enough call to getParameters)
+    denominator_metric: rtcrtpsender.count_setparameters_compat
+    bugs:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_reviews:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_sensitivity:
+      - technical
+    notification_emails:
+      - webrtc-telemetry-alerts@mozilla.com
+    expires: 120
+
+  blame_no_getparameters:
+    type: labeled_counter
+    description: >
+      The number of RTCRtpSenders that have warned at least once about
+      a `setParameters` call because `[[LastReturnedParameters]]` was not set,
+      broken down by the eTLD+1 of the site. (ie; there was not a recent
+      enough call to `getParameters`) Collected only on EARLY_BETA_OR_EARLIER.
+    bugs:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1831343
+    data_reviews:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1831343
+    data_sensitivity:
+      - web_activity
+    notification_emails:
+      - webrtc-telemetry-alerts@mozilla.com
+    expires: 120
+
+  warn_length_changed:
+    type: rate
+    description: >
+      The proportion of RTCRtpSenders configured with the setParameters compat
+      mode that have warned at least once about a setParameters call that
+      attempted to change the number of encodings.
+    denominator_metric: rtcrtpsender.count_setparameters_compat
+    bugs:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_reviews:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_sensitivity:
+      - technical
+    notification_emails:
+      - webrtc-telemetry-alerts@mozilla.com
+    expires: 120
+
+  blame_length_changed:
+    type: labeled_counter
+    description: >
+      The number of RTCRtpSenders that have warned at least once about a
+      `setParameters` call that attempted to change the number of encodings,
+      broken down by the eTLD+1 of the site. Collected only on
+      EARLY_BETA_OR_EARLIER.
+    bugs:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1831343
+    data_reviews:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1831343
+    data_sensitivity:
+      - web_activity
+    notification_emails:
+      - webrtc-telemetry-alerts@mozilla.com
+    expires: 120
+
+  warn_rid_changed:
+    type: rate
+    description: >
+      The proportion of RTCRtpSenders configured with the setParameters compat
+      mode that have warned at least once about a setParameters call that
+      attempted to change the rid on an encoding (note that we only check this
+      if the encoding count did not change, see warn_length_changed).
+    denominator_metric: rtcrtpsender.count_setparameters_compat
+    bugs:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+    data_reviews:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+    data_sensitivity:
+      - technical
+    notification_emails:
+      - webrtc-telemetry-alerts@mozilla.com
+    expires: 116
+
+  warn_no_transactionid:
+    type: rate
+    description: >
+      The proportion of RTCRtpSenders configured with the setParameters compat
+      mode that have warned at least once about a setParameters call that did
+      not set the transactionId field.
+    denominator_metric: rtcrtpsender.count_setparameters_compat
+    bugs:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_reviews:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_sensitivity:
+      - technical
+    notification_emails:
+      - webrtc-telemetry-alerts@mozilla.com
+    expires: 120
+
+  blame_no_transactionid:
+    type: labeled_counter
+    description: >
+      The number of RTCRtpSenders that have warned at least once about a
+      `setParameters` call that did not set the transactionId field, broken down
+      by the eTLD+1 of the site. Collected only on EARLY_BETA_OR_EARLIER.
+    bugs:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1831343
+    data_reviews:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1831343
+    data_sensitivity:
+      - web_activity
+    notification_emails:
+      - webrtc-telemetry-alerts@mozilla.com
+    expires: 120
+
+  warn_stale_transactionid:
+    type: rate
+    description: >
+      The proportion of RTCRtpSenders configured with the setParameters compat
+      mode that have warned at least once about a setParameters call that used
+      a stale transaction id.
+    denominator_metric: rtcrtpsender.count_setparameters_compat
+    bugs:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_reviews:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_sensitivity:
+      - technical
+    notification_emails:
+      - webrtc-telemetry-alerts@mozilla.com
+    expires: 120
+
+  blame_stale_transactionid:
+    type: labeled_counter
+    description: >
+      The number of RTCRtpSenders that have warned at least once about a
+      `setParameters` call that used a stale transaction id, broken down by the
+      eTLD+1 of the site. Collected only on EARLY_BETA_OR_EARLIER.
+    bugs:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1831343
+    data_reviews:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1831343
+    data_sensitivity:
+      - web_activity
+    notification_emails:
+      - webrtc-telemetry-alerts@mozilla.com
+    expires: 120
+
+  fail_length_changed:
+    type: rate
+    description: >
+      The proportion of RTCRtpSenders that have thrown an error at least once
+      about a setParameters call that attempted to change the number of
+      encodings.
+    denominator_metric: rtcrtpsender.count
+    bugs:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_reviews:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_sensitivity:
+      - technical
+    notification_emails:
+      - webrtc-telemetry-alerts@mozilla.com
+    expires: 120
+
+  fail_rid_changed:
+    type: rate
+    description: >
+      The proportion of RTCRtpSenders that have thrown an error at least once
+      about a setParameters call that attempted to change the rid on an
+      encoding (note that we only check this if the encoding count did not
+      change, see fail_length_changed).
+    denominator_metric: rtcrtpsender.count
+    bugs:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_reviews:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_sensitivity:
+      - technical
+    notification_emails:
+      - webrtc-telemetry-alerts@mozilla.com
+    expires: 120
+
+  fail_no_getparameters:
+    type: rate
+    description: >
+      The proportion of RTCRtpSenders that have thrown an error at least once
+      about a setParameters call because [[LastReturnedParameters]] was not set.
+      (ie; there was not a recent enough call to getParameters)
+    denominator_metric: rtcrtpsender.count
+    bugs:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_reviews:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_sensitivity:
+      - technical
+    notification_emails:
+      - webrtc-telemetry-alerts@mozilla.com
+    expires: 120
+
+  fail_no_transactionid:
+    type: rate
+    description: >
+      The proportion of RTCRtpSenders that have thrown an error at least once
+      about a setParameters call that did not set the transactionId field.
+    denominator_metric: rtcrtpsender.count
+    bugs:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_reviews:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_sensitivity:
+      - technical
+    notification_emails:
+      - webrtc-telemetry-alerts@mozilla.com
+    expires: 120
+
+  fail_stale_transactionid:
+    type: rate
+    description: >
+      The proportion of RTCRtpSenders that have thrown an error at least once
+      about a setParameters call that used a stale transaction id.
+    denominator_metric: rtcrtpsender.count
+    bugs:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_reviews:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_sensitivity:
+      - technical
+    notification_emails:
+      - webrtc-telemetry-alerts@mozilla.com
+    expires: 120
+
+  fail_no_encodings:
+    type: rate
+    description: >
+      The proportion of RTCRtpSenders configured with the setParameters compat
+      mode that have thrown an error at least once about a setParameters call
+      that had no encodings (we do not measure this against the general
+      population of RTCRtpSenders, since without the compat mode this failure
+      is never observed, because it fails the length change check).
+    denominator_metric: rtcrtpsender.count_setparameters_compat
+    bugs:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_reviews:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_sensitivity:
+      - technical
+    notification_emails:
+      - webrtc-telemetry-alerts@mozilla.com
+    expires: 120
+
+  fail_other:
+    type: rate
+    description: >
+      The proportion of RTCRtpSenders that have thrown an error at least once
+      about a setParameters call that had no encodings.
+    denominator_metric: rtcrtpsender.count
+    bugs:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_reviews:
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1401592
+      - https://bugzilla.mozilla.org/show_bug.cgi?id=1832459
+    data_sensitivity:
+      - technical
+    notification_emails:
+      - webrtc-telemetry-alerts@mozilla.com
+    expires: 120
diff --git a/dom/media/webrtc/moz.build b/dom/media/webrtc/moz.build
new file mode 100644
index 0000000000..cb8e363faa
--- /dev/null
+++ b/dom/media/webrtc/moz.build
@@ -0,0 +1,132 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+include("/dom/media/webrtc/third_party_build/webrtc.mozbuild")
+
+with Files("**"):
+    BUG_COMPONENT = ("Core", "WebRTC: Audio/Video")
+
+with Files("PeerIdentity.*"):
+    BUG_COMPONENT = ("Core", "WebRTC: Signaling")
+
+with Files("common/**"):
+    BUG_COMPONENT = ("Core", "WebRTC: Signaling")
+with Files("jsep/**"):
+    BUG_COMPONENT = ("Core", "WebRTC: Signaling")
+with Files("libwebrtcglue/**"):
+    BUG_COMPONENT = ("Core", "WebRTC: Signaling")
+with Files("transportbridge/**"):
+    BUG_COMPONENT = ("Core", "WebRTC: Signaling")
+with Files("jsapi/**"):
+    BUG_COMPONENT = ("Core", "WebRTC: Signaling")
+with Files("sdp/**"):
+    BUG_COMPONENT = ("Core", "WebRTC: Signaling")
+with Files("third_party_build/**"):
+    BUG_COMPONENT = ("Core", "WebRTC")
+
+
+EXPORTS += [
+    "CubebDeviceEnumerator.h",
+    "MediaEngine.h",
+    "MediaEngineFake.h",
+    "MediaEnginePrefs.h",
+    "MediaEngineSource.h",
+    "MediaTrackConstraints.h",
+    "SineWaveGenerator.h",
+]
+
+SOURCES += [
+    "CubebDeviceEnumerator.cpp",
+]
+
+if CONFIG["MOZ_WEBRTC"]:
+    EXPORTS += [
+        "MediaEngineRemoteVideoSource.h",
+        "MediaEngineWebRTC.h",
+        "MediaEngineWebRTCAudio.h",
+    ]
+    EXPORTS.mozilla.dom += ["RTCIdentityProviderRegistrar.h"]
+    UNIFIED_SOURCES += [
+        "MediaEngineRemoteVideoSource.cpp",
+        "MediaEngineWebRTCAudio.cpp",
+        "RTCCertificate.cpp",
+        "RTCIdentityProviderRegistrar.cpp",
+    ]
+    # MediaEngineWebRTC.cpp needs to be built separately.
+    SOURCES += [
+        "MediaEngineWebRTC.cpp",
+    ]
+    PREPROCESSED_IPDL_SOURCES += [
+        "PWebrtcGlobal.ipdl",
+    ]
+    LOCAL_INCLUDES += [
+        "..",
+        "/dom/base",
+        "/dom/media",
+        "/dom/media/webrtc/common",
+        "/dom/media/webrtc/common/browser_logging",
+        "/media/libyuv/libyuv/include",
+        "/third_party/libwebrtc",
+        "/third_party/libwebrtc/third_party/abseil-cpp",
+    ]
+
+if CONFIG["MOZ_WEBRTC_SIGNALING"]:
+    DIRS += [
+        "common",
+        "jsapi",
+        "jsep",
+        "libwebrtcglue",
+        "sdp",
+        "transportbridge",
+        "/third_party/libwebrtc",
+    ]
+
+    if CONFIG["MOZ_WIDGET_TOOLKIT"] == "gtk":
+        DIRS += [
+            "/third_party/drm/",
+            "/third_party/drm/libdrm",
+            "/third_party/gbm/",
+            "/third_party/gbm/libgbm",
+            "/third_party/libepoxy/",
+            "/third_party/pipewire/libpipewire",
+        ]
+
+    # Avoid warnings from third-party code that we can not modify.
+    if CONFIG["CC_TYPE"] == "clang-cl":
+        CXXFLAGS += ["-Wno-invalid-source-encoding"]
+
+
+PREPROCESSED_IPDL_SOURCES += [
+    "PMediaTransport.ipdl",
+]
+
+UNIFIED_SOURCES += [
+    "MediaEngineFake.cpp",
+    "MediaEngineSource.cpp",
+    "MediaTrackConstraints.cpp",
+    "PeerIdentity.cpp",
+]
+
+EXPORTS.mozilla += [
+    "PeerIdentity.h",
+]
+EXPORTS.mozilla.dom += [
+    "RTCCertificate.h",
+]
+
+include("/ipc/chromium/chromium-config.mozbuild")
+
+# Suppress some GCC/clang warnings being treated as errors:
+#  - about attributes on forward declarations for types that are already
+#    defined, which complains about important MOZ_EXPORT attributes for
+#    android API types
+CXXFLAGS += [
+    "-Wno-error=attributes",
+]
+
+FINAL_LIBRARY = "xul"
+
+# Add libFuzzer configuration directives
+include("/tools/fuzzing/libfuzzer-config.mozbuild")
diff --git a/dom/media/webrtc/sdp/HybridSdpParser.cpp b/dom/media/webrtc/sdp/HybridSdpParser.cpp
new file mode 100644
index 0000000000..d1f2d5c058
--- /dev/null
+++ b/dom/media/webrtc/sdp/HybridSdpParser.cpp
@@ -0,0 +1,88 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+#include "sdp/HybridSdpParser.h"
+#include "sdp/SdpLog.h"
+#include "sdp/SdpPref.h"
+#include "sdp/SdpTelemetry.h"
+#include "sdp/SipccSdpParser.h"
+#include "sdp/RsdparsaSdpParser.h"
+#include "sdp/ParsingResultComparer.h"
+
+#include "mozilla/Logging.h"
+#include "mozilla/Preferences.h"
+#include "mozilla/Telemetry.h"
+
+#include <unordered_map>
+
+namespace mozilla {
+
+using mozilla::LogLevel;
+
+const std::string& HybridSdpParser::ParserName() {
+  const static std::string PARSER_NAME = "hybrid";
+  return PARSER_NAME;
+}
+
+HybridSdpParser::HybridSdpParser()
+    : mStrictSuccess(SdpPref::StrictSuccess()),
+      mPrimary(SdpPref::Primary()),
+      mSecondary(SdpPref::Secondary()),
+      mFailover(SdpPref::Failover()) {
+  MOZ_ASSERT(!(mSecondary && mFailover),
+             "Can not have both a secondary and failover parser!");
+  MOZ_LOG(SdpLog, LogLevel::Info,
+          ("Primary SDP Parser: %s", mPrimary->Name().c_str()));
+  mSecondary.apply([](auto& parser) {
+    MOZ_LOG(SdpLog, LogLevel::Info,
+            ("Secondary SDP Logger: %s", parser->Name().c_str()));
+  });
+  mFailover.apply([](auto& parser) {
+    MOZ_LOG(SdpLog, LogLevel::Info,
+            ("Failover SDP Logger: %s", parser->Name().c_str()));
+  });
+}
+
+auto HybridSdpParser::Parse(const std::string& aText)
+    -> UniquePtr<SdpParser::Results> {
+  using Results = UniquePtr<SdpParser::Results>;
+  using Role = SdpTelemetry::Roles;
+  using Mode = SdpPref::AlternateParseModes;
+
+  Mode mode = Mode::Never;
+  auto results = mPrimary->Parse(aText);
+
+  auto successful = [&](Results& aRes) -> bool {
+    // In strict mode any reported error counts as failure
+    if (mStrictSuccess) {
+      return aRes->Ok();
+    }
+    return aRes->Sdp() != nullptr;
+  };
+  // Pass results on for comparison and return A if it was a success and B
+  // otherwise.
+  auto compare = [&](Results&& aResB) -> Results {
+    SdpTelemetry::RecordParse(aResB, mode, Role::Secondary);
+    ParsingResultComparer::Compare(results, aResB, aText, mode);
+    return std::move(successful(results) ? results : aResB);
+  };
+  // Run secondary parser, if there is one, and update selected results.
+  mSecondary.apply([&](auto& sec) {
+    mode = Mode::Parallel;
+    results = compare(std::move(sec->Parse(aText)));
+  });
+  // Run failover parser, if there is one, and update selected results.
+  mFailover.apply([&](auto& failover) {  // Only run if primary parser failed
+    mode = Mode::Failover;
+    if (!successful(results)) {
+      results = compare(std::move(failover->Parse(aText)));
+    }
+  });
+
+  SdpTelemetry::RecordParse(results, mode, Role::Primary);
+  return results;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/sdp/HybridSdpParser.h b/dom/media/webrtc/sdp/HybridSdpParser.h
new file mode 100644
index 0000000000..2b0ef399a5
--- /dev/null
+++ b/dom/media/webrtc/sdp/HybridSdpParser.h
@@ -0,0 +1,38 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _HYBRIDSDPPARSER_H_
+#define _HYBRIDSDPPARSER_H_
+
+#include "sdp/SdpParser.h"
+#include "sdp/SdpTelemetry.h"
+
+namespace mozilla {
+
+// This shim parser delegates parsing to WEbRTC-SDP and SIPCC, based on
+// preference flags. Additionally it handles collecting telemetry and fallback
+// behavior between the parsers.
+class HybridSdpParser : public SdpParser {
+  static const std::string& ParserName();
+
+ public:
+  HybridSdpParser();
+  virtual ~HybridSdpParser() = default;
+
+  auto Name() const -> const std::string& override { return ParserName(); }
+  auto Parse(const std::string& aText)
+      -> UniquePtr<SdpParser::Results> override;
+
+ private:
+  const bool mStrictSuccess;
+  const UniquePtr<SdpParser> mPrimary;
+  const Maybe<UniquePtr<SdpParser>> mSecondary;
+  const Maybe<UniquePtr<SdpParser>> mFailover;
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/sdp/ParsingResultComparer.cpp b/dom/media/webrtc/sdp/ParsingResultComparer.cpp
new file mode 100644
index 0000000000..8592fa52b3
--- /dev/null
+++ b/dom/media/webrtc/sdp/ParsingResultComparer.cpp
@@ -0,0 +1,331 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "sdp/Sdp.h"
+#include "sdp/ParsingResultComparer.h"
+#include "sdp/SipccSdpParser.h"
+#include "sdp/RsdparsaSdpParser.h"
+#include "sdp/SdpTelemetry.h"
+
+#include <string>
+#include <ostream>
+#include <regex>
+
+#include "mozilla/Assertions.h"
+#include "mozilla/Telemetry.h"
+#include "mozilla/Logging.h"
+
+using mozilla::LogLevel;
+static mozilla::LazyLogModule sSdpDiffLogger("sdpdiff_logger");
+
+#define LOGD(msg) MOZ_LOG(sSdpDiffLogger, LogLevel::Debug, msg)
+#define LOGE(msg) MOZ_LOG(sSdpDiffLogger, LogLevel::Error, msg)
+
+#define LOG_EXPECT(result, expect, msg)                         \
+  {                                                             \
+    if (((expect) == SdpComparisonResult::Equal) == (result)) { \
+      LOGD(msg);                                                \
+    } else {                                                    \
+      LOGE(("UNEXPECTED COMPARISON RESULT: vvvvvv"));           \
+      LOGE(msg);                                                \
+    }                                                           \
+  }
+
+namespace mozilla {
+
+using AttributeType = SdpAttribute::AttributeType;
+
+template <typename T>
+std::string ToString(const T& serializable) {
+  std::ostringstream os;
+
+  os << serializable;
+  return os.str();
+}
+bool ParsingResultComparer::Compare(const Results& aResA, const Results& aResB,
+                                    const std::string& aOriginalSdp,
+                                    const SdpPref::AlternateParseModes& aMode) {
+  MOZ_ASSERT(aResA, "aResA must not be a nullptr");
+  MOZ_ASSERT(aResB, "aResB must not be a nullptr");
+  MOZ_ASSERT(aResA->ParserName() != aResB->ParserName(),
+             "aResA and aResB must be from different parsers");
+  SdpTelemetry::RecordCompare(aResA, aResB, aMode);
+
+  ParsingResultComparer comparer;
+  if (!aResA->Sdp() || !aResB->Sdp()) {
+    return !aResA->Sdp() && !aResB->Sdp();
+  }
+  if (SipccSdpParser::IsNamed(aResA->ParserName())) {
+    MOZ_ASSERT(RsdparsaSdpParser::IsNamed(aResB->ParserName()));
+    return comparer.Compare(*aResB->Sdp(), *aResA->Sdp(), aOriginalSdp,
+                            SdpComparisonResult::Equal);
+  }
+  MOZ_ASSERT(SipccSdpParser::IsNamed(aResB->ParserName()));
+  MOZ_ASSERT(RsdparsaSdpParser::IsNamed(aResA->ParserName()));
+  return comparer.Compare(*aResA->Sdp(), *aResB->Sdp(), aOriginalSdp,
+                          SdpComparisonResult::Equal);
+}
+
+bool ParsingResultComparer::Compare(const Sdp& rsdparsaSdp, const Sdp& sipccSdp,
+                                    const std::string& originalSdp,
+                                    const SdpComparisonResult expect) {
+  mOriginalSdp = originalSdp;
+  const std::string sipccSdpStr = sipccSdp.ToString();
+  const std::string rsdparsaSdpStr = rsdparsaSdp.ToString();
+
+  bool result = rsdparsaSdpStr == sipccSdpStr;
+  LOG_EXPECT(result, expect, ("The original sdp: \n%s", mOriginalSdp.c_str()));
+  if (result) {
+    Telemetry::ScalarAdd(Telemetry::ScalarID::WEBRTC_SDP_PARSER_DIFF,
+                         u"serialization_is_equal"_ns, 1);
+    LOG_EXPECT(result, expect, ("Serialization is equal"));
+    return result;
+  }
+  // Do a deep comparison
+  result = true;
+
+  Telemetry::ScalarAdd(Telemetry::ScalarID::WEBRTC_SDP_PARSER_DIFF,
+                       u"serialization_is_not_equal"_ns, 1);
+  LOG_EXPECT(result, expect,
+             ("Serialization is not equal\n"
+              " --- Sipcc SDP ---\n"
+              "%s\n"
+              "--- Rsdparsa SDP ---\n"
+              "%s\n",
+              sipccSdpStr.c_str(), rsdparsaSdpStr.c_str()));
+
+  const std::string rsdparsaOriginStr = ToString(rsdparsaSdp.GetOrigin());
+  const std::string sipccOriginStr = ToString(sipccSdp.GetOrigin());
+
+  // Compare the session level
+  if (rsdparsaOriginStr != sipccOriginStr) {
+    Telemetry::ScalarAdd(Telemetry::ScalarID::WEBRTC_SDP_PARSER_DIFF, u"o="_ns,
+                         1);
+    result = false;
+    LOG_EXPECT(result, expect,
+               ("origin is not equal\nrust origin: %s\nsipcc origin: %s",
+                rsdparsaOriginStr.c_str(), sipccOriginStr.c_str()));
+  }
+
+  if (MOZ_LOG_TEST(sSdpDiffLogger, LogLevel::Debug)) {
+    const auto rust_sess_attr_count = rsdparsaSdp.GetAttributeList().Count();
+    const auto sipcc_sess_attr_count = sipccSdp.GetAttributeList().Count();
+
+    if (rust_sess_attr_count != sipcc_sess_attr_count) {
+      LOG_EXPECT(false, expect,
+                 ("Session level attribute count is NOT equal, rsdparsa: %u, "
+                  "sipcc: %u\n",
+                  rust_sess_attr_count, sipcc_sess_attr_count));
+    }
+  }
+
+  result &= CompareAttrLists(rsdparsaSdp.GetAttributeList(),
+                             sipccSdp.GetAttributeList(), -1);
+
+  const uint32_t sipccMediaSecCount =
+      static_cast<uint32_t>(sipccSdp.GetMediaSectionCount());
+  const uint32_t rsdparsaMediaSecCount =
+      static_cast<uint32_t>(rsdparsaSdp.GetMediaSectionCount());
+
+  if (sipccMediaSecCount != rsdparsaMediaSecCount) {
+    result = false;
+    Telemetry::ScalarAdd(Telemetry::ScalarID::WEBRTC_SDP_PARSER_DIFF,
+                         u"inequal_msec_count"_ns, 1);
+    LOG_EXPECT(result, expect,
+               ("Media section count is NOT equal, rsdparsa: %d, sipcc: %d \n",
+                rsdparsaMediaSecCount, sipccMediaSecCount));
+  }
+
+  for (size_t i = 0; i < std::min(sipccMediaSecCount, rsdparsaMediaSecCount);
+       i++) {
+    result &= CompareMediaSections(rsdparsaSdp.GetMediaSection(i),
+                                   sipccSdp.GetMediaSection(i));
+  }
+
+  return result;
+}
+
+bool ParsingResultComparer::CompareMediaSections(
+    const SdpMediaSection& rustMediaSection,
+    const SdpMediaSection& sipccMediaSection,
+    const SdpComparisonResult expect) const {
+  bool result = true;
+  auto trackMediaLineMismatch = [&result, &expect](
+                                    auto rustValue, auto sipccValue,
+                                    const nsString& valueDescription) {
+    result = false;
+    nsString typeStr = u"m="_ns;
+    typeStr += valueDescription;
+    Telemetry::ScalarAdd(Telemetry::ScalarID::WEBRTC_SDP_PARSER_DIFF, typeStr,
+                         1);
+    LOG_EXPECT(result, expect,
+               ("The media line values %s are not equal\n"
+                "rsdparsa value: %s\n"
+                "sipcc value: %s\n",
+                NS_LossyConvertUTF16toASCII(valueDescription).get(),
+                ToString(rustValue).c_str(), ToString(sipccValue).c_str()));
+  };
+
+  auto compareMediaLineValue = [trackMediaLineMismatch](
+                                   auto rustValue, auto sipccValue,
+                                   const nsString& valueDescription) {
+    if (rustValue != sipccValue) {
+      trackMediaLineMismatch(rustValue, sipccValue, valueDescription);
+    }
+  };
+
+  auto compareSimpleMediaLineValue =
+      [&rustMediaSection, &sipccMediaSection, compareMediaLineValue](
+          auto valGetFuncPtr, const nsString& valueDescription) {
+        compareMediaLineValue((rustMediaSection.*valGetFuncPtr)(),
+                              (sipccMediaSection.*valGetFuncPtr)(),
+                              valueDescription);
+      };
+
+  compareSimpleMediaLineValue(&SdpMediaSection::GetMediaType, u"media_type"_ns);
+  compareSimpleMediaLineValue(&SdpMediaSection::GetPort, u"port"_ns);
+  compareSimpleMediaLineValue(&SdpMediaSection::GetPortCount, u"port_count"_ns);
+  compareSimpleMediaLineValue(&SdpMediaSection::GetProtocol, u"protocol"_ns);
+  compareSimpleMediaLineValue(&SdpMediaSection::IsReceiving,
+                              u"is_receiving"_ns);
+  compareSimpleMediaLineValue(&SdpMediaSection::IsSending, u"is_sending"_ns);
+  compareSimpleMediaLineValue(&SdpMediaSection::GetDirection, u"direction"_ns);
+  compareSimpleMediaLineValue(&SdpMediaSection::GetLevel, u"level"_ns);
+
+  compareMediaLineValue(ToString(rustMediaSection.GetConnection()),
+                        ToString(sipccMediaSection.GetConnection()),
+                        u"connection"_ns);
+
+  result &= CompareAttrLists(rustMediaSection.GetAttributeList(),
+                             sipccMediaSection.GetAttributeList(),
+                             static_cast<int>(rustMediaSection.GetLevel()));
+  return result;
+}
+
+bool ParsingResultComparer::CompareAttrLists(
+    const SdpAttributeList& rustAttrlist, const SdpAttributeList& sipccAttrlist,
+    int level, const SdpComparisonResult expect) const {
+  bool result = true;
+
+  for (size_t i = AttributeType::kFirstAttribute;
+       i <= static_cast<size_t>(AttributeType::kLastAttribute); i++) {
+    const AttributeType type = static_cast<AttributeType>(i);
+    std::string attrStr;
+    if (type != AttributeType::kDirectionAttribute) {
+      attrStr = "a=" + SdpAttribute::GetAttributeTypeString(type);
+    } else {
+      attrStr = "a=_direction_attribute_";
+    }
+
+    if (sipccAttrlist.HasAttribute(type, false)) {
+      auto sipccAttrStr = ToString(*sipccAttrlist.GetAttribute(type, false));
+
+      if (!rustAttrlist.HasAttribute(type, false)) {
+        result = false;
+        nsString typeStr;
+        typeStr.AssignASCII(attrStr.c_str());
+        typeStr += u"_missing"_ns;
+        Telemetry::ScalarAdd(Telemetry::ScalarID::WEBRTC_SDP_PARSER_DIFF,
+                             typeStr, 1);
+        LOG_EXPECT(result, expect,
+                   ("Rust is missing the attribute: %s\n", attrStr.c_str()));
+        LOG_EXPECT(result, expect,
+                   ("Rust is missing: %s\n", sipccAttrStr.c_str()));
+
+        continue;
+      }
+
+      auto rustAttrStr = ToString(*rustAttrlist.GetAttribute(type, false));
+
+      if (rustAttrStr != sipccAttrStr) {
+        if (type == AttributeType::kFmtpAttribute) {
+          if (rustAttrlist.GetFmtp() == sipccAttrlist.GetFmtp()) {
+            continue;
+          }
+        }
+
+        std::string originalAttrStr = GetAttributeLines(attrStr, level);
+        if (rustAttrStr != originalAttrStr) {
+          result = false;
+          nsString typeStr;
+          typeStr.AssignASCII(attrStr.c_str());
+          typeStr += u"_inequal"_ns;
+          Telemetry::ScalarAdd(Telemetry::ScalarID::WEBRTC_SDP_PARSER_DIFF,
+                               typeStr, 1);
+          LOG_EXPECT(result, expect,
+                     ("%s is neither equal to sipcc nor to the orginal sdp\n"
+                      "--------------rsdparsa attribute---------------\n"
+                      "%s"
+                      "--------------sipcc attribute---------------\n"
+                      "%s"
+                      "--------------original attribute---------------\n"
+                      "%s\n",
+                      attrStr.c_str(), rustAttrStr.c_str(),
+                      sipccAttrStr.c_str(), originalAttrStr.c_str()));
+        } else {
+          LOG_EXPECT(
+              result, expect,
+              ("But the rust serialization is equal to the orignal sdp\n"));
+        }
+      }
+    } else {
+      if (rustAttrlist.HasAttribute(type, false)) {
+        nsString typeStr;
+        typeStr.AssignASCII(attrStr.c_str());
+        typeStr += u"_unexpected"_ns;
+        Telemetry::ScalarAdd(Telemetry::ScalarID::WEBRTC_SDP_PARSER_DIFF,
+                             typeStr, 1);
+      }
+    }
+  }
+
+  return result;
+}
+
+std::vector<std::string> SplitLines(const std::string& sdp) {
+  std::stringstream ss(sdp);
+  std::string to;
+  std::vector<std::string> lines;
+
+  while (std::getline(ss, to, '\n')) {
+    lines.push_back(to);
+  }
+
+  return lines;
+}
+
+std::string ParsingResultComparer::GetAttributeLines(
+    const std::string& attrType, int level) const {
+  std::vector<std::string> lines = SplitLines(mOriginalSdp);
+  std::string attrToFind = attrType + ":";
+  std::string attrLines;
+  int currentLevel = -1;
+  // Filters rtcp-fb lines that contain "x-..." types
+  // This is because every SDP from Edge contains these rtcp-fb x- types
+  // for example: a=rtcp-fb:121 x-foo
+  std::regex customRtcpFbLines(R"(a\=rtcp\-fb\:(\d+|\*).* x\-.*)");
+
+  for (auto& line : lines) {
+    if (line.find("m=") == 0) {
+      if (level > currentLevel) {
+        attrLines.clear();
+        currentLevel++;
+      } else {
+        break;
+      }
+    } else if (line.find(attrToFind) == 0) {
+      if (std::regex_match(line, customRtcpFbLines)) {
+        continue;
+      }
+
+      attrLines += (line + '\n');
+    }
+  }
+
+  return attrLines;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/sdp/ParsingResultComparer.h b/dom/media/webrtc/sdp/ParsingResultComparer.h
new file mode 100644
index 0000000000..5a1c2ac635
--- /dev/null
+++ b/dom/media/webrtc/sdp/ParsingResultComparer.h
@@ -0,0 +1,57 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _PARSINGRESULTCOMPARER_H_
+#define _PARSINGRESULTCOMPARER_H_
+
+#include "sdp/SdpParser.h"
+#include "sdp/SdpPref.h"
+
+#include <string>
+
+namespace mozilla {
+
+class Sdp;
+class SdpMediaSection;
+class SdpAttributeList;
+
+enum class SdpComparisonResult {
+  Inequal = false,
+  Equal = true,
+};
+
+class ParsingResultComparer {
+ public:
+  using Results = UniquePtr<SdpParser::Results>;
+
+  ParsingResultComparer() = default;
+
+  static bool Compare(const Results& aResA, const Results& aResB,
+                      const std::string& aOrignalSdp,
+                      const SdpPref::AlternateParseModes& aMode);
+  bool Compare(const Sdp& rsdparsaSdp, const Sdp& sipccSdp,
+               const std::string& aOriginalSdp,
+               const SdpComparisonResult expect = SdpComparisonResult::Equal);
+  bool CompareMediaSections(
+      const SdpMediaSection& rustMediaSection,
+      const SdpMediaSection& sipccMediaSection,
+      const SdpComparisonResult expect = SdpComparisonResult::Equal) const;
+  bool CompareAttrLists(
+      const SdpAttributeList& rustAttrlist,
+      const SdpAttributeList& sipccAttrlist, int level,
+      const SdpComparisonResult expect = SdpComparisonResult::Equal) const;
+  void TrackRustParsingFailed(size_t sipccErrorCount) const;
+  void TrackSipccParsingFailed(size_t rustErrorCount) const;
+
+ private:
+  std::string mOriginalSdp;
+
+  std::string GetAttributeLines(const std::string& attrType, int level) const;
+};
+
+}  // namespace mozilla
+
+#endif  // _PARSINGRESULTCOMPARER_H_
diff --git a/dom/media/webrtc/sdp/RsdparsaSdp.cpp b/dom/media/webrtc/sdp/RsdparsaSdp.cpp
new file mode 100644
index 0000000000..0e653dc17e
--- /dev/null
+++ b/dom/media/webrtc/sdp/RsdparsaSdp.cpp
@@ -0,0 +1,126 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "sdp/RsdparsaSdp.h"
+
+#include <cstdlib>
+#include "mozilla/UniquePtr.h"
+#include "mozilla/Assertions.h"
+#include "nsError.h"
+#include <iostream>
+
+#include "sdp/RsdparsaSdpInc.h"
+#include "sdp/RsdparsaSdpMediaSection.h"
+
+#ifdef CRLF
+#  undef CRLF
+#endif
+#define CRLF "\r\n"
+
+namespace mozilla {
+
+RsdparsaSdp::RsdparsaSdp(RsdparsaSessionHandle session, const SdpOrigin& origin)
+    : mSession(std::move(session)), mOrigin(origin) {
+  RsdparsaSessionHandle attributeSession(sdp_new_reference(mSession.get()));
+  mAttributeList.reset(
+      new RsdparsaSdpAttributeList(std::move(attributeSession)));
+
+  size_t section_count = sdp_media_section_count(mSession.get());
+  for (size_t level = 0; level < section_count; level++) {
+    RustMediaSection* mediaSection =
+        sdp_get_media_section(mSession.get(), level);
+    if (!mediaSection) {
+      MOZ_ASSERT(false,
+                 "sdp_get_media_section failed because level was out of"
+                 " bounds, but we did a bounds check!");
+      break;
+    }
+    RsdparsaSessionHandle newSession(sdp_new_reference(mSession.get()));
+    RsdparsaSdpMediaSection* sdpMediaSection;
+    sdpMediaSection = new RsdparsaSdpMediaSection(
+        level, std::move(newSession), mediaSection, mAttributeList.get());
+    mMediaSections.emplace_back(sdpMediaSection);
+  }
+}
+
+RsdparsaSdp::RsdparsaSdp(const RsdparsaSdp& aOrig)
+    : RsdparsaSdp(RsdparsaSessionHandle(create_sdp_clone(aOrig.mSession.get())),
+                  aOrig.mOrigin) {}
+
+Sdp* RsdparsaSdp::Clone() const { return new RsdparsaSdp(*this); }
+
+const SdpOrigin& RsdparsaSdp::GetOrigin() const { return mOrigin; }
+
+uint32_t RsdparsaSdp::GetBandwidth(const std::string& type) const {
+  return get_sdp_bandwidth(mSession.get(), type.c_str());
+}
+
+const SdpMediaSection& RsdparsaSdp::GetMediaSection(size_t level) const {
+  if (level > mMediaSections.size()) {
+    MOZ_CRASH();
+  }
+  return *mMediaSections[level];
+}
+
+SdpMediaSection& RsdparsaSdp::GetMediaSection(size_t level) {
+  if (level > mMediaSections.size()) {
+    MOZ_CRASH();
+  }
+  return *mMediaSections[level];
+}
+
+SdpMediaSection& RsdparsaSdp::AddMediaSection(
+    SdpMediaSection::MediaType mediaType, SdpDirectionAttribute::Direction dir,
+    uint16_t port, SdpMediaSection::Protocol protocol, sdp::AddrType addrType,
+    const std::string& addr) {
+  StringView rustAddr{addr.c_str(), addr.size()};
+  auto nr = sdp_add_media_section(mSession.get(), mediaType, dir, port,
+                                  protocol, addrType, rustAddr);
+
+  if (NS_SUCCEEDED(nr)) {
+    size_t level = mMediaSections.size();
+    RsdparsaSessionHandle newSessHandle(sdp_new_reference(mSession.get()));
+
+    auto rustMediaSection = sdp_get_media_section(mSession.get(), level);
+    auto mediaSection =
+        new RsdparsaSdpMediaSection(level, std::move(newSessHandle),
+                                    rustMediaSection, mAttributeList.get());
+    mMediaSections.emplace_back(mediaSection);
+
+    return *mediaSection;
+  } else {
+    // Return the last media section if the construction of this one fails
+    return GetMediaSection(mMediaSections.size() - 1);
+  }
+}
+
+void RsdparsaSdp::Serialize(std::ostream& os) const {
+  os << "v=0" << CRLF << mOrigin << "s=-" << CRLF;
+
+  // We don't support creating i=, u=, e=, p=
+  // We don't generate c= at the session level (only in media)
+
+  BandwidthVec* bwVec = sdp_get_session_bandwidth_vec(mSession.get());
+  char* bwString = sdp_serialize_bandwidth(bwVec);
+  if (bwString) {
+    os << bwString;
+    sdp_free_string(bwString);
+  }
+
+  os << "t=0 0" << CRLF;
+
+  // We don't support r= or z=
+
+  // attributes
+  os << *mAttributeList;
+
+  // media sections
+  for (const auto& msection : mMediaSections) {
+    os << *msection;
+  }
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/sdp/RsdparsaSdp.h b/dom/media/webrtc/sdp/RsdparsaSdp.h
new file mode 100644
index 0000000000..4942abb574
--- /dev/null
+++ b/dom/media/webrtc/sdp/RsdparsaSdp.h
@@ -0,0 +1,72 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _RSDPARSA_SDP_H_
+#define _RSDPARSA_SDP_H_
+
+#include "mozilla/UniquePtr.h"
+
+#include "sdp/Sdp.h"
+
+#include "sdp/RsdparsaSdpMediaSection.h"
+#include "sdp/RsdparsaSdpAttributeList.h"
+#include "sdp/RsdparsaSdpInc.h"
+#include "sdp/RsdparsaSdpGlue.h"
+
+namespace mozilla {
+
+class RsdparsaSdpParser;
+class SdpParser;
+
+class RsdparsaSdp final : public Sdp {
+  friend class RsdparsaSdpParser;
+
+ public:
+  explicit RsdparsaSdp(RsdparsaSessionHandle session, const SdpOrigin& origin);
+
+  Sdp* Clone() const override;
+
+  const SdpOrigin& GetOrigin() const override;
+
+  // Note: connection information is always retrieved from media sections
+  uint32_t GetBandwidth(const std::string& type) const override;
+
+  size_t GetMediaSectionCount() const override {
+    return sdp_media_section_count(mSession.get());
+  }
+
+  const SdpAttributeList& GetAttributeList() const override {
+    return *mAttributeList;
+  }
+
+  SdpAttributeList& GetAttributeList() override { return *mAttributeList; }
+
+  const SdpMediaSection& GetMediaSection(size_t level) const override;
+
+  SdpMediaSection& GetMediaSection(size_t level) override;
+
+  SdpMediaSection& AddMediaSection(SdpMediaSection::MediaType media,
+                                   SdpDirectionAttribute::Direction dir,
+                                   uint16_t port,
+                                   SdpMediaSection::Protocol proto,
+                                   sdp::AddrType addrType,
+                                   const std::string& addr) override;
+
+  void Serialize(std::ostream&) const override;
+
+ private:
+  RsdparsaSdp() : mOrigin("", 0, 0, sdp::kIPv4, "") {}
+  RsdparsaSdp(const RsdparsaSdp& aOrig);
+
+  RsdparsaSessionHandle mSession;
+  SdpOrigin mOrigin;
+  UniquePtr<RsdparsaSdpAttributeList> mAttributeList;
+  std::vector<UniquePtr<RsdparsaSdpMediaSection>> mMediaSections;
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/sdp/RsdparsaSdpAttributeList.cpp b/dom/media/webrtc/sdp/RsdparsaSdpAttributeList.cpp
new file mode 100644
index 0000000000..0b76757c44
--- /dev/null
+++ b/dom/media/webrtc/sdp/RsdparsaSdpAttributeList.cpp
@@ -0,0 +1,1301 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "nsCRT.h"
+
+#include "sdp/RsdparsaSdpAttributeList.h"
+#include "sdp/RsdparsaSdpInc.h"
+#include "sdp/RsdparsaSdpGlue.h"
+
+#include <ostream>
+#include "mozilla/Assertions.h"
+
+#include <limits>
+
+namespace mozilla {
+
+const std::string RsdparsaSdpAttributeList::kEmptyString = "";
+
+RsdparsaSdpAttributeList::~RsdparsaSdpAttributeList() {
+  for (size_t i = 0; i < kNumAttributeTypes; ++i) {
+    delete mAttributes[i];
+  }
+}
+
+bool RsdparsaSdpAttributeList::HasAttribute(AttributeType type,
+                                            bool sessionFallback) const {
+  return !!GetAttribute(type, sessionFallback);
+}
+
+const SdpAttribute* RsdparsaSdpAttributeList::GetAttribute(
+    AttributeType type, bool sessionFallback) const {
+  const SdpAttribute* value = mAttributes[static_cast<size_t>(type)];
+  // Only do fallback when the attribute can appear at both the media and
+  // session level
+  if (!value && !AtSessionLevel() && sessionFallback &&
+      SdpAttribute::IsAllowedAtSessionLevel(type) &&
+      SdpAttribute::IsAllowedAtMediaLevel(type)) {
+    return mSessionAttributes->GetAttribute(type, false);
+  }
+  return value;
+}
+
+void RsdparsaSdpAttributeList::RemoveAttribute(AttributeType type) {
+  delete mAttributes[static_cast<size_t>(type)];
+  mAttributes[static_cast<size_t>(type)] = nullptr;
+}
+
+void RsdparsaSdpAttributeList::Clear() {
+  for (size_t i = 0; i < kNumAttributeTypes; ++i) {
+    RemoveAttribute(static_cast<AttributeType>(i));
+  }
+}
+
+uint32_t RsdparsaSdpAttributeList::Count() const {
+  uint32_t count = 0;
+  for (size_t i = 0; i < kNumAttributeTypes; ++i) {
+    if (mAttributes[i]) {
+      count++;
+    }
+  }
+  return count;
+}
+
+void RsdparsaSdpAttributeList::SetAttribute(SdpAttribute* attr) {
+  if (!IsAllowedHere(attr->GetType())) {
+    MOZ_ASSERT(false, "This type of attribute is not allowed here");
+    delete attr;
+    return;
+  }
+  RemoveAttribute(attr->GetType());
+  mAttributes[attr->GetType()] = attr;
+}
+
+const std::vector<std::string>& RsdparsaSdpAttributeList::GetCandidate() const {
+  if (!HasAttribute(SdpAttribute::kCandidateAttribute)) {
+    MOZ_CRASH();
+  }
+
+  return static_cast<const SdpMultiStringAttribute*>(
+             GetAttribute(SdpAttribute::kCandidateAttribute))
+      ->mValues;
+}
+
+const SdpConnectionAttribute& RsdparsaSdpAttributeList::GetConnection() const {
+  if (!HasAttribute(SdpAttribute::kConnectionAttribute)) {
+    MOZ_CRASH();
+  }
+
+  return *static_cast<const SdpConnectionAttribute*>(
+      GetAttribute(SdpAttribute::kConnectionAttribute));
+}
+
+SdpDirectionAttribute::Direction RsdparsaSdpAttributeList::GetDirection()
+    const {
+  if (!HasAttribute(SdpAttribute::kDirectionAttribute)) {
+    MOZ_CRASH();
+  }
+
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kDirectionAttribute);
+  return static_cast<const SdpDirectionAttribute*>(attr)->mValue;
+}
+
+const SdpDtlsMessageAttribute& RsdparsaSdpAttributeList::GetDtlsMessage()
+    const {
+  if (!HasAttribute(SdpAttribute::kDtlsMessageAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kDtlsMessageAttribute);
+  return *static_cast<const SdpDtlsMessageAttribute*>(attr);
+}
+
+const SdpExtmapAttributeList& RsdparsaSdpAttributeList::GetExtmap() const {
+  if (!HasAttribute(SdpAttribute::kExtmapAttribute)) {
+    MOZ_CRASH();
+  }
+
+  return *static_cast<const SdpExtmapAttributeList*>(
+      GetAttribute(SdpAttribute::kExtmapAttribute));
+}
+
+const SdpFingerprintAttributeList& RsdparsaSdpAttributeList::GetFingerprint()
+    const {
+  if (!HasAttribute(SdpAttribute::kFingerprintAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kFingerprintAttribute);
+  return *static_cast<const SdpFingerprintAttributeList*>(attr);
+}
+
+const SdpFmtpAttributeList& RsdparsaSdpAttributeList::GetFmtp() const {
+  if (!HasAttribute(SdpAttribute::kFmtpAttribute)) {
+    MOZ_CRASH();
+  }
+
+  return *static_cast<const SdpFmtpAttributeList*>(
+      GetAttribute(SdpAttribute::kFmtpAttribute));
+}
+
+const SdpGroupAttributeList& RsdparsaSdpAttributeList::GetGroup() const {
+  if (!HasAttribute(SdpAttribute::kGroupAttribute)) {
+    MOZ_CRASH();
+  }
+
+  return *static_cast<const SdpGroupAttributeList*>(
+      GetAttribute(SdpAttribute::kGroupAttribute));
+}
+
+const SdpOptionsAttribute& RsdparsaSdpAttributeList::GetIceOptions() const {
+  if (!HasAttribute(SdpAttribute::kIceOptionsAttribute)) {
+    MOZ_CRASH();
+  }
+
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kIceOptionsAttribute);
+  return *static_cast<const SdpOptionsAttribute*>(attr);
+}
+
+const std::string& RsdparsaSdpAttributeList::GetIcePwd() const {
+  if (!HasAttribute(SdpAttribute::kIcePwdAttribute)) {
+    return kEmptyString;
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kIcePwdAttribute);
+  return static_cast<const SdpStringAttribute*>(attr)->mValue;
+}
+
+const std::string& RsdparsaSdpAttributeList::GetIceUfrag() const {
+  if (!HasAttribute(SdpAttribute::kIceUfragAttribute)) {
+    return kEmptyString;
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kIceUfragAttribute);
+  return static_cast<const SdpStringAttribute*>(attr)->mValue;
+}
+
+const std::string& RsdparsaSdpAttributeList::GetIdentity() const {
+  if (!HasAttribute(SdpAttribute::kIdentityAttribute)) {
+    return kEmptyString;
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kIdentityAttribute);
+  return static_cast<const SdpStringAttribute*>(attr)->mValue;
+}
+
+const SdpImageattrAttributeList& RsdparsaSdpAttributeList::GetImageattr()
+    const {
+  if (!HasAttribute(SdpAttribute::kImageattrAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kImageattrAttribute);
+  return *static_cast<const SdpImageattrAttributeList*>(attr);
+}
+
+const SdpSimulcastAttribute& RsdparsaSdpAttributeList::GetSimulcast() const {
+  if (!HasAttribute(SdpAttribute::kSimulcastAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kSimulcastAttribute);
+  return *static_cast<const SdpSimulcastAttribute*>(attr);
+}
+
+const std::string& RsdparsaSdpAttributeList::GetLabel() const {
+  if (!HasAttribute(SdpAttribute::kLabelAttribute)) {
+    return kEmptyString;
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kLabelAttribute);
+  return static_cast<const SdpStringAttribute*>(attr)->mValue;
+}
+
+uint32_t RsdparsaSdpAttributeList::GetMaxptime() const {
+  if (!HasAttribute(SdpAttribute::kMaxptimeAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kMaxptimeAttribute);
+  return static_cast<const SdpNumberAttribute*>(attr)->mValue;
+}
+
+const std::string& RsdparsaSdpAttributeList::GetMid() const {
+  if (!HasAttribute(SdpAttribute::kMidAttribute)) {
+    return kEmptyString;
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kMidAttribute);
+  return static_cast<const SdpStringAttribute*>(attr)->mValue;
+}
+
+const SdpMsidAttributeList& RsdparsaSdpAttributeList::GetMsid() const {
+  if (!HasAttribute(SdpAttribute::kMsidAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kMsidAttribute);
+  return *static_cast<const SdpMsidAttributeList*>(attr);
+}
+
+const SdpMsidSemanticAttributeList& RsdparsaSdpAttributeList::GetMsidSemantic()
+    const {
+  if (!HasAttribute(SdpAttribute::kMsidSemanticAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kMsidSemanticAttribute);
+  return *static_cast<const SdpMsidSemanticAttributeList*>(attr);
+}
+
+const SdpRidAttributeList& RsdparsaSdpAttributeList::GetRid() const {
+  if (!HasAttribute(SdpAttribute::kRidAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kRidAttribute);
+  return *static_cast<const SdpRidAttributeList*>(attr);
+}
+
+uint32_t RsdparsaSdpAttributeList::GetPtime() const {
+  if (!HasAttribute(SdpAttribute::kPtimeAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kPtimeAttribute);
+  return static_cast<const SdpNumberAttribute*>(attr)->mValue;
+}
+
+const SdpRtcpAttribute& RsdparsaSdpAttributeList::GetRtcp() const {
+  if (!HasAttribute(SdpAttribute::kRtcpAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kRtcpAttribute);
+  return *static_cast<const SdpRtcpAttribute*>(attr);
+}
+
+const SdpRtcpFbAttributeList& RsdparsaSdpAttributeList::GetRtcpFb() const {
+  if (!HasAttribute(SdpAttribute::kRtcpFbAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kRtcpFbAttribute);
+  return *static_cast<const SdpRtcpFbAttributeList*>(attr);
+}
+
+const SdpRemoteCandidatesAttribute&
+RsdparsaSdpAttributeList::GetRemoteCandidates() const {
+  MOZ_CRASH("Not yet implemented");
+}
+
+const SdpRtpmapAttributeList& RsdparsaSdpAttributeList::GetRtpmap() const {
+  if (!HasAttribute(SdpAttribute::kRtpmapAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kRtpmapAttribute);
+  return *static_cast<const SdpRtpmapAttributeList*>(attr);
+}
+
+const SdpSctpmapAttributeList& RsdparsaSdpAttributeList::GetSctpmap() const {
+  if (!HasAttribute(SdpAttribute::kSctpmapAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kSctpmapAttribute);
+  return *static_cast<const SdpSctpmapAttributeList*>(attr);
+}
+
+uint32_t RsdparsaSdpAttributeList::GetSctpPort() const {
+  if (!HasAttribute(SdpAttribute::kSctpPortAttribute)) {
+    MOZ_CRASH();
+  }
+
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kSctpPortAttribute);
+  return static_cast<const SdpNumberAttribute*>(attr)->mValue;
+}
+
+uint32_t RsdparsaSdpAttributeList::GetMaxMessageSize() const {
+  if (!HasAttribute(SdpAttribute::kMaxMessageSizeAttribute)) {
+    MOZ_CRASH();
+  }
+
+  const SdpAttribute* attr =
+      GetAttribute(SdpAttribute::kMaxMessageSizeAttribute);
+  return static_cast<const SdpNumberAttribute*>(attr)->mValue;
+}
+
+const SdpSetupAttribute& RsdparsaSdpAttributeList::GetSetup() const {
+  if (!HasAttribute(SdpAttribute::kSetupAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kSetupAttribute);
+  return *static_cast<const SdpSetupAttribute*>(attr);
+}
+
+const SdpSsrcAttributeList& RsdparsaSdpAttributeList::GetSsrc() const {
+  if (!HasAttribute(SdpAttribute::kSsrcAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kSsrcAttribute);
+  return *static_cast<const SdpSsrcAttributeList*>(attr);
+}
+
+const SdpSsrcGroupAttributeList& RsdparsaSdpAttributeList::GetSsrcGroup()
+    const {
+  if (!HasAttribute(SdpAttribute::kSsrcGroupAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kSsrcGroupAttribute);
+  return *static_cast<const SdpSsrcGroupAttributeList*>(attr);
+}
+
+void RsdparsaSdpAttributeList::LoadAttribute(RustAttributeList* attributeList,
+                                             AttributeType type) {
+  if (!mAttributes[type]) {
+    switch (type) {
+      case SdpAttribute::kIceUfragAttribute:
+        LoadIceUfrag(attributeList);
+        return;
+      case SdpAttribute::kIcePwdAttribute:
+        LoadIcePwd(attributeList);
+        return;
+      case SdpAttribute::kIceOptionsAttribute:
+        LoadIceOptions(attributeList);
+        return;
+      case SdpAttribute::kDtlsMessageAttribute:
+        LoadDtlsMessage(attributeList);
+        return;
+      case SdpAttribute::kFingerprintAttribute:
+        LoadFingerprint(attributeList);
+        return;
+      case SdpAttribute::kIdentityAttribute:
+        LoadIdentity(attributeList);
+        return;
+      case SdpAttribute::kSetupAttribute:
+        LoadSetup(attributeList);
+        return;
+      case SdpAttribute::kSsrcAttribute:
+        LoadSsrc(attributeList);
+        return;
+      case SdpAttribute::kRtpmapAttribute:
+        LoadRtpmap(attributeList);
+        return;
+      case SdpAttribute::kFmtpAttribute:
+        LoadFmtp(attributeList);
+        return;
+      case SdpAttribute::kPtimeAttribute:
+        LoadPtime(attributeList);
+        return;
+      case SdpAttribute::kIceLiteAttribute:
+      case SdpAttribute::kRtcpMuxAttribute:
+      case SdpAttribute::kRtcpRsizeAttribute:
+      case SdpAttribute::kBundleOnlyAttribute:
+      case SdpAttribute::kEndOfCandidatesAttribute:
+        LoadFlags(attributeList);
+        return;
+      case SdpAttribute::kMaxMessageSizeAttribute:
+        LoadMaxMessageSize(attributeList);
+        return;
+      case SdpAttribute::kMidAttribute:
+        LoadMid(attributeList);
+        return;
+      case SdpAttribute::kMsidAttribute:
+        LoadMsid(attributeList);
+        return;
+      case SdpAttribute::kMsidSemanticAttribute:
+        LoadMsidSemantics(attributeList);
+        return;
+      case SdpAttribute::kGroupAttribute:
+        LoadGroup(attributeList);
+        return;
+      case SdpAttribute::kRtcpAttribute:
+        LoadRtcp(attributeList);
+        return;
+      case SdpAttribute::kRtcpFbAttribute:
+        LoadRtcpFb(attributeList);
+        return;
+      case SdpAttribute::kImageattrAttribute:
+        LoadImageattr(attributeList);
+        return;
+      case SdpAttribute::kSctpmapAttribute:
+        LoadSctpmaps(attributeList);
+        return;
+      case SdpAttribute::kDirectionAttribute:
+        LoadDirection(attributeList);
+        return;
+      case SdpAttribute::kRemoteCandidatesAttribute:
+        LoadRemoteCandidates(attributeList);
+        return;
+      case SdpAttribute::kRidAttribute:
+        LoadRids(attributeList);
+        return;
+      case SdpAttribute::kSctpPortAttribute:
+        LoadSctpPort(attributeList);
+        return;
+      case SdpAttribute::kExtmapAttribute:
+        LoadExtmap(attributeList);
+        return;
+      case SdpAttribute::kSimulcastAttribute:
+        LoadSimulcast(attributeList);
+        return;
+      case SdpAttribute::kMaxptimeAttribute:
+        LoadMaxPtime(attributeList);
+        return;
+      case SdpAttribute::kCandidateAttribute:
+        LoadCandidate(attributeList);
+        return;
+      case SdpAttribute::kSsrcGroupAttribute:
+        LoadSsrcGroup(attributeList);
+        return;
+      case SdpAttribute::kConnectionAttribute:
+      case SdpAttribute::kIceMismatchAttribute:
+      case SdpAttribute::kLabelAttribute:
+        // These attributes are unused
+        return;
+    }
+  }
+}
+
+void RsdparsaSdpAttributeList::LoadAll(RustAttributeList* attributeList) {
+  for (int i = SdpAttribute::kFirstAttribute; i <= SdpAttribute::kLastAttribute;
+       i++) {
+    LoadAttribute(attributeList, static_cast<SdpAttribute::AttributeType>(i));
+  }
+}
+
+void RsdparsaSdpAttributeList::LoadIceUfrag(RustAttributeList* attributeList) {
+  StringView ufragStr;
+  nsresult nr = sdp_get_iceufrag(attributeList, &ufragStr);
+  if (NS_SUCCEEDED(nr)) {
+    std::string iceufrag = convertStringView(ufragStr);
+    SetAttribute(
+        new SdpStringAttribute(SdpAttribute::kIceUfragAttribute, iceufrag));
+  }
+}
+
+void RsdparsaSdpAttributeList::LoadIcePwd(RustAttributeList* attributeList) {
+  StringView pwdStr;
+  nsresult nr = sdp_get_icepwd(attributeList, &pwdStr);
+  if (NS_SUCCEEDED(nr)) {
+    std::string icePwd = convertStringView(pwdStr);
+    SetAttribute(
+        new SdpStringAttribute(SdpAttribute::kIcePwdAttribute, icePwd));
+  }
+}
+
+void RsdparsaSdpAttributeList::LoadIdentity(RustAttributeList* attributeList) {
+  StringView identityStr;
+  nsresult nr = sdp_get_identity(attributeList, &identityStr);
+  if (NS_SUCCEEDED(nr)) {
+    std::string identity = convertStringView(identityStr);
+    SetAttribute(
+        new SdpStringAttribute(SdpAttribute::kIdentityAttribute, identity));
+  }
+}
+
+void RsdparsaSdpAttributeList::LoadIceOptions(
+    RustAttributeList* attributeList) {
+  StringVec* options;
+  nsresult nr = sdp_get_iceoptions(attributeList, &options);
+  if (NS_SUCCEEDED(nr)) {
+    std::vector<std::string> optionsVec;
+    auto optionsAttr =
+        MakeUnique<SdpOptionsAttribute>(SdpAttribute::kIceOptionsAttribute);
+    for (size_t i = 0; i < string_vec_len(options); i++) {
+      StringView optionStr;
+      string_vec_get_view(options, i, &optionStr);
+      optionsAttr->PushEntry(convertStringView(optionStr));
+    }
+    SetAttribute(optionsAttr.release());
+  }
+}
+
+void RsdparsaSdpAttributeList::LoadFingerprint(
+    RustAttributeList* attributeList) {
+  size_t nFp = sdp_get_fingerprint_count(attributeList);
+  if (nFp == 0) {
+    return;
+  }
+  auto rustFingerprints = MakeUnique<RustSdpAttributeFingerprint[]>(nFp);
+  sdp_get_fingerprints(attributeList, nFp, rustFingerprints.get());
+  auto fingerprints = MakeUnique<SdpFingerprintAttributeList>();
+  for (size_t i = 0; i < nFp; i++) {
+    const RustSdpAttributeFingerprint& fingerprint = rustFingerprints[i];
+    std::string algorithm;
+    switch (fingerprint.hashAlgorithm) {
+      case RustSdpAttributeFingerprintHashAlgorithm::kSha1:
+        algorithm = "sha-1";
+        break;
+      case RustSdpAttributeFingerprintHashAlgorithm::kSha224:
+        algorithm = "sha-224";
+        break;
+      case RustSdpAttributeFingerprintHashAlgorithm::kSha256:
+        algorithm = "sha-256";
+        break;
+      case RustSdpAttributeFingerprintHashAlgorithm::kSha384:
+        algorithm = "sha-384";
+        break;
+      case RustSdpAttributeFingerprintHashAlgorithm::kSha512:
+        algorithm = "sha-512";
+        break;
+    }
+
+    std::vector<uint8_t> fingerprintBytes =
+        convertU8Vec(fingerprint.fingerprint);
+
+    fingerprints->PushEntry(algorithm, fingerprintBytes);
+  }
+  SetAttribute(fingerprints.release());
+}
+
+void RsdparsaSdpAttributeList::LoadDtlsMessage(
+    RustAttributeList* attributeList) {
+  RustSdpAttributeDtlsMessage rustDtlsMessage;
+  nsresult nr = sdp_get_dtls_message(attributeList, &rustDtlsMessage);
+  if (NS_SUCCEEDED(nr)) {
+    SdpDtlsMessageAttribute::Role role;
+    if (rustDtlsMessage.role == RustSdpAttributeDtlsMessageType::kClient) {
+      role = SdpDtlsMessageAttribute::kClient;
+    } else {
+      role = SdpDtlsMessageAttribute::kServer;
+    }
+
+    std::string value = convertStringView(rustDtlsMessage.value);
+
+    SetAttribute(new SdpDtlsMessageAttribute(role, value));
+  }
+}
+
+void RsdparsaSdpAttributeList::LoadSetup(RustAttributeList* attributeList) {
+  RustSdpSetup rustSetup;
+  nsresult nr = sdp_get_setup(attributeList, &rustSetup);
+  if (NS_SUCCEEDED(nr)) {
+    SdpSetupAttribute::Role setupEnum;
+    switch (rustSetup) {
+      case RustSdpSetup::kRustActive:
+        setupEnum = SdpSetupAttribute::kActive;
+        break;
+      case RustSdpSetup::kRustActpass:
+        setupEnum = SdpSetupAttribute::kActpass;
+        break;
+      case RustSdpSetup::kRustHoldconn:
+        setupEnum = SdpSetupAttribute::kHoldconn;
+        break;
+      case RustSdpSetup::kRustPassive:
+        setupEnum = SdpSetupAttribute::kPassive;
+        break;
+    }
+    SetAttribute(new SdpSetupAttribute(setupEnum));
+  }
+}
+
+void RsdparsaSdpAttributeList::LoadSsrc(RustAttributeList* attributeList) {
+  size_t numSsrc = sdp_get_ssrc_count(attributeList);
+  if (numSsrc == 0) {
+    return;
+  }
+  auto rustSsrcs = MakeUnique<RustSdpAttributeSsrc[]>(numSsrc);
+  sdp_get_ssrcs(attributeList, numSsrc, rustSsrcs.get());
+  auto ssrcs = MakeUnique<SdpSsrcAttributeList>();
+  for (size_t i = 0; i < numSsrc; i++) {
+    RustSdpAttributeSsrc& ssrc = rustSsrcs[i];
+    std::string attribute = convertStringView(ssrc.attribute);
+    std::string value = convertStringView(ssrc.value);
+    if (value.length() == 0) {
+      ssrcs->PushEntry(ssrc.id, attribute);
+    } else {
+      ssrcs->PushEntry(ssrc.id, attribute + ":" + value);
+    }
+  }
+  SetAttribute(ssrcs.release());
+}
+
+void RsdparsaSdpAttributeList::LoadSsrcGroup(RustAttributeList* attributeList) {
+  size_t numSsrcGroups = sdp_get_ssrc_group_count(attributeList);
+  if (numSsrcGroups == 0) {
+    return;
+  }
+  auto rustSsrcGroups = MakeUnique<RustSdpAttributeSsrcGroup[]>(numSsrcGroups);
+  sdp_get_ssrc_groups(attributeList, numSsrcGroups, rustSsrcGroups.get());
+  auto ssrcGroups = MakeUnique<SdpSsrcGroupAttributeList>();
+  for (size_t i = 0; i < numSsrcGroups; i++) {
+    RustSdpAttributeSsrcGroup& ssrcGroup = rustSsrcGroups[i];
+    SdpSsrcGroupAttributeList::Semantics semantic;
+    switch (ssrcGroup.semantic) {
+      case RustSdpAttributeSsrcGroupSemantic ::kRustDup:
+        semantic = SdpSsrcGroupAttributeList::kDup;
+        break;
+      case RustSdpAttributeSsrcGroupSemantic ::kRustFec:
+        semantic = SdpSsrcGroupAttributeList::kFec;
+        break;
+      case RustSdpAttributeSsrcGroupSemantic ::kRustFecFr:
+        semantic = SdpSsrcGroupAttributeList::kFecFr;
+        break;
+      case RustSdpAttributeSsrcGroupSemantic ::kRustFid:
+        semantic = SdpSsrcGroupAttributeList::kFid;
+        break;
+      case RustSdpAttributeSsrcGroupSemantic ::kRustSim:
+        semantic = SdpSsrcGroupAttributeList::kSim;
+        break;
+    }
+    std::vector<uint32_t> ssrcs;
+    for (size_t i = 0; i < ssrc_vec_len(ssrcGroup.ssrcs); ++i) {
+      uint32_t ssrc;
+      ssrc_vec_get_id(ssrcGroup.ssrcs, i, &ssrc);
+      ssrcs.push_back(ssrc);
+    }
+    ssrcGroups->PushEntry(semantic, ssrcs);
+  }
+  SetAttribute(ssrcGroups.release());
+}
+
+struct FmtDefaults {
+  uint32_t minimumChannels = 0;
+};
+
+std::tuple<SdpRtpmapAttributeList::CodecType, FmtDefaults> strToCodecType(
+    const std::string& name) {
+  auto codec = SdpRtpmapAttributeList::kOtherCodec;
+  FmtDefaults defaults = {0};  // This is tracked to match SIPCC behavior only
+  if (!nsCRT::strcasecmp(name.c_str(), "opus")) {
+    codec = SdpRtpmapAttributeList::kOpus;
+    defaults = {0};
+  } else if (!nsCRT::strcasecmp(name.c_str(), "G722")) {
+    codec = SdpRtpmapAttributeList::kG722;
+    defaults = {1};
+  } else if (!nsCRT::strcasecmp(name.c_str(), "PCMU")) {
+    codec = SdpRtpmapAttributeList::kPCMU;
+    defaults = {1};
+  } else if (!nsCRT::strcasecmp(name.c_str(), "PCMA")) {
+    codec = SdpRtpmapAttributeList::kPCMA;
+    defaults = {1};
+  } else if (!nsCRT::strcasecmp(name.c_str(), "VP8")) {
+    codec = SdpRtpmapAttributeList::kVP8;
+    defaults = {0};
+  } else if (!nsCRT::strcasecmp(name.c_str(), "VP9")) {
+    codec = SdpRtpmapAttributeList::kVP9;
+    defaults = {0};
+  } else if (!nsCRT::strcasecmp(name.c_str(), "iLBC")) {
+    codec = SdpRtpmapAttributeList::kiLBC;
+    defaults = {1};
+  } else if (!nsCRT::strcasecmp(name.c_str(), "iSAC")) {
+    codec = SdpRtpmapAttributeList::kiSAC;
+    defaults = {1};
+  } else if (!nsCRT::strcasecmp(name.c_str(), "H264")) {
+    codec = SdpRtpmapAttributeList::kH264;
+    defaults = {0};
+  } else if (!nsCRT::strcasecmp(name.c_str(), "red")) {
+    codec = SdpRtpmapAttributeList::kRed;
+    defaults = {0};
+  } else if (!nsCRT::strcasecmp(name.c_str(), "ulpfec")) {
+    codec = SdpRtpmapAttributeList::kUlpfec;
+    defaults = {0};
+  } else if (!nsCRT::strcasecmp(name.c_str(), "telephone-event")) {
+    codec = SdpRtpmapAttributeList::kTelephoneEvent;
+    defaults = {1};
+  } else if (!nsCRT::strcasecmp(name.c_str(), "rtx")) {
+    codec = SdpRtpmapAttributeList::kRtx;
+    defaults = {0};
+  }
+  return std::make_tuple(codec, defaults);
+}
+
+void RsdparsaSdpAttributeList::LoadRtpmap(RustAttributeList* attributeList) {
+  size_t numRtpmap = sdp_get_rtpmap_count(attributeList);
+  if (numRtpmap == 0) {
+    return;
+  }
+  auto rustRtpmaps = MakeUnique<RustSdpAttributeRtpmap[]>(numRtpmap);
+  sdp_get_rtpmaps(attributeList, numRtpmap, rustRtpmaps.get());
+  auto rtpmapList = MakeUnique<SdpRtpmapAttributeList>();
+  for (size_t i = 0; i < numRtpmap; i++) {
+    RustSdpAttributeRtpmap& rtpmap = rustRtpmaps[i];
+    std::string payloadType = std::to_string(rtpmap.payloadType);
+    std::string name = convertStringView(rtpmap.codecName);
+    auto [codec, defaults] = strToCodecType(name);
+    uint32_t channels = rtpmap.channels;
+    if (channels == 0) {
+      channels = defaults.minimumChannels;
+    }
+    rtpmapList->PushEntry(payloadType, codec, name, rtpmap.frequency, channels);
+  }
+  SetAttribute(rtpmapList.release());
+}
+
+void RsdparsaSdpAttributeList::LoadFmtp(RustAttributeList* attributeList) {
+  size_t numFmtp = sdp_get_fmtp_count(attributeList);
+  if (numFmtp == 0) {
+    return;
+  }
+  auto rustFmtps = MakeUnique<RustSdpAttributeFmtp[]>(numFmtp);
+  size_t numValidFmtp = sdp_get_fmtp(attributeList, numFmtp, rustFmtps.get());
+  auto fmtpList = MakeUnique<SdpFmtpAttributeList>();
+  for (size_t i = 0; i < numValidFmtp; i++) {
+    const RustSdpAttributeFmtp& fmtp = rustFmtps[i];
+    uint8_t payloadType = fmtp.payloadType;
+    std::string codecName = convertStringView(fmtp.codecName);
+    const RustSdpAttributeFmtpParameters& rustFmtpParameters = fmtp.parameters;
+
+    UniquePtr<SdpFmtpAttributeList::Parameters> fmtpParameters;
+
+    // use the upper case version of the codec name
+    std::transform(codecName.begin(), codecName.end(), codecName.begin(),
+                   ::toupper);
+
+    if (codecName == "H264") {
+      SdpFmtpAttributeList::H264Parameters h264Parameters;
+
+      h264Parameters.packetization_mode = rustFmtpParameters.packetization_mode;
+      h264Parameters.level_asymmetry_allowed =
+          rustFmtpParameters.level_asymmetry_allowed;
+      h264Parameters.profile_level_id = rustFmtpParameters.profile_level_id;
+      h264Parameters.max_mbps = rustFmtpParameters.max_mbps;
+      h264Parameters.max_fs = rustFmtpParameters.max_fs;
+      h264Parameters.max_cpb = rustFmtpParameters.max_cpb;
+      h264Parameters.max_dpb = rustFmtpParameters.max_dpb;
+      h264Parameters.max_br = rustFmtpParameters.max_br;
+
+      // TODO(bug 1466859): Support sprop-parameter-sets
+
+      fmtpParameters.reset(
+          new SdpFmtpAttributeList::H264Parameters(std::move(h264Parameters)));
+    } else if (codecName == "OPUS") {
+      SdpFmtpAttributeList::OpusParameters opusParameters;
+
+      opusParameters.maxplaybackrate = rustFmtpParameters.maxplaybackrate;
+      opusParameters.maxAverageBitrate = rustFmtpParameters.maxaveragebitrate;
+      opusParameters.useDTX = rustFmtpParameters.usedtx;
+      opusParameters.stereo = rustFmtpParameters.stereo;
+      opusParameters.useInBandFec = rustFmtpParameters.useinbandfec;
+      opusParameters.frameSizeMs = rustFmtpParameters.ptime;
+      opusParameters.minFrameSizeMs = rustFmtpParameters.minptime;
+      opusParameters.maxFrameSizeMs = rustFmtpParameters.maxptime;
+      opusParameters.useCbr = rustFmtpParameters.cbr;
+
+      fmtpParameters.reset(
+          new SdpFmtpAttributeList::OpusParameters(std::move(opusParameters)));
+    } else if ((codecName == "VP8") || (codecName == "VP9")) {
+      SdpFmtpAttributeList::VP8Parameters vp8Parameters(
+          codecName == "VP8" ? SdpRtpmapAttributeList::kVP8
+                             : SdpRtpmapAttributeList::kVP9);
+
+      vp8Parameters.max_fs = rustFmtpParameters.max_fs;
+      vp8Parameters.max_fr = rustFmtpParameters.max_fr;
+
+      fmtpParameters.reset(
+          new SdpFmtpAttributeList::VP8Parameters(std::move(vp8Parameters)));
+    } else if (codecName == "TELEPHONE-EVENT") {
+      SdpFmtpAttributeList::TelephoneEventParameters telephoneEventParameters;
+
+      telephoneEventParameters.dtmfTones =
+          convertStringView(rustFmtpParameters.dtmf_tones);
+
+      fmtpParameters.reset(new SdpFmtpAttributeList::TelephoneEventParameters(
+          std::move(telephoneEventParameters)));
+    } else if (codecName == "RED") {
+      SdpFmtpAttributeList::RedParameters redParameters;
+
+      redParameters.encodings = convertU8Vec(rustFmtpParameters.encodings);
+
+      fmtpParameters.reset(
+          new SdpFmtpAttributeList::RedParameters(std::move(redParameters)));
+    } else if (codecName == "RTX") {
+      SdpFmtpAttributeList::RtxParameters rtxParameters;
+
+      rtxParameters.apt = rustFmtpParameters.rtx.apt;
+      if (rustFmtpParameters.rtx.has_rtx_time) {
+        rtxParameters.rtx_time = Some(rustFmtpParameters.rtx.rtx_time);
+      }
+
+      fmtpParameters.reset(
+          new SdpFmtpAttributeList::RtxParameters(rtxParameters));
+    } else {
+      // The parameter set is unknown so skip it
+      continue;
+    }
+    fmtpList->PushEntry(std::to_string(payloadType), *fmtpParameters);
+  }
+  SetAttribute(fmtpList.release());
+}
+
+void RsdparsaSdpAttributeList::LoadPtime(RustAttributeList* attributeList) {
+  int64_t ptime = sdp_get_ptime(attributeList);
+  if (ptime >= 0) {
+    SetAttribute(new SdpNumberAttribute(SdpAttribute::kPtimeAttribute,
+                                        static_cast<uint32_t>(ptime)));
+  }
+}
+
+void RsdparsaSdpAttributeList::LoadFlags(RustAttributeList* attributeList) {
+  RustSdpAttributeFlags flags = sdp_get_attribute_flags(attributeList);
+  if (flags.iceLite) {
+    SetAttribute(new SdpFlagAttribute(SdpAttribute::kIceLiteAttribute));
+  }
+  if (flags.rtcpMux) {
+    SetAttribute(new SdpFlagAttribute(SdpAttribute::kRtcpMuxAttribute));
+  }
+  if (flags.rtcpRsize) {
+    SetAttribute(new SdpFlagAttribute(SdpAttribute::kRtcpRsizeAttribute));
+  }
+  if (flags.bundleOnly) {
+    SetAttribute(new SdpFlagAttribute(SdpAttribute::kBundleOnlyAttribute));
+  }
+  if (flags.endOfCandidates) {
+    SetAttribute(new SdpFlagAttribute(SdpAttribute::kEndOfCandidatesAttribute));
+  }
+}
+
+void RsdparsaSdpAttributeList::LoadMaxMessageSize(
+    RustAttributeList* attributeList) {
+  int64_t max_msg_size = sdp_get_max_msg_size(attributeList);
+  if (max_msg_size >= 0) {
+    SetAttribute(new SdpNumberAttribute(SdpAttribute::kMaxMessageSizeAttribute,
+                                        static_cast<uint32_t>(max_msg_size)));
+  }
+}
+
+void RsdparsaSdpAttributeList::LoadMid(RustAttributeList* attributeList) {
+  StringView rustMid;
+  if (NS_SUCCEEDED(sdp_get_mid(attributeList, &rustMid))) {
+    std::string mid = convertStringView(rustMid);
+    SetAttribute(new SdpStringAttribute(SdpAttribute::kMidAttribute, mid));
+  }
+}
+
+void RsdparsaSdpAttributeList::LoadMsid(RustAttributeList* attributeList) {
+  size_t numMsid = sdp_get_msid_count(attributeList);
+  if (numMsid == 0) {
+    return;
+  }
+  auto rustMsid = MakeUnique<RustSdpAttributeMsid[]>(numMsid);
+  sdp_get_msids(attributeList, numMsid, rustMsid.get());
+  auto msids = MakeUnique<SdpMsidAttributeList>();
+  for (size_t i = 0; i < numMsid; i++) {
+    RustSdpAttributeMsid& msid = rustMsid[i];
+    std::string id = convertStringView(msid.id);
+    std::string appdata = convertStringView(msid.appdata);
+    msids->PushEntry(id, appdata);
+  }
+  SetAttribute(msids.release());
+}
+
+void RsdparsaSdpAttributeList::LoadMsidSemantics(
+    RustAttributeList* attributeList) {
+  size_t numMsidSemantic = sdp_get_msid_semantic_count(attributeList);
+  if (numMsidSemantic == 0) {
+    return;
+  }
+  auto rustMsidSemantics =
+      MakeUnique<RustSdpAttributeMsidSemantic[]>(numMsidSemantic);
+  sdp_get_msid_semantics(attributeList, numMsidSemantic,
+                         rustMsidSemantics.get());
+  auto msidSemantics = MakeUnique<SdpMsidSemanticAttributeList>();
+  for (size_t i = 0; i < numMsidSemantic; i++) {
+    RustSdpAttributeMsidSemantic& rustMsidSemantic = rustMsidSemantics[i];
+    std::string semantic = convertStringView(rustMsidSemantic.semantic);
+    std::vector<std::string> msids = convertStringVec(rustMsidSemantic.msids);
+    msidSemantics->PushEntry(semantic, msids);
+  }
+  SetAttribute(msidSemantics.release());
+}
+
+void RsdparsaSdpAttributeList::LoadGroup(RustAttributeList* attributeList) {
+  size_t numGroup = sdp_get_group_count(attributeList);
+  if (numGroup == 0) {
+    return;
+  }
+  auto rustGroups = MakeUnique<RustSdpAttributeGroup[]>(numGroup);
+  sdp_get_groups(attributeList, numGroup, rustGroups.get());
+  auto groups = MakeUnique<SdpGroupAttributeList>();
+  for (size_t i = 0; i < numGroup; i++) {
+    RustSdpAttributeGroup& group = rustGroups[i];
+    SdpGroupAttributeList::Semantics semantic;
+    switch (group.semantic) {
+      case RustSdpAttributeGroupSemantic ::kRustLipSynchronization:
+        semantic = SdpGroupAttributeList::kLs;
+        break;
+      case RustSdpAttributeGroupSemantic ::kRustFlowIdentification:
+        semantic = SdpGroupAttributeList::kFid;
+        break;
+      case RustSdpAttributeGroupSemantic ::kRustSingleReservationFlow:
+        semantic = SdpGroupAttributeList::kSrf;
+        break;
+      case RustSdpAttributeGroupSemantic ::kRustAlternateNetworkAddressType:
+        semantic = SdpGroupAttributeList::kAnat;
+        break;
+      case RustSdpAttributeGroupSemantic ::kRustForwardErrorCorrection:
+        semantic = SdpGroupAttributeList::kFec;
+        break;
+      case RustSdpAttributeGroupSemantic ::kRustDecodingDependency:
+        semantic = SdpGroupAttributeList::kDdp;
+        break;
+      case RustSdpAttributeGroupSemantic ::kRustBundle:
+        semantic = SdpGroupAttributeList::kBundle;
+        break;
+    }
+    std::vector<std::string> tags = convertStringVec(group.tags);
+    groups->PushEntry(semantic, tags);
+  }
+  SetAttribute(groups.release());
+}
+
+void RsdparsaSdpAttributeList::LoadRtcp(RustAttributeList* attributeList) {
+  RustSdpAttributeRtcp rtcp;
+  if (NS_SUCCEEDED(sdp_get_rtcp(attributeList, &rtcp))) {
+    if (rtcp.has_address) {
+      auto address = convertExplicitlyTypedAddress(&rtcp.unicastAddr);
+      SetAttribute(new SdpRtcpAttribute(rtcp.port, sdp::kInternet,
+                                        address.first, address.second));
+    } else {
+      SetAttribute(new SdpRtcpAttribute(rtcp.port));
+    }
+  }
+}
+
+void RsdparsaSdpAttributeList::LoadRtcpFb(RustAttributeList* attributeList) {
+  auto rtcpfbsCount = sdp_get_rtcpfb_count(attributeList);
+  if (!rtcpfbsCount) {
+    return;
+  }
+
+  auto rustRtcpfbs = MakeUnique<RustSdpAttributeRtcpFb[]>(rtcpfbsCount);
+  sdp_get_rtcpfbs(attributeList, rtcpfbsCount, rustRtcpfbs.get());
+
+  auto rtcpfbList = MakeUnique<SdpRtcpFbAttributeList>();
+  for (size_t i = 0; i < rtcpfbsCount; i++) {
+    RustSdpAttributeRtcpFb& rtcpfb = rustRtcpfbs[i];
+    uint32_t payloadTypeU32 = rtcpfb.payloadType;
+
+    std::stringstream ss;
+    if (payloadTypeU32 == std::numeric_limits<uint32_t>::max()) {
+      ss << "*";
+    } else {
+      ss << payloadTypeU32;
+    }
+
+    uint32_t feedbackType = rtcpfb.feedbackType;
+    std::string parameter = convertStringView(rtcpfb.parameter);
+    std::string extra = convertStringView(rtcpfb.extra);
+
+    rtcpfbList->PushEntry(
+        ss.str(), static_cast<SdpRtcpFbAttributeList::Type>(feedbackType),
+        parameter, extra);
+  }
+
+  SetAttribute(rtcpfbList.release());
+}
+
+SdpSimulcastAttribute::Versions LoadSimulcastVersions(
+    const RustSdpAttributeSimulcastVersionVec* rustVersionList) {
+  size_t rustVersionCount = sdp_simulcast_get_version_count(rustVersionList);
+  auto rustVersionArray =
+      MakeUnique<RustSdpAttributeSimulcastVersion[]>(rustVersionCount);
+  sdp_simulcast_get_versions(rustVersionList, rustVersionCount,
+                             rustVersionArray.get());
+
+  SdpSimulcastAttribute::Versions versions;
+
+  for (size_t i = 0; i < rustVersionCount; i++) {
+    const RustSdpAttributeSimulcastVersion& rustVersion = rustVersionArray[i];
+    size_t rustIdCount = sdp_simulcast_get_ids_count(rustVersion.ids);
+    if (!rustIdCount) {
+      continue;
+    }
+
+    SdpSimulcastAttribute::Version version;
+    auto rustIdArray = MakeUnique<RustSdpAttributeSimulcastId[]>(rustIdCount);
+    sdp_simulcast_get_ids(rustVersion.ids, rustIdCount, rustIdArray.get());
+
+    for (size_t j = 0; j < rustIdCount; j++) {
+      const RustSdpAttributeSimulcastId& rustId = rustIdArray[j];
+      std::string id = convertStringView(rustId.id);
+      // TODO: Bug 1225877. Added support for 'paused'-state
+      version.choices.push_back(
+          SdpSimulcastAttribute::Encoding(id, rustId.paused));
+    }
+
+    versions.push_back(version);
+  }
+
+  return versions;
+}
+
+void RsdparsaSdpAttributeList::LoadSimulcast(RustAttributeList* attributeList) {
+  RustSdpAttributeSimulcast rustSimulcast;
+  if (NS_SUCCEEDED(sdp_get_simulcast(attributeList, &rustSimulcast))) {
+    auto simulcast = MakeUnique<SdpSimulcastAttribute>();
+
+    simulcast->sendVersions = LoadSimulcastVersions(rustSimulcast.send);
+    simulcast->recvVersions = LoadSimulcastVersions(rustSimulcast.recv);
+
+    SetAttribute(simulcast.release());
+  }
+}
+
+SdpImageattrAttributeList::XYRange LoadImageattrXYRange(
+    const RustSdpAttributeImageAttrXYRange& rustXYRange) {
+  SdpImageattrAttributeList::XYRange xyRange;
+
+  if (!rustXYRange.discrete_values) {
+    xyRange.min = rustXYRange.min;
+    xyRange.max = rustXYRange.max;
+    xyRange.step = rustXYRange.step;
+
+  } else {
+    xyRange.discreteValues = convertU32Vec(rustXYRange.discrete_values);
+  }
+
+  return xyRange;
+}
+
+std::vector<SdpImageattrAttributeList::Set> LoadImageattrSets(
+    const RustSdpAttributeImageAttrSetVec* rustSets) {
+  std::vector<SdpImageattrAttributeList::Set> sets;
+
+  size_t rustSetCount = sdp_imageattr_get_set_count(rustSets);
+  if (!rustSetCount) {
+    return sets;
+  }
+
+  auto rustSetArray = MakeUnique<RustSdpAttributeImageAttrSet[]>(rustSetCount);
+  sdp_imageattr_get_sets(rustSets, rustSetCount, rustSetArray.get());
+
+  for (size_t i = 0; i < rustSetCount; i++) {
+    const RustSdpAttributeImageAttrSet& rustSet = rustSetArray[i];
+    SdpImageattrAttributeList::Set set;
+
+    set.xRange = LoadImageattrXYRange(rustSet.x);
+    set.yRange = LoadImageattrXYRange(rustSet.y);
+
+    if (rustSet.has_sar) {
+      if (!rustSet.sar.discrete_values) {
+        set.sRange.min = rustSet.sar.min;
+        set.sRange.max = rustSet.sar.max;
+      } else {
+        set.sRange.discreteValues = convertF32Vec(rustSet.sar.discrete_values);
+      }
+    }
+
+    if (rustSet.has_par) {
+      set.pRange.min = rustSet.par.min;
+      set.pRange.max = rustSet.par.max;
+    }
+
+    set.qValue = rustSet.q;
+
+    sets.push_back(set);
+  }
+
+  return sets;
+}
+
+void RsdparsaSdpAttributeList::LoadImageattr(RustAttributeList* attributeList) {
+  size_t numImageattrs = sdp_get_imageattr_count(attributeList);
+  if (numImageattrs == 0) {
+    return;
+  }
+  auto rustImageattrs = MakeUnique<RustSdpAttributeImageAttr[]>(numImageattrs);
+  sdp_get_imageattrs(attributeList, numImageattrs, rustImageattrs.get());
+  auto imageattrList = MakeUnique<SdpImageattrAttributeList>();
+  for (size_t i = 0; i < numImageattrs; i++) {
+    const RustSdpAttributeImageAttr& rustImageAttr = rustImageattrs[i];
+
+    SdpImageattrAttributeList::Imageattr imageAttr;
+
+    if (rustImageAttr.payloadType != std::numeric_limits<uint32_t>::max()) {
+      imageAttr.pt = Some(rustImageAttr.payloadType);
+    }
+
+    if (rustImageAttr.send.sets) {
+      imageAttr.sendSets = LoadImageattrSets(rustImageAttr.send.sets);
+    } else {
+      imageAttr.sendAll = true;
+    }
+
+    if (rustImageAttr.recv.sets) {
+      imageAttr.recvSets = LoadImageattrSets(rustImageAttr.recv.sets);
+    } else {
+      imageAttr.recvAll = true;
+    }
+
+    imageattrList->mImageattrs.push_back(imageAttr);
+  }
+  SetAttribute(imageattrList.release());
+}
+
+void RsdparsaSdpAttributeList::LoadSctpmaps(RustAttributeList* attributeList) {
+  size_t numSctpmaps = sdp_get_sctpmap_count(attributeList);
+  if (numSctpmaps == 0) {
+    return;
+  }
+  auto rustSctpmaps = MakeUnique<RustSdpAttributeSctpmap[]>(numSctpmaps);
+  sdp_get_sctpmaps(attributeList, numSctpmaps, rustSctpmaps.get());
+  auto sctpmapList = MakeUnique<SdpSctpmapAttributeList>();
+  for (size_t i = 0; i < numSctpmaps; i++) {
+    RustSdpAttributeSctpmap& sctpmap = rustSctpmaps[i];
+    sctpmapList->PushEntry(std::to_string(sctpmap.port), "webrtc-datachannel",
+                           sctpmap.channels);
+  }
+  SetAttribute(sctpmapList.release());
+}
+
+void RsdparsaSdpAttributeList::LoadDirection(RustAttributeList* attributeList) {
+  SdpDirectionAttribute::Direction dir;
+  RustDirection rustDir = sdp_get_direction(attributeList);
+  switch (rustDir) {
+    case RustDirection::kRustRecvonly:
+      dir = SdpDirectionAttribute::kRecvonly;
+      break;
+    case RustDirection::kRustSendonly:
+      dir = SdpDirectionAttribute::kSendonly;
+      break;
+    case RustDirection::kRustSendrecv:
+      dir = SdpDirectionAttribute::kSendrecv;
+      break;
+    case RustDirection::kRustInactive:
+      dir = SdpDirectionAttribute::kInactive;
+      break;
+  }
+  SetAttribute(new SdpDirectionAttribute(dir));
+}
+
+void RsdparsaSdpAttributeList::LoadRemoteCandidates(
+    RustAttributeList* attributeList) {
+  size_t nC = sdp_get_remote_candidate_count(attributeList);
+  if (nC == 0) {
+    return;
+  }
+  auto rustCandidates = MakeUnique<RustSdpAttributeRemoteCandidate[]>(nC);
+  sdp_get_remote_candidates(attributeList, nC, rustCandidates.get());
+  std::vector<SdpRemoteCandidatesAttribute::Candidate> candidates;
+  for (size_t i = 0; i < nC; i++) {
+    RustSdpAttributeRemoteCandidate& rustCandidate = rustCandidates[i];
+    SdpRemoteCandidatesAttribute::Candidate candidate;
+    candidate.port = rustCandidate.port;
+    candidate.id = std::to_string(rustCandidate.component);
+    candidate.address = convertAddress(&rustCandidate.address);
+    candidates.push_back(candidate);
+  }
+  SdpRemoteCandidatesAttribute* candidatesList;
+  candidatesList = new SdpRemoteCandidatesAttribute(candidates);
+  SetAttribute(candidatesList);
+}
+
+void RsdparsaSdpAttributeList::LoadRids(RustAttributeList* attributeList) {
+  size_t numRids = sdp_get_rid_count(attributeList);
+  if (numRids == 0) {
+    return;
+  }
+
+  auto rustRids = MakeUnique<RustSdpAttributeRid[]>(numRids);
+  sdp_get_rids(attributeList, numRids, rustRids.get());
+
+  auto ridList = MakeUnique<SdpRidAttributeList>();
+  for (size_t i = 0; i < numRids; i++) {
+    const RustSdpAttributeRid& rid = rustRids[i];
+
+    std::string id = convertStringView(rid.id);
+    auto direction = static_cast<sdp::Direction>(rid.direction);
+    std::vector<uint16_t> formats = convertU16Vec(rid.formats);
+
+    EncodingConstraints parameters;
+    parameters.maxWidth = rid.params.max_width;
+    parameters.maxHeight = rid.params.max_height;
+    // Right now, we treat max-fps=0 and the absence of max-fps as no limit.
+    // We will eventually want to treat max-fps=0 as 0 frames per second, and
+    // the absence of max-fps as no limit (bug 1762632).
+    if (rid.params.max_fps) {
+      parameters.maxFps = Some(rid.params.max_fps);
+    }
+    parameters.maxFs = rid.params.max_fs;
+    parameters.maxBr = rid.params.max_br;
+    parameters.maxPps = rid.params.max_pps;
+
+    std::vector<std::string> depends = convertStringVec(rid.depends);
+
+    ridList->PushEntry(id, direction, formats, parameters, depends);
+  }
+
+  SetAttribute(ridList.release());
+}
+
+void RsdparsaSdpAttributeList::LoadSctpPort(RustAttributeList* attributeList) {
+  int64_t port = sdp_get_sctp_port(attributeList);
+  if (port >= 0) {
+    SetAttribute(new SdpNumberAttribute(SdpAttribute::kSctpPortAttribute,
+                                        static_cast<uint32_t>(port)));
+  }
+}
+
+void RsdparsaSdpAttributeList::LoadExtmap(RustAttributeList* attributeList) {
+  size_t numExtmap = sdp_get_extmap_count(attributeList);
+  if (numExtmap == 0) {
+    return;
+  }
+  auto rustExtmaps = MakeUnique<RustSdpAttributeExtmap[]>(numExtmap);
+  sdp_get_extmaps(attributeList, numExtmap, rustExtmaps.get());
+  auto extmaps = MakeUnique<SdpExtmapAttributeList>();
+  for (size_t i = 0; i < numExtmap; i++) {
+    RustSdpAttributeExtmap& rustExtmap = rustExtmaps[i];
+    std::string name = convertStringView(rustExtmap.url);
+    SdpDirectionAttribute::Direction direction;
+    bool directionSpecified = rustExtmap.direction_specified;
+    switch (rustExtmap.direction) {
+      case RustDirection::kRustRecvonly:
+        direction = SdpDirectionAttribute::kRecvonly;
+        break;
+      case RustDirection::kRustSendonly:
+        direction = SdpDirectionAttribute::kSendonly;
+        break;
+      case RustDirection::kRustSendrecv:
+        direction = SdpDirectionAttribute::kSendrecv;
+        break;
+      case RustDirection::kRustInactive:
+        direction = SdpDirectionAttribute::kInactive;
+        break;
+    }
+    std::string extensionAttributes;
+    extensionAttributes = convertStringView(rustExtmap.extensionAttributes);
+    extmaps->PushEntry((uint16_t)rustExtmap.id, direction, directionSpecified,
+                       name, extensionAttributes);
+  }
+  SetAttribute(extmaps.release());
+}
+
+void RsdparsaSdpAttributeList::LoadMaxPtime(RustAttributeList* attributeList) {
+  uint64_t maxPtime = 0;
+  nsresult nr = sdp_get_maxptime(attributeList, &maxPtime);
+  if (NS_SUCCEEDED(nr)) {
+    SetAttribute(
+        new SdpNumberAttribute(SdpAttribute::kMaxptimeAttribute, maxPtime));
+  }
+}
+
+void RsdparsaSdpAttributeList::LoadCandidate(RustAttributeList* attributeList) {
+  size_t candidatesCount = sdp_get_candidate_count(attributeList);
+  if (!candidatesCount) {
+    return;
+  }
+
+  StringVec* rustCandidatesStrings;
+  sdp_get_candidates(attributeList, candidatesCount, &rustCandidatesStrings);
+
+  std::vector<std::string> candidatesStrings =
+      convertStringVec(rustCandidatesStrings);
+  free_boxed_string_vec(rustCandidatesStrings);
+
+  auto candidates =
+      MakeUnique<SdpMultiStringAttribute>(SdpAttribute::kCandidateAttribute);
+  candidates->mValues = candidatesStrings;
+
+  SetAttribute(candidates.release());
+}
+
+bool RsdparsaSdpAttributeList::IsAllowedHere(SdpAttribute::AttributeType type) {
+  if (AtSessionLevel() && !SdpAttribute::IsAllowedAtSessionLevel(type)) {
+    return false;
+  }
+
+  if (!AtSessionLevel() && !SdpAttribute::IsAllowedAtMediaLevel(type)) {
+    return false;
+  }
+
+  return true;
+}
+
+void RsdparsaSdpAttributeList::Serialize(std::ostream& os) const {
+  for (size_t i = 0; i < kNumAttributeTypes; ++i) {
+    if (mAttributes[i]) {
+      os << *mAttributes[i];
+    }
+  }
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/sdp/RsdparsaSdpAttributeList.h b/dom/media/webrtc/sdp/RsdparsaSdpAttributeList.h
new file mode 100644
index 0000000000..812f15e112
--- /dev/null
+++ b/dom/media/webrtc/sdp/RsdparsaSdpAttributeList.h
@@ -0,0 +1,157 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _RSDPARSA_SDP_ATTRIBUTE_LIST_H_
+#define _RSDPARSA_SDP_ATTRIBUTE_LIST_H_
+
+#include "sdp/RsdparsaSdpGlue.h"
+#include "sdp/RsdparsaSdpInc.h"
+#include "sdp/SdpAttributeList.h"
+
+namespace mozilla {
+
+class RsdparsaSdp;
+class RsdparsaSdpMediaSection;
+class SdpParser;
+
+class RsdparsaSdpAttributeList : public SdpAttributeList {
+  friend class RsdparsaSdpMediaSection;
+  friend class RsdparsaSdp;
+
+ public:
+  // Make sure we don't hide the default arg thunks
+  using SdpAttributeList::GetAttribute;
+  using SdpAttributeList::HasAttribute;
+
+  bool HasAttribute(AttributeType type, bool sessionFallback) const override;
+  const SdpAttribute* GetAttribute(AttributeType type,
+                                   bool sessionFallback) const override;
+  void SetAttribute(SdpAttribute* attr) override;
+  void RemoveAttribute(AttributeType type) override;
+  void Clear() override;
+  uint32_t Count() const override;
+
+  const SdpConnectionAttribute& GetConnection() const override;
+  const SdpFingerprintAttributeList& GetFingerprint() const override;
+  const SdpGroupAttributeList& GetGroup() const override;
+  const SdpOptionsAttribute& GetIceOptions() const override;
+  const SdpRtcpAttribute& GetRtcp() const override;
+  const SdpRemoteCandidatesAttribute& GetRemoteCandidates() const override;
+  const SdpSetupAttribute& GetSetup() const override;
+  const SdpSsrcAttributeList& GetSsrc() const override;
+  const SdpSsrcGroupAttributeList& GetSsrcGroup() const override;
+  const SdpDtlsMessageAttribute& GetDtlsMessage() const override;
+
+  // These attributes can appear multiple times, so the returned
+  // classes actually represent a collection of values.
+  const std::vector<std::string>& GetCandidate() const override;
+  const SdpExtmapAttributeList& GetExtmap() const override;
+  const SdpFmtpAttributeList& GetFmtp() const override;
+  const SdpImageattrAttributeList& GetImageattr() const override;
+  const SdpSimulcastAttribute& GetSimulcast() const override;
+  const SdpMsidAttributeList& GetMsid() const override;
+  const SdpMsidSemanticAttributeList& GetMsidSemantic() const override;
+  const SdpRidAttributeList& GetRid() const override;
+  const SdpRtcpFbAttributeList& GetRtcpFb() const override;
+  const SdpRtpmapAttributeList& GetRtpmap() const override;
+  const SdpSctpmapAttributeList& GetSctpmap() const override;
+
+  // These attributes are effectively simple types, so we'll make life
+  // easy by just returning their value.
+  uint32_t GetSctpPort() const override;
+  uint32_t GetMaxMessageSize() const override;
+  const std::string& GetIcePwd() const override;
+  const std::string& GetIceUfrag() const override;
+  const std::string& GetIdentity() const override;
+  const std::string& GetLabel() const override;
+  unsigned int GetMaxptime() const override;
+  const std::string& GetMid() const override;
+  unsigned int GetPtime() const override;
+
+  SdpDirectionAttribute::Direction GetDirection() const override;
+
+  void Serialize(std::ostream&) const override;
+
+  virtual ~RsdparsaSdpAttributeList();
+
+ private:
+  explicit RsdparsaSdpAttributeList(RsdparsaSessionHandle session)
+      : mSession(std::move(session)),
+        mSessionAttributes(nullptr),
+        mIsVideo(false),
+        mAttributes() {
+    RustAttributeList* attributes = get_sdp_session_attributes(mSession.get());
+    LoadAll(attributes);
+  }
+
+  RsdparsaSdpAttributeList(RsdparsaSessionHandle session,
+                           const RustMediaSection* const msection,
+                           const RsdparsaSdpAttributeList* sessionAttributes)
+      : mSession(std::move(session)),
+        mSessionAttributes(sessionAttributes),
+        mAttributes() {
+    mIsVideo =
+        sdp_rust_get_media_type(msection) == RustSdpMediaValue::kRustVideo;
+    RustAttributeList* attributes = sdp_get_media_attribute_list(msection);
+    LoadAll(attributes);
+  }
+
+  static const std::string kEmptyString;
+  static const size_t kNumAttributeTypes = SdpAttribute::kLastAttribute + 1;
+
+  const RsdparsaSessionHandle mSession;
+  const RsdparsaSdpAttributeList* mSessionAttributes;
+  bool mIsVideo;
+
+  bool AtSessionLevel() const { return !mSessionAttributes; }
+
+  bool IsAllowedHere(SdpAttribute::AttributeType type);
+  void LoadAll(RustAttributeList* attributeList);
+  void LoadAttribute(RustAttributeList* attributeList, AttributeType type);
+  void LoadIceUfrag(RustAttributeList* attributeList);
+  void LoadIcePwd(RustAttributeList* attributeList);
+  void LoadIdentity(RustAttributeList* attributeList);
+  void LoadIceOptions(RustAttributeList* attributeList);
+  void LoadFingerprint(RustAttributeList* attributeList);
+  void LoadDtlsMessage(RustAttributeList* attributeList);
+  void LoadSetup(RustAttributeList* attributeList);
+  void LoadSsrc(RustAttributeList* attributeList);
+  void LoadSsrcGroup(RustAttributeList* attributeList);
+  void LoadRtpmap(RustAttributeList* attributeList);
+  void LoadFmtp(RustAttributeList* attributeList);
+  void LoadPtime(RustAttributeList* attributeList);
+  void LoadFlags(RustAttributeList* attributeList);
+  void LoadMaxMessageSize(RustAttributeList* attributeList);
+  void LoadMid(RustAttributeList* attributeList);
+  void LoadMsid(RustAttributeList* attributeList);
+  void LoadMsidSemantics(RustAttributeList* attributeList);
+  void LoadGroup(RustAttributeList* attributeList);
+  void LoadRtcp(RustAttributeList* attributeList);
+  void LoadRtcpFb(RustAttributeList* attributeList);
+  void LoadSctpPort(RustAttributeList* attributeList);
+  void LoadSimulcast(RustAttributeList* attributeList);
+  void LoadImageattr(RustAttributeList* attributeList);
+  void LoadSctpmaps(RustAttributeList* attributeList);
+  void LoadDirection(RustAttributeList* attributeList);
+  void LoadRemoteCandidates(RustAttributeList* attributeList);
+  void LoadRids(RustAttributeList* attributeList);
+  void LoadExtmap(RustAttributeList* attributeList);
+  void LoadMaxPtime(RustAttributeList* attributeList);
+  void LoadCandidate(RustAttributeList* attributeList);
+
+  void WarnAboutMisplacedAttribute(SdpAttribute::AttributeType type,
+                                   uint32_t lineNumber, SdpParser& errorHolder);
+
+  SdpAttribute* mAttributes[kNumAttributeTypes];
+
+  RsdparsaSdpAttributeList(const RsdparsaSdpAttributeList& orig) = delete;
+  RsdparsaSdpAttributeList& operator=(const RsdparsaSdpAttributeList& rhs) =
+      delete;
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/sdp/RsdparsaSdpGlue.cpp b/dom/media/webrtc/sdp/RsdparsaSdpGlue.cpp
new file mode 100644
index 0000000000..01a1a1d817
--- /dev/null
+++ b/dom/media/webrtc/sdp/RsdparsaSdpGlue.cpp
@@ -0,0 +1,106 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+#include <string>
+#include <cstdint>
+
+#include "sdp/RsdparsaSdpInc.h"
+#include "sdp/RsdparsaSdpGlue.h"
+namespace mozilla {
+
+std::string convertStringView(StringView str) {
+  if (nullptr == str.buf) {
+    return std::string();
+  } else {
+    return std::string(str.buf, str.len);
+  }
+}
+
+std::vector<std::string> convertStringVec(StringVec* vec) {
+  std::vector<std::string> ret;
+  size_t len = string_vec_len(vec);
+  for (size_t i = 0; i < len; i++) {
+    StringView view;
+    string_vec_get_view(vec, i, &view);
+    ret.push_back(convertStringView(view));
+  }
+  return ret;
+}
+
+sdp::AddrType convertAddressType(RustSdpAddressType addrType) {
+  switch (addrType) {
+    case RustSdpAddressType::kRustAddrIp4:
+      return sdp::kIPv4;
+    case RustSdpAddressType::kRustAddrIp6:
+      return sdp::kIPv6;
+  }
+
+  MOZ_CRASH("unknown address type");
+}
+
+std::string convertAddress(RustAddress* address) {
+  return address->isFqdn ? convertStringView(address->fqdn)
+                         : std::string(address->ipAddress);
+}
+
+std::pair<sdp::AddrType, std::string> convertExplicitlyTypedAddress(
+    RustExplicitlyTypedAddress* address) {
+  return std::make_pair(convertAddressType(address->addressType),
+                        convertAddress(&address->address));
+}
+
+std::vector<uint8_t> convertU8Vec(U8Vec* vec) {
+  std::vector<std::uint8_t> ret;
+
+  size_t len = u8_vec_len(vec);
+  for (size_t i = 0; i < len; i++) {
+    uint8_t byte;
+    u8_vec_get(vec, i, &byte);
+    ret.push_back(byte);
+  }
+
+  return ret;
+}
+
+std::vector<uint16_t> convertU16Vec(U16Vec* vec) {
+  std::vector<std::uint16_t> ret;
+
+  size_t len = u16_vec_len(vec);
+  for (size_t i = 0; i < len; i++) {
+    uint16_t word;
+    u16_vec_get(vec, i, &word);
+    ret.push_back(word);
+  }
+
+  return ret;
+}
+
+std::vector<uint32_t> convertU32Vec(U32Vec* vec) {
+  std::vector<std::uint32_t> ret;
+
+  size_t len = u32_vec_len(vec);
+  for (size_t i = 0; i < len; i++) {
+    uint32_t num;
+    u32_vec_get(vec, i, &num);
+    ret.push_back(num);
+  }
+
+  return ret;
+}
+
+std::vector<float> convertF32Vec(F32Vec* vec) {
+  std::vector<float> ret;
+
+  size_t len = f32_vec_len(vec);
+  for (size_t i = 0; i < len; i++) {
+    float flt;
+    f32_vec_get(vec, i, &flt);
+    ret.push_back(flt);
+  }
+
+  return ret;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/sdp/RsdparsaSdpGlue.h b/dom/media/webrtc/sdp/RsdparsaSdpGlue.h
new file mode 100644
index 0000000000..8ed275be75
--- /dev/null
+++ b/dom/media/webrtc/sdp/RsdparsaSdpGlue.h
@@ -0,0 +1,36 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+#ifndef _RUSTSDPGLUE_H_
+#define _RUSTSDPGLUE_H_
+
+#include <string>
+#include <vector>
+#include <utility>
+#include "SdpEnum.h"
+#include "mozilla/UniquePtr.h"
+#include "sdp/RsdparsaSdpInc.h"
+
+namespace mozilla {
+
+struct FreeRustSdpSession {
+  void operator()(RustSdpSession* aSess) { sdp_free_session(aSess); }
+};
+
+typedef UniquePtr<RustSdpSession, FreeRustSdpSession> RsdparsaSessionHandle;
+
+std::string convertStringView(StringView str);
+std::vector<std::string> convertStringVec(StringVec* vec);
+std::string convertAddress(RustAddress* address);
+std::pair<sdp::AddrType, std::string> convertExplicitlyTypedAddress(
+    RustExplicitlyTypedAddress* addr);
+std::vector<uint8_t> convertU8Vec(U8Vec* vec);
+std::vector<uint16_t> convertU16Vec(U16Vec* vec);
+std::vector<uint32_t> convertU32Vec(U32Vec* vec);
+std::vector<float> convertF32Vec(F32Vec* vec);
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/sdp/RsdparsaSdpInc.h b/dom/media/webrtc/sdp/RsdparsaSdpInc.h
new file mode 100644
index 0000000000..1237d2fdad
--- /dev/null
+++ b/dom/media/webrtc/sdp/RsdparsaSdpInc.h
@@ -0,0 +1,510 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+#ifndef _RUSTSDPINC_H_
+#define _RUSTSDPINC_H_
+
+#include "nsError.h"
+#include "mozilla/Maybe.h"
+
+#include <stdint.h>
+#include <stdbool.h>
+
+struct BandwidthVec;
+struct RustSdpSession;
+struct RustSdpError;
+struct RustMediaSection;
+struct RustAttributeList;
+struct StringVec;
+struct U8Vec;
+struct U32Vec;
+struct U16Vec;
+struct F32Vec;
+struct SsrcVec;
+struct RustHeapString;
+
+enum class RustSdpAddressType { kRustAddrIp4, kRustAddrIp6 };
+
+struct StringView {
+  const char* buf;
+  size_t len;
+};
+
+struct RustAddress {
+  char ipAddress[50];
+  StringView fqdn;
+  bool isFqdn;
+};
+
+struct RustExplicitlyTypedAddress {
+  RustSdpAddressType addressType;
+  RustAddress address;
+};
+
+struct RustSdpConnection {
+  RustExplicitlyTypedAddress addr;
+  uint8_t ttl;
+  uint64_t amount;
+};
+
+struct RustSdpOrigin {
+  StringView username;
+  uint64_t sessionId;
+  uint64_t sessionVersion;
+  RustExplicitlyTypedAddress addr;  // TODO address
+};
+
+enum class RustSdpMediaValue { kRustAudio, kRustVideo, kRustApplication };
+
+enum class RustSdpProtocolValue {
+  kRustRtpSavpf,
+  kRustUdpTlsRtpSavp,
+  kRustTcpDtlsRtpSavp,
+  kRustUdpTlsRtpSavpf,
+  kRustTcpDtlsRtpSavpf,
+  kRustDtlsSctp,
+  kRustUdpDtlsSctp,
+  kRustTcpDtlsSctp,
+  kRustRtpAvp,
+  kRustRtpAvpf,
+  kRustRtpSavp,
+};
+
+enum class RustSdpFormatType { kRustIntegers, kRustStrings };
+
+enum class RustSdpAttributeFingerprintHashAlgorithm : uint16_t {
+  kSha1,
+  kSha224,
+  kSha256,
+  kSha384,
+  kSha512,
+};
+
+struct RustSdpAttributeFingerprint {
+  RustSdpAttributeFingerprintHashAlgorithm hashAlgorithm;
+  U8Vec* fingerprint;
+};
+
+enum class RustSdpSetup {
+  kRustActive,
+  kRustActpass,
+  kRustHoldconn,
+  kRustPassive
+};
+
+enum class RustSdpAttributeDtlsMessageType : uint8_t {
+  kClient,
+  kServer,
+};
+
+struct RustSdpAttributeDtlsMessage {
+  RustSdpAttributeDtlsMessageType role;
+  StringView value;
+};
+
+struct RustSdpAttributeSsrc {
+  uint32_t id;
+  StringView attribute;
+  StringView value;
+};
+
+enum class RustSdpAttributeSsrcGroupSemantic {
+  kRustDup,
+  kRustFid,
+  kRustFec,
+  kRustFecFr,
+  kRustSim,
+};
+
+struct RustSdpAttributeSsrcGroup {
+  RustSdpAttributeSsrcGroupSemantic semantic;
+  SsrcVec* ssrcs;
+};
+
+struct RustSdpAttributeRtpmap {
+  uint8_t payloadType;
+  StringView codecName;
+  uint32_t frequency;
+  uint32_t channels;
+};
+
+struct RustSdpAttributeRtcpFb {
+  uint32_t payloadType;
+  uint32_t feedbackType;
+  StringView parameter;
+  StringView extra;
+};
+
+struct RustSdpAttributeRidParameters {
+  uint32_t max_width;
+  uint32_t max_height;
+  uint32_t max_fps;
+  uint32_t max_fs;
+  uint32_t max_br;
+  uint32_t max_pps;
+  StringVec* unknown;
+};
+
+struct RustSdpAttributeRid {
+  StringView id;
+  uint32_t direction;
+  U16Vec* formats;
+  RustSdpAttributeRidParameters params;
+  StringVec* depends;
+};
+
+struct RustSdpAttributeImageAttrXYRange {
+  uint32_t min;
+  uint32_t max;
+  uint32_t step;
+  U32Vec* discrete_values;
+};
+
+struct RustSdpAttributeImageAttrSRange {
+  float min;
+  float max;
+  F32Vec* discrete_values;
+};
+
+struct RustSdpAttributeImageAttrPRange {
+  float min;
+  float max;
+};
+
+struct RustSdpAttributeImageAttrSet {
+  RustSdpAttributeImageAttrXYRange x;
+  RustSdpAttributeImageAttrXYRange y;
+  bool has_sar;
+  RustSdpAttributeImageAttrSRange sar;
+  bool has_par;
+  RustSdpAttributeImageAttrPRange par;
+  float q;
+};
+
+struct RustSdpAttributeImageAttrSetVec;
+struct RustSdpAttributeImageAttrSetList {
+  RustSdpAttributeImageAttrSetVec* sets;
+};
+
+struct RustSdpAttributeImageAttr {
+  uint32_t payloadType;
+  RustSdpAttributeImageAttrSetList send;
+  RustSdpAttributeImageAttrSetList recv;
+};
+
+struct RustRtxFmtpParameters {
+  uint8_t apt;
+  bool has_rtx_time;
+  uint32_t rtx_time;
+};
+
+struct RustSdpAttributeFmtpParameters {
+  // H264
+  uint32_t packetization_mode;
+  bool level_asymmetry_allowed;
+  uint32_t profile_level_id;
+  uint32_t max_fs;
+  uint32_t max_cpb;
+  uint32_t max_dpb;
+  uint32_t max_br;
+  uint32_t max_mbps;
+
+  // VP8 and VP9
+  // max_fs, already defined in H264
+  uint32_t max_fr;
+
+  // Opus
+  uint32_t maxplaybackrate;
+  uint32_t maxaveragebitrate;
+  bool usedtx;
+  bool stereo;
+  bool useinbandfec;
+  bool cbr;
+  uint32_t ptime;
+  uint32_t minptime;
+  uint32_t maxptime;
+
+  // telephone-event
+  StringView dtmf_tones;
+
+  // RTX
+  RustRtxFmtpParameters rtx;
+
+  // Red codecs
+  U8Vec* encodings;
+
+  // Unknown
+  StringVec* unknown_tokens;
+};
+
+struct RustSdpAttributeFmtp {
+  uint8_t payloadType;
+  StringView codecName;
+  RustSdpAttributeFmtpParameters parameters;
+};
+
+struct RustSdpAttributeFlags {
+  bool iceLite;
+  bool rtcpMux;
+  bool rtcpRsize;
+  bool bundleOnly;
+  bool endOfCandidates;
+};
+
+struct RustSdpAttributeMsid {
+  StringView id;
+  StringView appdata;
+};
+
+struct RustSdpAttributeMsidSemantic {
+  StringView semantic;
+  StringVec* msids;
+};
+
+enum class RustSdpAttributeGroupSemantic {
+  kRustLipSynchronization,
+  kRustFlowIdentification,
+  kRustSingleReservationFlow,
+  kRustAlternateNetworkAddressType,
+  kRustForwardErrorCorrection,
+  kRustDecodingDependency,
+  kRustBundle,
+};
+
+struct RustSdpAttributeGroup {
+  RustSdpAttributeGroupSemantic semantic;
+  StringVec* tags;
+};
+
+struct RustSdpAttributeRtcp {
+  uint32_t port;
+  RustExplicitlyTypedAddress unicastAddr;
+  bool has_address;
+};
+
+struct RustSdpAttributeSctpmap {
+  uint32_t port;
+  uint32_t channels;
+};
+
+struct RustSdpAttributeSimulcastId {
+  StringView id;
+  bool paused;
+};
+
+struct RustSdpAttributeSimulcastIdVec;
+struct RustSdpAttributeSimulcastVersion {
+  RustSdpAttributeSimulcastIdVec* ids;
+};
+
+struct RustSdpAttributeSimulcastVersionVec;
+struct RustSdpAttributeSimulcast {
+  RustSdpAttributeSimulcastVersionVec* send;
+  RustSdpAttributeSimulcastVersionVec* recv;
+};
+
+enum class RustDirection {
+  kRustRecvonly,
+  kRustSendonly,
+  kRustSendrecv,
+  kRustInactive
+};
+
+struct RustSdpAttributeRemoteCandidate {
+  uint32_t component;
+  RustAddress address;
+  uint32_t port;
+};
+
+struct RustSdpAttributeExtmap {
+  uint16_t id;
+  bool direction_specified;
+  RustDirection direction;
+  StringView url;
+  StringView extensionAttributes;
+};
+
+extern "C" {
+
+size_t string_vec_len(const StringVec* vec);
+nsresult string_vec_get_view(const StringVec* vec, size_t index,
+                             StringView* str);
+nsresult free_boxed_string_vec(StringVec* vec);
+
+size_t f32_vec_len(const F32Vec* vec);
+nsresult f32_vec_get(const F32Vec* vec, size_t index, float* ret);
+
+size_t u32_vec_len(const U32Vec* vec);
+nsresult u32_vec_get(const U32Vec* vec, size_t index, uint32_t* ret);
+
+size_t u16_vec_len(const U16Vec* vec);
+nsresult u16_vec_get(const U16Vec* vec, size_t index, uint16_t* ret);
+
+size_t u8_vec_len(const U8Vec* vec);
+nsresult u8_vec_get(const U8Vec* vec, size_t index, uint8_t* ret);
+
+size_t ssrc_vec_len(const SsrcVec* vec);
+nsresult ssrc_vec_get_id(const SsrcVec* vec, size_t index, uint32_t* ret);
+
+void sdp_free_string(char* string);
+
+nsresult parse_sdp(StringView sdp, bool fail_on_warning, RustSdpSession** ret,
+                   RustSdpError** err);
+RustSdpSession* sdp_new_reference(RustSdpSession* aSess);
+RustSdpSession* create_sdp_clone(const RustSdpSession* aSess);
+void sdp_free_session(RustSdpSession* ret);
+size_t sdp_get_error_line_num(const RustSdpError* err);
+char* sdp_get_error_message(const RustSdpError* err);
+void sdp_free_error_message(char* message);
+void sdp_free_error(RustSdpError* err);
+
+RustSdpOrigin sdp_get_origin(const RustSdpSession* aSess);
+
+uint32_t get_sdp_bandwidth(const RustSdpSession* aSess,
+                           const char* aBandwidthType);
+BandwidthVec* sdp_get_session_bandwidth_vec(const RustSdpSession* aSess);
+BandwidthVec* sdp_get_media_bandwidth_vec(const RustMediaSection* aMediaSec);
+char* sdp_serialize_bandwidth(const BandwidthVec* bandwidths);
+bool sdp_session_has_connection(const RustSdpSession* aSess);
+nsresult sdp_get_session_connection(const RustSdpSession* aSess,
+                                    RustSdpConnection* ret);
+RustAttributeList* get_sdp_session_attributes(const RustSdpSession* aSess);
+
+size_t sdp_media_section_count(const RustSdpSession* aSess);
+RustMediaSection* sdp_get_media_section(const RustSdpSession* aSess,
+                                        size_t aLevel);
+nsresult sdp_add_media_section(RustSdpSession* aSess, uint32_t aMediaType,
+                               uint32_t aDirection, uint16_t aPort,
+                               uint32_t aProtocol, uint32_t aAddrType,
+                               StringView aAddr);
+RustSdpMediaValue sdp_rust_get_media_type(const RustMediaSection* aMediaSec);
+RustSdpProtocolValue sdp_get_media_protocol(const RustMediaSection* aMediaSec);
+RustSdpFormatType sdp_get_format_type(const RustMediaSection* aMediaSec);
+StringVec* sdp_get_format_string_vec(const RustMediaSection* aMediaSec);
+U32Vec* sdp_get_format_u32_vec(const RustMediaSection* aMediaSec);
+void sdp_set_media_port(const RustMediaSection* aMediaSec, uint32_t aPort);
+uint32_t sdp_get_media_port(const RustMediaSection* aMediaSec);
+uint32_t sdp_get_media_port_count(const RustMediaSection* aMediaSec);
+uint32_t sdp_get_media_bandwidth(const RustMediaSection* aMediaSec,
+                                 const char* aBandwidthType);
+bool sdp_media_has_connection(const RustMediaSection* aMediaSec);
+nsresult sdp_get_media_connection(const RustMediaSection* aMediaSec,
+                                  RustSdpConnection* ret);
+
+RustAttributeList* sdp_get_media_attribute_list(
+    const RustMediaSection* aMediaSec);
+
+nsresult sdp_media_add_codec(const RustMediaSection* aMediaSec, uint8_t aPT,
+                             StringView aCodecName, uint32_t aClockrate,
+                             uint16_t channels);
+void sdp_media_clear_codecs(const RustMediaSection* aMediaSec);
+nsresult sdp_media_add_datachannel(const RustMediaSection* aMediaSec,
+                                   StringView aName, uint16_t aPort,
+                                   uint16_t streams, uint32_t aMessageSize);
+
+nsresult sdp_get_iceufrag(const RustAttributeList* aList, StringView* ret);
+nsresult sdp_get_icepwd(const RustAttributeList* aList, StringView* ret);
+nsresult sdp_get_identity(const RustAttributeList* aList, StringView* ret);
+nsresult sdp_get_iceoptions(const RustAttributeList* aList, StringVec** ret);
+
+nsresult sdp_get_dtls_message(const RustAttributeList* aList,
+                              RustSdpAttributeDtlsMessage* ret);
+
+size_t sdp_get_fingerprint_count(const RustAttributeList* aList);
+void sdp_get_fingerprints(const RustAttributeList* aList, size_t listSize,
+                          RustSdpAttributeFingerprint* ret);
+
+nsresult sdp_get_setup(const RustAttributeList* aList, RustSdpSetup* ret);
+
+size_t sdp_get_ssrc_count(const RustAttributeList* aList);
+void sdp_get_ssrcs(const RustAttributeList* aList, size_t listSize,
+                   RustSdpAttributeSsrc* ret);
+
+size_t sdp_get_ssrc_group_count(const RustAttributeList* aList);
+void sdp_get_ssrc_groups(const RustAttributeList* aList, size_t listSize,
+                         RustSdpAttributeSsrcGroup* ret);
+
+size_t sdp_get_rtpmap_count(const RustAttributeList* aList);
+void sdp_get_rtpmaps(const RustAttributeList* aList, size_t listSize,
+                     RustSdpAttributeRtpmap* ret);
+
+size_t sdp_get_fmtp_count(const RustAttributeList* aList);
+size_t sdp_get_fmtp(const RustAttributeList* aList, size_t listSize,
+                    RustSdpAttributeFmtp* ret);
+
+int64_t sdp_get_ptime(const RustAttributeList* aList);
+int64_t sdp_get_max_msg_size(const RustAttributeList* aList);
+int64_t sdp_get_sctp_port(const RustAttributeList* aList);
+nsresult sdp_get_maxptime(const RustAttributeList* aList, uint64_t* aMaxPtime);
+
+RustSdpAttributeFlags sdp_get_attribute_flags(const RustAttributeList* aList);
+
+nsresult sdp_get_mid(const RustAttributeList* aList, StringView* ret);
+
+size_t sdp_get_msid_count(const RustAttributeList* aList);
+void sdp_get_msids(const RustAttributeList* aList, size_t listSize,
+                   RustSdpAttributeMsid* ret);
+
+size_t sdp_get_msid_semantic_count(RustAttributeList* aList);
+void sdp_get_msid_semantics(const RustAttributeList* aList, size_t listSize,
+                            RustSdpAttributeMsidSemantic* ret);
+
+size_t sdp_get_group_count(const RustAttributeList* aList);
+void sdp_get_groups(const RustAttributeList* aList, size_t listSize,
+                    RustSdpAttributeGroup* ret);
+
+nsresult sdp_get_rtcp(const RustAttributeList* aList,
+                      RustSdpAttributeRtcp* ret);
+
+size_t sdp_get_rtcpfb_count(const RustAttributeList* aList);
+void sdp_get_rtcpfbs(const RustAttributeList* aList, size_t listSize,
+                     RustSdpAttributeRtcpFb* ret);
+
+size_t sdp_get_imageattr_count(const RustAttributeList* aList);
+void sdp_get_imageattrs(const RustAttributeList* aList, size_t listSize,
+                        RustSdpAttributeImageAttr* ret);
+
+size_t sdp_imageattr_get_set_count(const RustSdpAttributeImageAttrSetVec* sets);
+void sdp_imageattr_get_sets(const RustSdpAttributeImageAttrSetVec* sets,
+                            size_t listSize, RustSdpAttributeImageAttrSet* ret);
+
+size_t sdp_get_sctpmap_count(const RustAttributeList* aList);
+void sdp_get_sctpmaps(const RustAttributeList* aList, size_t listSize,
+                      RustSdpAttributeSctpmap* ret);
+
+nsresult sdp_get_simulcast(const RustAttributeList* aList,
+                           RustSdpAttributeSimulcast* ret);
+
+size_t sdp_simulcast_get_version_count(
+    const RustSdpAttributeSimulcastVersionVec* aVersionList);
+void sdp_simulcast_get_versions(
+    const RustSdpAttributeSimulcastVersionVec* aversionList, size_t listSize,
+    RustSdpAttributeSimulcastVersion* ret);
+
+size_t sdp_simulcast_get_ids_count(const RustSdpAttributeSimulcastIdVec* aAlts);
+void sdp_simulcast_get_ids(const RustSdpAttributeSimulcastIdVec* aAlts,
+                           size_t listSize, RustSdpAttributeSimulcastId* ret);
+
+RustDirection sdp_get_direction(const RustAttributeList* aList);
+
+size_t sdp_get_remote_candidate_count(const RustAttributeList* aList);
+void sdp_get_remote_candidates(const RustAttributeList* aList, size_t listSize,
+                               RustSdpAttributeRemoteCandidate* ret);
+
+size_t sdp_get_candidate_count(const RustAttributeList* aList);
+void sdp_get_candidates(const RustAttributeList* aLisst, size_t listSize,
+                        StringVec** ret);
+
+size_t sdp_get_rid_count(const RustAttributeList* aList);
+void sdp_get_rids(const RustAttributeList* aList, size_t listSize,
+                  RustSdpAttributeRid* ret);
+
+size_t sdp_get_extmap_count(const RustAttributeList* aList);
+void sdp_get_extmaps(const RustAttributeList* aList, size_t listSize,
+                     RustSdpAttributeExtmap* ret);
+
+}  // extern "C"
+
+#endif
diff --git a/dom/media/webrtc/sdp/RsdparsaSdpMediaSection.cpp b/dom/media/webrtc/sdp/RsdparsaSdpMediaSection.cpp
new file mode 100644
index 0000000000..25084db2ad
--- /dev/null
+++ b/dom/media/webrtc/sdp/RsdparsaSdpMediaSection.cpp
@@ -0,0 +1,253 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "sdp/SdpMediaSection.h"
+#include "sdp/RsdparsaSdpMediaSection.h"
+
+#include "sdp/RsdparsaSdpGlue.h"
+#include "sdp/RsdparsaSdpInc.h"
+
+#include <ostream>
+
+#ifdef CRLF
+#  undef CRLF
+#endif
+#define CRLF "\r\n"
+
+namespace mozilla {
+
+RsdparsaSdpMediaSection::RsdparsaSdpMediaSection(
+    size_t level, RsdparsaSessionHandle session,
+    const RustMediaSection* const section,
+    const RsdparsaSdpAttributeList* sessionLevel)
+    : SdpMediaSection(level), mSession(std::move(session)), mSection(section) {
+  switch (sdp_rust_get_media_type(section)) {
+    case RustSdpMediaValue::kRustAudio:
+      mMediaType = kAudio;
+      break;
+    case RustSdpMediaValue::kRustVideo:
+      mMediaType = kVideo;
+      break;
+    case RustSdpMediaValue::kRustApplication:
+      mMediaType = kApplication;
+      break;
+  }
+
+  RsdparsaSessionHandle attributeSession(sdp_new_reference(mSession.get()));
+  mAttributeList.reset(new RsdparsaSdpAttributeList(std::move(attributeSession),
+                                                    section, sessionLevel));
+
+  LoadFormats();
+  LoadConnection();
+}
+
+unsigned int RsdparsaSdpMediaSection::GetPort() const {
+  return sdp_get_media_port(mSection);
+}
+
+void RsdparsaSdpMediaSection::SetPort(unsigned int port) {
+  sdp_set_media_port(mSection, port);
+}
+
+unsigned int RsdparsaSdpMediaSection::GetPortCount() const {
+  return sdp_get_media_port_count(mSection);
+}
+
+SdpMediaSection::Protocol RsdparsaSdpMediaSection::GetProtocol() const {
+  switch (sdp_get_media_protocol(mSection)) {
+    case RustSdpProtocolValue::kRustRtpSavpf:
+      return kRtpSavpf;
+    case RustSdpProtocolValue::kRustUdpTlsRtpSavp:
+      return kUdpTlsRtpSavp;
+    case RustSdpProtocolValue::kRustTcpDtlsRtpSavp:
+      return kTcpDtlsRtpSavp;
+    case RustSdpProtocolValue::kRustUdpTlsRtpSavpf:
+      return kUdpTlsRtpSavpf;
+    case RustSdpProtocolValue::kRustTcpDtlsRtpSavpf:
+      return kTcpDtlsRtpSavpf;
+    case RustSdpProtocolValue::kRustDtlsSctp:
+      return kDtlsSctp;
+    case RustSdpProtocolValue::kRustUdpDtlsSctp:
+      return kUdpDtlsSctp;
+    case RustSdpProtocolValue::kRustTcpDtlsSctp:
+      return kTcpDtlsSctp;
+    case RustSdpProtocolValue::kRustRtpAvp:
+      return kRtpAvp;
+    case RustSdpProtocolValue::kRustRtpAvpf:
+      return kRtpAvpf;
+    case RustSdpProtocolValue::kRustRtpSavp:
+      return kRtpSavp;
+  }
+  MOZ_CRASH("invalid media protocol");
+}
+
+const SdpConnection& RsdparsaSdpMediaSection::GetConnection() const {
+  MOZ_ASSERT(mConnection);
+  return *mConnection;
+}
+
+SdpConnection& RsdparsaSdpMediaSection::GetConnection() {
+  MOZ_ASSERT(mConnection);
+  return *mConnection;
+}
+
+uint32_t RsdparsaSdpMediaSection::GetBandwidth(const std::string& type) const {
+  return sdp_get_media_bandwidth(mSection, type.c_str());
+}
+
+const std::vector<std::string>& RsdparsaSdpMediaSection::GetFormats() const {
+  return mFormats;
+}
+
+const SdpAttributeList& RsdparsaSdpMediaSection::GetAttributeList() const {
+  return *mAttributeList;
+}
+
+SdpAttributeList& RsdparsaSdpMediaSection::GetAttributeList() {
+  return *mAttributeList;
+}
+
+SdpDirectionAttribute RsdparsaSdpMediaSection::GetDirectionAttribute() const {
+  return SdpDirectionAttribute(mAttributeList->GetDirection());
+}
+
+void RsdparsaSdpMediaSection::AddCodec(const std::string& pt,
+                                       const std::string& name,
+                                       uint32_t clockrate, uint16_t channels) {
+  StringView rustName{name.c_str(), name.size()};
+
+  // call the rust interface
+  auto nr = sdp_media_add_codec(mSection, std::stoul(pt), rustName, clockrate,
+                                channels);
+
+  if (NS_SUCCEEDED(nr)) {
+    // If the rust call was successful, adjust the shadow C++ structures
+    mFormats.push_back(pt);
+
+    // Add a rtpmap in mAttributeList
+    SdpRtpmapAttributeList* rtpmap = new SdpRtpmapAttributeList();
+    if (mAttributeList->HasAttribute(SdpAttribute::kRtpmapAttribute)) {
+      const SdpRtpmapAttributeList& old = mAttributeList->GetRtpmap();
+      for (auto it = old.mRtpmaps.begin(); it != old.mRtpmaps.end(); ++it) {
+        rtpmap->mRtpmaps.push_back(*it);
+      }
+    }
+
+    SdpRtpmapAttributeList::CodecType codec =
+        SdpRtpmapAttributeList::kOtherCodec;
+    if (name == "opus") {
+      codec = SdpRtpmapAttributeList::kOpus;
+    } else if (name == "VP8") {
+      codec = SdpRtpmapAttributeList::kVP8;
+    } else if (name == "VP9") {
+      codec = SdpRtpmapAttributeList::kVP9;
+    } else if (name == "H264") {
+      codec = SdpRtpmapAttributeList::kH264;
+    }
+
+    rtpmap->PushEntry(pt, codec, name, clockrate, channels);
+    mAttributeList->SetAttribute(rtpmap);
+  }
+}
+
+void RsdparsaSdpMediaSection::ClearCodecs() {
+  // Clear the codecs in rust
+  sdp_media_clear_codecs(mSection);
+
+  mFormats.clear();
+  mAttributeList->RemoveAttribute(SdpAttribute::kRtpmapAttribute);
+  mAttributeList->RemoveAttribute(SdpAttribute::kFmtpAttribute);
+  mAttributeList->RemoveAttribute(SdpAttribute::kSctpmapAttribute);
+  mAttributeList->RemoveAttribute(SdpAttribute::kRtcpFbAttribute);
+}
+
+void RsdparsaSdpMediaSection::AddDataChannel(const std::string& name,
+                                             uint16_t port, uint16_t streams,
+                                             uint32_t message_size) {
+  StringView rustName{name.c_str(), name.size()};
+  auto nr = sdp_media_add_datachannel(mSection, rustName, port, streams,
+                                      message_size);
+  if (NS_SUCCEEDED(nr)) {
+    // Update the formats
+    mFormats.clear();
+    LoadFormats();
+
+    // Update the attribute list
+    RsdparsaSessionHandle sessHandle(sdp_new_reference(mSession.get()));
+    auto sessAttributes = mAttributeList->mSessionAttributes;
+    mAttributeList.reset(new RsdparsaSdpAttributeList(
+        std::move(sessHandle), mSection, sessAttributes));
+  }
+}
+
+void RsdparsaSdpMediaSection::Serialize(std::ostream& os) const {
+  os << "m=" << mMediaType << " " << GetPort();
+  if (GetPortCount()) {
+    os << "/" << GetPortCount();
+  }
+  os << " " << GetProtocol();
+  for (auto i = mFormats.begin(); i != mFormats.end(); ++i) {
+    os << " " << (*i);
+  }
+  os << CRLF;
+
+  // We dont do i=
+
+  if (mConnection) {
+    os << *mConnection;
+  }
+
+  BandwidthVec* bwVec = sdp_get_media_bandwidth_vec(mSection);
+  char* bwString = sdp_serialize_bandwidth(bwVec);
+  if (bwString) {
+    os << bwString;
+    sdp_free_string(bwString);
+  }
+
+  // We dont do k= because they're evil
+
+  os << *mAttributeList;
+}
+
+void RsdparsaSdpMediaSection::LoadFormats() {
+  RustSdpFormatType formatType = sdp_get_format_type(mSection);
+  if (formatType == RustSdpFormatType::kRustIntegers) {
+    U32Vec* vec = sdp_get_format_u32_vec(mSection);
+    size_t len = u32_vec_len(vec);
+    for (size_t i = 0; i < len; i++) {
+      uint32_t val;
+      u32_vec_get(vec, i, &val);
+      mFormats.push_back(std::to_string(val));
+    }
+  } else {
+    StringVec* vec = sdp_get_format_string_vec(mSection);
+    mFormats = convertStringVec(vec);
+  }
+}
+
+UniquePtr<SdpConnection> convertRustConnection(RustSdpConnection conn) {
+  auto address = convertExplicitlyTypedAddress(&conn.addr);
+  return MakeUnique<SdpConnection>(address.first, address.second, conn.ttl,
+                                   conn.amount);
+}
+
+void RsdparsaSdpMediaSection::LoadConnection() {
+  RustSdpConnection conn;
+  nsresult nr;
+  if (sdp_media_has_connection(mSection)) {
+    nr = sdp_get_media_connection(mSection, &conn);
+    if (NS_SUCCEEDED(nr)) {
+      mConnection = convertRustConnection(conn);
+    }
+  } else if (sdp_session_has_connection(mSession.get())) {
+    nr = sdp_get_session_connection(mSession.get(), &conn);
+    if (NS_SUCCEEDED(nr)) {
+      mConnection = convertRustConnection(conn);
+    }
+  }
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/sdp/RsdparsaSdpMediaSection.h b/dom/media/webrtc/sdp/RsdparsaSdpMediaSection.h
new file mode 100644
index 0000000000..3c193bd99c
--- /dev/null
+++ b/dom/media/webrtc/sdp/RsdparsaSdpMediaSection.h
@@ -0,0 +1,71 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _RUSTSDPMEDIASECTION_H_
+#define _RUSTSDPMEDIASECTION_H_
+
+#include "mozilla/UniquePtr.h"
+#include "sdp/RsdparsaSdpInc.h"
+#include "sdp/RsdparsaSdpGlue.h"
+#include "sdp/SdpMediaSection.h"
+#include "sdp/RsdparsaSdpAttributeList.h"
+
+namespace mozilla {
+
+class RsdparsaSdp;
+class SdpParser;
+
+class RsdparsaSdpMediaSection final : public SdpMediaSection {
+  friend class RsdparsaSdp;
+
+ public:
+  ~RsdparsaSdpMediaSection() {}
+
+  MediaType GetMediaType() const override { return mMediaType; }
+
+  unsigned int GetPort() const override;
+  void SetPort(unsigned int port) override;
+  unsigned int GetPortCount() const override;
+  Protocol GetProtocol() const override;
+  const SdpConnection& GetConnection() const override;
+  SdpConnection& GetConnection() override;
+  uint32_t GetBandwidth(const std::string& type) const override;
+  const std::vector<std::string>& GetFormats() const override;
+
+  const SdpAttributeList& GetAttributeList() const override;
+  SdpAttributeList& GetAttributeList() override;
+  SdpDirectionAttribute GetDirectionAttribute() const override;
+
+  void AddCodec(const std::string& pt, const std::string& name,
+                uint32_t clockrate, uint16_t channels) override;
+  void ClearCodecs() override;
+
+  void AddDataChannel(const std::string& name, uint16_t port, uint16_t streams,
+                      uint32_t message_size) override;
+
+  void Serialize(std::ostream&) const override;
+
+ private:
+  RsdparsaSdpMediaSection(size_t level, RsdparsaSessionHandle session,
+                          const RustMediaSection* const section,
+                          const RsdparsaSdpAttributeList* sessionLevel);
+
+  void LoadFormats();
+  void LoadConnection();
+
+  RsdparsaSessionHandle mSession;
+  const RustMediaSection* mSection;
+
+  MediaType mMediaType;
+  std::vector<std::string> mFormats;
+
+  UniquePtr<SdpConnection> mConnection;
+
+  UniquePtr<RsdparsaSdpAttributeList> mAttributeList;
+};
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/sdp/RsdparsaSdpParser.cpp b/dom/media/webrtc/sdp/RsdparsaSdpParser.cpp
new file mode 100644
index 0000000000..b955bcd46b
--- /dev/null
+++ b/dom/media/webrtc/sdp/RsdparsaSdpParser.cpp
@@ -0,0 +1,73 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "nsError.h"
+
+#include "mozilla/UniquePtr.h"
+
+#include "sdp/Sdp.h"
+#include "sdp/RsdparsaSdp.h"
+#include "sdp/RsdparsaSdpParser.h"
+#include "sdp/RsdparsaSdpInc.h"
+#include "sdp/RsdparsaSdpGlue.h"
+
+namespace mozilla {
+
+const std::string& RsdparsaSdpParser::ParserName() {
+  static const std::string& WEBRTC_SDP_NAME = "WEBRTCSDP";
+  return WEBRTC_SDP_NAME;
+}
+
+UniquePtr<SdpParser::Results> RsdparsaSdpParser::Parse(
+    const std::string& aText) {
+  UniquePtr<SdpParser::InternalResults> results(
+      new SdpParser::InternalResults(Name()));
+  RustSdpSession* result = nullptr;
+  RustSdpError* err = nullptr;
+  StringView sdpTextView{aText.c_str(), aText.length()};
+  nsresult rv = parse_sdp(sdpTextView, false, &result, &err);
+  if (rv != NS_OK) {
+    size_t line = sdp_get_error_line_num(err);
+    char* cString = sdp_get_error_message(err);
+    if (cString) {
+      std::string errMsg(cString);
+      sdp_free_error_message(cString);
+      sdp_free_error(err);
+      results->AddParseError(line, errMsg);
+    } else {
+      results->AddParseError(line, "Unable to retreive parse error.");
+    }
+    return results;
+  }
+
+  if (err) {
+    size_t line = sdp_get_error_line_num(err);
+    char* cString = sdp_get_error_message(err);
+    if (cString) {
+      std::string warningMsg(cString);
+      results->AddParseWarning(line, warningMsg);
+      sdp_free_error_message(cString);
+      sdp_free_error(err);
+    } else {
+      results->AddParseWarning(line, "Unable to retreive parse warning.");
+    }
+  }
+
+  RsdparsaSessionHandle uniqueResult(result);
+  RustSdpOrigin rustOrigin = sdp_get_origin(uniqueResult.get());
+  auto address = convertExplicitlyTypedAddress(&rustOrigin.addr);
+  SdpOrigin origin(convertStringView(rustOrigin.username), rustOrigin.sessionId,
+                   rustOrigin.sessionVersion, address.first, address.second);
+
+  results->SetSdp(MakeUnique<RsdparsaSdp>(std::move(uniqueResult), origin));
+  return results;
+}
+
+bool RsdparsaSdpParser::IsNamed(const std::string& aName) {
+  return aName == ParserName();
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/sdp/RsdparsaSdpParser.h b/dom/media/webrtc/sdp/RsdparsaSdpParser.h
new file mode 100644
index 0000000000..443aa46557
--- /dev/null
+++ b/dom/media/webrtc/sdp/RsdparsaSdpParser.h
@@ -0,0 +1,34 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _RUSTSDPPARSER_H_
+#define _RUSTSDPPARSER_H_
+
+#include <string>
+
+#include "mozilla/UniquePtr.h"
+
+#include "sdp/SdpParser.h"
+
+namespace mozilla {
+
+class RsdparsaSdpParser final : public SdpParser {
+  static const std::string& ParserName();
+
+ public:
+  RsdparsaSdpParser() = default;
+  virtual ~RsdparsaSdpParser() = default;
+
+  const std::string& Name() const override { return ParserName(); }
+
+  UniquePtr<SdpParser::Results> Parse(const std::string& text) override;
+
+  static bool IsNamed(const std::string& aName);
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/sdp/Sdp.h b/dom/media/webrtc/sdp/Sdp.h
new file mode 100644
index 0000000000..7576cac46f
--- /dev/null
+++ b/dom/media/webrtc/sdp/Sdp.h
@@ -0,0 +1,166 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+/*
+
+         ,-----.                  ,--.  ,--.
+        '  .--./ ,--,--.,--.,--.,-'  '-.`--' ,---. ,--,--,
+        |  |    ' ,-.  ||  ||  |'-.  .-',--.| .-. ||      `
+        '  '--'\\ '-'  |'  ''  '  |  |  |  |' '-' '|  ||  |
+         `-----' `--`--' `----'   `--'  `--' `---' `--''--'
+
+                        :+o+-
+                      -dNNNNNd.
+                      yNNNNNNNs
+                      :mNNNNNm-
+                       `/sso/``-://-
+                        .:+sydNNNNNNms:                      `://`
+                 `-/+shmNNNNNNNNNNNNNNNms-                  :mNNNm/
+           `-/oydmNNNNNNNNNNNNNNNNNNNNNNNNdo-              +NNNNNN+
+       .shmNNNNNNNNNNNmdyo/:dNNNNNNNNNNNNNNNNdo.         `sNNNNNm+
+       hNNNNNNNNmhs+:-`   .dNNNNNNNNNNNNNNNNNNNNh+-`    `hNNNNNm:
+       -yddyo/:.         -dNNNNm::ymNNNNNNNNNNNNNNNmdy+/dNNNNNd.
+                        :mNNNNd.   `/ymNNNNNNNNNNNNNNNNNNNNNNh`
+                       +NNNNNh`       `+hNNNNNNNNNNNNNNNNNNNs
+                      sNNNNNy`           .yNNNNNm`-/oymNNNm+
+                    `yNNNNNo              oNNNNNm`     `-.
+                   .dNNNNm/               oNNNNNm`
+                   oNNNNm:                +NNNNNm`
+                   `+yho.                 +NNNNNm`
+                                          +NNNNNNs.
+                                          `yNNNNNNmy-
+                                            -smNNNNNNh:
+                                              .smNNNNNNh/
+                                                `omNNNNNNd:
+                                                  `+dNNNNNd
+                             ````......````          /hmdy-
+            `.:/+osyhddmNNMMMMMMMMMMMMMMMMMMMMNNmddhyso+/:.`
+     `-+shmNMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMNmhs+-`
+  -smMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMds-
+ hMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMh
+ yMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMs
+  .ohNMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMNh+.
+      ./oydmMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMMmhyo:.
+             `.:/+osyyhddmmNNMMMMMMMMMMMMMMNNmmddhyyso+/:.`
+
+            ,--------.,--.     ,--.           ,--.
+            '--.  .--'|  ,---. `--' ,---.     |  | ,---.
+               |  |   |  .-.  |,--.(  .-'     |  |(  .-'
+               |  |   |  | |  ||  |.-'  `)    |  |.-'  `)
+               `--'   `--' `--'`--'`----'     `--'`----'
+                                                                ,--.
+       ,---.  ,------.  ,------.                  ,--.          |  |
+      '   .-' |  .-.  \ |  .--. ' ,--,--.,--.--.,-'  '-. ,--,--.|  |
+      `.  `-. |  |  \  :|  '--' |' ,-.  ||  .--''-.  .-'' ,-.  ||  |
+      .-'    ||  '--'  /|  | --' \ '-'  ||  |     |  |  \ '-'  |`--'
+      `-----' `-------' `--'      `--`--'`--'     `--'   `--`--'.--.
+                                                                '__'
+*/
+
+#ifndef _SDP_H_
+#define _SDP_H_
+
+#include <ostream>
+#include <vector>
+#include <sstream>
+#include "mozilla/UniquePtr.h"
+#include "mozilla/Maybe.h"
+#include "sdp/SdpMediaSection.h"
+#include "sdp/SdpAttributeList.h"
+#include "sdp/SdpEnum.h"
+
+namespace mozilla {
+
+class SdpOrigin;
+class SdpEncryptionKey;
+class SdpMediaSection;
+
+/**
+ * Base class for an SDP
+ */
+class Sdp {
+ public:
+  Sdp() = default;
+  virtual ~Sdp() = default;
+
+  virtual Sdp* Clone() const = 0;
+
+  virtual const SdpOrigin& GetOrigin() const = 0;
+  // Note: connection information is always retrieved from media sections
+  virtual uint32_t GetBandwidth(const std::string& type) const = 0;
+
+  virtual const SdpAttributeList& GetAttributeList() const = 0;
+  virtual SdpAttributeList& GetAttributeList() = 0;
+
+  virtual size_t GetMediaSectionCount() const = 0;
+  virtual const SdpMediaSection& GetMediaSection(size_t level) const = 0;
+  virtual SdpMediaSection& GetMediaSection(size_t level) = 0;
+
+  virtual SdpMediaSection& AddMediaSection(SdpMediaSection::MediaType media,
+                                           SdpDirectionAttribute::Direction dir,
+                                           uint16_t port,
+                                           SdpMediaSection::Protocol proto,
+                                           sdp::AddrType addrType,
+                                           const std::string& addr) = 0;
+
+  virtual void Serialize(std::ostream&) const = 0;
+
+  std::string ToString() const;
+};
+
+inline std::ostream& operator<<(std::ostream& os, const Sdp& sdp) {
+  sdp.Serialize(os);
+  return os;
+}
+
+inline std::string Sdp::ToString() const {
+  std::stringstream s;
+  s << *this;
+  return s.str();
+}
+
+class SdpOrigin {
+ public:
+  SdpOrigin(const std::string& username, uint64_t sessId, uint64_t sessVer,
+            sdp::AddrType addrType, const std::string& addr)
+      : mUsername(username),
+        mSessionId(sessId),
+        mSessionVersion(sessVer),
+        mAddrType(addrType),
+        mAddress(addr) {}
+
+  const std::string& GetUsername() const { return mUsername; }
+
+  uint64_t GetSessionId() const { return mSessionId; }
+
+  uint64_t GetSessionVersion() const { return mSessionVersion; }
+
+  sdp::AddrType GetAddrType() const { return mAddrType; }
+
+  const std::string& GetAddress() const { return mAddress; }
+
+  void Serialize(std::ostream& os) const {
+    sdp::NetType netType = sdp::kInternet;
+    os << "o=" << mUsername << " " << mSessionId << " " << mSessionVersion
+       << " " << netType << " " << mAddrType << " " << mAddress << "\r\n";
+  }
+
+ private:
+  std::string mUsername;
+  uint64_t mSessionId;
+  uint64_t mSessionVersion;
+  sdp::AddrType mAddrType;
+  std::string mAddress;
+};
+
+inline std::ostream& operator<<(std::ostream& os, const SdpOrigin& origin) {
+  origin.Serialize(os);
+  return os;
+}
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/sdp/SdpAttribute.cpp b/dom/media/webrtc/sdp/SdpAttribute.cpp
new file mode 100644
index 0000000000..cacb25e6b1
--- /dev/null
+++ b/dom/media/webrtc/sdp/SdpAttribute.cpp
@@ -0,0 +1,1562 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "sdp/SdpAttribute.h"
+#include "sdp/SdpHelper.h"
+#include <iomanip>
+#include <bitset>
+
+#ifdef CRLF
+#  undef CRLF
+#endif
+#define CRLF "\r\n"
+
+namespace mozilla {
+
+static unsigned char PeekChar(std::istream& is, std::string* error) {
+  int next = is.peek();
+  if (next == EOF) {
+    *error = "Truncated";
+    return 0;
+  }
+
+  return next;
+}
+
+static std::string ParseToken(std::istream& is, const std::string& delims,
+                              std::string* error) {
+  std::string token;
+  while (is) {
+    unsigned char c = PeekChar(is, error);
+    if (!c || (delims.find(c) != std::string::npos)) {
+      break;
+    }
+    token.push_back(std::tolower(is.get()));
+  }
+  return token;
+}
+
+static bool SkipChar(std::istream& is, unsigned char c, std::string* error) {
+  if (PeekChar(is, error) != c) {
+    *error = "Expected \'";
+    error->push_back(c);
+    error->push_back('\'');
+    return false;
+  }
+
+  is.get();
+  return true;
+}
+
+void SdpConnectionAttribute::Serialize(std::ostream& os) const {
+  os << "a=" << mType << ":" << mValue << CRLF;
+}
+
+void SdpDirectionAttribute::Serialize(std::ostream& os) const {
+  os << "a=" << mValue << CRLF;
+}
+
+void SdpDtlsMessageAttribute::Serialize(std::ostream& os) const {
+  os << "a=" << mType << ":" << mRole << " " << mValue << CRLF;
+}
+
+bool SdpDtlsMessageAttribute::Parse(std::istream& is, std::string* error) {
+  std::string roleToken = ParseToken(is, " ", error);
+  if (roleToken == "server") {
+    mRole = kServer;
+  } else if (roleToken == "client") {
+    mRole = kClient;
+  } else {
+    *error = "Invalid dtls-message role; must be either client or server";
+    return false;
+  }
+
+  is >> std::ws;
+
+  std::string s(std::istreambuf_iterator<char>(is), {});
+  mValue = s;
+
+  return true;
+}
+
+void SdpExtmapAttributeList::Serialize(std::ostream& os) const {
+  for (auto i = mExtmaps.begin(); i != mExtmaps.end(); ++i) {
+    os << "a=" << mType << ":" << i->entry;
+    if (i->direction_specified) {
+      os << "/" << i->direction;
+    }
+    os << " " << i->extensionname;
+    if (i->extensionattributes.length()) {
+      os << " " << i->extensionattributes;
+    }
+    os << CRLF;
+  }
+}
+
+void SdpFingerprintAttributeList::Serialize(std::ostream& os) const {
+  for (auto i = mFingerprints.begin(); i != mFingerprints.end(); ++i) {
+    os << "a=" << mType << ":" << i->hashFunc << " "
+       << FormatFingerprint(i->fingerprint) << CRLF;
+  }
+}
+
+// Format the fingerprint in RFC 4572 Section 5 attribute format
+std::string SdpFingerprintAttributeList::FormatFingerprint(
+    const std::vector<uint8_t>& fp) {
+  if (fp.empty()) {
+    MOZ_ASSERT(false, "Cannot format an empty fingerprint.");
+    return "";
+  }
+
+  std::ostringstream os;
+  for (auto i = fp.begin(); i != fp.end(); ++i) {
+    os << ":" << std::hex << std::uppercase << std::setw(2) << std::setfill('0')
+       << static_cast<uint32_t>(*i);
+  }
+  return os.str().substr(1);
+}
+
+static uint8_t FromUppercaseHex(char ch) {
+  if ((ch >= '0') && (ch <= '9')) {
+    return ch - '0';
+  }
+  if ((ch >= 'A') && (ch <= 'F')) {
+    return ch - 'A' + 10;
+  }
+  return 16;  // invalid
+}
+
+// Parse the fingerprint from RFC 4572 Section 5 attribute format
+std::vector<uint8_t> SdpFingerprintAttributeList::ParseFingerprint(
+    const std::string& str) {
+  size_t targetSize = (str.length() + 1) / 3;
+  std::vector<uint8_t> fp(targetSize);
+  size_t fpIndex = 0;
+
+  if (str.length() % 3 != 2) {
+    fp.clear();
+    return fp;
+  }
+
+  for (size_t i = 0; i < str.length(); i += 3) {
+    uint8_t high = FromUppercaseHex(str[i]);
+    uint8_t low = FromUppercaseHex(str[i + 1]);
+    if (high > 0xf || low > 0xf ||
+        (i + 2 < str.length() && str[i + 2] != ':')) {
+      fp.clear();  // error
+      return fp;
+    }
+    fp[fpIndex++] = high << 4 | low;
+  }
+  return fp;
+}
+
+bool SdpFmtpAttributeList::operator==(const SdpFmtpAttributeList& other) const {
+  return mFmtps == other.mFmtps;
+}
+
+void SdpFmtpAttributeList::Serialize(std::ostream& os) const {
+  for (auto i = mFmtps.begin(); i != mFmtps.end(); ++i) {
+    if (i->parameters) {
+      os << "a=" << mType << ":" << i->format << " ";
+      i->parameters->Serialize(os);
+      os << CRLF;
+    }
+  }
+}
+
+void SdpGroupAttributeList::Serialize(std::ostream& os) const {
+  for (auto i = mGroups.begin(); i != mGroups.end(); ++i) {
+    os << "a=" << mType << ":" << i->semantics;
+    for (auto j = i->tags.begin(); j != i->tags.end(); ++j) {
+      os << " " << (*j);
+    }
+    os << CRLF;
+  }
+}
+
+// We're just using an SdpStringAttribute for this right now
+#if 0
+void SdpIdentityAttribute::Serialize(std::ostream& os) const
+{
+  os << "a=" << mType << ":" << mAssertion;
+  for (auto i = mExtensions.begin(); i != mExtensions.end(); i++) {
+    os << (i == mExtensions.begin() ? " " : ";") << (*i);
+  }
+  os << CRLF;
+}
+#endif
+
+// Class to help with omitting a leading delimiter for the first item in a list
+class SkipFirstDelimiter {
+ public:
+  explicit SkipFirstDelimiter(const std::string& delim)
+      : mDelim(delim), mFirst(true) {}
+
+  std::ostream& print(std::ostream& os) {
+    if (!mFirst) {
+      os << mDelim;
+    }
+    mFirst = false;
+    return os;
+  }
+
+ private:
+  std::string mDelim;
+  bool mFirst;
+};
+
+static std::ostream& operator<<(std::ostream& os, SkipFirstDelimiter& delim) {
+  return delim.print(os);
+}
+
+void SdpImageattrAttributeList::XYRange::Serialize(std::ostream& os) const {
+  if (discreteValues.empty()) {
+    os << "[" << min << ":";
+    if (step != 1) {
+      os << step << ":";
+    }
+    os << max << "]";
+  } else if (discreteValues.size() == 1) {
+    os << discreteValues.front();
+  } else {
+    os << "[";
+    SkipFirstDelimiter comma(",");
+    for (auto value : discreteValues) {
+      os << comma << value;
+    }
+    os << "]";
+  }
+}
+
+template <typename T>
+bool GetUnsigned(std::istream& is, T min, T max, T* value, std::string* error) {
+  if (PeekChar(is, error) == '-') {
+    *error = "Value is less than 0";
+    return false;
+  }
+
+  is >> std::noskipws >> *value;
+
+  if (is.fail()) {
+    *error = "Malformed";
+    return false;
+  }
+
+  if (*value < min) {
+    *error = "Value too small";
+    return false;
+  }
+
+  if (*value > max) {
+    *error = "Value too large";
+    return false;
+  }
+
+  return true;
+}
+
+static bool GetXYValue(std::istream& is, uint32_t* value, std::string* error) {
+  return GetUnsigned<uint32_t>(is, 1, 999999, value, error);
+}
+
+bool SdpImageattrAttributeList::XYRange::ParseDiscreteValues(
+    std::istream& is, std::string* error) {
+  do {
+    uint32_t value;
+    if (!GetXYValue(is, &value, error)) {
+      return false;
+    }
+    discreteValues.push_back(value);
+  } while (SkipChar(is, ',', error));
+
+  return SkipChar(is, ']', error);
+}
+
+bool SdpImageattrAttributeList::XYRange::ParseAfterMin(std::istream& is,
+                                                       std::string* error) {
+  // We have already parsed "[320:", and now expect another uint
+  uint32_t value;
+  if (!GetXYValue(is, &value, error)) {
+    return false;
+  }
+
+  if (SkipChar(is, ':', error)) {
+    // Range with step eg [320:16:640]
+    step = value;
+    // Now |value| should be the max
+    if (!GetXYValue(is, &value, error)) {
+      return false;
+    }
+  }
+
+  max = value;
+  if (min >= max) {
+    *error = "Min is not smaller than max";
+    return false;
+  }
+
+  return SkipChar(is, ']', error);
+}
+
+bool SdpImageattrAttributeList::XYRange::ParseAfterBracket(std::istream& is,
+                                                           std::string* error) {
+  // Either a range, or a list of discrete values
+  // [320:640], [320:16:640], or [320,640]
+  uint32_t value;
+  if (!GetXYValue(is, &value, error)) {
+    return false;
+  }
+
+  if (SkipChar(is, ':', error)) {
+    // Range - [640:480] or [640:16:480]
+    min = value;
+    return ParseAfterMin(is, error);
+  }
+
+  if (SkipChar(is, ',', error)) {
+    discreteValues.push_back(value);
+    return ParseDiscreteValues(is, error);
+  }
+
+  *error = "Expected \':\' or \',\'";
+  return false;
+}
+
+bool SdpImageattrAttributeList::XYRange::Parse(std::istream& is,
+                                               std::string* error) {
+  if (SkipChar(is, '[', error)) {
+    return ParseAfterBracket(is, error);
+  }
+
+  // Single discrete value
+  uint32_t value;
+  if (!GetXYValue(is, &value, error)) {
+    return false;
+  }
+  discreteValues.push_back(value);
+
+  return true;
+}
+
+static bool GetSPValue(std::istream& is, float* value, std::string* error) {
+  return GetUnsigned<float>(is, 0.1f, 9.9999f, value, error);
+}
+
+static bool GetQValue(std::istream& is, float* value, std::string* error) {
+  return GetUnsigned<float>(is, 0.0f, 1.0f, value, error);
+}
+
+bool SdpImageattrAttributeList::SRange::ParseDiscreteValues(
+    std::istream& is, std::string* error) {
+  do {
+    float value;
+    if (!GetSPValue(is, &value, error)) {
+      return false;
+    }
+    discreteValues.push_back(value);
+  } while (SkipChar(is, ',', error));
+
+  return SkipChar(is, ']', error);
+}
+
+bool SdpImageattrAttributeList::SRange::ParseAfterMin(std::istream& is,
+                                                      std::string* error) {
+  if (!GetSPValue(is, &max, error)) {
+    return false;
+  }
+
+  if (min >= max) {
+    *error = "Min is not smaller than max";
+    return false;
+  }
+
+  return SkipChar(is, ']', error);
+}
+
+bool SdpImageattrAttributeList::SRange::ParseAfterBracket(std::istream& is,
+                                                          std::string* error) {
+  // Either a range, or a list of discrete values
+  float value;
+  if (!GetSPValue(is, &value, error)) {
+    return false;
+  }
+
+  if (SkipChar(is, '-', error)) {
+    min = value;
+    return ParseAfterMin(is, error);
+  }
+
+  if (SkipChar(is, ',', error)) {
+    discreteValues.push_back(value);
+    return ParseDiscreteValues(is, error);
+  }
+
+  *error = "Expected either \'-\' or \',\'";
+  return false;
+}
+
+bool SdpImageattrAttributeList::SRange::Parse(std::istream& is,
+                                              std::string* error) {
+  if (SkipChar(is, '[', error)) {
+    return ParseAfterBracket(is, error);
+  }
+
+  // Single discrete value
+  float value;
+  if (!GetSPValue(is, &value, error)) {
+    return false;
+  }
+  discreteValues.push_back(value);
+  return true;
+}
+
+bool SdpImageattrAttributeList::PRange::Parse(std::istream& is,
+                                              std::string* error) {
+  if (!SkipChar(is, '[', error)) {
+    return false;
+  }
+
+  if (!GetSPValue(is, &min, error)) {
+    return false;
+  }
+
+  if (!SkipChar(is, '-', error)) {
+    return false;
+  }
+
+  if (!GetSPValue(is, &max, error)) {
+    return false;
+  }
+
+  if (min >= max) {
+    *error = "min must be smaller than max";
+    return false;
+  }
+
+  if (!SkipChar(is, ']', error)) {
+    return false;
+  }
+  return true;
+}
+
+void SdpImageattrAttributeList::SRange::Serialize(std::ostream& os) const {
+  os << std::setprecision(4) << std::fixed;
+  if (discreteValues.empty()) {
+    os << "[" << min << "-" << max << "]";
+  } else if (discreteValues.size() == 1) {
+    os << discreteValues.front();
+  } else {
+    os << "[";
+    SkipFirstDelimiter comma(",");
+    for (auto value : discreteValues) {
+      os << comma << value;
+    }
+    os << "]";
+  }
+}
+
+void SdpImageattrAttributeList::PRange::Serialize(std::ostream& os) const {
+  os << std::setprecision(4) << std::fixed;
+  os << "[" << min << "-" << max << "]";
+}
+
+static std::string ParseKey(std::istream& is, std::string* error) {
+  std::string token = ParseToken(is, "=", error);
+  if (!SkipChar(is, '=', error)) {
+    return "";
+  }
+  return token;
+}
+
+static bool SkipBraces(std::istream& is, std::string* error) {
+  if (PeekChar(is, error) != '[') {
+    *error = "Expected \'[\'";
+    return false;
+  }
+
+  size_t braceCount = 0;
+  do {
+    switch (PeekChar(is, error)) {
+      case '[':
+        ++braceCount;
+        break;
+      case ']':
+        --braceCount;
+        break;
+      default:
+        break;
+    }
+    is.get();
+  } while (braceCount && is);
+
+  if (!is) {
+    *error = "Expected closing brace";
+    return false;
+  }
+
+  return true;
+}
+
+// Assumptions:
+// 1. If the value contains '[' or ']', they are balanced.
+// 2. The value contains no ',' outside of brackets.
+static bool SkipValue(std::istream& is, std::string* error) {
+  while (is) {
+    switch (PeekChar(is, error)) {
+      case ',':
+      case ']':
+        return true;
+      case '[':
+        if (!SkipBraces(is, error)) {
+          return false;
+        }
+        break;
+      default:
+        is.get();
+    }
+  }
+
+  *error = "No closing \']\' on set";
+  return false;
+}
+
+bool SdpImageattrAttributeList::Set::Parse(std::istream& is,
+                                           std::string* error) {
+  if (!SkipChar(is, '[', error)) {
+    return false;
+  }
+
+  if (ParseKey(is, error) != "x") {
+    *error = "Expected x=";
+    return false;
+  }
+
+  if (!xRange.Parse(is, error)) {
+    return false;
+  }
+
+  if (!SkipChar(is, ',', error)) {
+    return false;
+  }
+
+  if (ParseKey(is, error) != "y") {
+    *error = "Expected y=";
+    return false;
+  }
+
+  if (!yRange.Parse(is, error)) {
+    return false;
+  }
+
+  qValue = 0.5f;  // default
+
+  bool gotSar = false;
+  bool gotPar = false;
+  bool gotQ = false;
+
+  while (SkipChar(is, ',', error)) {
+    std::string key = ParseKey(is, error);
+    if (key.empty()) {
+      *error = "Expected key-value";
+      return false;
+    }
+
+    if (key == "sar") {
+      if (gotSar) {
+        *error = "Extra sar parameter";
+        return false;
+      }
+      gotSar = true;
+      if (!sRange.Parse(is, error)) {
+        return false;
+      }
+    } else if (key == "par") {
+      if (gotPar) {
+        *error = "Extra par parameter";
+        return false;
+      }
+      gotPar = true;
+      if (!pRange.Parse(is, error)) {
+        return false;
+      }
+    } else if (key == "q") {
+      if (gotQ) {
+        *error = "Extra q parameter";
+        return false;
+      }
+      gotQ = true;
+      if (!GetQValue(is, &qValue, error)) {
+        return false;
+      }
+    } else {
+      if (!SkipValue(is, error)) {
+        return false;
+      }
+    }
+  }
+
+  return SkipChar(is, ']', error);
+}
+
+void SdpImageattrAttributeList::Set::Serialize(std::ostream& os) const {
+  os << "[x=";
+  xRange.Serialize(os);
+  os << ",y=";
+  yRange.Serialize(os);
+  if (sRange.IsSet()) {
+    os << ",sar=";
+    sRange.Serialize(os);
+  }
+  if (pRange.IsSet()) {
+    os << ",par=";
+    pRange.Serialize(os);
+  }
+  if (qValue >= 0) {
+    os << std::setprecision(2) << std::fixed << ",q=" << qValue;
+  }
+  os << "]";
+}
+
+bool SdpImageattrAttributeList::Imageattr::ParseSets(std::istream& is,
+                                                     std::string* error) {
+  std::string type = ParseToken(is, " \t", error);
+
+  bool* isAll = nullptr;
+  std::vector<Set>* sets = nullptr;
+
+  if (type == "send") {
+    isAll = &sendAll;
+    sets = &sendSets;
+  } else if (type == "recv") {
+    isAll = &recvAll;
+    sets = &recvSets;
+  } else {
+    *error = "Unknown type, must be either send or recv";
+    return false;
+  }
+
+  if (*isAll || !sets->empty()) {
+    *error = "Multiple send or recv set lists";
+    return false;
+  }
+
+  is >> std::ws;
+  if (SkipChar(is, '*', error)) {
+    *isAll = true;
+    return true;
+  }
+
+  do {
+    Set set;
+    if (!set.Parse(is, error)) {
+      return false;
+    }
+
+    sets->push_back(set);
+    is >> std::ws;
+  } while (PeekChar(is, error) == '[');
+
+  return true;
+}
+
+bool SdpImageattrAttributeList::Imageattr::Parse(std::istream& is,
+                                                 std::string* error) {
+  if (!SkipChar(is, '*', error)) {
+    uint16_t value;
+    if (!GetUnsigned<uint16_t>(is, 0, UINT16_MAX, &value, error)) {
+      return false;
+    }
+    pt = Some(value);
+  }
+
+  is >> std::ws;
+  if (!ParseSets(is, error)) {
+    return false;
+  }
+
+  // There might be a second one
+  is >> std::ws;
+  if (is.eof()) {
+    return true;
+  }
+
+  if (!ParseSets(is, error)) {
+    return false;
+  }
+
+  is >> std::ws;
+  if (!is.eof()) {
+    *error = "Trailing characters";
+    return false;
+  }
+
+  return true;
+}
+
+void SdpImageattrAttributeList::Imageattr::Serialize(std::ostream& os) const {
+  if (pt.isSome()) {
+    os << *pt;
+  } else {
+    os << "*";
+  }
+
+  if (sendAll) {
+    os << " send *";
+  } else if (!sendSets.empty()) {
+    os << " send";
+    for (auto& set : sendSets) {
+      os << " ";
+      set.Serialize(os);
+    }
+  }
+
+  if (recvAll) {
+    os << " recv *";
+  } else if (!recvSets.empty()) {
+    os << " recv";
+    for (auto& set : recvSets) {
+      os << " ";
+      set.Serialize(os);
+    }
+  }
+}
+
+void SdpImageattrAttributeList::Serialize(std::ostream& os) const {
+  for (auto& imageattr : mImageattrs) {
+    os << "a=" << mType << ":";
+    imageattr.Serialize(os);
+    os << CRLF;
+  }
+}
+
+bool SdpImageattrAttributeList::PushEntry(const std::string& raw,
+                                          std::string* error,
+                                          size_t* errorPos) {
+  std::istringstream is(raw);
+
+  Imageattr imageattr;
+  if (!imageattr.Parse(is, error)) {
+    is.clear();
+    *errorPos = is.tellg();
+    return false;
+  }
+
+  mImageattrs.push_back(imageattr);
+  return true;
+}
+
+void SdpMsidAttributeList::Serialize(std::ostream& os) const {
+  for (auto i = mMsids.begin(); i != mMsids.end(); ++i) {
+    os << "a=" << mType << ":" << i->identifier;
+    if (i->appdata.length()) {
+      os << " " << i->appdata;
+    }
+    os << CRLF;
+  }
+}
+
+void SdpMsidSemanticAttributeList::Serialize(std::ostream& os) const {
+  for (auto i = mMsidSemantics.begin(); i != mMsidSemantics.end(); ++i) {
+    os << "a=" << mType << ":" << i->semantic;
+    for (auto j = i->msids.begin(); j != i->msids.end(); ++j) {
+      os << " " << *j;
+    }
+    os << CRLF;
+  }
+}
+
+void SdpRemoteCandidatesAttribute::Serialize(std::ostream& os) const {
+  if (mCandidates.empty()) {
+    return;
+  }
+
+  os << "a=" << mType;
+  for (auto i = mCandidates.begin(); i != mCandidates.end(); i++) {
+    os << (i == mCandidates.begin() ? ":" : " ") << i->id << " " << i->address
+       << " " << i->port;
+  }
+  os << CRLF;
+}
+
+// Remove this function. See Bug 1469702
+bool SdpRidAttributeList::Rid::ParseParameters(std::istream& is,
+                                               std::string* error) {
+  if (!PeekChar(is, error)) {
+    // No parameters
+    return true;
+  }
+
+  do {
+    is >> std::ws;
+    std::string key = ParseKey(is, error);
+    if (key.empty()) {
+      return false;  // Illegal trailing cruft
+    }
+
+    // This allows pt= to appear anywhere, instead of only at the beginning, but
+    // this ends up being significantly less code.
+    if (key == "pt") {
+      if (!ParseFormats(is, error)) {
+        return false;
+      }
+    } else if (key == "max-width") {
+      if (!GetUnsigned<uint32_t>(is, 0, UINT32_MAX, &constraints.maxWidth,
+                                 error)) {
+        return false;
+      }
+    } else if (key == "max-height") {
+      if (!GetUnsigned<uint32_t>(is, 0, UINT32_MAX, &constraints.maxHeight,
+                                 error)) {
+        return false;
+      }
+    } else if (key == "max-fps") {
+      uint32_t maxFps;
+      if (!GetUnsigned<uint32_t>(is, 0, UINT32_MAX, &maxFps, error)) {
+        return false;
+      }
+      constraints.maxFps = Some(maxFps);
+    } else if (key == "max-fs") {
+      if (!GetUnsigned<uint32_t>(is, 0, UINT32_MAX, &constraints.maxFs,
+                                 error)) {
+        return false;
+      }
+    } else if (key == "max-br") {
+      if (!GetUnsigned<uint32_t>(is, 0, UINT32_MAX, &constraints.maxBr,
+                                 error)) {
+        return false;
+      }
+    } else if (key == "max-pps") {
+      if (!GetUnsigned<uint32_t>(is, 0, UINT32_MAX, &constraints.maxPps,
+                                 error)) {
+        return false;
+      }
+    } else if (key == "depend") {
+      if (!ParseDepend(is, error)) {
+        return false;
+      }
+    } else {
+      (void)ParseToken(is, ";", error);
+    }
+  } while (SkipChar(is, ';', error));
+  return true;
+}
+
+// Remove this function. See Bug 1469702
+bool SdpRidAttributeList::Rid::ParseDepend(std::istream& is,
+                                           std::string* error) {
+  do {
+    std::string id = ParseToken(is, ",;", error);
+    if (id.empty()) {
+      return false;
+    }
+    dependIds.push_back(id);
+  } while (SkipChar(is, ',', error));
+
+  return true;
+}
+
+// Remove this function. See Bug 1469702
+bool SdpRidAttributeList::Rid::ParseFormats(std::istream& is,
+                                            std::string* error) {
+  do {
+    uint16_t fmt;
+    if (!GetUnsigned<uint16_t>(is, 0, 127, &fmt, error)) {
+      return false;
+    }
+    formats.push_back(fmt);
+  } while (SkipChar(is, ',', error));
+
+  return true;
+}
+
+void SdpRidAttributeList::Rid::SerializeParameters(std::ostream& os) const {
+  if (!HasParameters()) {
+    return;
+  }
+
+  os << " ";
+
+  SkipFirstDelimiter semic(";");
+
+  if (!formats.empty()) {
+    os << semic << "pt=";
+    SkipFirstDelimiter comma(",");
+    for (uint16_t fmt : formats) {
+      os << comma << fmt;
+    }
+  }
+
+  if (constraints.maxWidth) {
+    os << semic << "max-width=" << constraints.maxWidth;
+  }
+
+  if (constraints.maxHeight) {
+    os << semic << "max-height=" << constraints.maxHeight;
+  }
+
+  if (constraints.maxFps) {
+    os << semic << "max-fps=" << constraints.maxFps;
+  }
+
+  if (constraints.maxFs) {
+    os << semic << "max-fs=" << constraints.maxFs;
+  }
+
+  if (constraints.maxBr) {
+    os << semic << "max-br=" << constraints.maxBr;
+  }
+
+  if (constraints.maxPps) {
+    os << semic << "max-pps=" << constraints.maxPps;
+  }
+
+  if (!dependIds.empty()) {
+    os << semic << "depend=";
+    SkipFirstDelimiter comma(",");
+    for (const std::string& id : dependIds) {
+      os << comma << id;
+    }
+  }
+}
+
+// Remove this function. See Bug 1469702
+bool SdpRidAttributeList::Rid::Parse(std::istream& is, std::string* error) {
+  id = ParseToken(is, " ", error);
+  if (!CheckRidValidity(id, error)) {
+    return false;
+  }
+
+  is >> std::ws;
+  std::string directionToken = ParseToken(is, " ", error);
+  if (directionToken == "send") {
+    direction = sdp::kSend;
+  } else if (directionToken == "recv") {
+    direction = sdp::kRecv;
+  } else {
+    *error = "Invalid direction, must be either send or recv";
+    return false;
+  }
+
+  return ParseParameters(is, error);
+}
+
+static std::bitset<256> GetAllowedRidCharacters() {
+  // From RFC 8851:
+  // rid-id            = 1*(alpha-numeric / "-" / "_")
+  std::bitset<256> result;
+  for (unsigned char c = 'a'; c <= 'z'; ++c) {
+    result.set(c);
+  }
+  for (unsigned char c = 'A'; c <= 'Z'; ++c) {
+    result.set(c);
+  }
+  for (unsigned char c = '0'; c <= '9'; ++c) {
+    result.set(c);
+  }
+  // NOTE: RFC 8851 says these are allowed, but RFC 8852 says they are not
+  // https://www.rfc-editor.org/errata/eid7132
+  // result.set('-');
+  // result.set('_');
+  return result;
+}
+
+/* static */
+bool SdpRidAttributeList::CheckRidValidity(const std::string& aRid,
+                                           std::string* aError) {
+  if (aRid.empty()) {
+    *aError = "Rid must be non-empty (according to RFC 8851)";
+    return false;
+  }
+
+  // We need to check against a maximum length, but that is nowhere
+  // specified in webrtc-pc right now.
+  if (aRid.size() > 255) {
+    *aError = "Rid can be at most 255 characters long (according to RFC 8852)";
+    return false;
+  }
+
+  // TODO: Right now, if the rid is longer than kMaxRidLength, we don't treat it
+  // as a parse error, since the grammar does not have this restriction.
+  // Instead, our JSEP code ignores rids that exceed this limit. However, there
+  // is a possibility that the IETF grammar (in RFC 8852) will change the limit
+  // from 255 to 16, in which case we will need to revise this code.
+
+  static const std::bitset<256> allowed = GetAllowedRidCharacters();
+  for (unsigned char c : aRid) {
+    if (!allowed[c]) {
+      *aError =
+          "Rid can contain only alphanumeric characters (according to RFC "
+          "8852)";
+      return false;
+    }
+  }
+
+  return true;
+}
+
+// This can be overridden if necessary
+size_t SdpRidAttributeList::kMaxRidLength = 255;
+
+void SdpRidAttributeList::Rid::Serialize(std::ostream& os) const {
+  os << id << " " << direction;
+  SerializeParameters(os);
+}
+
+bool SdpRidAttributeList::Rid::HasFormat(const std::string& format) const {
+  if (formats.empty()) {
+    return true;
+  }
+
+  uint16_t formatAsInt;
+  if (!SdpHelper::GetPtAsInt(format, &formatAsInt)) {
+    return false;
+  }
+
+  return (std::find(formats.begin(), formats.end(), formatAsInt) !=
+          formats.end());
+}
+
+void SdpRidAttributeList::Serialize(std::ostream& os) const {
+  for (const Rid& rid : mRids) {
+    os << "a=" << mType << ":";
+    rid.Serialize(os);
+    os << CRLF;
+  }
+}
+
+// Remove this function. See Bug 1469702
+bool SdpRidAttributeList::PushEntry(const std::string& raw, std::string* error,
+                                    size_t* errorPos) {
+  std::istringstream is(raw);
+
+  Rid rid;
+  if (!rid.Parse(is, error)) {
+    is.clear();
+    *errorPos = is.tellg();
+    return false;
+  }
+
+  mRids.push_back(rid);
+  return true;
+}
+
+void SdpRidAttributeList::PushEntry(const std::string& id, sdp::Direction dir,
+                                    const std::vector<uint16_t>& formats,
+                                    const EncodingConstraints& constraints,
+                                    const std::vector<std::string>& dependIds) {
+  SdpRidAttributeList::Rid rid;
+
+  rid.id = id;
+  rid.direction = dir;
+  rid.formats = formats;
+  rid.constraints = constraints;
+  rid.dependIds = dependIds;
+
+  mRids.push_back(std::move(rid));
+}
+
+void SdpRtcpAttribute::Serialize(std::ostream& os) const {
+  os << "a=" << mType << ":" << mPort;
+  if (!mAddress.empty()) {
+    os << " " << mNetType << " " << mAddrType << " " << mAddress;
+  }
+  os << CRLF;
+}
+
+const char* SdpRtcpFbAttributeList::pli = "pli";
+const char* SdpRtcpFbAttributeList::sli = "sli";
+const char* SdpRtcpFbAttributeList::rpsi = "rpsi";
+const char* SdpRtcpFbAttributeList::app = "app";
+
+const char* SdpRtcpFbAttributeList::fir = "fir";
+const char* SdpRtcpFbAttributeList::tmmbr = "tmmbr";
+const char* SdpRtcpFbAttributeList::tstr = "tstr";
+const char* SdpRtcpFbAttributeList::vbcm = "vbcm";
+
+void SdpRtcpFbAttributeList::Serialize(std::ostream& os) const {
+  for (auto i = mFeedbacks.begin(); i != mFeedbacks.end(); ++i) {
+    os << "a=" << mType << ":" << i->pt << " " << i->type;
+    if (i->parameter.length()) {
+      os << " " << i->parameter;
+      if (i->extra.length()) {
+        os << " " << i->extra;
+      }
+    }
+    os << CRLF;
+  }
+}
+
+static bool ShouldSerializeChannels(SdpRtpmapAttributeList::CodecType type) {
+  switch (type) {
+    case SdpRtpmapAttributeList::kOpus:
+    case SdpRtpmapAttributeList::kG722:
+      return true;
+    case SdpRtpmapAttributeList::kPCMU:
+    case SdpRtpmapAttributeList::kPCMA:
+    case SdpRtpmapAttributeList::kVP8:
+    case SdpRtpmapAttributeList::kVP9:
+    case SdpRtpmapAttributeList::kiLBC:
+    case SdpRtpmapAttributeList::kiSAC:
+    case SdpRtpmapAttributeList::kH264:
+    case SdpRtpmapAttributeList::kRed:
+    case SdpRtpmapAttributeList::kUlpfec:
+    case SdpRtpmapAttributeList::kTelephoneEvent:
+    case SdpRtpmapAttributeList::kRtx:
+      return false;
+    case SdpRtpmapAttributeList::kOtherCodec:
+      return true;
+  }
+  MOZ_CRASH();
+}
+
+void SdpRtpmapAttributeList::Serialize(std::ostream& os) const {
+  for (auto i = mRtpmaps.begin(); i != mRtpmaps.end(); ++i) {
+    os << "a=" << mType << ":" << i->pt << " " << i->name << "/" << i->clock;
+    if (i->channels && ShouldSerializeChannels(i->codec)) {
+      os << "/" << i->channels;
+    }
+    os << CRLF;
+  }
+}
+
+void SdpSctpmapAttributeList::Serialize(std::ostream& os) const {
+  for (auto i = mSctpmaps.begin(); i != mSctpmaps.end(); ++i) {
+    os << "a=" << mType << ":" << i->pt << " " << i->name << " " << i->streams
+       << CRLF;
+  }
+}
+
+void SdpSetupAttribute::Serialize(std::ostream& os) const {
+  os << "a=" << mType << ":" << mRole << CRLF;
+}
+
+void SdpSimulcastAttribute::Version::Serialize(std::ostream& os) const {
+  SkipFirstDelimiter comma(",");
+  for (const Encoding& choice : choices) {
+    os << comma;
+    if (choice.paused) {
+      os << '~';
+    }
+    os << choice.rid;
+  }
+}
+
+bool SdpSimulcastAttribute::Version::Parse(std::istream& is,
+                                           std::string* error) {
+  do {
+    bool paused = SkipChar(is, '~', error);
+    std::string value = ParseToken(is, ",; ", error);
+    if (value.empty()) {
+      *error = "Missing rid";
+      return false;
+    }
+    if (!SdpRidAttributeList::CheckRidValidity(value, error)) {
+      return false;
+    }
+    choices.push_back(Encoding(value, paused));
+  } while (SkipChar(is, ',', error));
+
+  return true;
+}
+
+void SdpSimulcastAttribute::Versions::Serialize(std::ostream& os) const {
+  SkipFirstDelimiter semic(";");
+  for (const Version& version : *this) {
+    if (!version.IsSet()) {
+      continue;
+    }
+    os << semic;
+    version.Serialize(os);
+  }
+}
+
+bool SdpSimulcastAttribute::Versions::Parse(std::istream& is,
+                                            std::string* error) {
+  do {
+    Version version;
+    if (!version.Parse(is, error)) {
+      return false;
+    }
+    push_back(version);
+  } while (SkipChar(is, ';', error));
+
+  return true;
+}
+
+void SdpSimulcastAttribute::Serialize(std::ostream& os) const {
+  MOZ_ASSERT(sendVersions.IsSet() || recvVersions.IsSet());
+
+  os << "a=" << mType << ":";
+
+  if (sendVersions.IsSet()) {
+    os << "send ";
+    sendVersions.Serialize(os);
+  }
+
+  if (recvVersions.IsSet()) {
+    if (sendVersions.IsSet()) {
+      os << " ";
+    }
+    os << "recv ";
+    recvVersions.Serialize(os);
+  }
+
+  os << CRLF;
+}
+
+bool SdpSimulcastAttribute::Parse(std::istream& is, std::string* error) {
+  bool gotRecv = false;
+  bool gotSend = false;
+
+  while (true) {
+    is >> std::ws;
+    std::string token = ParseToken(is, " \t", error);
+    if (token.empty()) {
+      break;
+    }
+
+    if (token == "send") {
+      if (gotSend) {
+        *error = "Already got a send list";
+        return false;
+      }
+      gotSend = true;
+
+      is >> std::ws;
+      if (!sendVersions.Parse(is, error)) {
+        return false;
+      }
+    } else if (token == "recv") {
+      if (gotRecv) {
+        *error = "Already got a recv list";
+        return false;
+      }
+      gotRecv = true;
+
+      is >> std::ws;
+      if (!recvVersions.Parse(is, error)) {
+        return false;
+      }
+    } else {
+      *error = "Type must be either 'send' or 'recv'";
+      return false;
+    }
+  }
+
+  if (!gotSend && !gotRecv) {
+    *error = "Empty simulcast attribute";
+    return false;
+  }
+
+  return true;
+}
+
+void SdpSsrcAttributeList::Serialize(std::ostream& os) const {
+  for (auto i = mSsrcs.begin(); i != mSsrcs.end(); ++i) {
+    os << "a=" << mType << ":" << i->ssrc << " " << i->attribute << CRLF;
+  }
+}
+
+void SdpSsrcGroupAttributeList::Serialize(std::ostream& os) const {
+  for (auto i = mSsrcGroups.begin(); i != mSsrcGroups.end(); ++i) {
+    os << "a=" << mType << ":" << i->semantics;
+    for (auto j = i->ssrcs.begin(); j != i->ssrcs.end(); ++j) {
+      os << " " << (*j);
+    }
+    os << CRLF;
+  }
+}
+
+void SdpMultiStringAttribute::Serialize(std::ostream& os) const {
+  for (auto i = mValues.begin(); i != mValues.end(); ++i) {
+    os << "a=" << mType << ":" << *i << CRLF;
+  }
+}
+
+void SdpOptionsAttribute::Serialize(std::ostream& os) const {
+  if (mValues.empty()) {
+    return;
+  }
+
+  os << "a=" << mType << ":";
+
+  for (auto i = mValues.begin(); i != mValues.end(); ++i) {
+    if (i != mValues.begin()) {
+      os << " ";
+    }
+    os << *i;
+  }
+  os << CRLF;
+}
+
+void SdpOptionsAttribute::Load(const std::string& value) {
+  size_t start = 0;
+  size_t end = value.find(' ');
+  while (end != std::string::npos) {
+    PushEntry(value.substr(start, end));
+    start = end + 1;
+    end = value.find(' ', start);
+  }
+  PushEntry(value.substr(start));
+}
+
+void SdpFlagAttribute::Serialize(std::ostream& os) const {
+  os << "a=" << mType << CRLF;
+}
+
+void SdpStringAttribute::Serialize(std::ostream& os) const {
+  os << "a=" << mType << ":" << mValue << CRLF;
+}
+
+void SdpNumberAttribute::Serialize(std::ostream& os) const {
+  os << "a=" << mType << ":" << mValue << CRLF;
+}
+
+bool SdpAttribute::IsAllowedAtMediaLevel(AttributeType type) {
+  switch (type) {
+    case kBundleOnlyAttribute:
+      return true;
+    case kCandidateAttribute:
+      return true;
+    case kConnectionAttribute:
+      return true;
+    case kDirectionAttribute:
+      return true;
+    case kDtlsMessageAttribute:
+      return false;
+    case kEndOfCandidatesAttribute:
+      return true;
+    case kExtmapAttribute:
+      return true;
+    case kFingerprintAttribute:
+      return true;
+    case kFmtpAttribute:
+      return true;
+    case kGroupAttribute:
+      return false;
+    case kIceLiteAttribute:
+      return false;
+    case kIceMismatchAttribute:
+      return true;
+    // RFC 5245 says this is session-level only, but
+    // draft-ietf-mmusic-ice-sip-sdp-03 updates this to allow at the media
+    // level.
+    case kIceOptionsAttribute:
+      return true;
+    case kIcePwdAttribute:
+      return true;
+    case kIceUfragAttribute:
+      return true;
+    case kIdentityAttribute:
+      return false;
+    case kImageattrAttribute:
+      return true;
+    case kLabelAttribute:
+      return true;
+    case kMaxptimeAttribute:
+      return true;
+    case kMidAttribute:
+      return true;
+    case kMsidAttribute:
+      return true;
+    case kMsidSemanticAttribute:
+      return false;
+    case kPtimeAttribute:
+      return true;
+    case kRemoteCandidatesAttribute:
+      return true;
+    case kRidAttribute:
+      return true;
+    case kRtcpAttribute:
+      return true;
+    case kRtcpFbAttribute:
+      return true;
+    case kRtcpMuxAttribute:
+      return true;
+    case kRtcpRsizeAttribute:
+      return true;
+    case kRtpmapAttribute:
+      return true;
+    case kSctpmapAttribute:
+      return true;
+    case kSetupAttribute:
+      return true;
+    case kSimulcastAttribute:
+      return true;
+    case kSsrcAttribute:
+      return true;
+    case kSsrcGroupAttribute:
+      return true;
+    case kSctpPortAttribute:
+      return true;
+    case kMaxMessageSizeAttribute:
+      return true;
+  }
+  MOZ_CRASH("Unknown attribute type");
+}
+
+bool SdpAttribute::IsAllowedAtSessionLevel(AttributeType type) {
+  switch (type) {
+    case kBundleOnlyAttribute:
+      return false;
+    case kCandidateAttribute:
+      return false;
+    case kConnectionAttribute:
+      return true;
+    case kDirectionAttribute:
+      return true;
+    case kDtlsMessageAttribute:
+      return true;
+    case kEndOfCandidatesAttribute:
+      return true;
+    case kExtmapAttribute:
+      return true;
+    case kFingerprintAttribute:
+      return true;
+    case kFmtpAttribute:
+      return false;
+    case kGroupAttribute:
+      return true;
+    case kIceLiteAttribute:
+      return true;
+    case kIceMismatchAttribute:
+      return false;
+    case kIceOptionsAttribute:
+      return true;
+    case kIcePwdAttribute:
+      return true;
+    case kIceUfragAttribute:
+      return true;
+    case kIdentityAttribute:
+      return true;
+    case kImageattrAttribute:
+      return false;
+    case kLabelAttribute:
+      return false;
+    case kMaxptimeAttribute:
+      return false;
+    case kMidAttribute:
+      return false;
+    case kMsidSemanticAttribute:
+      return true;
+    case kMsidAttribute:
+      return false;
+    case kPtimeAttribute:
+      return false;
+    case kRemoteCandidatesAttribute:
+      return false;
+    case kRidAttribute:
+      return false;
+    case kRtcpAttribute:
+      return false;
+    case kRtcpFbAttribute:
+      return false;
+    case kRtcpMuxAttribute:
+      return false;
+    case kRtcpRsizeAttribute:
+      return false;
+    case kRtpmapAttribute:
+      return false;
+    case kSctpmapAttribute:
+      return false;
+    case kSetupAttribute:
+      return true;
+    case kSimulcastAttribute:
+      return false;
+    case kSsrcAttribute:
+      return false;
+    case kSsrcGroupAttribute:
+      return false;
+    case kSctpPortAttribute:
+      return false;
+    case kMaxMessageSizeAttribute:
+      return false;
+  }
+  MOZ_CRASH("Unknown attribute type");
+}
+
+const std::string SdpAttribute::GetAttributeTypeString(AttributeType type) {
+  switch (type) {
+    case kBundleOnlyAttribute:
+      return "bundle-only";
+    case kCandidateAttribute:
+      return "candidate";
+    case kConnectionAttribute:
+      return "connection";
+    case kDtlsMessageAttribute:
+      return "dtls-message";
+    case kEndOfCandidatesAttribute:
+      return "end-of-candidates";
+    case kExtmapAttribute:
+      return "extmap";
+    case kFingerprintAttribute:
+      return "fingerprint";
+    case kFmtpAttribute:
+      return "fmtp";
+    case kGroupAttribute:
+      return "group";
+    case kIceLiteAttribute:
+      return "ice-lite";
+    case kIceMismatchAttribute:
+      return "ice-mismatch";
+    case kIceOptionsAttribute:
+      return "ice-options";
+    case kIcePwdAttribute:
+      return "ice-pwd";
+    case kIceUfragAttribute:
+      return "ice-ufrag";
+    case kIdentityAttribute:
+      return "identity";
+    case kImageattrAttribute:
+      return "imageattr";
+    case kLabelAttribute:
+      return "label";
+    case kMaxptimeAttribute:
+      return "maxptime";
+    case kMidAttribute:
+      return "mid";
+    case kMsidAttribute:
+      return "msid";
+    case kMsidSemanticAttribute:
+      return "msid-semantic";
+    case kPtimeAttribute:
+      return "ptime";
+    case kRemoteCandidatesAttribute:
+      return "remote-candidates";
+    case kRidAttribute:
+      return "rid";
+    case kRtcpAttribute:
+      return "rtcp";
+    case kRtcpFbAttribute:
+      return "rtcp-fb";
+    case kRtcpMuxAttribute:
+      return "rtcp-mux";
+    case kRtcpRsizeAttribute:
+      return "rtcp-rsize";
+    case kRtpmapAttribute:
+      return "rtpmap";
+    case kSctpmapAttribute:
+      return "sctpmap";
+    case kSetupAttribute:
+      return "setup";
+    case kSimulcastAttribute:
+      return "simulcast";
+    case kSsrcAttribute:
+      return "ssrc";
+    case kSsrcGroupAttribute:
+      return "ssrc-group";
+    case kSctpPortAttribute:
+      return "sctp-port";
+    case kMaxMessageSizeAttribute:
+      return "max-message-size";
+    case kDirectionAttribute:
+      MOZ_CRASH("kDirectionAttribute not valid here");
+  }
+  MOZ_CRASH("Unknown attribute type");
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/sdp/SdpAttribute.h b/dom/media/webrtc/sdp/SdpAttribute.h
new file mode 100644
index 0000000000..6af497ac18
--- /dev/null
+++ b/dom/media/webrtc/sdp/SdpAttribute.h
@@ -0,0 +1,1907 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _SDPATTRIBUTE_H_
+#define _SDPATTRIBUTE_H_
+
+#include <algorithm>
+#include <cctype>
+#include <vector>
+#include <ostream>
+#include <sstream>
+#include <cstring>
+#include <iomanip>
+#include <string>
+
+#include "mozilla/UniquePtr.h"
+#include "mozilla/Attributes.h"
+#include "mozilla/Assertions.h"
+#include "mozilla/Maybe.h"
+
+#include "sdp/SdpEnum.h"
+#include "common/EncodingConstraints.h"
+
+namespace mozilla {
+
+/**
+ * Base class for SDP attributes
+ */
+class SdpAttribute {
+ public:
+  enum AttributeType {
+    kFirstAttribute = 0,
+    kBundleOnlyAttribute = 0,
+    kCandidateAttribute,
+    kConnectionAttribute,
+    kDirectionAttribute,
+    kDtlsMessageAttribute,
+    kEndOfCandidatesAttribute,
+    kExtmapAttribute,
+    kFingerprintAttribute,
+    kFmtpAttribute,
+    kGroupAttribute,
+    kIceLiteAttribute,
+    kIceMismatchAttribute,
+    kIceOptionsAttribute,
+    kIcePwdAttribute,
+    kIceUfragAttribute,
+    kIdentityAttribute,
+    kImageattrAttribute,
+    kLabelAttribute,
+    kMaxptimeAttribute,
+    kMidAttribute,
+    kMsidAttribute,
+    kMsidSemanticAttribute,
+    kPtimeAttribute,
+    kRemoteCandidatesAttribute,
+    kRidAttribute,
+    kRtcpAttribute,
+    kRtcpFbAttribute,
+    kRtcpMuxAttribute,
+    kRtcpRsizeAttribute,
+    kRtpmapAttribute,
+    kSctpmapAttribute,
+    kSetupAttribute,
+    kSimulcastAttribute,
+    kSsrcAttribute,
+    kSsrcGroupAttribute,
+    kSctpPortAttribute,
+    kMaxMessageSizeAttribute,
+    kLastAttribute = kMaxMessageSizeAttribute
+  };
+
+  explicit SdpAttribute(AttributeType type) : mType(type) {}
+  virtual ~SdpAttribute() {}
+
+  virtual SdpAttribute* Clone() const = 0;
+
+  AttributeType GetType() const { return mType; }
+
+  virtual void Serialize(std::ostream&) const = 0;
+
+  static bool IsAllowedAtSessionLevel(AttributeType type);
+  static bool IsAllowedAtMediaLevel(AttributeType type);
+  static const std::string GetAttributeTypeString(AttributeType type);
+
+ protected:
+  AttributeType mType;
+};
+
+inline std::ostream& operator<<(std::ostream& os, const SdpAttribute& attr) {
+  attr.Serialize(os);
+  return os;
+}
+
+inline std::ostream& operator<<(std::ostream& os,
+                                const SdpAttribute::AttributeType type) {
+  os << SdpAttribute::GetAttributeTypeString(type);
+  return os;
+}
+
+///////////////////////////////////////////////////////////////////////////
+// a=candidate, RFC5245
+//-------------------------------------------------------------------------
+//
+// candidate-attribute   = "candidate" ":" foundation SP component-id SP
+//                          transport SP
+//                          priority SP
+//                          connection-address SP     ;from RFC 4566
+//                          port         ;port from RFC 4566
+//                          SP cand-type
+//                          [SP rel-addr]
+//                          [SP rel-port]
+//                          *(SP extension-att-name SP
+//                               extension-att-value)
+// foundation            = 1*32ice-char
+// component-id          = 1*5DIGIT
+// transport             = "UDP" / transport-extension
+// transport-extension   = token              ; from RFC 3261
+// priority              = 1*10DIGIT
+// cand-type             = "typ" SP candidate-types
+// candidate-types       = "host" / "srflx" / "prflx" / "relay" / token
+// rel-addr              = "raddr" SP connection-address
+// rel-port              = "rport" SP port
+// extension-att-name    = byte-string    ;from RFC 4566
+// extension-att-value   = byte-string
+// ice-char              = ALPHA / DIGIT / "+" / "/"
+
+// We use a SdpMultiStringAttribute for candidates
+
+///////////////////////////////////////////////////////////////////////////
+// a=connection, RFC4145
+//-------------------------------------------------------------------------
+//         connection-attr        = "a=connection:" conn-value
+//         conn-value             = "new" / "existing"
+class SdpConnectionAttribute : public SdpAttribute {
+ public:
+  enum ConnValue { kNew, kExisting };
+
+  explicit SdpConnectionAttribute(SdpConnectionAttribute::ConnValue value)
+      : SdpAttribute(kConnectionAttribute), mValue(value) {}
+
+  SdpAttribute* Clone() const override {
+    return new SdpConnectionAttribute(*this);
+  }
+
+  virtual void Serialize(std::ostream& os) const override;
+
+  ConnValue mValue;
+};
+
+inline std::ostream& operator<<(std::ostream& os,
+                                SdpConnectionAttribute::ConnValue c) {
+  switch (c) {
+    case SdpConnectionAttribute::kNew:
+      os << "new";
+      break;
+    case SdpConnectionAttribute::kExisting:
+      os << "existing";
+      break;
+    default:
+      MOZ_ASSERT(false);
+      os << "?";
+  }
+  return os;
+}
+
+///////////////////////////////////////////////////////////////////////////
+// a=sendrecv / a=sendonly / a=recvonly / a=inactive, RFC 4566
+//-------------------------------------------------------------------------
+class SdpDirectionAttribute : public SdpAttribute {
+ public:
+  enum Direction {
+    kInactive = 0,
+    kSendonly = sdp::kSend,
+    kRecvonly = sdp::kRecv,
+    kSendrecv = sdp::kSend | sdp::kRecv
+  };
+
+  explicit SdpDirectionAttribute(Direction value)
+      : SdpAttribute(kDirectionAttribute), mValue(value) {}
+
+  SdpAttribute* Clone() const override {
+    return new SdpDirectionAttribute(*this);
+  }
+
+  virtual void Serialize(std::ostream& os) const override;
+
+  Direction mValue;
+};
+
+inline std::ostream& operator<<(std::ostream& os,
+                                SdpDirectionAttribute::Direction d) {
+  switch (d) {
+    case SdpDirectionAttribute::kSendonly:
+      os << "sendonly";
+      break;
+    case SdpDirectionAttribute::kRecvonly:
+      os << "recvonly";
+      break;
+    case SdpDirectionAttribute::kSendrecv:
+      os << "sendrecv";
+      break;
+    case SdpDirectionAttribute::kInactive:
+      os << "inactive";
+      break;
+    default:
+      MOZ_ASSERT(false);
+      os << "?";
+  }
+  return os;
+}
+
+inline SdpDirectionAttribute::Direction reverse(
+    SdpDirectionAttribute::Direction d) {
+  switch (d) {
+    case SdpDirectionAttribute::Direction::kInactive:
+      return SdpDirectionAttribute::Direction::kInactive;
+    case SdpDirectionAttribute::Direction::kSendonly:
+      return SdpDirectionAttribute::Direction::kRecvonly;
+    case SdpDirectionAttribute::Direction::kRecvonly:
+      return SdpDirectionAttribute::Direction::kSendonly;
+    case SdpDirectionAttribute::Direction::kSendrecv:
+      return SdpDirectionAttribute::Direction::kSendrecv;
+  }
+  MOZ_MAKE_COMPILER_ASSUME_IS_UNREACHABLE("Invalid direction!");
+  MOZ_RELEASE_ASSERT(false);
+}
+
+inline SdpDirectionAttribute::Direction operator|(
+    SdpDirectionAttribute::Direction d1, SdpDirectionAttribute::Direction d2) {
+  return (SdpDirectionAttribute::Direction)((unsigned)d1 | (unsigned)d2);
+}
+
+inline SdpDirectionAttribute::Direction operator&(
+    SdpDirectionAttribute::Direction d1, SdpDirectionAttribute::Direction d2) {
+  return (SdpDirectionAttribute::Direction)((unsigned)d1 & (unsigned)d2);
+}
+
+inline SdpDirectionAttribute::Direction operator|=(
+    SdpDirectionAttribute::Direction& d1, SdpDirectionAttribute::Direction d2) {
+  d1 = d1 | d2;
+  return d1;
+}
+
+inline SdpDirectionAttribute::Direction operator&=(
+    SdpDirectionAttribute::Direction& d1, SdpDirectionAttribute::Direction d2) {
+  d1 = d1 & d2;
+  return d1;
+}
+
+///////////////////////////////////////////////////////////////////////////
+// a=dtls-message, draft-rescorla-dtls-in-sdp
+//-------------------------------------------------------------------------
+//   attribute               =/   dtls-message-attribute
+//
+//   dtls-message-attribute  =    "dtls-message" ":" role SP value
+//
+//   role                    =    "client" / "server"
+//
+//   value                   =    1*(ALPHA / DIGIT / "+" / "/" / "=" )
+//                                ; base64 encoded message
+class SdpDtlsMessageAttribute : public SdpAttribute {
+ public:
+  enum Role { kClient, kServer };
+
+  explicit SdpDtlsMessageAttribute(Role role, const std::string& value)
+      : SdpAttribute(kDtlsMessageAttribute), mRole(role), mValue(value) {}
+
+  // TODO: remove this, Bug 1469702
+  explicit SdpDtlsMessageAttribute(const std::string& unparsed)
+      : SdpAttribute(kDtlsMessageAttribute), mRole(kClient) {
+    std::istringstream is(unparsed);
+    std::string error;
+    // We're not really worried about errors here if we don't parse;
+    // this attribute is a pure optimization.
+    Parse(is, &error);
+  }
+
+  SdpAttribute* Clone() const override {
+    return new SdpDtlsMessageAttribute(*this);
+  }
+
+  virtual void Serialize(std::ostream& os) const override;
+
+  // TODO: remove this, Bug 1469702
+  bool Parse(std::istream& is, std::string* error);
+
+  Role mRole;
+  std::string mValue;
+};
+
+inline std::ostream& operator<<(std::ostream& os,
+                                SdpDtlsMessageAttribute::Role r) {
+  switch (r) {
+    case SdpDtlsMessageAttribute::kClient:
+      os << "client";
+      break;
+    case SdpDtlsMessageAttribute::kServer:
+      os << "server";
+      break;
+    default:
+      MOZ_ASSERT(false);
+      os << "?";
+  }
+  return os;
+}
+
+///////////////////////////////////////////////////////////////////////////
+// a=extmap, RFC5285
+//-------------------------------------------------------------------------
+// RFC5285
+//        extmap = mapentry SP extensionname [SP extensionattributes]
+//
+//        extensionname = URI
+//
+//        direction = "sendonly" / "recvonly" / "sendrecv" / "inactive"
+//
+//        mapentry = "extmap:" 1*5DIGIT ["/" direction]
+//
+//        extensionattributes = byte-string
+//
+//        URI = <Defined in RFC 3986>
+//
+//        byte-string = <Defined in RFC 4566>
+//
+//        SP = <Defined in RFC 5234>
+//
+//        DIGIT = <Defined in RFC 5234>
+class SdpExtmapAttributeList : public SdpAttribute {
+ public:
+  SdpExtmapAttributeList() : SdpAttribute(kExtmapAttribute) {}
+
+  struct Extmap {
+    uint16_t entry;
+    SdpDirectionAttribute::Direction direction;
+    bool direction_specified;
+    std::string extensionname;
+    std::string extensionattributes;
+  };
+
+  void PushEntry(uint16_t entry, SdpDirectionAttribute::Direction direction,
+                 bool direction_specified, const std::string& extensionname,
+                 const std::string& extensionattributes = "") {
+    Extmap value = {entry, direction, direction_specified, extensionname,
+                    extensionattributes};
+    mExtmaps.push_back(value);
+  }
+
+  SdpAttribute* Clone() const override {
+    return new SdpExtmapAttributeList(*this);
+  }
+
+  virtual void Serialize(std::ostream& os) const override;
+
+  std::vector<Extmap> mExtmaps;
+};
+
+///////////////////////////////////////////////////////////////////////////
+// a=fingerprint, RFC4572
+//-------------------------------------------------------------------------
+//   fingerprint-attribute  =  "fingerprint" ":" hash-func SP fingerprint
+//
+//   hash-func              =  "sha-1" / "sha-224" / "sha-256" /
+//                             "sha-384" / "sha-512" /
+//                             "md5" / "md2" / token
+//                             ; Additional hash functions can only come
+//                             ; from updates to RFC 3279
+//
+//   fingerprint            =  2UHEX *(":" 2UHEX)
+//                             ; Each byte in upper-case hex, separated
+//                             ; by colons.
+//
+//   UHEX                   =  DIGIT / %x41-46 ; A-F uppercase
+class SdpFingerprintAttributeList : public SdpAttribute {
+ public:
+  SdpFingerprintAttributeList() : SdpAttribute(kFingerprintAttribute) {}
+
+  enum HashAlgorithm {
+    kSha1,
+    kSha224,
+    kSha256,
+    kSha384,
+    kSha512,
+    kMd5,
+    kMd2,
+    kUnknownAlgorithm
+  };
+
+  struct Fingerprint {
+    HashAlgorithm hashFunc;
+    std::vector<uint8_t> fingerprint;
+  };
+
+  // For use by application programmers. Enforces that it's a known and
+  // reasonable algorithm.
+  void PushEntry(std::string algorithm_str,
+                 const std::vector<uint8_t>& fingerprint,
+                 bool enforcePlausible = true) {
+    std::transform(algorithm_str.begin(), algorithm_str.end(),
+                   algorithm_str.begin(), ::tolower);
+
+    SdpFingerprintAttributeList::HashAlgorithm algorithm =
+        SdpFingerprintAttributeList::kUnknownAlgorithm;
+
+    if (algorithm_str == "sha-1") {
+      algorithm = SdpFingerprintAttributeList::kSha1;
+    } else if (algorithm_str == "sha-224") {
+      algorithm = SdpFingerprintAttributeList::kSha224;
+    } else if (algorithm_str == "sha-256") {
+      algorithm = SdpFingerprintAttributeList::kSha256;
+    } else if (algorithm_str == "sha-384") {
+      algorithm = SdpFingerprintAttributeList::kSha384;
+    } else if (algorithm_str == "sha-512") {
+      algorithm = SdpFingerprintAttributeList::kSha512;
+    } else if (algorithm_str == "md5") {
+      algorithm = SdpFingerprintAttributeList::kMd5;
+    } else if (algorithm_str == "md2") {
+      algorithm = SdpFingerprintAttributeList::kMd2;
+    }
+
+    if ((algorithm == SdpFingerprintAttributeList::kUnknownAlgorithm) ||
+        fingerprint.empty()) {
+      if (enforcePlausible) {
+        MOZ_ASSERT(false, "Unknown fingerprint algorithm");
+      } else {
+        return;
+      }
+    }
+
+    PushEntry(algorithm, fingerprint);
+  }
+
+  void PushEntry(HashAlgorithm hashFunc,
+                 const std::vector<uint8_t>& fingerprint) {
+    Fingerprint value = {hashFunc, fingerprint};
+    mFingerprints.push_back(value);
+  }
+
+  SdpAttribute* Clone() const override {
+    return new SdpFingerprintAttributeList(*this);
+  }
+
+  virtual void Serialize(std::ostream& os) const override;
+
+  std::vector<Fingerprint> mFingerprints;
+
+  static std::string FormatFingerprint(const std::vector<uint8_t>& fp);
+  static std::vector<uint8_t> ParseFingerprint(const std::string& str);
+};
+
+inline std::ostream& operator<<(std::ostream& os,
+                                SdpFingerprintAttributeList::HashAlgorithm a) {
+  switch (a) {
+    case SdpFingerprintAttributeList::kSha1:
+      os << "sha-1";
+      break;
+    case SdpFingerprintAttributeList::kSha224:
+      os << "sha-224";
+      break;
+    case SdpFingerprintAttributeList::kSha256:
+      os << "sha-256";
+      break;
+    case SdpFingerprintAttributeList::kSha384:
+      os << "sha-384";
+      break;
+    case SdpFingerprintAttributeList::kSha512:
+      os << "sha-512";
+      break;
+    case SdpFingerprintAttributeList::kMd5:
+      os << "md5";
+      break;
+    case SdpFingerprintAttributeList::kMd2:
+      os << "md2";
+      break;
+    default:
+      MOZ_ASSERT(false);
+      os << "?";
+  }
+  return os;
+}
+
+///////////////////////////////////////////////////////////////////////////
+// a=group, RFC5888
+//-------------------------------------------------------------------------
+//         group-attribute     = "a=group:" semantics
+//                               *(SP identification-tag)
+//         semantics           = "LS" / "FID" / semantics-extension
+//         semantics-extension = token
+//         identification-tag  = token
+class SdpGroupAttributeList : public SdpAttribute {
+ public:
+  SdpGroupAttributeList() : SdpAttribute(kGroupAttribute) {}
+
+  enum Semantics {
+    kLs,     // RFC5888
+    kFid,    // RFC5888
+    kSrf,    // RFC3524
+    kAnat,   // RFC4091
+    kFec,    // RFC5956
+    kFecFr,  // RFC5956
+    kCs,     // draft-mehta-rmt-flute-sdp-05
+    kDdp,    // RFC5583
+    kDup,    // RFC7104
+    kBundle  // draft-ietf-mmusic-bundle
+  };
+
+  struct Group {
+    Semantics semantics;
+    std::vector<std::string> tags;
+  };
+
+  void PushEntry(Semantics semantics, const std::vector<std::string>& tags) {
+    Group value = {semantics, tags};
+    mGroups.push_back(value);
+  }
+
+  void RemoveMid(const std::string& mid) {
+    for (auto i = mGroups.begin(); i != mGroups.end();) {
+      auto tag = std::find(i->tags.begin(), i->tags.end(), mid);
+      if (tag != i->tags.end()) {
+        i->tags.erase(tag);
+      }
+
+      if (i->tags.empty()) {
+        i = mGroups.erase(i);
+      } else {
+        ++i;
+      }
+    }
+  }
+
+  SdpAttribute* Clone() const override {
+    return new SdpGroupAttributeList(*this);
+  }
+
+  virtual void Serialize(std::ostream& os) const override;
+
+  std::vector<Group> mGroups;
+};
+
+inline std::ostream& operator<<(std::ostream& os,
+                                SdpGroupAttributeList::Semantics s) {
+  switch (s) {
+    case SdpGroupAttributeList::kLs:
+      os << "LS";
+      break;
+    case SdpGroupAttributeList::kFid:
+      os << "FID";
+      break;
+    case SdpGroupAttributeList::kSrf:
+      os << "SRF";
+      break;
+    case SdpGroupAttributeList::kAnat:
+      os << "ANAT";
+      break;
+    case SdpGroupAttributeList::kFec:
+      os << "FEC";
+      break;
+    case SdpGroupAttributeList::kFecFr:
+      os << "FEC-FR";
+      break;
+    case SdpGroupAttributeList::kCs:
+      os << "CS";
+      break;
+    case SdpGroupAttributeList::kDdp:
+      os << "DDP";
+      break;
+    case SdpGroupAttributeList::kDup:
+      os << "DUP";
+      break;
+    case SdpGroupAttributeList::kBundle:
+      os << "BUNDLE";
+      break;
+    default:
+      MOZ_ASSERT(false);
+      os << "?";
+  }
+  return os;
+}
+
+///////////////////////////////////////////////////////////////////////////
+// a=identity, draft-ietf-rtcweb-security-arch
+//-------------------------------------------------------------------------
+//   identity-attribute  = "identity:" identity-assertion
+//                         [ SP identity-extension
+//                           *(";" [ SP ] identity-extension) ]
+//   identity-assertion  = base64
+//   base64              = 1*(ALPHA / DIGIT / "+" / "/" / "=" )
+//   identity-extension  = extension-att-name [ "=" extension-att-value ]
+//   extension-att-name  = token
+//   extension-att-value = 1*(%x01-09 / %x0b-0c / %x0e-3a / %x3c-ff)
+//                         ; byte-string from [RFC4566] omitting ";"
+
+// We're just using an SdpStringAttribute for this right now
+#if 0
+class SdpIdentityAttribute : public SdpAttribute
+{
+public:
+  explicit SdpIdentityAttribute(const std::string &assertion,
+                                const std::vector<std::string> &extensions =
+                                    std::vector<std::string>()) :
+    SdpAttribute(kIdentityAttribute),
+    mAssertion(assertion),
+    mExtensions(extensions) {}
+
+  virtual void Serialize(std::ostream& os) const override;
+
+  std::string mAssertion;
+  std::vector<std::string> mExtensions;
+}
+#endif
+
+///////////////////////////////////////////////////////////////////////////
+// a=imageattr, RFC6236
+//-------------------------------------------------------------------------
+//     image-attr = "imageattr:" PT 1*2( 1*WSP ( "send" / "recv" )
+//                                       1*WSP attr-list )
+//     PT = 1*DIGIT / "*"
+//     attr-list = ( set *(1*WSP set) ) / "*"
+//       ;  WSP and DIGIT defined in [RFC5234]
+//
+//     set= "[" "x=" xyrange "," "y=" xyrange *( "," key-value ) "]"
+//                ; x is the horizontal image size range (pixel count)
+//                ; y is the vertical image size range (pixel count)
+//
+//     key-value = ( "sar=" srange )
+//               / ( "par=" prange )
+//               / ( "q=" qvalue )
+//                ; Key-value MAY be extended with other keyword
+//                ;  parameters.
+//                ; At most, one instance each of sar, par, or q
+//                ;  is allowed in a set.
+//                ;
+//                ; sar (sample aspect ratio) is the sample aspect ratio
+//                ;  associated with the set (optional, MAY be ignored)
+//                ; par (picture aspect ratio) is the allowed
+//                ;  ratio between the display's x and y physical
+//                ;  size (optional)
+//                ; q (optional, range [0.0..1.0], default value 0.5)
+//                ;  is the preference for the given set,
+//                ;  a higher value means a higher preference
+//
+//     onetonine = "1" / "2" / "3" / "4" / "5" / "6" / "7" / "8" / "9"
+//                ; Digit between 1 and 9
+//     xyvalue = onetonine *5DIGIT
+//                ; Digit between 1 and 9 that is
+//                ; followed by 0 to 5 other digits
+//     step = xyvalue
+//     xyrange = ( "[" xyvalue ":" [ step ":" ] xyvalue "]" )
+//                ; Range between a lower and an upper value
+//                ; with an optional step, default step = 1
+//                ; The rightmost occurrence of xyvalue MUST have a
+//                ; higher value than the leftmost occurrence.
+//             / ( "[" xyvalue 1*( "," xyvalue ) "]" )
+//                ; Discrete values separated by ','
+//             / ( xyvalue )
+//                ; A single value
+//     spvalue = ( "0" "." onetonine *3DIGIT )
+//                ; Values between 0.1000 and 0.9999
+//             / ( onetonine "." 1*4DIGIT )
+//                ; Values between 1.0000 and 9.9999
+//     srange =  ( "[" spvalue 1*( "," spvalue ) "]" )
+//                ; Discrete values separated by ','.
+//                ; Each occurrence of spvalue MUST be
+//                ; greater than the previous occurrence.
+//             / ( "[" spvalue "-" spvalue "]" )
+//                ; Range between a lower and an upper level (inclusive)
+//                ; The second occurrence of spvalue MUST have a higher
+//                ; value than the first
+//             / ( spvalue )
+//                ; A single value
+//
+//     prange =  ( "[" spvalue "-" spvalue "]" )
+//                ; Range between a lower and an upper level (inclusive)
+//                ; The second occurrence of spvalue MUST have a higher
+//                ; value than the first
+//
+//     qvalue  = ( "0" "." 1*2DIGIT )
+//             / ( "1" "." 1*2("0") )
+//                ; Values between 0.00 and 1.00
+//
+//  XXX TBD -- We don't use this yet, and it's a project unto itself.
+//
+
+class SdpImageattrAttributeList : public SdpAttribute {
+ public:
+  SdpImageattrAttributeList() : SdpAttribute(kImageattrAttribute) {}
+
+  class XYRange {
+   public:
+    XYRange() : min(0), max(0), step(1) {}
+    void Serialize(std::ostream& os) const;
+    // TODO: Remove this Bug 1469702
+    bool Parse(std::istream& is, std::string* error);
+    // TODO: Remove this Bug 1469702
+    bool ParseAfterBracket(std::istream& is, std::string* error);
+    // TODO: Remove this Bug 1469702
+    bool ParseAfterMin(std::istream& is, std::string* error);
+    // TODO: Remove this Bug 1469702
+    bool ParseDiscreteValues(std::istream& is, std::string* error);
+    std::vector<uint32_t> discreteValues;
+    // min/max are used iff discreteValues is empty
+    uint32_t min;
+    uint32_t max;
+    uint32_t step;
+  };
+
+  class SRange {
+   public:
+    SRange() : min(0), max(0) {}
+    void Serialize(std::ostream& os) const;
+    // TODO: Remove this Bug 1469702
+    bool Parse(std::istream& is, std::string* error);
+    // TODO: Remove this Bug 1469702
+    bool ParseAfterBracket(std::istream& is, std::string* error);
+    // TODO: Remove this Bug 1469702
+    bool ParseAfterMin(std::istream& is, std::string* error);
+    // TODO: Remove this Bug 1469702
+    bool ParseDiscreteValues(std::istream& is, std::string* error);
+    bool IsSet() const { return !discreteValues.empty() || (min && max); }
+    std::vector<float> discreteValues;
+    // min/max are used iff discreteValues is empty
+    float min;
+    float max;
+  };
+
+  class PRange {
+   public:
+    PRange() : min(0), max(0) {}
+    void Serialize(std::ostream& os) const;
+    // TODO: Remove this Bug 1469702
+    bool Parse(std::istream& is, std::string* error);
+    bool IsSet() const { return min && max; }
+    float min;
+    float max;
+  };
+
+  class Set {
+   public:
+    Set() : qValue(-1) {}
+    void Serialize(std::ostream& os) const;
+    // TODO: Remove this Bug 1469702
+    bool Parse(std::istream& is, std::string* error);
+    XYRange xRange;
+    XYRange yRange;
+    SRange sRange;
+    PRange pRange;
+    float qValue;
+  };
+
+  class Imageattr {
+   public:
+    Imageattr() : pt(), sendAll(false), recvAll(false) {}
+    void Serialize(std::ostream& os) const;
+    // TODO: Remove this Bug 1469702
+    bool Parse(std::istream& is, std::string* error);
+    // TODO: Remove this Bug 1469702
+    bool ParseSets(std::istream& is, std::string* error);
+    // If not set, this means all payload types
+    Maybe<uint16_t> pt;
+    bool sendAll;
+    std::vector<Set> sendSets;
+    bool recvAll;
+    std::vector<Set> recvSets;
+  };
+
+  SdpAttribute* Clone() const override {
+    return new SdpImageattrAttributeList(*this);
+  }
+
+  virtual void Serialize(std::ostream& os) const override;
+
+  // TODO: Remove this Bug 1469702
+  bool PushEntry(const std::string& raw, std::string* error, size_t* errorPos);
+
+  std::vector<Imageattr> mImageattrs;
+};
+
+///////////////////////////////////////////////////////////////////////////
+// a=msid, draft-ietf-mmusic-msid
+//-------------------------------------------------------------------------
+//   msid-attr = "msid:" identifier [ SP appdata ]
+//   identifier = 1*64token-char ; see RFC 4566
+//   appdata = 1*64token-char  ; see RFC 4566
+class SdpMsidAttributeList : public SdpAttribute {
+ public:
+  SdpMsidAttributeList() : SdpAttribute(kMsidAttribute) {}
+
+  struct Msid {
+    std::string identifier;
+    std::string appdata;
+  };
+
+  void PushEntry(const std::string& identifier,
+                 const std::string& appdata = "") {
+    Msid value = {identifier, appdata};
+    mMsids.push_back(value);
+  }
+
+  SdpAttribute* Clone() const override {
+    return new SdpMsidAttributeList(*this);
+  }
+
+  virtual void Serialize(std::ostream& os) const override;
+
+  std::vector<Msid> mMsids;
+};
+
+///////////////////////////////////////////////////////////////////////////
+// a=msid-semantic, draft-ietf-mmusic-msid
+//-------------------------------------------------------------------------
+//   msid-semantic-attr = "msid-semantic:" msid-semantic msid-list
+//   msid-semantic = token ; see RFC 4566
+//   msid-list = *(" " msid-id) / " *"
+class SdpMsidSemanticAttributeList : public SdpAttribute {
+ public:
+  SdpMsidSemanticAttributeList() : SdpAttribute(kMsidSemanticAttribute) {}
+
+  struct MsidSemantic {
+    // TODO: Once we have some more of these, we might want to make an enum
+    std::string semantic;
+    std::vector<std::string> msids;
+  };
+
+  void PushEntry(const std::string& semantic,
+                 const std::vector<std::string>& msids) {
+    MsidSemantic value = {semantic, msids};
+    mMsidSemantics.push_back(value);
+  }
+
+  SdpAttribute* Clone() const override {
+    return new SdpMsidSemanticAttributeList(*this);
+  }
+
+  virtual void Serialize(std::ostream& os) const override;
+
+  std::vector<MsidSemantic> mMsidSemantics;
+};
+
+///////////////////////////////////////////////////////////////////////////
+// a=remote-candiate, RFC5245
+//-------------------------------------------------------------------------
+//   remote-candidate-att = "remote-candidates" ":" remote-candidate
+//                           0*(SP remote-candidate)
+//   remote-candidate = component-ID SP connection-address SP port
+class SdpRemoteCandidatesAttribute : public SdpAttribute {
+ public:
+  struct Candidate {
+    std::string id;
+    std::string address;
+    uint16_t port;
+  };
+
+  explicit SdpRemoteCandidatesAttribute(
+      const std::vector<Candidate>& candidates)
+      : SdpAttribute(kRemoteCandidatesAttribute), mCandidates(candidates) {}
+
+  SdpAttribute* Clone() const override {
+    return new SdpRemoteCandidatesAttribute(*this);
+  }
+
+  virtual void Serialize(std::ostream& os) const override;
+
+  std::vector<Candidate> mCandidates;
+};
+
+/*
+a=rid, draft-pthatcher-mmusic-rid-01
+
+   rid-syntax        = "a=rid:" rid-identifier SP rid-dir
+                       [ rid-pt-param-list / rid-param-list ]
+
+   rid-identifier    = 1*(alpha-numeric / "-" / "_")
+
+   rid-dir           = "send" / "recv"
+
+   rid-pt-param-list = SP rid-fmt-list *(";" rid-param)
+
+   rid-param-list    = SP rid-param *(";" rid-param)
+
+   rid-fmt-list      = "pt=" fmt *( "," fmt )
+                        ; fmt defined in {{RFC4566}}
+
+   rid-param         = rid-width-param
+                       / rid-height-param
+                       / rid-fps-param
+                       / rid-fs-param
+                       / rid-br-param
+                       / rid-pps-param
+                       / rid-depend-param
+                       / rid-param-other
+
+   rid-width-param   = "max-width" [ "=" int-param-val ]
+
+   rid-height-param  = "max-height" [ "=" int-param-val ]
+
+   rid-fps-param     = "max-fps" [ "=" int-param-val ]
+
+   rid-fs-param      = "max-fs" [ "=" int-param-val ]
+
+   rid-br-param      = "max-br" [ "=" int-param-val ]
+
+   rid-pps-param     = "max-pps" [ "=" int-param-val ]
+
+   rid-depend-param  = "depend=" rid-list
+
+   rid-param-other   = 1*(alpha-numeric / "-") [ "=" param-val ]
+
+   rid-list          = rid-identifier *( "," rid-identifier )
+
+   int-param-val     = 1*DIGIT
+
+   param-val         = *( %x20-58 / %x60-7E )
+                       ; Any printable character except semicolon
+*/
+class SdpRidAttributeList : public SdpAttribute {
+ public:
+  explicit SdpRidAttributeList() : SdpAttribute(kRidAttribute) {}
+
+  struct Rid {
+    Rid() : direction(sdp::kSend) {}
+
+    // Remove this function. See Bug 1469702
+    bool Parse(std::istream& is, std::string* error);
+    // Remove this function. See Bug 1469702
+    bool ParseParameters(std::istream& is, std::string* error);
+    // Remove this function. See Bug 1469702
+    bool ParseDepend(std::istream& is, std::string* error);
+    // Remove this function. See Bug 1469702
+    bool ParseFormats(std::istream& is, std::string* error);
+
+    void Serialize(std::ostream& os) const;
+    void SerializeParameters(std::ostream& os) const;
+    bool HasFormat(const std::string& format) const;
+    bool HasParameters() const {
+      return !formats.empty() || constraints.maxWidth ||
+             constraints.maxHeight || constraints.maxFps || constraints.maxFs ||
+             constraints.maxBr || constraints.maxPps || !dependIds.empty();
+    }
+
+    std::string id;
+    sdp::Direction direction;
+    std::vector<uint16_t> formats;  // Empty implies all
+    EncodingConstraints constraints;
+    std::vector<std::string> dependIds;
+  };
+
+  SdpAttribute* Clone() const override {
+    return new SdpRidAttributeList(*this);
+  }
+
+  static bool CheckRidValidity(const std::string& aRid, std::string* aError);
+  static size_t kMaxRidLength;
+
+  virtual void Serialize(std::ostream& os) const override;
+
+  // Remove this function. See Bug 1469702
+  bool PushEntry(const std::string& raw, std::string* error, size_t* errorPos);
+
+  void PushEntry(const std::string& id, sdp::Direction dir,
+                 const std::vector<uint16_t>& formats,
+                 const EncodingConstraints& constraints,
+                 const std::vector<std::string>& dependIds);
+
+  std::vector<Rid> mRids;
+};
+
+///////////////////////////////////////////////////////////////////////////
+// a=rtcp, RFC3605
+//-------------------------------------------------------------------------
+//   rtcp-attribute =  "a=rtcp:" port  [nettype space addrtype space
+//                         connection-address] CRLF
+class SdpRtcpAttribute : public SdpAttribute {
+ public:
+  explicit SdpRtcpAttribute(uint16_t port)
+      : SdpAttribute(kRtcpAttribute),
+        mPort(port),
+        mNetType(sdp::kNetTypeNone),
+        mAddrType(sdp::kAddrTypeNone) {}
+
+  SdpRtcpAttribute(uint16_t port, sdp::NetType netType, sdp::AddrType addrType,
+                   const std::string& address)
+      : SdpAttribute(kRtcpAttribute),
+        mPort(port),
+        mNetType(netType),
+        mAddrType(addrType),
+        mAddress(address) {
+    MOZ_ASSERT(netType != sdp::kNetTypeNone);
+    MOZ_ASSERT(addrType != sdp::kAddrTypeNone);
+    MOZ_ASSERT(!address.empty());
+  }
+
+  SdpAttribute* Clone() const override { return new SdpRtcpAttribute(*this); }
+
+  virtual void Serialize(std::ostream& os) const override;
+
+  uint16_t mPort;
+  sdp::NetType mNetType;
+  sdp::AddrType mAddrType;
+  std::string mAddress;
+};
+
+///////////////////////////////////////////////////////////////////////////
+// a=rtcp-fb, RFC4585
+//-------------------------------------------------------------------------
+//    rtcp-fb-syntax = "a=rtcp-fb:" rtcp-fb-pt SP rtcp-fb-val CRLF
+//
+//    rtcp-fb-pt         = "*"   ; wildcard: applies to all formats
+//                       / fmt   ; as defined in SDP spec
+//
+//    rtcp-fb-val        = "ack" rtcp-fb-ack-param
+//                       / "nack" rtcp-fb-nack-param
+//                       / "trr-int" SP 1*DIGIT
+//                       / rtcp-fb-id rtcp-fb-param
+//
+//    rtcp-fb-id         = 1*(alpha-numeric / "-" / "_")
+//
+//    rtcp-fb-param      = SP "app" [SP byte-string]
+//                       / SP token [SP byte-string]
+//                       / ; empty
+//
+//    rtcp-fb-ack-param  = SP "rpsi"
+//                       / SP "app" [SP byte-string]
+//                       / SP token [SP byte-string]
+//                       / ; empty
+//
+//    rtcp-fb-nack-param = SP "pli"
+//                       / SP "sli"
+//                       / SP "rpsi"
+//                       / SP "app" [SP byte-string]
+//                       / SP token [SP byte-string]
+//                       / ; empty
+//
+class SdpRtcpFbAttributeList : public SdpAttribute {
+ public:
+  SdpRtcpFbAttributeList() : SdpAttribute(kRtcpFbAttribute) {}
+
+  enum Type { kAck, kApp, kCcm, kNack, kTrrInt, kRemb, kTransportCC };
+
+  static const char* pli;
+  static const char* sli;
+  static const char* rpsi;
+  static const char* app;
+
+  static const char* fir;
+  static const char* tmmbr;
+  static const char* tstr;
+  static const char* vbcm;
+
+  struct Feedback {
+    std::string pt;
+    Type type;
+    std::string parameter;
+    std::string extra;
+    // TODO(bug 1744307): Use =default here once it is supported
+    bool operator==(const Feedback& aOther) const {
+      return pt == aOther.pt && type == aOther.type &&
+             parameter == aOther.parameter && extra == aOther.extra;
+    }
+  };
+
+  void PushEntry(const std::string& pt, Type type,
+                 const std::string& parameter = "",
+                 const std::string& extra = "") {
+    Feedback value = {pt, type, parameter, extra};
+    mFeedbacks.push_back(value);
+  }
+
+  SdpAttribute* Clone() const override {
+    return new SdpRtcpFbAttributeList(*this);
+  }
+
+  virtual void Serialize(std::ostream& os) const override;
+
+  std::vector<Feedback> mFeedbacks;
+};
+
+inline std::ostream& operator<<(std::ostream& os,
+                                SdpRtcpFbAttributeList::Type type) {
+  switch (type) {
+    case SdpRtcpFbAttributeList::kAck:
+      os << "ack";
+      break;
+    case SdpRtcpFbAttributeList::kApp:
+      os << "app";
+      break;
+    case SdpRtcpFbAttributeList::kCcm:
+      os << "ccm";
+      break;
+    case SdpRtcpFbAttributeList::kNack:
+      os << "nack";
+      break;
+    case SdpRtcpFbAttributeList::kTrrInt:
+      os << "trr-int";
+      break;
+    case SdpRtcpFbAttributeList::kRemb:
+      os << "goog-remb";
+      break;
+    case SdpRtcpFbAttributeList::kTransportCC:
+      os << "transport-cc";
+      break;
+    default:
+      MOZ_ASSERT(false);
+      os << "?";
+  }
+  return os;
+}
+
+///////////////////////////////////////////////////////////////////////////
+// a=rtpmap, RFC4566
+//-------------------------------------------------------------------------
+// a=rtpmap:<payload type> <encoding name>/<clock rate> [/<encoding parameters>]
+class SdpRtpmapAttributeList : public SdpAttribute {
+ public:
+  SdpRtpmapAttributeList() : SdpAttribute(kRtpmapAttribute) {}
+
+  // Minimal set to get going
+  enum CodecType {
+    kOpus,
+    kG722,
+    kPCMU,
+    kPCMA,
+    kVP8,
+    kVP9,
+    kiLBC,
+    kiSAC,
+    kH264,
+    kRed,
+    kUlpfec,
+    kTelephoneEvent,
+    kRtx,
+    kOtherCodec
+  };
+
+  struct Rtpmap {
+    std::string pt;
+    CodecType codec;
+    std::string name;
+    uint32_t clock;
+    // Technically, this could mean something else in the future.
+    // In practice, that's probably not going to happen.
+    uint32_t channels;
+  };
+
+  void PushEntry(const std::string& pt, CodecType codec,
+                 const std::string& name, uint32_t clock,
+                 uint32_t channels = 0) {
+    Rtpmap value = {pt, codec, name, clock, channels};
+    mRtpmaps.push_back(value);
+  }
+
+  SdpAttribute* Clone() const override {
+    return new SdpRtpmapAttributeList(*this);
+  }
+
+  virtual void Serialize(std::ostream& os) const override;
+
+  bool HasEntry(const std::string& pt) const {
+    for (auto it = mRtpmaps.begin(); it != mRtpmaps.end(); ++it) {
+      if (it->pt == pt) {
+        return true;
+      }
+    }
+    return false;
+  }
+
+  const Rtpmap& GetEntry(const std::string& pt) const {
+    for (auto it = mRtpmaps.begin(); it != mRtpmaps.end(); ++it) {
+      if (it->pt == pt) {
+        return *it;
+      }
+    }
+    MOZ_CRASH();
+  }
+
+  std::vector<Rtpmap> mRtpmaps;
+};
+
+inline std::ostream& operator<<(std::ostream& os,
+                                SdpRtpmapAttributeList::CodecType c) {
+  switch (c) {
+    case SdpRtpmapAttributeList::kOpus:
+      os << "opus";
+      break;
+    case SdpRtpmapAttributeList::kG722:
+      os << "G722";
+      break;
+    case SdpRtpmapAttributeList::kPCMU:
+      os << "PCMU";
+      break;
+    case SdpRtpmapAttributeList::kPCMA:
+      os << "PCMA";
+      break;
+    case SdpRtpmapAttributeList::kVP8:
+      os << "VP8";
+      break;
+    case SdpRtpmapAttributeList::kVP9:
+      os << "VP9";
+      break;
+    case SdpRtpmapAttributeList::kiLBC:
+      os << "iLBC";
+      break;
+    case SdpRtpmapAttributeList::kiSAC:
+      os << "iSAC";
+      break;
+    case SdpRtpmapAttributeList::kH264:
+      os << "H264";
+      break;
+    case SdpRtpmapAttributeList::kRed:
+      os << "red";
+      break;
+    case SdpRtpmapAttributeList::kUlpfec:
+      os << "ulpfec";
+      break;
+    case SdpRtpmapAttributeList::kTelephoneEvent:
+      os << "telephone-event";
+      break;
+    case SdpRtpmapAttributeList::kRtx:
+      os << "rtx";
+      break;
+    default:
+      MOZ_ASSERT(false);
+      os << "?";
+  }
+  return os;
+}
+
+///////////////////////////////////////////////////////////////////////////
+// a=fmtp, RFC4566, RFC5576
+//-------------------------------------------------------------------------
+//       a=fmtp:<format> <format specific parameters>
+//
+class SdpFmtpAttributeList : public SdpAttribute {
+ public:
+  SdpFmtpAttributeList() : SdpAttribute(kFmtpAttribute) {}
+
+  // Base class for format parameters
+  class Parameters {
+   public:
+    explicit Parameters(SdpRtpmapAttributeList::CodecType aCodec)
+        : codec_type(aCodec) {}
+
+    virtual ~Parameters() {}
+    virtual Parameters* Clone() const = 0;
+    virtual void Serialize(std::ostream& os) const = 0;
+    virtual bool CompareEq(const Parameters& other) const = 0;
+
+    bool operator==(const Parameters& other) const {
+      return codec_type == other.codec_type && CompareEq(other);
+    }
+
+    SdpRtpmapAttributeList::CodecType codec_type;
+  };
+
+  class RedParameters : public Parameters {
+   public:
+    RedParameters() : Parameters(SdpRtpmapAttributeList::kRed) {}
+
+    virtual Parameters* Clone() const override {
+      return new RedParameters(*this);
+    }
+
+    virtual void Serialize(std::ostream& os) const override {
+      for (size_t i = 0; i < encodings.size(); ++i) {
+        os << (i != 0 ? "/" : "") << std::to_string(encodings[i]);
+      }
+    }
+
+    virtual bool CompareEq(const Parameters& other) const override {
+      return encodings == static_cast<const RedParameters&>(other).encodings;
+    }
+
+    std::vector<uint8_t> encodings;
+  };
+
+  class RtxParameters : public Parameters {
+   public:
+    uint8_t apt = 255;  // Valid payload types are 0 - 127, use 255 to represent
+                        // unset value.
+    Maybe<uint32_t> rtx_time;
+
+    RtxParameters() : Parameters(SdpRtpmapAttributeList::kRtx) {}
+
+    virtual ~RtxParameters() {}
+
+    virtual Parameters* Clone() const override {
+      return new RtxParameters(*this);
+    }
+
+    virtual void Serialize(std::ostream& os) const override {
+      if (apt <= 127) {
+        os << "apt=" << static_cast<uint32_t>(apt);
+        rtx_time.apply([&](const auto& time) { os << ";rtx-time=" << time; });
+      }
+    }
+
+    virtual bool CompareEq(const Parameters& aOther) const override {
+      if (aOther.codec_type != codec_type) {
+        return false;
+      }
+      auto other = static_cast<const RtxParameters&>(aOther);
+      return other.apt == apt && other.rtx_time == rtx_time;
+    }
+  };
+
+  class H264Parameters : public Parameters {
+   public:
+    static const uint32_t kDefaultProfileLevelId = 0x420010;
+
+    H264Parameters()
+        : Parameters(SdpRtpmapAttributeList::kH264),
+          packetization_mode(0),
+          level_asymmetry_allowed(false),
+          profile_level_id(kDefaultProfileLevelId),
+          max_mbps(0),
+          max_fs(0),
+          max_cpb(0),
+          max_dpb(0),
+          max_br(0) {
+      memset(sprop_parameter_sets, 0, sizeof(sprop_parameter_sets));
+    }
+
+    virtual Parameters* Clone() const override {
+      return new H264Parameters(*this);
+    }
+
+    virtual void Serialize(std::ostream& os) const override {
+      // Note: don't move this, since having an unconditional param up top
+      // lets us avoid a whole bunch of conditional streaming of ';' below
+      os << "profile-level-id=" << std::hex << std::setfill('0') << std::setw(6)
+         << profile_level_id << std::dec << std::setfill(' ');
+
+      os << ";level-asymmetry-allowed=" << (level_asymmetry_allowed ? 1 : 0);
+
+      if (strlen(sprop_parameter_sets)) {
+        os << ";sprop-parameter-sets=" << sprop_parameter_sets;
+      }
+
+      if (packetization_mode != 0) {
+        os << ";packetization-mode=" << packetization_mode;
+      }
+
+      if (max_mbps != 0) {
+        os << ";max-mbps=" << max_mbps;
+      }
+
+      if (max_fs != 0) {
+        os << ";max-fs=" << max_fs;
+      }
+
+      if (max_cpb != 0) {
+        os << ";max-cpb=" << max_cpb;
+      }
+
+      if (max_dpb != 0) {
+        os << ";max-dpb=" << max_dpb;
+      }
+
+      if (max_br != 0) {
+        os << ";max-br=" << max_br;
+      }
+    }
+
+    virtual bool CompareEq(const Parameters& other) const override {
+      const auto& otherH264 = static_cast<const H264Parameters&>(other);
+
+      // sprop is not comapred here as it does not get parsed in the rsdparsa
+      return packetization_mode == otherH264.packetization_mode &&
+             level_asymmetry_allowed == otherH264.level_asymmetry_allowed &&
+             profile_level_id == otherH264.profile_level_id &&
+             max_mbps == otherH264.max_mbps && max_fs == otherH264.max_fs &&
+             max_cpb == otherH264.max_cpb && max_dpb == otherH264.max_dpb &&
+             max_br == otherH264.max_br;
+    }
+
+    static const size_t max_sprop_len = 128;
+    char sprop_parameter_sets[max_sprop_len];
+    unsigned int packetization_mode;
+    bool level_asymmetry_allowed;
+    unsigned int profile_level_id;
+    unsigned int max_mbps;
+    unsigned int max_fs;
+    unsigned int max_cpb;
+    unsigned int max_dpb;
+    unsigned int max_br;
+  };
+
+  // Also used for VP9 since they share parameters
+  class VP8Parameters : public Parameters {
+   public:
+    explicit VP8Parameters(SdpRtpmapAttributeList::CodecType type)
+        : Parameters(type), max_fs(0), max_fr(0) {}
+
+    virtual Parameters* Clone() const override {
+      return new VP8Parameters(*this);
+    }
+
+    virtual void Serialize(std::ostream& os) const override {
+      // draft-ietf-payload-vp8-11 says these are mandatory, upper layer
+      // needs to ensure they're set properly.
+      os << "max-fs=" << max_fs;
+      os << ";max-fr=" << max_fr;
+    }
+
+    virtual bool CompareEq(const Parameters& other) const override {
+      const auto& otherVP8 = static_cast<const VP8Parameters&>(other);
+
+      return max_fs == otherVP8.max_fs && max_fr == otherVP8.max_fr;
+    }
+
+    unsigned int max_fs;
+    unsigned int max_fr;
+  };
+
+  class OpusParameters : public Parameters {
+   public:
+    enum {
+      kDefaultMaxPlaybackRate = 48000,
+      kDefaultStereo = 0,
+      kDefaultUseInBandFec = 0,
+      kDefaultMaxAverageBitrate = 0,
+      kDefaultUseDTX = 0,
+      kDefaultFrameSize = 0,
+      kDefaultMinFrameSize = 0,
+      kDefaultMaxFrameSize = 0,
+      kDefaultUseCbr = 0
+    };
+    OpusParameters()
+        : Parameters(SdpRtpmapAttributeList::kOpus),
+          maxplaybackrate(kDefaultMaxPlaybackRate),
+          stereo(kDefaultStereo),
+          useInBandFec(kDefaultUseInBandFec),
+          maxAverageBitrate(kDefaultMaxAverageBitrate),
+          useDTX(kDefaultUseDTX),
+          frameSizeMs(kDefaultFrameSize),
+          minFrameSizeMs(kDefaultMinFrameSize),
+          maxFrameSizeMs(kDefaultMaxFrameSize),
+          useCbr(kDefaultUseCbr) {}
+
+    Parameters* Clone() const override { return new OpusParameters(*this); }
+
+    void Serialize(std::ostream& os) const override {
+      os << "maxplaybackrate=" << maxplaybackrate << ";stereo=" << stereo
+         << ";useinbandfec=" << useInBandFec;
+
+      if (useDTX) {
+        os << ";usedtx=1";
+      }
+      if (maxAverageBitrate) {
+        os << ";maxaveragebitrate=" << maxAverageBitrate;
+      }
+      if (frameSizeMs) {
+        os << ";ptime=" << frameSizeMs;
+      }
+      if (minFrameSizeMs) {
+        os << ";minptime=" << minFrameSizeMs;
+      }
+      if (maxFrameSizeMs) {
+        os << ";maxptime=" << maxFrameSizeMs;
+      }
+      if (useCbr) {
+        os << ";cbr=1";
+      }
+    }
+
+    virtual bool CompareEq(const Parameters& other) const override {
+      const auto& otherOpus = static_cast<const OpusParameters&>(other);
+
+      bool maxplaybackrateIsEq = (maxplaybackrate == otherOpus.maxplaybackrate);
+
+      // This is due to a bug in sipcc that causes maxplaybackrate to
+      // always be 0 if it appears in the fmtp
+      if (((maxplaybackrate == 0) && (otherOpus.maxplaybackrate != 0)) ||
+          ((maxplaybackrate != 0) && (otherOpus.maxplaybackrate == 0))) {
+        maxplaybackrateIsEq = true;
+      }
+
+      return maxplaybackrateIsEq && stereo == otherOpus.stereo &&
+             useInBandFec == otherOpus.useInBandFec &&
+             maxAverageBitrate == otherOpus.maxAverageBitrate &&
+             useDTX == otherOpus.useDTX &&
+             frameSizeMs == otherOpus.frameSizeMs &&
+             minFrameSizeMs == otherOpus.minFrameSizeMs &&
+             maxFrameSizeMs == otherOpus.maxFrameSizeMs &&
+             useCbr == otherOpus.useCbr;
+    }
+
+    unsigned int maxplaybackrate;
+    unsigned int stereo;
+    unsigned int useInBandFec;
+    uint32_t maxAverageBitrate;
+    bool useDTX;
+    uint32_t frameSizeMs;
+    uint32_t minFrameSizeMs;
+    uint32_t maxFrameSizeMs;
+    bool useCbr;
+  };
+
+  class TelephoneEventParameters : public Parameters {
+   public:
+    TelephoneEventParameters()
+        : Parameters(SdpRtpmapAttributeList::kTelephoneEvent),
+          dtmfTones("0-15") {}
+
+    virtual Parameters* Clone() const override {
+      return new TelephoneEventParameters(*this);
+    }
+
+    void Serialize(std::ostream& os) const override { os << dtmfTones; }
+
+    virtual bool CompareEq(const Parameters& other) const override {
+      return dtmfTones ==
+             static_cast<const TelephoneEventParameters&>(other).dtmfTones;
+    }
+
+    std::string dtmfTones;
+  };
+
+  class Fmtp {
+   public:
+    Fmtp(const std::string& aFormat, const Parameters& aParameters)
+        : format(aFormat), parameters(aParameters.Clone()) {}
+
+    // TODO: Rip all of this out when we have move semantics in the stl.
+    Fmtp(const Fmtp& orig) { *this = orig; }
+
+    Fmtp& operator=(const Fmtp& rhs) {
+      if (this != &rhs) {
+        format = rhs.format;
+        parameters.reset(rhs.parameters ? rhs.parameters->Clone() : nullptr);
+      }
+      return *this;
+    }
+
+    bool operator==(const Fmtp& other) const {
+      return format == other.format && *parameters == *other.parameters;
+    }
+
+    // The contract around these is as follows:
+    // * |parameters| is only set if we recognized the media type and had
+    //   a subclass of Parameters to represent that type of parameters
+    // * |parameters| is a best-effort representation; it might be missing
+    //   stuff
+    // * Parameters::codec_type tells you the concrete class, eg
+    //   kH264 -> H264Parameters
+    std::string format;
+    UniquePtr<Parameters> parameters;
+  };
+
+  bool operator==(const SdpFmtpAttributeList& other) const;
+
+  SdpAttribute* Clone() const override {
+    return new SdpFmtpAttributeList(*this);
+  }
+
+  virtual void Serialize(std::ostream& os) const override;
+
+  void PushEntry(const std::string& format, const Parameters& parameters) {
+    mFmtps.push_back(Fmtp(format, parameters));
+  }
+
+  std::vector<Fmtp> mFmtps;
+};
+
+///////////////////////////////////////////////////////////////////////////
+// a=sctpmap, draft-ietf-mmusic-sctp-sdp-05
+//-------------------------------------------------------------------------
+//      sctpmap-attr        =  "a=sctpmap:" sctpmap-number media-subtypes
+// [streams]
+//      sctpmap-number      =  1*DIGIT
+//      protocol            =  labelstring
+//        labelstring         =  text
+//        text                =  byte-string
+//      streams      =  1*DIGIT
+//
+// We're going to pretend that there are spaces where they make sense.
+class SdpSctpmapAttributeList : public SdpAttribute {
+ public:
+  SdpSctpmapAttributeList() : SdpAttribute(kSctpmapAttribute) {}
+
+  struct Sctpmap {
+    std::string pt;
+    std::string name;
+    uint32_t streams;
+  };
+
+  void PushEntry(const std::string& pt, const std::string& name,
+                 uint32_t streams = 0) {
+    Sctpmap value = {pt, name, streams};
+    mSctpmaps.push_back(value);
+  }
+
+  SdpAttribute* Clone() const override {
+    return new SdpSctpmapAttributeList(*this);
+  }
+
+  virtual void Serialize(std::ostream& os) const override;
+
+  bool HasEntry(const std::string& pt) const {
+    for (auto it = mSctpmaps.begin(); it != mSctpmaps.end(); ++it) {
+      if (it->pt == pt) {
+        return true;
+      }
+    }
+    return false;
+  }
+
+  const Sctpmap& GetFirstEntry() const { return mSctpmaps[0]; }
+
+  std::vector<Sctpmap> mSctpmaps;
+};
+
+///////////////////////////////////////////////////////////////////////////
+// a=setup, RFC4145
+//-------------------------------------------------------------------------
+//       setup-attr           =  "a=setup:" role
+//       role                 =  "active" / "passive" / "actpass" / "holdconn"
+class SdpSetupAttribute : public SdpAttribute {
+ public:
+  enum Role { kActive, kPassive, kActpass, kHoldconn };
+
+  explicit SdpSetupAttribute(Role role)
+      : SdpAttribute(kSetupAttribute), mRole(role) {}
+
+  SdpAttribute* Clone() const override { return new SdpSetupAttribute(*this); }
+
+  virtual void Serialize(std::ostream& os) const override;
+
+  Role mRole;
+};
+
+inline std::ostream& operator<<(std::ostream& os, SdpSetupAttribute::Role r) {
+  switch (r) {
+    case SdpSetupAttribute::kActive:
+      os << "active";
+      break;
+    case SdpSetupAttribute::kPassive:
+      os << "passive";
+      break;
+    case SdpSetupAttribute::kActpass:
+      os << "actpass";
+      break;
+    case SdpSetupAttribute::kHoldconn:
+      os << "holdconn";
+      break;
+    default:
+      MOZ_ASSERT(false);
+      os << "?";
+  }
+  return os;
+}
+
+// Old draft-04
+// sc-attr     = "a=simulcast:" 1*2( WSP sc-str-list ) [WSP sc-pause-list]
+// sc-str-list = sc-dir WSP sc-id-type "=" sc-alt-list *( ";" sc-alt-list )
+// sc-pause-list = "paused=" sc-alt-list
+// sc-dir      = "send" / "recv"
+// sc-id-type  = "pt" / "rid" / token
+// sc-alt-list = sc-id *( "," sc-id )
+// sc-id       = fmt / rid-identifier / token
+// ; WSP defined in [RFC5234]
+// ; fmt, token defined in [RFC4566]
+// ; rid-identifier defined in [I-D.pthatcher-mmusic-rid]
+//
+// New draft 14, need to parse this for now, will eventually emit it
+// sc-value     = ( sc-send [SP sc-recv] ) / ( sc-recv [SP sc-send] )
+// sc-send      = %s"send" SP sc-str-list
+// sc-recv      = %s"recv" SP sc-str-list
+// sc-str-list  = sc-alt-list *( ";" sc-alt-list )
+// sc-alt-list  = sc-id *( "," sc-id )
+// sc-id-paused = "~"
+// sc-id        = [sc-id-paused] rid-id
+// ; SP defined in [RFC5234]
+// ; rid-id defined in [I-D.ietf-mmusic-rid]
+
+class SdpSimulcastAttribute : public SdpAttribute {
+ public:
+  SdpSimulcastAttribute() : SdpAttribute(kSimulcastAttribute) {}
+
+  SdpAttribute* Clone() const override {
+    return new SdpSimulcastAttribute(*this);
+  }
+
+  void Serialize(std::ostream& os) const override;
+  bool Parse(std::istream& is, std::string* error);
+
+  class Encoding {
+   public:
+    Encoding(const std::string& aRid, bool aPaused)
+        : rid(aRid), paused(aPaused) {}
+    std::string rid;
+    bool paused = false;
+  };
+
+  class Version {
+   public:
+    void Serialize(std::ostream& os) const;
+    bool IsSet() const { return !choices.empty(); }
+    bool Parse(std::istream& is, std::string* error);
+
+    std::vector<Encoding> choices;
+  };
+
+  class Versions : public std::vector<Version> {
+   public:
+    void Serialize(std::ostream& os) const;
+    bool IsSet() const {
+      if (empty()) {
+        return false;
+      }
+
+      for (const Version& version : *this) {
+        if (version.IsSet()) {
+          return true;
+        }
+      }
+
+      return false;
+    }
+
+    bool Parse(std::istream& is, std::string* error);
+  };
+
+  Versions sendVersions;
+  Versions recvVersions;
+};
+
+///////////////////////////////////////////////////////////////////////////
+// a=ssrc, RFC5576
+//-------------------------------------------------------------------------
+// ssrc-attr = "ssrc:" ssrc-id SP attribute
+// ; The base definition of "attribute" is in RFC 4566.
+// ; (It is the content of "a=" lines.)
+//
+// ssrc-id = integer ; 0 .. 2**32 - 1
+//-------------------------------------------------------------------------
+// TODO -- In the future, it might be nice if we ran a parse on the
+// attribute section of this so that we could interpret it semantically.
+// For WebRTC, the key use case for a=ssrc is assocaiting SSRCs with
+// media sections, and we're not really going to care about the attribute
+// itself. So we're just going to store it as a string for the time being.
+// Issue 187.
+class SdpSsrcAttributeList : public SdpAttribute {
+ public:
+  SdpSsrcAttributeList() : SdpAttribute(kSsrcAttribute) {}
+
+  struct Ssrc {
+    uint32_t ssrc;
+    std::string attribute;
+  };
+
+  void PushEntry(uint32_t ssrc, const std::string& attribute) {
+    Ssrc value = {ssrc, attribute};
+    mSsrcs.push_back(value);
+  }
+
+  SdpAttribute* Clone() const override {
+    return new SdpSsrcAttributeList(*this);
+  }
+
+  virtual void Serialize(std::ostream& os) const override;
+
+  std::vector<Ssrc> mSsrcs;
+};
+
+///////////////////////////////////////////////////////////////////////////
+// a=ssrc-group, RFC5576
+//-------------------------------------------------------------------------
+// ssrc-group-attr = "ssrc-group:" semantics *(SP ssrc-id)
+//
+// semantics       = "FEC" / "FID" / token
+//
+// ssrc-id = integer ; 0 .. 2**32 - 1
+class SdpSsrcGroupAttributeList : public SdpAttribute {
+ public:
+  enum Semantics {
+    kFec,    // RFC5576
+    kFid,    // RFC5576
+    kFecFr,  // RFC5956
+    kDup,    // RFC7104
+    kSim     // non-standard, used by hangouts
+  };
+
+  struct SsrcGroup {
+    Semantics semantics;
+    std::vector<uint32_t> ssrcs;
+  };
+
+  SdpSsrcGroupAttributeList() : SdpAttribute(kSsrcGroupAttribute) {}
+
+  void PushEntry(Semantics semantics, const std::vector<uint32_t>& ssrcs) {
+    SsrcGroup value = {semantics, ssrcs};
+    mSsrcGroups.push_back(value);
+  }
+
+  SdpAttribute* Clone() const override {
+    return new SdpSsrcGroupAttributeList(*this);
+  }
+
+  virtual void Serialize(std::ostream& os) const override;
+
+  std::vector<SsrcGroup> mSsrcGroups;
+};
+
+inline std::ostream& operator<<(std::ostream& os,
+                                SdpSsrcGroupAttributeList::Semantics s) {
+  switch (s) {
+    case SdpSsrcGroupAttributeList::kFec:
+      os << "FEC";
+      break;
+    case SdpSsrcGroupAttributeList::kFid:
+      os << "FID";
+      break;
+    case SdpSsrcGroupAttributeList::kFecFr:
+      os << "FEC-FR";
+      break;
+    case SdpSsrcGroupAttributeList::kDup:
+      os << "DUP";
+      break;
+    case SdpSsrcGroupAttributeList::kSim:
+      os << "SIM";
+      break;
+    default:
+      MOZ_ASSERT(false);
+      os << "?";
+  }
+  return os;
+}
+
+///////////////////////////////////////////////////////////////////////////
+class SdpMultiStringAttribute : public SdpAttribute {
+ public:
+  explicit SdpMultiStringAttribute(AttributeType type) : SdpAttribute(type) {}
+
+  void PushEntry(const std::string& entry) { mValues.push_back(entry); }
+
+  SdpAttribute* Clone() const override {
+    return new SdpMultiStringAttribute(*this);
+  }
+
+  virtual void Serialize(std::ostream& os) const override;
+
+  std::vector<std::string> mValues;
+};
+
+// otherwise identical to SdpMultiStringAttribute, this is used for
+// ice-options and other places where the value is serialized onto
+// a single line with space separating tokens
+class SdpOptionsAttribute : public SdpAttribute {
+ public:
+  explicit SdpOptionsAttribute(AttributeType type) : SdpAttribute(type) {}
+
+  void PushEntry(const std::string& entry) { mValues.push_back(entry); }
+
+  void Load(const std::string& value);
+
+  SdpAttribute* Clone() const override {
+    return new SdpOptionsAttribute(*this);
+  }
+
+  virtual void Serialize(std::ostream& os) const override;
+
+  std::vector<std::string> mValues;
+};
+
+// Used for attributes that take no value (eg; a=ice-lite)
+class SdpFlagAttribute : public SdpAttribute {
+ public:
+  explicit SdpFlagAttribute(AttributeType type) : SdpAttribute(type) {}
+
+  SdpAttribute* Clone() const override { return new SdpFlagAttribute(*this); }
+
+  virtual void Serialize(std::ostream& os) const override;
+};
+
+// Used for any other kind of single-valued attribute not otherwise specialized
+class SdpStringAttribute : public SdpAttribute {
+ public:
+  explicit SdpStringAttribute(AttributeType type, const std::string& value)
+      : SdpAttribute(type), mValue(value) {}
+
+  SdpAttribute* Clone() const override { return new SdpStringAttribute(*this); }
+
+  virtual void Serialize(std::ostream& os) const override;
+
+  std::string mValue;
+};
+
+// Used for any purely (non-negative) numeric attribute
+class SdpNumberAttribute : public SdpAttribute {
+ public:
+  explicit SdpNumberAttribute(AttributeType type, uint32_t value = 0)
+      : SdpAttribute(type), mValue(value) {}
+
+  SdpAttribute* Clone() const override { return new SdpNumberAttribute(*this); }
+
+  virtual void Serialize(std::ostream& os) const override;
+
+  uint32_t mValue;
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/sdp/SdpAttributeList.h b/dom/media/webrtc/sdp/SdpAttributeList.h
new file mode 100644
index 0000000000..23088dc967
--- /dev/null
+++ b/dom/media/webrtc/sdp/SdpAttributeList.h
@@ -0,0 +1,90 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _SDPATTRIBUTELIST_H_
+#define _SDPATTRIBUTELIST_H_
+
+#include "mozilla/UniquePtr.h"
+#include "mozilla/Attributes.h"
+
+#include "sdp/SdpAttribute.h"
+
+namespace mozilla {
+
+class SdpAttributeList {
+ public:
+  virtual ~SdpAttributeList() {}
+  typedef SdpAttribute::AttributeType AttributeType;
+
+  // Avoid default params on virtual functions
+  bool HasAttribute(AttributeType type) const {
+    return HasAttribute(type, true);
+  }
+
+  const SdpAttribute* GetAttribute(AttributeType type) const {
+    return GetAttribute(type, true);
+  }
+
+  virtual bool HasAttribute(AttributeType type, bool sessionFallback) const = 0;
+  virtual const SdpAttribute* GetAttribute(AttributeType type,
+                                           bool sessionFallback) const = 0;
+  // The setter takes an attribute of any type, and takes ownership
+  virtual void SetAttribute(SdpAttribute* attr) = 0;
+  virtual void RemoveAttribute(AttributeType type) = 0;
+  virtual void Clear() = 0;
+  virtual uint32_t Count() const = 0;
+
+  virtual const SdpConnectionAttribute& GetConnection() const = 0;
+  virtual const SdpOptionsAttribute& GetIceOptions() const = 0;
+  virtual const SdpRtcpAttribute& GetRtcp() const = 0;
+  virtual const SdpRemoteCandidatesAttribute& GetRemoteCandidates() const = 0;
+  virtual const SdpSetupAttribute& GetSetup() const = 0;
+  virtual const SdpDtlsMessageAttribute& GetDtlsMessage() const = 0;
+
+  // These attributes can appear multiple times, so the returned
+  // classes actually represent a collection of values.
+  virtual const std::vector<std::string>& GetCandidate() const = 0;
+  virtual const SdpExtmapAttributeList& GetExtmap() const = 0;
+  virtual const SdpFingerprintAttributeList& GetFingerprint() const = 0;
+  virtual const SdpFmtpAttributeList& GetFmtp() const = 0;
+  virtual const SdpGroupAttributeList& GetGroup() const = 0;
+  virtual const SdpImageattrAttributeList& GetImageattr() const = 0;
+  virtual const SdpSimulcastAttribute& GetSimulcast() const = 0;
+  virtual const SdpMsidAttributeList& GetMsid() const = 0;
+  virtual const SdpMsidSemanticAttributeList& GetMsidSemantic() const = 0;
+  virtual const SdpRidAttributeList& GetRid() const = 0;
+  virtual const SdpRtcpFbAttributeList& GetRtcpFb() const = 0;
+  virtual const SdpRtpmapAttributeList& GetRtpmap() const = 0;
+  virtual const SdpSctpmapAttributeList& GetSctpmap() const = 0;
+  virtual uint32_t GetSctpPort() const = 0;
+  virtual uint32_t GetMaxMessageSize() const = 0;
+  virtual const SdpSsrcAttributeList& GetSsrc() const = 0;
+  virtual const SdpSsrcGroupAttributeList& GetSsrcGroup() const = 0;
+
+  // These attributes are effectively simple types, so we'll make life
+  // easy by just returning their value.
+  virtual const std::string& GetIcePwd() const = 0;
+  virtual const std::string& GetIceUfrag() const = 0;
+  virtual const std::string& GetIdentity() const = 0;
+  virtual const std::string& GetLabel() const = 0;
+  virtual unsigned int GetMaxptime() const = 0;
+  virtual const std::string& GetMid() const = 0;
+  virtual unsigned int GetPtime() const = 0;
+
+  // This is "special", because it's multiple things
+  virtual SdpDirectionAttribute::Direction GetDirection() const = 0;
+
+  virtual void Serialize(std::ostream&) const = 0;
+};
+
+inline std::ostream& operator<<(std::ostream& os, const SdpAttributeList& al) {
+  al.Serialize(os);
+  return os;
+}
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/sdp/SdpEnum.h b/dom/media/webrtc/sdp/SdpEnum.h
new file mode 100644
index 0000000000..c5d67352e0
--- /dev/null
+++ b/dom/media/webrtc/sdp/SdpEnum.h
@@ -0,0 +1,64 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _SDPENUM_H_
+#define _SDPENUM_H_
+
+#include <ostream>
+
+#include "mozilla/Assertions.h"
+
+namespace mozilla::sdp {
+
+enum NetType { kNetTypeNone, kInternet };
+
+inline std::ostream& operator<<(std::ostream& os, sdp::NetType t) {
+  switch (t) {
+    case sdp::kNetTypeNone:
+      MOZ_ASSERT(false);
+      return os << "NONE";
+    case sdp::kInternet:
+      return os << "IN";
+  }
+  MOZ_CRASH("Unknown NetType");
+}
+
+enum AddrType { kAddrTypeNone, kIPv4, kIPv6 };
+
+inline std::ostream& operator<<(std::ostream& os, sdp::AddrType t) {
+  switch (t) {
+    case sdp::kAddrTypeNone:
+      MOZ_ASSERT(false);
+      return os << "NONE";
+    case sdp::kIPv4:
+      return os << "IP4";
+    case sdp::kIPv6:
+      return os << "IP6";
+  }
+  MOZ_CRASH("Unknown AddrType");
+}
+
+enum Direction {
+  // Start at 1 so these can be used as flags
+  kSend = 1,
+  kRecv = 2
+};
+
+inline std::ostream& operator<<(std::ostream& os, sdp::Direction d) {
+  switch (d) {
+    case sdp::kSend:
+      return os << "send";
+    case sdp::kRecv:
+      return os << "recv";
+  }
+  MOZ_CRASH("Unknown Direction");
+}
+
+enum SdpType { kOffer, kAnswer };
+
+}  // namespace mozilla::sdp
+
+#endif
diff --git a/dom/media/webrtc/sdp/SdpHelper.cpp b/dom/media/webrtc/sdp/SdpHelper.cpp
new file mode 100644
index 0000000000..d24a7d199d
--- /dev/null
+++ b/dom/media/webrtc/sdp/SdpHelper.cpp
@@ -0,0 +1,801 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "sdp/SdpHelper.h"
+
+#include "sdp/Sdp.h"
+#include "sdp/SdpMediaSection.h"
+#include "transport/logging.h"
+
+#include "nsDebug.h"
+#include "nsError.h"
+#include "prprf.h"
+
+#include <string.h>
+#include <set>
+
+namespace mozilla {
+MOZ_MTLOG_MODULE("sdp")
+
+#define SDP_SET_ERROR(error)         \
+  do {                               \
+    std::ostringstream os;           \
+    os << error;                     \
+    mLastError = os.str();           \
+    MOZ_MTLOG(ML_ERROR, mLastError); \
+  } while (0);
+
+nsresult SdpHelper::CopyTransportParams(size_t numComponents,
+                                        const SdpMediaSection& oldLocal,
+                                        SdpMediaSection* newLocal) {
+  const SdpAttributeList& oldLocalAttrs = oldLocal.GetAttributeList();
+  // Copy over m-section details
+  if (!oldLocalAttrs.HasAttribute(SdpAttribute::kBundleOnlyAttribute)) {
+    // Do not copy port 0 from an offer with a=bundle-only; this could cause
+    // an answer msection to be erroneously rejected.
+    newLocal->SetPort(oldLocal.GetPort());
+  }
+  newLocal->GetConnection() = oldLocal.GetConnection();
+
+  SdpAttributeList& newLocalAttrs = newLocal->GetAttributeList();
+
+  // Now we copy over attributes that won't be added by the usual logic
+  if (oldLocalAttrs.HasAttribute(SdpAttribute::kCandidateAttribute) &&
+      numComponents) {
+    UniquePtr<SdpMultiStringAttribute> candidateAttrs(
+        new SdpMultiStringAttribute(SdpAttribute::kCandidateAttribute));
+    for (const std::string& candidate : oldLocalAttrs.GetCandidate()) {
+      size_t component;
+      nsresult rv = GetComponent(candidate, &component);
+      NS_ENSURE_SUCCESS(rv, rv);
+      if (numComponents >= component) {
+        candidateAttrs->mValues.push_back(candidate);
+      }
+    }
+    if (!candidateAttrs->mValues.empty()) {
+      newLocalAttrs.SetAttribute(candidateAttrs.release());
+    }
+  }
+
+  if (oldLocalAttrs.HasAttribute(SdpAttribute::kEndOfCandidatesAttribute)) {
+    newLocalAttrs.SetAttribute(
+        new SdpFlagAttribute(SdpAttribute::kEndOfCandidatesAttribute));
+  }
+
+  if (numComponents == 2 &&
+      oldLocalAttrs.HasAttribute(SdpAttribute::kRtcpAttribute)) {
+    // copy rtcp attribute if we had one that we are using
+    newLocalAttrs.SetAttribute(new SdpRtcpAttribute(oldLocalAttrs.GetRtcp()));
+  }
+
+  return NS_OK;
+}
+
+bool SdpHelper::AreOldTransportParamsValid(const Sdp& oldAnswer,
+                                           const Sdp& offerersPreviousSdp,
+                                           const Sdp& newOffer, size_t level) {
+  if (MsectionIsDisabled(oldAnswer.GetMediaSection(level)) ||
+      MsectionIsDisabled(newOffer.GetMediaSection(level))) {
+    // Obvious
+    return false;
+  }
+
+  if (!OwnsTransport(oldAnswer, level, sdp::kAnswer)) {
+    // The transport attributes on this m-section were thrown away, because it
+    // was bundled.
+    return false;
+  }
+
+  if (!OwnsTransport(newOffer, level, sdp::kOffer)) {
+    return false;
+  }
+
+  if (IceCredentialsDiffer(newOffer.GetMediaSection(level),
+                           offerersPreviousSdp.GetMediaSection(level))) {
+    return false;
+  }
+
+  return true;
+}
+
+bool SdpHelper::IceCredentialsDiffer(const SdpMediaSection& msection1,
+                                     const SdpMediaSection& msection2) {
+  const SdpAttributeList& attrs1(msection1.GetAttributeList());
+  const SdpAttributeList& attrs2(msection2.GetAttributeList());
+
+  if ((attrs1.GetIceUfrag() != attrs2.GetIceUfrag()) ||
+      (attrs1.GetIcePwd() != attrs2.GetIcePwd())) {
+    return true;
+  }
+
+  return false;
+}
+
+nsresult SdpHelper::GetComponent(const std::string& candidate,
+                                 size_t* component) {
+  unsigned int temp;
+  int32_t result = PR_sscanf(candidate.c_str(), "%*s %u", &temp);
+  if (result == 1) {
+    *component = temp;
+    return NS_OK;
+  }
+  SDP_SET_ERROR("Malformed ICE candidate: " << candidate);
+  return NS_ERROR_INVALID_ARG;
+}
+
+bool SdpHelper::MsectionIsDisabled(const SdpMediaSection& msection) const {
+  return !msection.GetPort() && !msection.GetAttributeList().HasAttribute(
+                                    SdpAttribute::kBundleOnlyAttribute);
+}
+
+void SdpHelper::DisableMsection(Sdp* sdp, SdpMediaSection* msection) {
+  std::string mid;
+
+  // Make sure to remove the mid from any group attributes
+  if (msection->GetAttributeList().HasAttribute(SdpAttribute::kMidAttribute)) {
+    mid = msection->GetAttributeList().GetMid();
+    if (sdp->GetAttributeList().HasAttribute(SdpAttribute::kGroupAttribute)) {
+      UniquePtr<SdpGroupAttributeList> newGroupAttr(
+          new SdpGroupAttributeList(sdp->GetAttributeList().GetGroup()));
+      newGroupAttr->RemoveMid(mid);
+      sdp->GetAttributeList().SetAttribute(newGroupAttr.release());
+    }
+  }
+
+  // Clear out attributes.
+  msection->GetAttributeList().Clear();
+
+  auto* direction = new SdpDirectionAttribute(SdpDirectionAttribute::kInactive);
+  msection->GetAttributeList().SetAttribute(direction);
+  msection->SetPort(0);
+
+  // maintain the mid for easier identification on other side
+  if (!mid.empty()) {
+    msection->GetAttributeList().SetAttribute(
+        new SdpStringAttribute(SdpAttribute::kMidAttribute, mid));
+  }
+
+  msection->ClearCodecs();
+
+  auto mediaType = msection->GetMediaType();
+  switch (mediaType) {
+    case SdpMediaSection::kAudio:
+      msection->AddCodec("0", "PCMU", 8000, 1);
+      break;
+    case SdpMediaSection::kVideo:
+      msection->AddCodec("120", "VP8", 90000, 1);
+      break;
+    case SdpMediaSection::kApplication:
+      msection->AddDataChannel("webrtc-datachannel", 0, 0, 0);
+      break;
+    default:
+      // We need to have something here to fit the grammar, this seems safe
+      // and 19 is a reserved payload type which should not be used by anyone.
+      msection->AddCodec("19", "reserved", 8000, 1);
+  }
+}
+
+void SdpHelper::GetBundleGroups(
+    const Sdp& sdp,
+    std::vector<SdpGroupAttributeList::Group>* bundleGroups) const {
+  if (sdp.GetAttributeList().HasAttribute(SdpAttribute::kGroupAttribute)) {
+    for (auto& group : sdp.GetAttributeList().GetGroup().mGroups) {
+      if (group.semantics == SdpGroupAttributeList::kBundle) {
+        bundleGroups->push_back(group);
+      }
+    }
+  }
+}
+
+nsresult SdpHelper::GetBundledMids(const Sdp& sdp, BundledMids* bundledMids) {
+  std::vector<SdpGroupAttributeList::Group> bundleGroups;
+  GetBundleGroups(sdp, &bundleGroups);
+
+  for (SdpGroupAttributeList::Group& group : bundleGroups) {
+    if (group.tags.empty()) {
+      continue;
+    }
+
+    const SdpMediaSection* msection(FindMsectionByMid(sdp, group.tags[0]));
+
+    if (!msection) {
+      SDP_SET_ERROR(
+          "mid specified for bundle transport in group attribute"
+          " does not exist in the SDP. (mid="
+          << group.tags[0] << ")");
+      return NS_ERROR_INVALID_ARG;
+    }
+
+    if (MsectionIsDisabled(*msection)) {
+      SDP_SET_ERROR(
+          "mid specified for bundle transport in group attribute"
+          " points at a disabled m-section. (mid="
+          << group.tags[0] << ")");
+      return NS_ERROR_INVALID_ARG;
+    }
+
+    for (const std::string& mid : group.tags) {
+      if (bundledMids->count(mid)) {
+        SDP_SET_ERROR("mid \'" << mid
+                               << "\' appears more than once in a "
+                                  "BUNDLE group");
+        return NS_ERROR_INVALID_ARG;
+      }
+
+      (*bundledMids)[mid] = msection;
+    }
+  }
+
+  return NS_OK;
+}
+
+bool SdpHelper::OwnsTransport(const Sdp& sdp, uint16_t level,
+                              sdp::SdpType type) {
+  auto& msection = sdp.GetMediaSection(level);
+
+  BundledMids bundledMids;
+  nsresult rv = GetBundledMids(sdp, &bundledMids);
+  if (NS_FAILED(rv)) {
+    // Should have been caught sooner.
+    MOZ_ASSERT(false);
+    return true;
+  }
+
+  return OwnsTransport(msection, bundledMids, type);
+}
+
+bool SdpHelper::OwnsTransport(const SdpMediaSection& msection,
+                              const BundledMids& bundledMids,
+                              sdp::SdpType type) {
+  if (MsectionIsDisabled(msection)) {
+    return false;
+  }
+
+  if (!msection.GetAttributeList().HasAttribute(SdpAttribute::kMidAttribute)) {
+    // No mid, definitely no bundle for this m-section
+    return true;
+  }
+  std::string mid(msection.GetAttributeList().GetMid());
+  if (type != sdp::kOffer || msection.GetAttributeList().HasAttribute(
+                                 SdpAttribute::kBundleOnlyAttribute)) {
+    // If this is an answer, or this m-section is marked bundle-only, the group
+    // attribute is authoritative. Otherwise, we aren't sure.
+    if (bundledMids.count(mid) && &msection != bundledMids.at(mid)) {
+      // mid is bundled, and isn't the bundle m-section
+      return false;
+    }
+  }
+
+  return true;
+}
+
+nsresult SdpHelper::GetMidFromLevel(const Sdp& sdp, uint16_t level,
+                                    std::string* mid) {
+  if (level >= sdp.GetMediaSectionCount()) {
+    SDP_SET_ERROR("Index " << level << " out of range");
+    return NS_ERROR_INVALID_ARG;
+  }
+
+  const SdpMediaSection& msection = sdp.GetMediaSection(level);
+  const SdpAttributeList& attrList = msection.GetAttributeList();
+
+  // grab the mid and set the outparam
+  if (attrList.HasAttribute(SdpAttribute::kMidAttribute)) {
+    *mid = attrList.GetMid();
+  }
+
+  return NS_OK;
+}
+
+nsresult SdpHelper::AddCandidateToSdp(Sdp* sdp,
+                                      const std::string& candidateUntrimmed,
+                                      uint16_t level,
+                                      const std::string& ufrag) {
+  if (level >= sdp->GetMediaSectionCount()) {
+    SDP_SET_ERROR("Index " << level << " out of range");
+    return NS_ERROR_INVALID_ARG;
+  }
+
+  SdpMediaSection& msection = sdp->GetMediaSection(level);
+  SdpAttributeList& attrList = msection.GetAttributeList();
+
+  if (!ufrag.empty()) {
+    if (!attrList.HasAttribute(SdpAttribute::kIceUfragAttribute) ||
+        attrList.GetIceUfrag() != ufrag) {
+      SDP_SET_ERROR("Unknown ufrag (" << ufrag << ")");
+      return NS_ERROR_INVALID_ARG;
+    }
+  }
+
+  if (candidateUntrimmed.empty()) {
+    SetIceGatheringComplete(sdp, level, ufrag);
+    return NS_OK;
+  }
+
+  // Trim off '[a=]candidate:'
+  size_t begin = candidateUntrimmed.find(':');
+  if (begin == std::string::npos) {
+    SDP_SET_ERROR("Invalid candidate, no ':' (" << candidateUntrimmed << ")");
+    return NS_ERROR_INVALID_ARG;
+  }
+  ++begin;
+
+  std::string candidate = candidateUntrimmed.substr(begin);
+
+  UniquePtr<SdpMultiStringAttribute> candidates;
+  if (!attrList.HasAttribute(SdpAttribute::kCandidateAttribute)) {
+    // Create new
+    candidates.reset(
+        new SdpMultiStringAttribute(SdpAttribute::kCandidateAttribute));
+  } else {
+    // Copy existing
+    candidates.reset(new SdpMultiStringAttribute(
+        *static_cast<const SdpMultiStringAttribute*>(
+            attrList.GetAttribute(SdpAttribute::kCandidateAttribute))));
+  }
+  candidates->PushEntry(candidate);
+  attrList.SetAttribute(candidates.release());
+
+  return NS_OK;
+}
+
+nsresult SdpHelper::SetIceGatheringComplete(Sdp* sdp,
+                                            const std::string& ufrag) {
+  for (uint16_t i = 0; i < sdp->GetMediaSectionCount(); ++i) {
+    nsresult rv = SetIceGatheringComplete(sdp, i, ufrag);
+    NS_ENSURE_SUCCESS(rv, rv);
+  }
+  return NS_OK;
+}
+
+nsresult SdpHelper::SetIceGatheringComplete(Sdp* sdp, uint16_t level,
+                                            const std::string& ufrag) {
+  if (level >= sdp->GetMediaSectionCount()) {
+    SDP_SET_ERROR("Index " << level << " out of range");
+    return NS_ERROR_INVALID_ARG;
+  }
+
+  SdpMediaSection& msection = sdp->GetMediaSection(level);
+  SdpAttributeList& attrList = msection.GetAttributeList();
+
+  if (!ufrag.empty()) {
+    if (!attrList.HasAttribute(SdpAttribute::kIceUfragAttribute) ||
+        attrList.GetIceUfrag() != ufrag) {
+      SDP_SET_ERROR("Unknown ufrag (" << ufrag << ")");
+      return NS_ERROR_INVALID_ARG;
+    }
+  }
+
+  attrList.SetAttribute(
+      new SdpFlagAttribute(SdpAttribute::kEndOfCandidatesAttribute));
+  // Remove trickle-ice option
+  attrList.RemoveAttribute(SdpAttribute::kIceOptionsAttribute);
+  return NS_OK;
+}
+
+void SdpHelper::SetDefaultAddresses(const std::string& defaultCandidateAddr,
+                                    uint16_t defaultCandidatePort,
+                                    const std::string& defaultRtcpCandidateAddr,
+                                    uint16_t defaultRtcpCandidatePort,
+                                    SdpMediaSection* msection) {
+  SdpAttributeList& attrList = msection->GetAttributeList();
+
+  msection->GetConnection().SetAddress(defaultCandidateAddr);
+  msection->SetPort(defaultCandidatePort);
+  if (!defaultRtcpCandidateAddr.empty()) {
+    sdp::AddrType ipVersion = sdp::kIPv4;
+    if (defaultRtcpCandidateAddr.find(':') != std::string::npos) {
+      ipVersion = sdp::kIPv6;
+    }
+    attrList.SetAttribute(new SdpRtcpAttribute(defaultRtcpCandidatePort,
+                                               sdp::kInternet, ipVersion,
+                                               defaultRtcpCandidateAddr));
+  }
+}
+
+nsresult SdpHelper::GetIdsFromMsid(const Sdp& sdp,
+                                   const SdpMediaSection& msection,
+                                   std::vector<std::string>* streamIds) {
+  std::vector<SdpMsidAttributeList::Msid> allMsids;
+  nsresult rv = GetMsids(msection, &allMsids);
+  NS_ENSURE_SUCCESS(rv, rv);
+
+  if (allMsids.empty()) {
+    return NS_ERROR_NOT_AVAILABLE;
+  }
+
+  streamIds->clear();
+  for (const auto& msid : allMsids) {
+    // "-" means no stream, see draft-ietf-mmusic-msid
+    // Remove duplicates, but leave order the same
+    if (msid.identifier != "-" &&
+        !std::count(streamIds->begin(), streamIds->end(), msid.identifier)) {
+      streamIds->push_back(msid.identifier);
+    }
+  }
+
+  return NS_OK;
+}
+
+nsresult SdpHelper::GetMsids(const SdpMediaSection& msection,
+                             std::vector<SdpMsidAttributeList::Msid>* msids) {
+  if (msection.GetAttributeList().HasAttribute(SdpAttribute::kMsidAttribute)) {
+    *msids = msection.GetAttributeList().GetMsid().mMsids;
+    return NS_OK;
+  }
+
+  // If there are no a=msid, can we find msids in ssrc attributes?
+  // (Chrome does not put plain-old msid attributes in its SDP)
+  if (msection.GetAttributeList().HasAttribute(SdpAttribute::kSsrcAttribute)) {
+    auto& ssrcs = msection.GetAttributeList().GetSsrc().mSsrcs;
+
+    for (auto i = ssrcs.begin(); i != ssrcs.end(); ++i) {
+      if (i->attribute.find("msid:") == 0) {
+        std::string streamId;
+        std::string trackId;
+        nsresult rv = ParseMsid(i->attribute, &streamId, &trackId);
+        NS_ENSURE_SUCCESS(rv, rv);
+        msids->push_back({streamId, trackId});
+      }
+    }
+  }
+
+  return NS_OK;
+}
+
+nsresult SdpHelper::ParseMsid(const std::string& msidAttribute,
+                              std::string* streamId, std::string* trackId) {
+  // Would be nice if SdpSsrcAttributeList could parse out the contained
+  // attribute, but at least the parse here is simple.
+  // We are being very forgiving here wrt whitespace; tabs are not actually
+  // allowed, nor is leading/trailing whitespace.
+  size_t streamIdStart = msidAttribute.find_first_not_of(" \t", 5);
+  // We do not assume the appdata token is here, since this is not
+  // necessarily a webrtc msid
+  if (streamIdStart == std::string::npos) {
+    SDP_SET_ERROR("Malformed source-level msid attribute: " << msidAttribute);
+    return NS_ERROR_INVALID_ARG;
+  }
+
+  size_t streamIdEnd = msidAttribute.find_first_of(" \t", streamIdStart);
+  if (streamIdEnd == std::string::npos) {
+    streamIdEnd = msidAttribute.size();
+  }
+
+  size_t trackIdStart = msidAttribute.find_first_not_of(" \t", streamIdEnd);
+  if (trackIdStart == std::string::npos) {
+    trackIdStart = msidAttribute.size();
+  }
+
+  size_t trackIdEnd = msidAttribute.find_first_of(" \t", trackIdStart);
+  if (trackIdEnd == std::string::npos) {
+    trackIdEnd = msidAttribute.size();
+  }
+
+  size_t streamIdSize = streamIdEnd - streamIdStart;
+  size_t trackIdSize = trackIdEnd - trackIdStart;
+
+  *streamId = msidAttribute.substr(streamIdStart, streamIdSize);
+  *trackId = msidAttribute.substr(trackIdStart, trackIdSize);
+  return NS_OK;
+}
+
+void SdpHelper::SetupMsidSemantic(const std::vector<std::string>& msids,
+                                  Sdp* sdp) const {
+  if (!msids.empty()) {
+    UniquePtr<SdpMsidSemanticAttributeList> msidSemantics(
+        new SdpMsidSemanticAttributeList);
+    msidSemantics->PushEntry("WMS", msids);
+    sdp->GetAttributeList().SetAttribute(msidSemantics.release());
+  }
+}
+
+std::string SdpHelper::GetCNAME(const SdpMediaSection& msection) const {
+  if (msection.GetAttributeList().HasAttribute(SdpAttribute::kSsrcAttribute)) {
+    auto& ssrcs = msection.GetAttributeList().GetSsrc().mSsrcs;
+    for (auto i = ssrcs.begin(); i != ssrcs.end(); ++i) {
+      if (i->attribute.find("cname:") == 0) {
+        return i->attribute.substr(6);
+      }
+    }
+  }
+  return "";
+}
+
+const SdpMediaSection* SdpHelper::FindMsectionByMid(
+    const Sdp& sdp, const std::string& mid) const {
+  for (size_t i = 0; i < sdp.GetMediaSectionCount(); ++i) {
+    auto& attrs = sdp.GetMediaSection(i).GetAttributeList();
+    if (attrs.HasAttribute(SdpAttribute::kMidAttribute) &&
+        attrs.GetMid() == mid) {
+      return &sdp.GetMediaSection(i);
+    }
+  }
+  return nullptr;
+}
+
+SdpMediaSection* SdpHelper::FindMsectionByMid(Sdp& sdp,
+                                              const std::string& mid) const {
+  for (size_t i = 0; i < sdp.GetMediaSectionCount(); ++i) {
+    auto& attrs = sdp.GetMediaSection(i).GetAttributeList();
+    if (attrs.HasAttribute(SdpAttribute::kMidAttribute) &&
+        attrs.GetMid() == mid) {
+      return &sdp.GetMediaSection(i);
+    }
+  }
+  return nullptr;
+}
+
+nsresult SdpHelper::CopyStickyParams(const SdpMediaSection& source,
+                                     SdpMediaSection* dest) {
+  auto& sourceAttrs = source.GetAttributeList();
+  auto& destAttrs = dest->GetAttributeList();
+
+  // There's no reason to renegotiate rtcp-mux
+  if (sourceAttrs.HasAttribute(SdpAttribute::kRtcpMuxAttribute)) {
+    destAttrs.SetAttribute(
+        new SdpFlagAttribute(SdpAttribute::kRtcpMuxAttribute));
+  }
+
+  // mid should stay the same
+  if (sourceAttrs.HasAttribute(SdpAttribute::kMidAttribute)) {
+    destAttrs.SetAttribute(new SdpStringAttribute(SdpAttribute::kMidAttribute,
+                                                  sourceAttrs.GetMid()));
+  }
+
+  // Keep RTCP mode setting
+  if (sourceAttrs.HasAttribute(SdpAttribute::kRtcpRsizeAttribute) &&
+      source.GetMediaType() == SdpMediaSection::kVideo) {
+    destAttrs.SetAttribute(
+        new SdpFlagAttribute(SdpAttribute::kRtcpRsizeAttribute));
+  }
+
+  return NS_OK;
+}
+
+bool SdpHelper::HasRtcp(SdpMediaSection::Protocol proto) const {
+  switch (proto) {
+    case SdpMediaSection::kRtpAvpf:
+    case SdpMediaSection::kDccpRtpAvpf:
+    case SdpMediaSection::kDccpRtpSavpf:
+    case SdpMediaSection::kRtpSavpf:
+    case SdpMediaSection::kUdpTlsRtpSavpf:
+    case SdpMediaSection::kTcpDtlsRtpSavpf:
+    case SdpMediaSection::kDccpTlsRtpSavpf:
+      return true;
+    case SdpMediaSection::kRtpAvp:
+    case SdpMediaSection::kUdp:
+    case SdpMediaSection::kVat:
+    case SdpMediaSection::kRtp:
+    case SdpMediaSection::kUdptl:
+    case SdpMediaSection::kTcp:
+    case SdpMediaSection::kTcpRtpAvp:
+    case SdpMediaSection::kRtpSavp:
+    case SdpMediaSection::kTcpBfcp:
+    case SdpMediaSection::kTcpTlsBfcp:
+    case SdpMediaSection::kTcpTls:
+    case SdpMediaSection::kFluteUdp:
+    case SdpMediaSection::kTcpMsrp:
+    case SdpMediaSection::kTcpTlsMsrp:
+    case SdpMediaSection::kDccp:
+    case SdpMediaSection::kDccpRtpAvp:
+    case SdpMediaSection::kDccpRtpSavp:
+    case SdpMediaSection::kUdpTlsRtpSavp:
+    case SdpMediaSection::kTcpDtlsRtpSavp:
+    case SdpMediaSection::kDccpTlsRtpSavp:
+    case SdpMediaSection::kUdpMbmsFecRtpAvp:
+    case SdpMediaSection::kUdpMbmsFecRtpSavp:
+    case SdpMediaSection::kUdpMbmsRepair:
+    case SdpMediaSection::kFecUdp:
+    case SdpMediaSection::kUdpFec:
+    case SdpMediaSection::kTcpMrcpv2:
+    case SdpMediaSection::kTcpTlsMrcpv2:
+    case SdpMediaSection::kPstn:
+    case SdpMediaSection::kUdpTlsUdptl:
+    case SdpMediaSection::kSctp:
+    case SdpMediaSection::kDtlsSctp:
+    case SdpMediaSection::kUdpDtlsSctp:
+    case SdpMediaSection::kTcpDtlsSctp:
+      return false;
+  }
+  MOZ_CRASH("Unknown protocol, probably corruption.");
+}
+
+SdpMediaSection::Protocol SdpHelper::GetProtocolForMediaType(
+    SdpMediaSection::MediaType type) {
+  if (type == SdpMediaSection::kApplication) {
+    return SdpMediaSection::kUdpDtlsSctp;
+  }
+
+  return SdpMediaSection::kUdpTlsRtpSavpf;
+}
+
+void SdpHelper::AppendSdpParseErrors(
+    const std::vector<std::pair<size_t, std::string> >& aErrors,
+    std::string* aErrorString) {
+  std::ostringstream os;
+  for (auto i = aErrors.begin(); i != aErrors.end(); ++i) {
+    os << "SDP Parse Error on line " << i->first << ": " + i->second
+       << std::endl;
+  }
+  *aErrorString += os.str();
+}
+
+/* static */
+bool SdpHelper::GetPtAsInt(const std::string& ptString, uint16_t* ptOutparam) {
+  char* end;
+  unsigned long pt = strtoul(ptString.c_str(), &end, 10);
+  size_t length = static_cast<size_t>(end - ptString.c_str());
+  if ((pt > UINT16_MAX) || (length != ptString.size())) {
+    return false;
+  }
+  *ptOutparam = pt;
+  return true;
+}
+
+void SdpHelper::NegotiateAndAddExtmaps(
+    const SdpMediaSection& remoteMsection,
+    std::vector<SdpExtmapAttributeList::Extmap>& localExtensions,
+    SdpMediaSection* localMsection) {
+  if (!remoteMsection.GetAttributeList().HasAttribute(
+          SdpAttribute::kExtmapAttribute)) {
+    return;
+  }
+
+  UniquePtr<SdpExtmapAttributeList> localExtmap(new SdpExtmapAttributeList);
+  auto& theirExtmap = remoteMsection.GetAttributeList().GetExtmap().mExtmaps;
+  for (const auto& theirExt : theirExtmap) {
+    for (auto& ourExt : localExtensions) {
+      if (theirExt.entry == 0) {
+        // 0 is invalid, ignore it
+        continue;
+      }
+
+      if (theirExt.extensionname != ourExt.extensionname) {
+        continue;
+      }
+
+      ourExt.direction = reverse(theirExt.direction) & ourExt.direction;
+      if (ourExt.direction == SdpDirectionAttribute::Direction::kInactive) {
+        continue;
+      }
+
+      // RFC 5285 says that ids >= 4096 can be used by the offerer to
+      // force the answerer to pick, otherwise the value in the offer is
+      // used.
+      if (theirExt.entry < 4096) {
+        ourExt.entry = theirExt.entry;
+      }
+
+      localExtmap->mExtmaps.push_back(ourExt);
+    }
+  }
+
+  if (!localExtmap->mExtmaps.empty()) {
+    localMsection->GetAttributeList().SetAttribute(localExtmap.release());
+  }
+}
+
+static bool AttributeListMatch(const SdpAttributeList& list1,
+                               const SdpAttributeList& list2) {
+  // TODO: Consider adding telemetry in this function to record which
+  // attributes don't match. See Bug 1432955.
+  for (int i = SdpAttribute::kFirstAttribute; i <= SdpAttribute::kLastAttribute;
+       i++) {
+    auto attributeType = static_cast<SdpAttribute::AttributeType>(i);
+    // TODO: We should do more thorough checking here, e.g. serialize and
+    // compare strings. See Bug 1439690.
+    if (list1.HasAttribute(attributeType, false) !=
+        list2.HasAttribute(attributeType, false)) {
+      return false;
+    }
+  }
+  return true;
+}
+
+static bool MediaSectionMatch(const SdpMediaSection& mediaSection1,
+                              const SdpMediaSection& mediaSection2) {
+  // TODO: We should do more thorough checking in this function.
+  // See Bug 1439690.
+  if (!AttributeListMatch(mediaSection1.GetAttributeList(),
+                          mediaSection2.GetAttributeList())) {
+    return false;
+  }
+  if (mediaSection1.GetPort() != mediaSection2.GetPort()) {
+    return false;
+  }
+  const std::vector<std::string>& formats1 = mediaSection1.GetFormats();
+  const std::vector<std::string>& formats2 = mediaSection2.GetFormats();
+  auto formats1Set = std::set<std::string>(formats1.begin(), formats1.end());
+  auto formats2Set = std::set<std::string>(formats2.begin(), formats2.end());
+  if (formats1Set != formats2Set) {
+    return false;
+  }
+  return true;
+}
+
+bool SdpHelper::SdpMatch(const Sdp& sdp1, const Sdp& sdp2) {
+  if (sdp1.GetMediaSectionCount() != sdp2.GetMediaSectionCount()) {
+    return false;
+  }
+  if (!AttributeListMatch(sdp1.GetAttributeList(), sdp2.GetAttributeList())) {
+    return false;
+  }
+  for (size_t i = 0; i < sdp1.GetMediaSectionCount(); i++) {
+    const SdpMediaSection& mediaSection1 = sdp1.GetMediaSection(i);
+    const SdpMediaSection& mediaSection2 = sdp2.GetMediaSection(i);
+    if (!MediaSectionMatch(mediaSection1, mediaSection2)) {
+      return false;
+    }
+  }
+
+  return true;
+}
+
+nsresult SdpHelper::ValidateTransportAttributes(const Sdp& aSdp,
+                                                sdp::SdpType aType) {
+  BundledMids bundledMids;
+  nsresult rv = GetBundledMids(aSdp, &bundledMids);
+  NS_ENSURE_SUCCESS(rv, rv);
+
+  for (size_t level = 0; level < aSdp.GetMediaSectionCount(); ++level) {
+    const auto& msection = aSdp.GetMediaSection(level);
+    if (OwnsTransport(msection, bundledMids, aType)) {
+      const auto& mediaAttrs = msection.GetAttributeList();
+      if (mediaAttrs.GetIceUfrag().empty()) {
+        SDP_SET_ERROR("Invalid description, no ice-ufrag attribute at level "
+                      << level);
+        return NS_ERROR_INVALID_ARG;
+      }
+
+      if (mediaAttrs.GetIcePwd().empty()) {
+        SDP_SET_ERROR("Invalid description, no ice-pwd attribute at level "
+                      << level);
+        return NS_ERROR_INVALID_ARG;
+      }
+
+      if (!mediaAttrs.HasAttribute(SdpAttribute::kFingerprintAttribute)) {
+        SDP_SET_ERROR("Invalid description, no fingerprint attribute at level "
+                      << level);
+        return NS_ERROR_INVALID_ARG;
+      }
+
+      const SdpFingerprintAttributeList& fingerprints(
+          mediaAttrs.GetFingerprint());
+      if (fingerprints.mFingerprints.empty()) {
+        SDP_SET_ERROR(
+            "Invalid description, no supported fingerprint algorithms present "
+            "at level "
+            << level);
+        return NS_ERROR_INVALID_ARG;
+      }
+
+      if (mediaAttrs.HasAttribute(SdpAttribute::kSetupAttribute, true)) {
+        if (mediaAttrs.GetSetup().mRole == SdpSetupAttribute::kHoldconn) {
+          SDP_SET_ERROR(
+              "Invalid description, illegal setup attribute \"holdconn\" "
+              "at level "
+              << level);
+          return NS_ERROR_INVALID_ARG;
+        }
+
+        if (aType == sdp::kAnswer &&
+            mediaAttrs.GetSetup().mRole == SdpSetupAttribute::kActpass) {
+          SDP_SET_ERROR(
+              "Invalid answer, illegal setup attribute \"actpass\" at level "
+              << level);
+          return NS_ERROR_INVALID_ARG;
+        }
+      } else if (aType == sdp::kOffer) {
+        SDP_SET_ERROR("Invalid offer, no setup attribute at level " << level);
+        return NS_ERROR_INVALID_ARG;
+      }
+    }
+  }
+  return NS_OK;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/sdp/SdpHelper.h b/dom/media/webrtc/sdp/SdpHelper.h
new file mode 100644
index 0000000000..3c65d01442
--- /dev/null
+++ b/dom/media/webrtc/sdp/SdpHelper.h
@@ -0,0 +1,109 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _SDPHELPER_H_
+#define _SDPHELPER_H_
+
+#include "nsError.h"
+
+#include "sdp/SdpMediaSection.h"
+#include "sdp/SdpAttribute.h"
+
+#include "transport/m_cpp_utils.h"
+
+#include <string>
+#include <map>
+#include <vector>
+
+namespace mozilla {
+class SdpMediaSection;
+class Sdp;
+
+class SdpHelper {
+ public:
+  // Takes a std::string* into which error strings will be written for the
+  // lifetime of the SdpHelper.
+  explicit SdpHelper(std::string* errorDest) : mLastError(*errorDest) {}
+  ~SdpHelper() {}
+
+  nsresult GetComponent(const std::string& candidate, size_t* component);
+  nsresult CopyTransportParams(size_t numComponents,
+                               const SdpMediaSection& source,
+                               SdpMediaSection* dest);
+  bool AreOldTransportParamsValid(const Sdp& oldAnswer,
+                                  const Sdp& offerersPreviousSdp,
+                                  const Sdp& newOffer, size_t level);
+  bool IceCredentialsDiffer(const SdpMediaSection& msection1,
+                            const SdpMediaSection& msection2);
+
+  bool MsectionIsDisabled(const SdpMediaSection& msection) const;
+  static void DisableMsection(Sdp* sdp, SdpMediaSection* msection);
+
+  // Maps each mid to the m-section that owns its bundle transport.
+  // Mids that do not appear in an a=group:BUNDLE do not appear here.
+  typedef std::map<std::string, const SdpMediaSection*> BundledMids;
+
+  nsresult GetBundledMids(const Sdp& sdp, BundledMids* bundledMids);
+
+  bool OwnsTransport(const Sdp& localSdp, uint16_t level, sdp::SdpType type);
+  bool OwnsTransport(const SdpMediaSection& msection,
+                     const BundledMids& bundledMids, sdp::SdpType type);
+  void GetBundleGroups(const Sdp& sdp,
+                       std::vector<SdpGroupAttributeList::Group>* groups) const;
+
+  nsresult GetMidFromLevel(const Sdp& sdp, uint16_t level, std::string* mid);
+  nsresult GetIdsFromMsid(const Sdp& sdp, const SdpMediaSection& msection,
+                          std::vector<std::string>* streamId);
+  nsresult GetMsids(const SdpMediaSection& msection,
+                    std::vector<SdpMsidAttributeList::Msid>* msids);
+  nsresult ParseMsid(const std::string& msidAttribute, std::string* streamId,
+                     std::string* trackId);
+  nsresult AddCandidateToSdp(Sdp* sdp, const std::string& candidate,
+                             uint16_t level, const std::string& ufrag);
+  nsresult SetIceGatheringComplete(Sdp* sdp, const std::string& ufrag);
+  nsresult SetIceGatheringComplete(Sdp* sdp, uint16_t level,
+                                   const std::string& ufrag);
+  void SetDefaultAddresses(const std::string& defaultCandidateAddr,
+                           uint16_t defaultCandidatePort,
+                           const std::string& defaultRtcpCandidateAddr,
+                           uint16_t defaultRtcpCandidatePort,
+                           SdpMediaSection* msection);
+  void SetupMsidSemantic(const std::vector<std::string>& msids, Sdp* sdp) const;
+
+  std::string GetCNAME(const SdpMediaSection& msection) const;
+
+  SdpMediaSection* FindMsectionByMid(Sdp& sdp, const std::string& mid) const;
+
+  const SdpMediaSection* FindMsectionByMid(const Sdp& sdp,
+                                           const std::string& mid) const;
+
+  nsresult CopyStickyParams(const SdpMediaSection& source,
+                            SdpMediaSection* dest);
+  bool HasRtcp(SdpMediaSection::Protocol proto) const;
+  static SdpMediaSection::Protocol GetProtocolForMediaType(
+      SdpMediaSection::MediaType type);
+  void AppendSdpParseErrors(
+      const std::vector<std::pair<size_t, std::string> >& aErrors,
+      std::string* aErrorString);
+
+  static bool GetPtAsInt(const std::string& ptString, uint16_t* ptOutparam);
+
+  void NegotiateAndAddExtmaps(
+      const SdpMediaSection& remoteMsection,
+      std::vector<SdpExtmapAttributeList::Extmap>& localExtensions,
+      SdpMediaSection* localMsection);
+
+  bool SdpMatch(const Sdp& sdp1, const Sdp& sdp2);
+  nsresult ValidateTransportAttributes(const Sdp& aSdp, sdp::SdpType aType);
+
+ private:
+  std::string& mLastError;
+
+  DISALLOW_COPY_ASSIGN(SdpHelper);
+};
+}  // namespace mozilla
+
+#endif  // _SDPHELPER_H_
diff --git a/dom/media/webrtc/sdp/SdpLog.cpp b/dom/media/webrtc/sdp/SdpLog.cpp
new file mode 100644
index 0000000000..a841ac7412
--- /dev/null
+++ b/dom/media/webrtc/sdp/SdpLog.cpp
@@ -0,0 +1,68 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+#include <type_traits>
+
+#include "sdp/SdpLog.h"
+#include "common/browser_logging/CSFLog.h"
+
+namespace mozilla {
+LazyLogModule SdpLog("sdp");
+}  // namespace mozilla
+
+// For compile time enum comparison
+template <typename E, typename F>
+constexpr bool compareEnum(E e, F f) {
+  return static_cast<typename std::underlying_type<E>::type>(e) ==
+         static_cast<typename std::underlying_type<F>::type>(f);
+}
+
+CSFLogLevel SDPToCSFLogLevel(const SDPLogLevel priority) {
+  static_assert(compareEnum(SDP_LOG_ERROR, CSF_LOG_ERROR));
+  static_assert(compareEnum(SDP_LOG_WARNING, CSF_LOG_WARNING));
+  static_assert(compareEnum(SDP_LOG_INFO, CSF_LOG_INFO));
+  static_assert(compareEnum(SDP_LOG_DEBUG, CSF_LOG_DEBUG));
+  static_assert(compareEnum(SDP_LOG_VERBOSE, CSF_LOG_VERBOSE));
+
+  // Check that all SDP_LOG_* cases are covered. It compiles to nothing.
+  switch (priority) {
+    case SDP_LOG_ERROR:
+    case SDP_LOG_WARNING:
+    case SDP_LOG_INFO:
+    case SDP_LOG_DEBUG:
+    case SDP_LOG_VERBOSE:
+      break;
+  }
+
+  // Ditto for CSF_LOG_*
+  switch (static_cast<CSFLogLevel>(priority)) {
+    case CSF_LOG_ERROR:
+    case CSF_LOG_WARNING:
+    case CSF_LOG_INFO:
+    case CSF_LOG_DEBUG:
+    case CSF_LOG_VERBOSE:
+      break;
+  }
+
+  return static_cast<CSFLogLevel>(priority);
+}
+
+void SDPLog(SDPLogLevel priority, const char* sourceFile, int sourceLine,
+            const char* tag, const char* format, ...) {
+  va_list ap;
+  va_start(ap, format);
+  CSFLogV(SDPToCSFLogLevel(priority), sourceFile, sourceLine, tag, format, ap);
+  va_end(ap);
+}
+
+void SDPLogV(SDPLogLevel priority, const char* sourceFile, int sourceLine,
+             const char* tag, const char* format, va_list args) {
+  CSFLogV(SDPToCSFLogLevel(priority), sourceFile, sourceLine, tag, format,
+          args);
+}
+
+int SDPLogTestLevel(SDPLogLevel priority) {
+  return CSFLogTestLevel(SDPToCSFLogLevel(priority));
+}
diff --git a/dom/media/webrtc/sdp/SdpLog.h b/dom/media/webrtc/sdp/SdpLog.h
new file mode 100644
index 0000000000..63bceff889
--- /dev/null
+++ b/dom/media/webrtc/sdp/SdpLog.h
@@ -0,0 +1,17 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _SDP_LOG_H_
+#define _SDP_LOG_H_
+
+#include "mozilla/Logging.h"
+#include "sdp_log.h"
+
+namespace mozilla {
+extern mozilla::LazyLogModule SdpLog;
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/sdp/SdpMediaSection.cpp b/dom/media/webrtc/sdp/SdpMediaSection.cpp
new file mode 100644
index 0000000000..dd8f6e54fe
--- /dev/null
+++ b/dom/media/webrtc/sdp/SdpMediaSection.cpp
@@ -0,0 +1,197 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "sdp/SdpMediaSection.h"
+
+namespace mozilla {
+const SdpFmtpAttributeList::Parameters* SdpMediaSection::FindFmtp(
+    const std::string& pt) const {
+  const SdpAttributeList& attrs = GetAttributeList();
+
+  if (attrs.HasAttribute(SdpAttribute::kFmtpAttribute)) {
+    for (auto& fmtpAttr : attrs.GetFmtp().mFmtps) {
+      if (fmtpAttr.format == pt && fmtpAttr.parameters) {
+        return fmtpAttr.parameters.get();
+      }
+    }
+  }
+  return nullptr;
+}
+
+void SdpMediaSection::SetFmtp(const SdpFmtpAttributeList::Fmtp& fmtpToSet) {
+  UniquePtr<SdpFmtpAttributeList> fmtps(new SdpFmtpAttributeList);
+
+  if (GetAttributeList().HasAttribute(SdpAttribute::kFmtpAttribute)) {
+    *fmtps = GetAttributeList().GetFmtp();
+  }
+
+  bool found = false;
+  for (SdpFmtpAttributeList::Fmtp& fmtp : fmtps->mFmtps) {
+    if (fmtp.format == fmtpToSet.format) {
+      fmtp = fmtpToSet;
+      found = true;
+    }
+  }
+
+  if (!found) {
+    fmtps->mFmtps.push_back(fmtpToSet);
+  }
+
+  GetAttributeList().SetAttribute(fmtps.release());
+}
+
+void SdpMediaSection::RemoveFmtp(const std::string& pt) {
+  UniquePtr<SdpFmtpAttributeList> fmtps(new SdpFmtpAttributeList);
+
+  SdpAttributeList& attrList = GetAttributeList();
+  if (attrList.HasAttribute(SdpAttribute::kFmtpAttribute)) {
+    *fmtps = attrList.GetFmtp();
+  }
+
+  for (size_t i = 0; i < fmtps->mFmtps.size(); ++i) {
+    if (pt == fmtps->mFmtps[i].format) {
+      fmtps->mFmtps.erase(fmtps->mFmtps.begin() + i);
+      break;
+    }
+  }
+
+  attrList.SetAttribute(fmtps.release());
+}
+
+const SdpRtpmapAttributeList::Rtpmap* SdpMediaSection::FindRtpmap(
+    const std::string& pt) const {
+  auto& attrs = GetAttributeList();
+  if (!attrs.HasAttribute(SdpAttribute::kRtpmapAttribute)) {
+    return nullptr;
+  }
+
+  const SdpRtpmapAttributeList& rtpmap = attrs.GetRtpmap();
+  if (!rtpmap.HasEntry(pt)) {
+    return nullptr;
+  }
+
+  return &rtpmap.GetEntry(pt);
+}
+
+const SdpSctpmapAttributeList::Sctpmap* SdpMediaSection::GetSctpmap() const {
+  auto& attrs = GetAttributeList();
+  if (!attrs.HasAttribute(SdpAttribute::kSctpmapAttribute)) {
+    return nullptr;
+  }
+
+  const SdpSctpmapAttributeList& sctpmap = attrs.GetSctpmap();
+  if (sctpmap.mSctpmaps.empty()) {
+    return nullptr;
+  }
+
+  return &sctpmap.GetFirstEntry();
+}
+
+uint32_t SdpMediaSection::GetSctpPort() const {
+  auto& attrs = GetAttributeList();
+  if (!attrs.HasAttribute(SdpAttribute::kSctpPortAttribute)) {
+    return 0;
+  }
+
+  return attrs.GetSctpPort();
+}
+
+bool SdpMediaSection::GetMaxMessageSize(uint32_t* size) const {
+  *size = 0;
+
+  auto& attrs = GetAttributeList();
+  if (!attrs.HasAttribute(SdpAttribute::kMaxMessageSizeAttribute)) {
+    return false;
+  }
+
+  *size = attrs.GetMaxMessageSize();
+  return true;
+}
+
+bool SdpMediaSection::HasRtcpFb(const std::string& pt,
+                                SdpRtcpFbAttributeList::Type type,
+                                const std::string& subType) const {
+  const SdpAttributeList& attrs(GetAttributeList());
+
+  if (!attrs.HasAttribute(SdpAttribute::kRtcpFbAttribute)) {
+    return false;
+  }
+
+  for (auto& rtcpfb : attrs.GetRtcpFb().mFeedbacks) {
+    if (rtcpfb.type == type) {
+      if (rtcpfb.pt == "*" || rtcpfb.pt == pt) {
+        if (rtcpfb.parameter == subType) {
+          return true;
+        }
+      }
+    }
+  }
+
+  return false;
+}
+
+SdpRtcpFbAttributeList SdpMediaSection::GetRtcpFbs() const {
+  SdpRtcpFbAttributeList result;
+  if (GetAttributeList().HasAttribute(SdpAttribute::kRtcpFbAttribute)) {
+    result = GetAttributeList().GetRtcpFb();
+  }
+  return result;
+}
+
+void SdpMediaSection::SetRtcpFbs(const SdpRtcpFbAttributeList& rtcpfbs) {
+  if (rtcpfbs.mFeedbacks.empty()) {
+    GetAttributeList().RemoveAttribute(SdpAttribute::kRtcpFbAttribute);
+    return;
+  }
+
+  GetAttributeList().SetAttribute(new SdpRtcpFbAttributeList(rtcpfbs));
+}
+
+void SdpMediaSection::SetSsrcs(const std::vector<uint32_t>& ssrcs,
+                               const std::string& cname) {
+  if (ssrcs.empty()) {
+    GetAttributeList().RemoveAttribute(SdpAttribute::kSsrcAttribute);
+    return;
+  }
+
+  UniquePtr<SdpSsrcAttributeList> ssrcAttr(new SdpSsrcAttributeList);
+  for (auto ssrc : ssrcs) {
+    // When using ssrc attributes, we are required to at least have a cname.
+    // (See https://tools.ietf.org/html/rfc5576#section-6.1)
+    std::string cnameAttr("cname:");
+    cnameAttr += cname;
+    ssrcAttr->PushEntry(ssrc, cnameAttr);
+  }
+
+  GetAttributeList().SetAttribute(ssrcAttr.release());
+}
+
+void SdpMediaSection::AddMsid(const std::string& id,
+                              const std::string& appdata) {
+  UniquePtr<SdpMsidAttributeList> msids(new SdpMsidAttributeList);
+  if (GetAttributeList().HasAttribute(SdpAttribute::kMsidAttribute)) {
+    msids->mMsids = GetAttributeList().GetMsid().mMsids;
+  }
+  msids->PushEntry(id, appdata);
+  GetAttributeList().SetAttribute(msids.release());
+}
+
+const SdpRidAttributeList::Rid* SdpMediaSection::FindRid(
+    const std::string& id) const {
+  if (!GetAttributeList().HasAttribute(SdpAttribute::kRidAttribute)) {
+    return nullptr;
+  }
+
+  for (const auto& rid : GetAttributeList().GetRid().mRids) {
+    if (rid.id == id) {
+      return &rid;
+    }
+  }
+
+  return nullptr;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/sdp/SdpMediaSection.h b/dom/media/webrtc/sdp/SdpMediaSection.h
new file mode 100644
index 0000000000..a205551a1d
--- /dev/null
+++ b/dom/media/webrtc/sdp/SdpMediaSection.h
@@ -0,0 +1,317 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _SDPMEDIASECTION_H_
+#define _SDPMEDIASECTION_H_
+
+#include "mozilla/Maybe.h"
+#include "sdp/SdpEnum.h"
+#include "sdp/SdpAttributeList.h"
+#include <string>
+#include <vector>
+#include <sstream>
+
+namespace mozilla {
+
+class SdpAttributeList;
+
+class SdpConnection;
+
+class SdpMediaSection {
+ public:
+  enum MediaType { kAudio, kVideo, kText, kApplication, kMessage };
+  // don't add to enum to avoid warnings about unhandled enum values
+  static const size_t kMediaTypes = static_cast<size_t>(kMessage) + 1;
+
+  enum Protocol {
+    kRtpAvp,             // RTP/AVP [RFC4566]
+    kUdp,                // udp [RFC4566]
+    kVat,                // vat [historic]
+    kRtp,                // rtp [historic]
+    kUdptl,              // udptl [ITU-T]
+    kTcp,                // TCP [RFC4145]
+    kRtpAvpf,            // RTP/AVPF [RFC4585]
+    kTcpRtpAvp,          // TCP/RTP/AVP [RFC4571]
+    kRtpSavp,            // RTP/SAVP [RFC3711]
+    kTcpBfcp,            // TCP/BFCP [RFC4583]
+    kTcpTlsBfcp,         // TCP/TLS/BFCP [RFC4583]
+    kTcpTls,             // TCP/TLS [RFC4572]
+    kFluteUdp,           // FLUTE/UDP [RFC-mehta-rmt-flute-sdp-05]
+    kTcpMsrp,            // TCP/MSRP [RFC4975]
+    kTcpTlsMsrp,         // TCP/TLS/MSRP [RFC4975]
+    kDccp,               // DCCP [RFC5762]
+    kDccpRtpAvp,         // DCCP/RTP/AVP [RFC5762]
+    kDccpRtpSavp,        // DCCP/RTP/SAVP [RFC5762]
+    kDccpRtpAvpf,        // DCCP/RTP/AVPF [RFC5762]
+    kDccpRtpSavpf,       // DCCP/RTP/SAVPF [RFC5762]
+    kRtpSavpf,           // RTP/SAVPF [RFC5124]
+    kUdpTlsRtpSavp,      // UDP/TLS/RTP/SAVP [RFC5764]
+    kTcpDtlsRtpSavp,     // TCP/DTLS/RTP/SAVP [RFC7850]
+    kDccpTlsRtpSavp,     // DCCP/TLS/RTP/SAVP [RFC5764]
+    kUdpTlsRtpSavpf,     // UDP/TLS/RTP/SAVPF [RFC5764]
+    kTcpDtlsRtpSavpf,    // TCP/DTLS/RTP/SAVPF [RFC7850]
+    kDccpTlsRtpSavpf,    // DCCP/TLS/RTP/SAVPF [RFC5764]
+    kUdpMbmsFecRtpAvp,   // UDP/MBMS-FEC/RTP/AVP [RFC6064]
+    kUdpMbmsFecRtpSavp,  // UDP/MBMS-FEC/RTP/SAVP [RFC6064]
+    kUdpMbmsRepair,      // UDP/MBMS-REPAIR [RFC6064]
+    kFecUdp,             // FEC/UDP [RFC6364]
+    kUdpFec,             // UDP/FEC [RFC6364]
+    kTcpMrcpv2,          // TCP/MRCPv2 [RFC6787]
+    kTcpTlsMrcpv2,       // TCP/TLS/MRCPv2 [RFC6787]
+    kPstn,               // PSTN [RFC7195]
+    kUdpTlsUdptl,        // UDP/TLS/UDPTL [RFC7345]
+    kSctp,               // SCTP [draft-ietf-mmusic-sctp-sdp-07]
+    kDtlsSctp,           // DTLS/SCTP [draft-ietf-mmusic-sctp-sdp-07]
+    kUdpDtlsSctp,        // UDP/DTLS/SCTP [draft-ietf-mmusic-sctp-sdp-21]
+    kTcpDtlsSctp         // TCP/DTLS/SCTP [draft-ietf-mmusic-sctp-sdp-21]
+  };
+
+  explicit SdpMediaSection(size_t level) : mLevel(level) {}
+
+  virtual MediaType GetMediaType() const = 0;
+  virtual unsigned int GetPort() const = 0;
+  virtual void SetPort(unsigned int port) = 0;
+  virtual unsigned int GetPortCount() const = 0;
+  virtual Protocol GetProtocol() const = 0;
+  virtual const SdpConnection& GetConnection() const = 0;
+  virtual SdpConnection& GetConnection() = 0;
+  virtual uint32_t GetBandwidth(const std::string& type) const = 0;
+  virtual const std::vector<std::string>& GetFormats() const = 0;
+
+  std::vector<std::string> GetFormatsForSimulcastVersion(
+      size_t simulcastVersion, bool send, bool recv) const;
+  virtual const SdpAttributeList& GetAttributeList() const = 0;
+  virtual SdpAttributeList& GetAttributeList() = 0;
+
+  virtual SdpDirectionAttribute GetDirectionAttribute() const = 0;
+
+  virtual void Serialize(std::ostream&) const = 0;
+
+  virtual void AddCodec(const std::string& pt, const std::string& name,
+                        uint32_t clockrate, uint16_t channels) = 0;
+  virtual void ClearCodecs() = 0;
+
+  virtual void AddDataChannel(const std::string& name, uint16_t port,
+                              uint16_t streams, uint32_t message_size) = 0;
+
+  size_t GetLevel() const { return mLevel; }
+
+  inline bool IsReceiving() const { return GetDirection() & sdp::kRecv; }
+
+  inline bool IsSending() const { return GetDirection() & sdp::kSend; }
+
+  inline void SetReceiving(bool receiving) {
+    auto direction = GetDirection();
+    if (direction & sdp::kSend) {
+      SetDirection(receiving ? SdpDirectionAttribute::kSendrecv
+                             : SdpDirectionAttribute::kSendonly);
+    } else {
+      SetDirection(receiving ? SdpDirectionAttribute::kRecvonly
+                             : SdpDirectionAttribute::kInactive);
+    }
+  }
+
+  inline void SetSending(bool sending) {
+    auto direction = GetDirection();
+    if (direction & sdp::kRecv) {
+      SetDirection(sending ? SdpDirectionAttribute::kSendrecv
+                           : SdpDirectionAttribute::kRecvonly);
+    } else {
+      SetDirection(sending ? SdpDirectionAttribute::kSendonly
+                           : SdpDirectionAttribute::kInactive);
+    }
+  }
+
+  inline void SetDirection(SdpDirectionAttribute::Direction direction) {
+    GetAttributeList().SetAttribute(new SdpDirectionAttribute(direction));
+  }
+
+  inline SdpDirectionAttribute::Direction GetDirection() const {
+    return GetDirectionAttribute().mValue;
+  }
+
+  const SdpFmtpAttributeList::Parameters* FindFmtp(const std::string& pt) const;
+  void SetFmtp(const SdpFmtpAttributeList::Fmtp& fmtp);
+  void RemoveFmtp(const std::string& pt);
+  const SdpRtpmapAttributeList::Rtpmap* FindRtpmap(const std::string& pt) const;
+  const SdpSctpmapAttributeList::Sctpmap* GetSctpmap() const;
+  uint32_t GetSctpPort() const;
+  bool GetMaxMessageSize(uint32_t* size) const;
+  bool HasRtcpFb(const std::string& pt, SdpRtcpFbAttributeList::Type type,
+                 const std::string& subType) const;
+  SdpRtcpFbAttributeList GetRtcpFbs() const;
+  void SetRtcpFbs(const SdpRtcpFbAttributeList& rtcpfbs);
+  bool HasFormat(const std::string& format) const {
+    return std::find(GetFormats().begin(), GetFormats().end(), format) !=
+           GetFormats().end();
+  }
+  void SetSsrcs(const std::vector<uint32_t>& ssrcs, const std::string& cname);
+  void AddMsid(const std::string& id, const std::string& appdata);
+  const SdpRidAttributeList::Rid* FindRid(const std::string& id) const;
+
+ private:
+  size_t mLevel;
+};
+
+inline std::ostream& operator<<(std::ostream& os, const SdpMediaSection& ms) {
+  ms.Serialize(os);
+  return os;
+}
+
+inline std::ostream& operator<<(std::ostream& os,
+                                SdpMediaSection::MediaType t) {
+  switch (t) {
+    case SdpMediaSection::kAudio:
+      return os << "audio";
+    case SdpMediaSection::kVideo:
+      return os << "video";
+    case SdpMediaSection::kText:
+      return os << "text";
+    case SdpMediaSection::kApplication:
+      return os << "application";
+    case SdpMediaSection::kMessage:
+      return os << "message";
+  }
+  MOZ_ASSERT(false, "Unknown MediaType");
+  return os << "?";
+}
+
+inline std::ostream& operator<<(std::ostream& os, SdpMediaSection::Protocol p) {
+  switch (p) {
+    case SdpMediaSection::kRtpAvp:
+      return os << "RTP/AVP";
+    case SdpMediaSection::kUdp:
+      return os << "udp";
+    case SdpMediaSection::kVat:
+      return os << "vat";
+    case SdpMediaSection::kRtp:
+      return os << "rtp";
+    case SdpMediaSection::kUdptl:
+      return os << "udptl";
+    case SdpMediaSection::kTcp:
+      return os << "TCP";
+    case SdpMediaSection::kRtpAvpf:
+      return os << "RTP/AVPF";
+    case SdpMediaSection::kTcpRtpAvp:
+      return os << "TCP/RTP/AVP";
+    case SdpMediaSection::kRtpSavp:
+      return os << "RTP/SAVP";
+    case SdpMediaSection::kTcpBfcp:
+      return os << "TCP/BFCP";
+    case SdpMediaSection::kTcpTlsBfcp:
+      return os << "TCP/TLS/BFCP";
+    case SdpMediaSection::kTcpTls:
+      return os << "TCP/TLS";
+    case SdpMediaSection::kFluteUdp:
+      return os << "FLUTE/UDP";
+    case SdpMediaSection::kTcpMsrp:
+      return os << "TCP/MSRP";
+    case SdpMediaSection::kTcpTlsMsrp:
+      return os << "TCP/TLS/MSRP";
+    case SdpMediaSection::kDccp:
+      return os << "DCCP";
+    case SdpMediaSection::kDccpRtpAvp:
+      return os << "DCCP/RTP/AVP";
+    case SdpMediaSection::kDccpRtpSavp:
+      return os << "DCCP/RTP/SAVP";
+    case SdpMediaSection::kDccpRtpAvpf:
+      return os << "DCCP/RTP/AVPF";
+    case SdpMediaSection::kDccpRtpSavpf:
+      return os << "DCCP/RTP/SAVPF";
+    case SdpMediaSection::kRtpSavpf:
+      return os << "RTP/SAVPF";
+    case SdpMediaSection::kUdpTlsRtpSavp:
+      return os << "UDP/TLS/RTP/SAVP";
+    case SdpMediaSection::kTcpDtlsRtpSavp:
+      return os << "TCP/DTLS/RTP/SAVP";
+    case SdpMediaSection::kDccpTlsRtpSavp:
+      return os << "DCCP/TLS/RTP/SAVP";
+    case SdpMediaSection::kUdpTlsRtpSavpf:
+      return os << "UDP/TLS/RTP/SAVPF";
+    case SdpMediaSection::kTcpDtlsRtpSavpf:
+      return os << "TCP/DTLS/RTP/SAVPF";
+    case SdpMediaSection::kDccpTlsRtpSavpf:
+      return os << "DCCP/TLS/RTP/SAVPF";
+    case SdpMediaSection::kUdpMbmsFecRtpAvp:
+      return os << "UDP/MBMS-FEC/RTP/AVP";
+    case SdpMediaSection::kUdpMbmsFecRtpSavp:
+      return os << "UDP/MBMS-FEC/RTP/SAVP";
+    case SdpMediaSection::kUdpMbmsRepair:
+      return os << "UDP/MBMS-REPAIR";
+    case SdpMediaSection::kFecUdp:
+      return os << "FEC/UDP";
+    case SdpMediaSection::kUdpFec:
+      return os << "UDP/FEC";
+    case SdpMediaSection::kTcpMrcpv2:
+      return os << "TCP/MRCPv2";
+    case SdpMediaSection::kTcpTlsMrcpv2:
+      return os << "TCP/TLS/MRCPv2";
+    case SdpMediaSection::kPstn:
+      return os << "PSTN";
+    case SdpMediaSection::kUdpTlsUdptl:
+      return os << "UDP/TLS/UDPTL";
+    case SdpMediaSection::kSctp:
+      return os << "SCTP";
+    case SdpMediaSection::kDtlsSctp:
+      return os << "DTLS/SCTP";
+    case SdpMediaSection::kUdpDtlsSctp:
+      return os << "UDP/DTLS/SCTP";
+    case SdpMediaSection::kTcpDtlsSctp:
+      return os << "TCP/DTLS/SCTP";
+  }
+  MOZ_ASSERT(false, "Unknown Protocol");
+  return os << "?";
+}
+
+class SdpConnection {
+ public:
+  SdpConnection(sdp::AddrType addrType, std::string addr, uint8_t ttl = 0,
+                uint32_t count = 0)
+      : mAddrType(addrType), mAddr(addr), mTtl(ttl), mCount(count) {}
+  ~SdpConnection() {}
+
+  sdp::AddrType GetAddrType() const { return mAddrType; }
+  const std::string& GetAddress() const { return mAddr; }
+  void SetAddress(const std::string& address) {
+    mAddr = address;
+    if (mAddr.find(':') != std::string::npos) {
+      mAddrType = sdp::kIPv6;
+    } else {
+      mAddrType = sdp::kIPv4;
+    }
+  }
+  uint8_t GetTtl() const { return mTtl; }
+  uint32_t GetCount() const { return mCount; }
+
+  void Serialize(std::ostream& os) const {
+    sdp::NetType netType = sdp::kInternet;
+
+    os << "c=" << netType << " " << mAddrType << " " << mAddr;
+
+    if (mTtl) {
+      os << "/" << static_cast<uint32_t>(mTtl);
+      if (mCount) {
+        os << "/" << mCount;
+      }
+    }
+    os << "\r\n";
+  }
+
+ private:
+  sdp::AddrType mAddrType;
+  std::string mAddr;
+  uint8_t mTtl;     // 0-255; 0 when unset
+  uint32_t mCount;  // 0 when unset
+};
+
+inline std::ostream& operator<<(std::ostream& os, const SdpConnection& c) {
+  c.Serialize(os);
+  return os;
+}
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/sdp/SdpParser.h b/dom/media/webrtc/sdp/SdpParser.h
new file mode 100644
index 0000000000..4b23e93705
--- /dev/null
+++ b/dom/media/webrtc/sdp/SdpParser.h
@@ -0,0 +1,81 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _SDPPARSER_H_
+#define _SDPPARSER_H_
+
+#include <vector>
+#include <string>
+#include "sdp/Sdp.h"
+#include "sdp/SdpLog.h"
+
+#include "mozilla/Telemetry.h"
+
+namespace mozilla {
+
+class SdpParser {
+ public:
+  SdpParser() = default;
+  virtual ~SdpParser() = default;
+
+  class Results {
+   public:
+    typedef std::pair<size_t, std::string> Anomaly;
+    typedef std::vector<Anomaly> AnomalyVec;
+    virtual ~Results() = default;
+    UniquePtr<mozilla::Sdp>& Sdp() { return mSdp; }
+    AnomalyVec& Errors() { return mErrors; }
+    AnomalyVec& Warnings() { return mWarnings; }
+    virtual const std::string& ParserName() const = 0;
+    bool Ok() const { return mErrors.empty(); }
+
+   protected:
+    UniquePtr<mozilla::Sdp> mSdp;
+    AnomalyVec mErrors;
+    AnomalyVec mWarnings;
+  };
+
+  // The name of the parser implementation
+  virtual const std::string& Name() const = 0;
+
+  /**
+   * This parses the provided text into an SDP object.
+   * This returns a nullptr-valued pointer if things go poorly.
+   */
+  virtual UniquePtr<SdpParser::Results> Parse(const std::string& aText) = 0;
+
+  class InternalResults : public Results {
+   public:
+    explicit InternalResults(const std::string& aParserName)
+        : mParserName(aParserName) {}
+    virtual ~InternalResults() = default;
+
+    void SetSdp(UniquePtr<mozilla::Sdp>&& aSdp) { mSdp = std::move(aSdp); }
+
+    void AddParseError(size_t line, const std::string& message) {
+      MOZ_LOG(SdpLog, LogLevel::Error,
+              ("%s: parser error %s, at line %zu", mParserName.c_str(),
+               message.c_str(), line));
+      mErrors.push_back(std::make_pair(line, message));
+    }
+
+    void AddParseWarning(size_t line, const std::string& message) {
+      MOZ_LOG(SdpLog, LogLevel::Warning,
+              ("%s: parser warning %s, at line %zu", mParserName.c_str(),
+               message.c_str(), line));
+      mWarnings.push_back(std::make_pair(line, message));
+    }
+
+    const std::string& ParserName() const override { return mParserName; }
+
+   private:
+    const std::string mParserName;
+  };
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/sdp/SdpPref.cpp b/dom/media/webrtc/sdp/SdpPref.cpp
new file mode 100644
index 0000000000..d0fc03c4d0
--- /dev/null
+++ b/dom/media/webrtc/sdp/SdpPref.cpp
@@ -0,0 +1,107 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "sdp/SdpPref.h"
+#include "sdp/RsdparsaSdpParser.h"
+#include "sdp/SipccSdpParser.h"
+
+namespace mozilla {
+
+const std::string SdpPref::PRIMARY_PREF = "media.peerconnection.sdp.parser";
+const std::string SdpPref::ALTERNATE_PREF =
+    "media.peerconnection.sdp.alternate_parse_mode";
+const std::string SdpPref::STRICT_SUCCESS_PREF =
+    "media.peerconnection.sdp.strict_success";
+const std::string SdpPref::DEFAULT = "default";
+
+auto SdpPref::ToString(const Parsers& aParser) -> std::string {
+  switch (aParser) {
+    case Parsers::Sipcc:
+      return "sipcc";
+    case Parsers::WebRtcSdp:
+      return "webrtc-sdp";
+  };
+  MOZ_CRASH("ALL Parsers CASES ARE NOT COVERED");
+  return "";
+}
+
+auto SdpPref::ToString(const AlternateParseModes& aMode) -> std::string {
+  switch (aMode) {
+    case AlternateParseModes::Parallel:
+      return "parallel";
+    case AlternateParseModes::Failover:
+      return "failover";
+    case AlternateParseModes::Never:
+      return "never";
+  };
+  MOZ_CRASH("ALL AlternateParseModes CASES ARE NOT COVERED");
+  return "";
+}
+
+auto SdpPref::Parser() -> Parsers {
+  static const auto values = std::unordered_map<std::string, Parsers>{
+      {"sipcc", Parsers::Sipcc},
+      {"webrtc-sdp", Parsers::WebRtcSdp},
+      {DEFAULT, Parsers::Sipcc},
+  };
+  return Pref(PRIMARY_PREF, values);
+}
+
+auto SdpPref::AlternateParseMode() -> AlternateParseModes {
+  static const auto values =
+      std::unordered_map<std::string, AlternateParseModes>{
+          {"parallel", AlternateParseModes::Parallel},
+          {"failover", AlternateParseModes::Failover},
+          {"never", AlternateParseModes::Never},
+          {DEFAULT, AlternateParseModes::Parallel},
+      };
+  return Pref(ALTERNATE_PREF, values);
+}
+
+auto SdpPref::Primary() -> UniquePtr<SdpParser> {
+  switch (Parser()) {
+    case Parsers::Sipcc:
+      return UniquePtr<SdpParser>(new SipccSdpParser());
+    case Parsers::WebRtcSdp:
+      return UniquePtr<SdpParser>(new RsdparsaSdpParser());
+  }
+  MOZ_CRASH("ALL Parsers CASES ARE NOT COVERED");
+  return nullptr;
+}
+
+auto SdpPref::Secondary() -> Maybe<UniquePtr<SdpParser>> {
+  if (AlternateParseMode() != AlternateParseModes::Parallel) {
+    return Nothing();
+  }
+  switch (Parser()) {  // Choose whatever the primary parser isn't
+    case Parsers::Sipcc:
+      return Some(UniquePtr<SdpParser>(new RsdparsaSdpParser()));
+    case Parsers::WebRtcSdp:
+      return Some(UniquePtr<SdpParser>(new SipccSdpParser()));
+  }
+  MOZ_CRASH("ALL Parsers CASES ARE NOT COVERED");
+  return Nothing();
+}
+
+auto SdpPref::Failover() -> Maybe<UniquePtr<SdpParser>> {
+  if (AlternateParseMode() != AlternateParseModes::Failover) {
+    return Nothing();
+  }
+  switch (Parser()) {
+    case Parsers::Sipcc:
+      return Some(UniquePtr<SdpParser>(new RsdparsaSdpParser()));
+    case Parsers::WebRtcSdp:
+      return Some(UniquePtr<SdpParser>(new SipccSdpParser()));
+  }
+  MOZ_CRASH("ALL Parsers CASES ARE NOT COVERED");
+  return Nothing();
+}
+
+auto SdpPref::StrictSuccess() -> bool {
+  return Preferences::GetBool(STRICT_SUCCESS_PREF.c_str(), false);
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/sdp/SdpPref.h b/dom/media/webrtc/sdp/SdpPref.h
new file mode 100644
index 0000000000..5f24fb12a7
--- /dev/null
+++ b/dom/media/webrtc/sdp/SdpPref.h
@@ -0,0 +1,82 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _SDPPREF_H_
+#define _SDPPREF_H_
+
+#include "mozilla/Maybe.h"
+#include "mozilla/Preferences.h"
+
+#include <string>
+#include <unordered_map>
+
+namespace mozilla {
+
+class SdpParser;
+
+// Interprets about:config SDP parsing preferences
+class SdpPref {
+ private:
+  static const std::string PRIMARY_PREF;
+  static const std::string ALTERNATE_PREF;
+  static const std::string STRICT_SUCCESS_PREF;
+  static const std::string DEFAULT;
+
+ public:
+  // Supported Parsers
+  enum class Parsers {
+    Sipcc,
+    WebRtcSdp,
+  };
+  static auto ToString(const Parsers& aParser) -> std::string;
+
+  // How is the alternate used
+  enum class AlternateParseModes {
+    Parallel,  // Alternate is always run, if A succedes it is used, otherwise B
+               // is used
+    Failover,  // Alternate is only run on failure of the primary to parse
+    Never,     // Alternate is never run; this is effectively a kill switch
+  };
+  static auto ToString(const AlternateParseModes& aMode) -> std::string;
+
+ private:
+  // Finds the mapping between a pref string and pref value, if none exists the
+  // default is used
+  template <class T>
+  static auto Pref(const std::string& aPrefName,
+                   const std::unordered_map<std::string, T>& aMap) -> T {
+    MOZ_ASSERT(aMap.find(DEFAULT) != aMap.end());
+
+    nsCString value;
+    if (NS_FAILED(Preferences::GetCString(aPrefName.c_str(), value))) {
+      return aMap.at(DEFAULT);
+    }
+    const auto found = aMap.find(value.get());
+    if (found != aMap.end()) {
+      return found->second;
+    }
+    return aMap.at(DEFAULT);
+  }
+  // The value of the parser pref
+  static auto Parser() -> Parsers;
+
+  // The value of the alternate parse mode pref
+  static auto AlternateParseMode() -> AlternateParseModes;
+
+ public:
+  // Do non-fatal parsing errors count as failure
+  static auto StrictSuccess() -> bool;
+  // Functions to create the primary, secondary and failover parsers.
+
+  // Reads about:config to choose the primary Parser
+  static auto Primary() -> UniquePtr<SdpParser>;
+  static auto Secondary() -> Maybe<UniquePtr<SdpParser>>;
+  static auto Failover() -> Maybe<UniquePtr<SdpParser>>;
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/sdp/SdpTelemetry.cpp b/dom/media/webrtc/sdp/SdpTelemetry.cpp
new file mode 100644
index 0000000000..7b3a6cec18
--- /dev/null
+++ b/dom/media/webrtc/sdp/SdpTelemetry.cpp
@@ -0,0 +1,63 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "sdp/SdpTelemetry.h"
+
+#include "mozilla/Telemetry.h"
+
+namespace mozilla {
+
+auto SdpTelemetry::RecordParse(const SdpTelemetry::Results& aResult,
+                               const SdpTelemetry::Modes& aMode,
+                               const SdpTelemetry::Roles& aRole) -> void {
+  Telemetry::ScalarAdd(Telemetry::ScalarID::WEBRTC_SDP_PARSER_DIFF,
+                       BucketNameFragment(aResult, aMode, aRole), 1);
+}
+
+auto SdpTelemetry::RecordCompare(const SdpTelemetry::Results& aFirst,
+                                 const SdpTelemetry::Results& aSecond,
+                                 const SdpTelemetry::Modes& aMode) -> void {
+  const nsAutoString bucket =
+      BucketNameFragment(aFirst, aMode, Roles::Primary) +
+      NS_ConvertASCIItoUTF16("__") +
+      BucketNameFragment(aSecond, aMode, Roles::Secondary);
+  Telemetry::ScalarAdd(Telemetry::ScalarID::WEBRTC_SDP_PARSER_DIFF, bucket, 1);
+}
+
+auto SdpTelemetry::BucketNameFragment(const SdpTelemetry::Results& aResult,
+                                      const SdpTelemetry::Modes& aMode,
+                                      const SdpTelemetry::Roles& aRole)
+    -> nsAutoString {
+  auto mode = [&]() -> std::string {
+    switch (aMode) {
+      case Modes::Parallel:
+        return "parallel";
+      case Modes::Failover:
+        return "failover";
+      case Modes::Never:
+        return "standalone";
+    }
+    MOZ_CRASH("Unknown SDP Parse Mode!");
+  };
+  auto role = [&]() -> std::string {
+    switch (aRole) {
+      case Roles::Primary:
+        return "primary";
+      case Roles::Secondary:
+        return "secondary";
+    }
+    MOZ_CRASH("Unknown SDP Parse Role!");
+  };
+  auto success = [&]() -> std::string {
+    return aResult->Ok() ? "success" : "failure";
+  };
+  nsAutoString name;
+  name.AssignASCII(nsCString(aResult->ParserName() + "_" + mode() + "_" +
+                             role() + "_" + success()));
+  return name;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/sdp/SdpTelemetry.h b/dom/media/webrtc/sdp/SdpTelemetry.h
new file mode 100644
index 0000000000..ce308a04ec
--- /dev/null
+++ b/dom/media/webrtc/sdp/SdpTelemetry.h
@@ -0,0 +1,43 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _SDPTELEMETRY_H_
+#define _SDPTELEMETRY_H_
+
+#include "sdp/SdpParser.h"
+#include "sdp/SdpPref.h"
+
+namespace mozilla {
+
+class SdpTelemetry {
+ public:
+  SdpTelemetry() = delete;
+
+  using Results = UniquePtr<SdpParser::Results>;
+  using Modes = SdpPref::AlternateParseModes;
+
+  enum class Roles {
+    Primary,
+    Secondary,
+  };
+
+  static auto RecordParse(const Results& aResults, const Modes& aMode,
+                          const Roles& aRole) -> void;
+
+  static auto RecordSecondaryParse(const Results& aResult, const Modes& aMode)
+      -> void;
+
+  static auto RecordCompare(const Results& aFirst, const Results& aSecond,
+                            const Modes& aMode) -> void;
+
+ private:
+  static auto BucketNameFragment(const Results& aResult, const Modes& aModes,
+                                 const Roles& aRoles) -> nsAutoString;
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/sdp/SipccSdp.cpp b/dom/media/webrtc/sdp/SipccSdp.cpp
new file mode 100644
index 0000000000..7e36f9e930
--- /dev/null
+++ b/dom/media/webrtc/sdp/SipccSdp.cpp
@@ -0,0 +1,173 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "sdp/SipccSdp.h"
+
+#include <cstdlib>
+#include "mozilla/UniquePtr.h"
+#include "mozilla/Assertions.h"
+#include "sdp/SdpParser.h"
+
+#ifdef CRLF
+#  undef CRLF
+#endif
+#define CRLF "\r\n"
+
+namespace mozilla {
+
+SipccSdp::SipccSdp(const SipccSdp& aOrig)
+    : mOrigin(aOrig.mOrigin),
+      mBandwidths(aOrig.mBandwidths),
+      mAttributeList(aOrig.mAttributeList, nullptr) {
+  for (const auto& msection : aOrig.mMediaSections) {
+    mMediaSections.emplace_back(
+        new SipccSdpMediaSection(*msection, &mAttributeList));
+  }
+}
+
+Sdp* SipccSdp::Clone() const { return new SipccSdp(*this); }
+
+const SdpOrigin& SipccSdp::GetOrigin() const { return mOrigin; }
+
+uint32_t SipccSdp::GetBandwidth(const std::string& type) const {
+  auto found = mBandwidths.find(type);
+  if (found == mBandwidths.end()) {
+    return 0;
+  }
+  return found->second;
+}
+
+const SdpMediaSection& SipccSdp::GetMediaSection(size_t level) const {
+  if (level > mMediaSections.size()) {
+    MOZ_CRASH();
+  }
+  return *mMediaSections[level];
+}
+
+SdpMediaSection& SipccSdp::GetMediaSection(size_t level) {
+  if (level > mMediaSections.size()) {
+    MOZ_CRASH();
+  }
+  return *mMediaSections[level];
+}
+
+SdpMediaSection& SipccSdp::AddMediaSection(SdpMediaSection::MediaType mediaType,
+                                           SdpDirectionAttribute::Direction dir,
+                                           uint16_t port,
+                                           SdpMediaSection::Protocol protocol,
+                                           sdp::AddrType addrType,
+                                           const std::string& addr) {
+  size_t level = mMediaSections.size();
+  SipccSdpMediaSection* media =
+      new SipccSdpMediaSection(level, &mAttributeList);
+  media->mMediaType = mediaType;
+  media->mPort = port;
+  media->mPortCount = 0;
+  media->mProtocol = protocol;
+  media->mConnection = MakeUnique<SdpConnection>(addrType, addr);
+  media->GetAttributeList().SetAttribute(new SdpDirectionAttribute(dir));
+  mMediaSections.emplace_back(media);
+  return *media;
+}
+
+bool SipccSdp::LoadOrigin(sdp_t* sdp, InternalResults& results) {
+  std::string username = sdp_get_owner_username(sdp);
+  uint64_t sessId = strtoull(sdp_get_owner_sessionid(sdp), nullptr, 10);
+  uint64_t sessVer = strtoull(sdp_get_owner_version(sdp), nullptr, 10);
+
+  sdp_nettype_e type = sdp_get_owner_network_type(sdp);
+  if (type != SDP_NT_INTERNET) {
+    results.AddParseError(2, "Unsupported network type");
+    return false;
+  }
+
+  sdp::AddrType addrType;
+  switch (sdp_get_owner_address_type(sdp)) {
+    case SDP_AT_IP4:
+      addrType = sdp::kIPv4;
+      break;
+    case SDP_AT_IP6:
+      addrType = sdp::kIPv6;
+      break;
+    default:
+      results.AddParseError(2, "Unsupported address type");
+      return false;
+  }
+
+  std::string address = sdp_get_owner_address(sdp);
+  mOrigin = SdpOrigin(username, sessId, sessVer, addrType, address);
+  return true;
+}
+
+bool SipccSdp::Load(sdp_t* sdp, InternalResults& results) {
+  // Believe it or not, SDP_SESSION_LEVEL is 0xFFFF
+  if (!mAttributeList.Load(sdp, SDP_SESSION_LEVEL, results)) {
+    return false;
+  }
+
+  if (!LoadOrigin(sdp, results)) {
+    return false;
+  }
+
+  if (!mBandwidths.Load(sdp, SDP_SESSION_LEVEL, results)) {
+    return false;
+  }
+
+  for (int i = 0; i < sdp_get_num_media_lines(sdp); ++i) {
+    // note that we pass a "level" here that is one higher
+    // sipcc counts media sections from 1, using 0xFFFF as the "session"
+    UniquePtr<SipccSdpMediaSection> section(
+        new SipccSdpMediaSection(i, &mAttributeList));
+    if (!section->Load(sdp, i + 1, results)) {
+      return false;
+    }
+    mMediaSections.push_back(std::move(section));
+  }
+  return true;
+}
+
+void SipccSdp::Serialize(std::ostream& os) const {
+  os << "v=0" << CRLF << mOrigin << "s=-" << CRLF;
+
+  // We don't support creating i=, u=, e=, p=
+  // We don't generate c= at the session level (only in media)
+
+  mBandwidths.Serialize(os);
+  os << "t=0 0" << CRLF;
+
+  // We don't support r= or z=
+
+  // attributes
+  os << mAttributeList;
+
+  // media sections
+  for (const auto& msection : mMediaSections) {
+    os << *msection;
+  }
+}
+
+bool SipccSdpBandwidths::Load(sdp_t* sdp, uint16_t level,
+                              InternalResults& results) {
+  size_t count = sdp_get_num_bw_lines(sdp, level);
+  for (size_t i = 1; i <= count; ++i) {
+    sdp_bw_modifier_e bwtype = sdp_get_bw_modifier(sdp, level, i);
+    uint32_t bandwidth = sdp_get_bw_value(sdp, level, i);
+    if (bwtype != SDP_BW_MODIFIER_UNSUPPORTED) {
+      const char* typeName = sdp_get_bw_modifier_name(bwtype);
+      (*this)[typeName] = bandwidth;
+    }
+  }
+
+  return true;
+}
+
+void SipccSdpBandwidths::Serialize(std::ostream& os) const {
+  for (auto i = begin(); i != end(); ++i) {
+    os << "b=" << i->first << ":" << i->second << CRLF;
+  }
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/sdp/SipccSdp.h b/dom/media/webrtc/sdp/SipccSdp.h
new file mode 100644
index 0000000000..4915821cee
--- /dev/null
+++ b/dom/media/webrtc/sdp/SipccSdp.h
@@ -0,0 +1,82 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _SIPCCSDP_H_
+#define _SIPCCSDP_H_
+
+#include <map>
+#include <vector>
+#include "mozilla/Attributes.h"
+
+#include "sdp/Sdp.h"
+#include "sdp/SdpParser.h"
+#include "sdp/SipccSdpMediaSection.h"
+#include "sdp/SipccSdpAttributeList.h"
+extern "C" {
+#include "sipcc_sdp.h"
+}
+
+namespace mozilla {
+
+class SipccSdpParser;
+
+class SipccSdp final : public Sdp {
+  friend class SipccSdpParser;
+
+ public:
+  explicit SipccSdp(const SdpOrigin& origin)
+      : mOrigin(origin), mAttributeList(nullptr) {}
+  SipccSdp(const SipccSdp& aOrig);
+
+  virtual Sdp* Clone() const override;
+
+  virtual const SdpOrigin& GetOrigin() const override;
+
+  // Note: connection information is always retrieved from media sections
+  virtual uint32_t GetBandwidth(const std::string& type) const override;
+
+  virtual size_t GetMediaSectionCount() const override {
+    return mMediaSections.size();
+  }
+
+  virtual const SdpAttributeList& GetAttributeList() const override {
+    return mAttributeList;
+  }
+
+  virtual SdpAttributeList& GetAttributeList() override {
+    return mAttributeList;
+  }
+
+  virtual const SdpMediaSection& GetMediaSection(size_t level) const override;
+
+  virtual SdpMediaSection& GetMediaSection(size_t level) override;
+
+  virtual SdpMediaSection& AddMediaSection(SdpMediaSection::MediaType media,
+                                           SdpDirectionAttribute::Direction dir,
+                                           uint16_t port,
+                                           SdpMediaSection::Protocol proto,
+                                           sdp::AddrType addrType,
+                                           const std::string& addr) override;
+
+  virtual void Serialize(std::ostream&) const override;
+
+ private:
+  using InternalResults = SdpParser::InternalResults;
+
+  SipccSdp() : mOrigin("", 0, 0, sdp::kIPv4, ""), mAttributeList(nullptr) {}
+
+  bool Load(sdp_t* sdp, InternalResults& results);
+  bool LoadOrigin(sdp_t* sdp, InternalResults& results);
+
+  SdpOrigin mOrigin;
+  SipccSdpBandwidths mBandwidths;
+  SipccSdpAttributeList mAttributeList;
+  std::vector<UniquePtr<SipccSdpMediaSection>> mMediaSections;
+};
+
+}  // namespace mozilla
+
+#endif  // _sdp_h_
diff --git a/dom/media/webrtc/sdp/SipccSdpAttributeList.cpp b/dom/media/webrtc/sdp/SipccSdpAttributeList.cpp
new file mode 100644
index 0000000000..15572b6dd3
--- /dev/null
+++ b/dom/media/webrtc/sdp/SipccSdpAttributeList.cpp
@@ -0,0 +1,1386 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "sdp/SipccSdpAttributeList.h"
+
+#include <ostream>
+#include "mozilla/Assertions.h"
+
+extern "C" {
+#include "sdp_private.h"
+}
+
+namespace mozilla {
+
+using InternalResults = SdpParser::InternalResults;
+
+/* static */
+const std::string SipccSdpAttributeList::kEmptyString = "";
+
+SipccSdpAttributeList::SipccSdpAttributeList(
+    const SipccSdpAttributeList* sessionLevel)
+    : mSessionLevel(sessionLevel) {
+  memset(&mAttributes, 0, sizeof(mAttributes));
+}
+
+SipccSdpAttributeList::SipccSdpAttributeList(
+    const SipccSdpAttributeList& aOrig,
+    const SipccSdpAttributeList* sessionLevel)
+    : SipccSdpAttributeList(sessionLevel) {
+  for (size_t i = 0; i < kNumAttributeTypes; ++i) {
+    if (aOrig.mAttributes[i]) {
+      mAttributes[i] = aOrig.mAttributes[i]->Clone();
+    }
+  }
+}
+
+SipccSdpAttributeList::~SipccSdpAttributeList() {
+  for (size_t i = 0; i < kNumAttributeTypes; ++i) {
+    delete mAttributes[i];
+  }
+}
+
+bool SipccSdpAttributeList::HasAttribute(AttributeType type,
+                                         bool sessionFallback) const {
+  return !!GetAttribute(type, sessionFallback);
+}
+
+const SdpAttribute* SipccSdpAttributeList::GetAttribute(
+    AttributeType type, bool sessionFallback) const {
+  const SdpAttribute* value = mAttributes[static_cast<size_t>(type)];
+  // Only do fallback when the attribute can appear at both the media and
+  // session level
+  if (!value && !AtSessionLevel() && sessionFallback &&
+      SdpAttribute::IsAllowedAtSessionLevel(type) &&
+      SdpAttribute::IsAllowedAtMediaLevel(type)) {
+    return mSessionLevel->GetAttribute(type, false);
+  }
+  return value;
+}
+
+void SipccSdpAttributeList::RemoveAttribute(AttributeType type) {
+  delete mAttributes[static_cast<size_t>(type)];
+  mAttributes[static_cast<size_t>(type)] = nullptr;
+}
+
+void SipccSdpAttributeList::Clear() {
+  for (size_t i = 0; i < kNumAttributeTypes; ++i) {
+    RemoveAttribute(static_cast<AttributeType>(i));
+  }
+}
+
+uint32_t SipccSdpAttributeList::Count() const {
+  uint32_t count = 0;
+  for (size_t i = 0; i < kNumAttributeTypes; ++i) {
+    if (mAttributes[i]) {
+      count++;
+    }
+  }
+  return count;
+}
+
+void SipccSdpAttributeList::SetAttribute(SdpAttribute* attr) {
+  if (!IsAllowedHere(attr->GetType())) {
+    MOZ_ASSERT(false, "This type of attribute is not allowed here");
+    return;
+  }
+  RemoveAttribute(attr->GetType());
+  mAttributes[attr->GetType()] = attr;
+}
+
+void SipccSdpAttributeList::LoadSimpleString(sdp_t* sdp, uint16_t level,
+                                             sdp_attr_e attr,
+                                             AttributeType targetType,
+                                             InternalResults& results) {
+  const char* value = sdp_attr_get_simple_string(sdp, attr, level, 0, 1);
+  if (value) {
+    if (!IsAllowedHere(targetType)) {
+      uint32_t lineNumber = sdp_attr_line_number(sdp, attr, level, 0, 1);
+      WarnAboutMisplacedAttribute(targetType, lineNumber, results);
+    } else {
+      SetAttribute(new SdpStringAttribute(targetType, std::string(value)));
+    }
+  }
+}
+
+void SipccSdpAttributeList::LoadSimpleStrings(sdp_t* sdp, uint16_t level,
+                                              InternalResults& results) {
+  LoadSimpleString(sdp, level, SDP_ATTR_MID, SdpAttribute::kMidAttribute,
+                   results);
+  LoadSimpleString(sdp, level, SDP_ATTR_LABEL, SdpAttribute::kLabelAttribute,
+                   results);
+}
+
+void SipccSdpAttributeList::LoadSimpleNumber(sdp_t* sdp, uint16_t level,
+                                             sdp_attr_e attr,
+                                             AttributeType targetType,
+                                             InternalResults& results) {
+  if (sdp_attr_valid(sdp, attr, level, 0, 1)) {
+    if (!IsAllowedHere(targetType)) {
+      uint32_t lineNumber = sdp_attr_line_number(sdp, attr, level, 0, 1);
+      WarnAboutMisplacedAttribute(targetType, lineNumber, results);
+    } else {
+      uint32_t value = sdp_attr_get_simple_u32(sdp, attr, level, 0, 1);
+      SetAttribute(new SdpNumberAttribute(targetType, value));
+    }
+  }
+}
+
+void SipccSdpAttributeList::LoadSimpleNumbers(sdp_t* sdp, uint16_t level,
+                                              InternalResults& results) {
+  LoadSimpleNumber(sdp, level, SDP_ATTR_PTIME, SdpAttribute::kPtimeAttribute,
+                   results);
+  LoadSimpleNumber(sdp, level, SDP_ATTR_MAXPTIME,
+                   SdpAttribute::kMaxptimeAttribute, results);
+  LoadSimpleNumber(sdp, level, SDP_ATTR_SCTPPORT,
+                   SdpAttribute::kSctpPortAttribute, results);
+  LoadSimpleNumber(sdp, level, SDP_ATTR_MAXMESSAGESIZE,
+                   SdpAttribute::kMaxMessageSizeAttribute, results);
+}
+
+void SipccSdpAttributeList::LoadFlags(sdp_t* sdp, uint16_t level) {
+  if (AtSessionLevel()) {
+    if (sdp_attr_valid(sdp, SDP_ATTR_ICE_LITE, level, 0, 1)) {
+      SetAttribute(new SdpFlagAttribute(SdpAttribute::kIceLiteAttribute));
+    }
+  } else {  // media-level
+    if (sdp_attr_valid(sdp, SDP_ATTR_RTCP_MUX, level, 0, 1)) {
+      SetAttribute(new SdpFlagAttribute(SdpAttribute::kRtcpMuxAttribute));
+    }
+    if (sdp_attr_valid(sdp, SDP_ATTR_END_OF_CANDIDATES, level, 0, 1)) {
+      SetAttribute(
+          new SdpFlagAttribute(SdpAttribute::kEndOfCandidatesAttribute));
+    }
+    if (sdp_attr_valid(sdp, SDP_ATTR_BUNDLE_ONLY, level, 0, 1)) {
+      SetAttribute(new SdpFlagAttribute(SdpAttribute::kBundleOnlyAttribute));
+    }
+    if (sdp_attr_valid(sdp, SDP_ATTR_RTCP_RSIZE, level, 0, 1))
+      SetAttribute(new SdpFlagAttribute(SdpAttribute::kRtcpRsizeAttribute));
+  }
+}
+
+static void ConvertDirection(sdp_direction_e sipcc_direction,
+                             SdpDirectionAttribute::Direction* dir_outparam) {
+  switch (sipcc_direction) {
+    case SDP_DIRECTION_SENDRECV:
+      *dir_outparam = SdpDirectionAttribute::kSendrecv;
+      return;
+    case SDP_DIRECTION_SENDONLY:
+      *dir_outparam = SdpDirectionAttribute::kSendonly;
+      return;
+    case SDP_DIRECTION_RECVONLY:
+      *dir_outparam = SdpDirectionAttribute::kRecvonly;
+      return;
+    case SDP_DIRECTION_INACTIVE:
+      *dir_outparam = SdpDirectionAttribute::kInactive;
+      return;
+    case SDP_MAX_QOS_DIRECTIONS:
+      // Nothing actually sets this value.
+      // Fall through to MOZ_CRASH below.
+      {}
+  }
+
+  MOZ_CRASH("Invalid direction from sipcc; this is probably corruption");
+}
+
+void SipccSdpAttributeList::LoadDirection(sdp_t* sdp, uint16_t level,
+                                          InternalResults& results) {
+  SdpDirectionAttribute::Direction dir;
+  ConvertDirection(sdp_get_media_direction(sdp, level, 0), &dir);
+  SetAttribute(new SdpDirectionAttribute(dir));
+}
+
+void SipccSdpAttributeList::LoadIceAttributes(sdp_t* sdp, uint16_t level) {
+  char* value;
+  sdp_result_e sdpres =
+      sdp_attr_get_ice_attribute(sdp, level, 0, SDP_ATTR_ICE_UFRAG, 1, &value);
+  if (sdpres == SDP_SUCCESS) {
+    SetAttribute(new SdpStringAttribute(SdpAttribute::kIceUfragAttribute,
+                                        std::string(value)));
+  }
+  sdpres =
+      sdp_attr_get_ice_attribute(sdp, level, 0, SDP_ATTR_ICE_PWD, 1, &value);
+  if (sdpres == SDP_SUCCESS) {
+    SetAttribute(new SdpStringAttribute(SdpAttribute::kIcePwdAttribute,
+                                        std::string(value)));
+  }
+
+  const char* iceOptVal =
+      sdp_attr_get_simple_string(sdp, SDP_ATTR_ICE_OPTIONS, level, 0, 1);
+  if (iceOptVal) {
+    auto* iceOptions =
+        new SdpOptionsAttribute(SdpAttribute::kIceOptionsAttribute);
+    iceOptions->Load(iceOptVal);
+    SetAttribute(iceOptions);
+  }
+}
+
+bool SipccSdpAttributeList::LoadFingerprint(sdp_t* sdp, uint16_t level,
+                                            InternalResults& results) {
+  char* value;
+  UniquePtr<SdpFingerprintAttributeList> fingerprintAttrs;
+
+  for (uint16_t i = 1; i < UINT16_MAX; ++i) {
+    sdp_result_e result = sdp_attr_get_dtls_fingerprint_attribute(
+        sdp, level, 0, SDP_ATTR_DTLS_FINGERPRINT, i, &value);
+
+    if (result != SDP_SUCCESS) {
+      break;
+    }
+
+    std::string fingerprintAttr(value);
+    uint32_t lineNumber =
+        sdp_attr_line_number(sdp, SDP_ATTR_DTLS_FINGERPRINT, level, 0, i);
+
+    // sipcc does not expose parse code for this
+    size_t start = fingerprintAttr.find_first_not_of(" \t");
+    if (start == std::string::npos) {
+      results.AddParseError(lineNumber, "Empty fingerprint attribute");
+      return false;
+    }
+
+    size_t end = fingerprintAttr.find_first_of(" \t", start);
+    if (end == std::string::npos) {
+      // One token, no trailing ws
+      results.AddParseError(lineNumber,
+                            "Only one token in fingerprint attribute");
+      return false;
+    }
+
+    std::string algorithmToken(fingerprintAttr.substr(start, end - start));
+
+    start = fingerprintAttr.find_first_not_of(" \t", end);
+    if (start == std::string::npos) {
+      // One token, trailing ws
+      results.AddParseError(lineNumber,
+                            "Only one token in fingerprint attribute");
+      return false;
+    }
+
+    std::string fingerprintToken(fingerprintAttr.substr(start));
+
+    std::vector<uint8_t> fingerprint =
+        SdpFingerprintAttributeList::ParseFingerprint(fingerprintToken);
+    if (fingerprint.empty()) {
+      results.AddParseError(lineNumber, "Malformed fingerprint token");
+      return false;
+    }
+
+    if (!fingerprintAttrs) {
+      fingerprintAttrs.reset(new SdpFingerprintAttributeList);
+    }
+
+    // Don't assert on unknown algorithm, just skip
+    fingerprintAttrs->PushEntry(algorithmToken, fingerprint, false);
+  }
+
+  if (fingerprintAttrs) {
+    SetAttribute(fingerprintAttrs.release());
+  }
+
+  return true;
+}
+
+void SipccSdpAttributeList::LoadCandidate(sdp_t* sdp, uint16_t level) {
+  char* value;
+  auto candidates =
+      MakeUnique<SdpMultiStringAttribute>(SdpAttribute::kCandidateAttribute);
+
+  for (uint16_t i = 1; i < UINT16_MAX; ++i) {
+    sdp_result_e result = sdp_attr_get_ice_attribute(
+        sdp, level, 0, SDP_ATTR_ICE_CANDIDATE, i, &value);
+
+    if (result != SDP_SUCCESS) {
+      break;
+    }
+
+    candidates->mValues.push_back(value);
+  }
+
+  if (!candidates->mValues.empty()) {
+    SetAttribute(candidates.release());
+  }
+}
+
+bool SipccSdpAttributeList::LoadSctpmap(sdp_t* sdp, uint16_t level,
+                                        InternalResults& results) {
+  auto sctpmap = MakeUnique<SdpSctpmapAttributeList>();
+  for (uint16_t i = 0; i < UINT16_MAX; ++i) {
+    sdp_attr_t* attr = sdp_find_attr(sdp, level, 0, SDP_ATTR_SCTPMAP, i + 1);
+
+    if (!attr) {
+      break;
+    }
+
+    // Yeah, this is a little weird, but for now we'll just store this as a
+    // payload type.
+    uint16_t payloadType = attr->attr.sctpmap.port;
+    uint16_t streams = attr->attr.sctpmap.streams;
+    const char* name = attr->attr.sctpmap.protocol;
+
+    std::ostringstream osPayloadType;
+    osPayloadType << payloadType;
+    sctpmap->PushEntry(osPayloadType.str(), name, streams);
+  }
+
+  if (!sctpmap->mSctpmaps.empty()) {
+    SetAttribute(sctpmap.release());
+  }
+
+  return true;
+}
+
+SdpRtpmapAttributeList::CodecType SipccSdpAttributeList::GetCodecType(
+    rtp_ptype type) {
+  switch (type) {
+    case RTP_PCMU:
+      return SdpRtpmapAttributeList::kPCMU;
+    case RTP_PCMA:
+      return SdpRtpmapAttributeList::kPCMA;
+    case RTP_G722:
+      return SdpRtpmapAttributeList::kG722;
+    case RTP_H264_P0:
+    case RTP_H264_P1:
+      return SdpRtpmapAttributeList::kH264;
+    case RTP_OPUS:
+      return SdpRtpmapAttributeList::kOpus;
+    case RTP_VP8:
+      return SdpRtpmapAttributeList::kVP8;
+    case RTP_VP9:
+      return SdpRtpmapAttributeList::kVP9;
+    case RTP_RED:
+      return SdpRtpmapAttributeList::kRed;
+    case RTP_ULPFEC:
+      return SdpRtpmapAttributeList::kUlpfec;
+    case RTP_RTX:
+      return SdpRtpmapAttributeList::kRtx;
+    case RTP_TELEPHONE_EVENT:
+      return SdpRtpmapAttributeList::kTelephoneEvent;
+    case RTP_NONE:
+    // Happens when sipcc doesn't know how to translate to the enum
+    case RTP_CELP:
+    case RTP_G726:
+    case RTP_GSM:
+    case RTP_G723:
+    case RTP_DVI4:
+    case RTP_DVI4_II:
+    case RTP_LPC:
+    case RTP_G728:
+    case RTP_G729:
+    case RTP_JPEG:
+    case RTP_NV:
+    case RTP_H261:
+    case RTP_L16:
+    case RTP_H263:
+    case RTP_ILBC:
+    case RTP_I420:
+      return SdpRtpmapAttributeList::kOtherCodec;
+  }
+  MOZ_CRASH("Invalid codec type from sipcc. Probably corruption.");
+}
+
+bool SipccSdpAttributeList::LoadRtpmap(sdp_t* sdp, uint16_t level,
+                                       InternalResults& results) {
+  auto rtpmap = MakeUnique<SdpRtpmapAttributeList>();
+  uint16_t count;
+  sdp_result_e result =
+      sdp_attr_num_instances(sdp, level, 0, SDP_ATTR_RTPMAP, &count);
+  if (result != SDP_SUCCESS) {
+    MOZ_ASSERT(false, "Unable to get rtpmap size");
+    results.AddParseError(sdp_get_media_line_number(sdp, level),
+                          "Unable to get rtpmap size");
+    return false;
+  }
+  for (uint16_t i = 0; i < count; ++i) {
+    uint16_t pt = sdp_attr_get_rtpmap_payload_type(sdp, level, 0, i + 1);
+    const char* ccName = sdp_attr_get_rtpmap_encname(sdp, level, 0, i + 1);
+
+    if (!ccName) {
+      // Probably no rtpmap attribute for a pt in an m-line
+      results.AddParseError(sdp_get_media_line_number(sdp, level),
+                            "No rtpmap attribute for payload type");
+      continue;
+    }
+
+    std::string name(ccName);
+
+    SdpRtpmapAttributeList::CodecType codec =
+        GetCodecType(sdp_get_known_payload_type(sdp, level, pt));
+
+    uint32_t clock = sdp_attr_get_rtpmap_clockrate(sdp, level, 0, i + 1);
+    uint16_t channels = 0;
+
+    // sipcc gives us a channels value of "1" for video
+    if (sdp_get_media_type(sdp, level) == SDP_MEDIA_AUDIO) {
+      channels = sdp_attr_get_rtpmap_num_chan(sdp, level, 0, i + 1);
+    }
+
+    std::ostringstream osPayloadType;
+    osPayloadType << pt;
+    rtpmap->PushEntry(osPayloadType.str(), codec, name, clock, channels);
+  }
+
+  if (!rtpmap->mRtpmaps.empty()) {
+    SetAttribute(rtpmap.release());
+  }
+
+  return true;
+}
+
+void SipccSdpAttributeList::LoadSetup(sdp_t* sdp, uint16_t level) {
+  sdp_setup_type_e setupType;
+  auto sdpres = sdp_attr_get_setup_attribute(sdp, level, 0, 1, &setupType);
+
+  if (sdpres != SDP_SUCCESS) {
+    return;
+  }
+
+  switch (setupType) {
+    case SDP_SETUP_ACTIVE:
+      SetAttribute(new SdpSetupAttribute(SdpSetupAttribute::kActive));
+      return;
+    case SDP_SETUP_PASSIVE:
+      SetAttribute(new SdpSetupAttribute(SdpSetupAttribute::kPassive));
+      return;
+    case SDP_SETUP_ACTPASS:
+      SetAttribute(new SdpSetupAttribute(SdpSetupAttribute::kActpass));
+      return;
+    case SDP_SETUP_HOLDCONN:
+      SetAttribute(new SdpSetupAttribute(SdpSetupAttribute::kHoldconn));
+      return;
+    case SDP_SETUP_UNKNOWN:
+      return;
+    case SDP_SETUP_NOT_FOUND:
+    case SDP_MAX_SETUP:
+      // There is no code that will set these.
+      // Fall through to MOZ_CRASH() below.
+      {}
+  }
+
+  MOZ_CRASH("Invalid setup type from sipcc. This is probably corruption.");
+}
+
+void SipccSdpAttributeList::LoadSsrc(sdp_t* sdp, uint16_t level) {
+  auto ssrcs = MakeUnique<SdpSsrcAttributeList>();
+
+  for (uint16_t i = 1; i < UINT16_MAX; ++i) {
+    sdp_attr_t* attr = sdp_find_attr(sdp, level, 0, SDP_ATTR_SSRC, i);
+
+    if (!attr) {
+      break;
+    }
+
+    sdp_ssrc_t* ssrc = &(attr->attr.ssrc);
+    ssrcs->PushEntry(ssrc->ssrc, ssrc->attribute);
+  }
+
+  if (!ssrcs->mSsrcs.empty()) {
+    SetAttribute(ssrcs.release());
+  }
+}
+
+void SipccSdpAttributeList::LoadSsrcGroup(sdp_t* sdp, uint16_t level) {
+  auto ssrcGroups = MakeUnique<SdpSsrcGroupAttributeList>();
+
+  for (uint16_t i = 1; i < UINT16_MAX; ++i) {
+    sdp_attr_t* attr = sdp_find_attr(sdp, level, 0, SDP_ATTR_SSRC_GROUP, i);
+
+    if (!attr) {
+      break;
+    }
+
+    sdp_ssrc_group_t* ssrc_group = &(attr->attr.ssrc_group);
+
+    SdpSsrcGroupAttributeList::Semantics semantic;
+    switch (ssrc_group->semantic) {
+      case SDP_SSRC_GROUP_ATTR_FEC:
+        semantic = SdpSsrcGroupAttributeList::kFec;
+        break;
+      case SDP_SSRC_GROUP_ATTR_FID:
+        semantic = SdpSsrcGroupAttributeList::kFid;
+        break;
+      case SDP_SSRC_GROUP_ATTR_FECFR:
+        semantic = SdpSsrcGroupAttributeList::kFecFr;
+        break;
+      case SDP_SSRC_GROUP_ATTR_DUP:
+        semantic = SdpSsrcGroupAttributeList::kDup;
+        break;
+      case SDP_SSRC_GROUP_ATTR_SIM:
+        semantic = SdpSsrcGroupAttributeList::kSim;
+        break;
+      case SDP_MAX_SSRC_GROUP_ATTR_VAL:
+        continue;
+      case SDP_SSRC_GROUP_ATTR_UNSUPPORTED:
+        continue;
+    }
+
+    std::vector<uint32_t> ssrcs;
+    ssrcs.reserve(ssrc_group->num_ssrcs);
+    for (int i = 0; i < ssrc_group->num_ssrcs; ++i) {
+      ssrcs.push_back(ssrc_group->ssrcs[i]);
+    }
+
+    ssrcGroups->PushEntry(semantic, ssrcs);
+  }
+
+  if (!ssrcGroups->mSsrcGroups.empty()) {
+    SetAttribute(ssrcGroups.release());
+  }
+}
+
+bool SipccSdpAttributeList::LoadImageattr(sdp_t* sdp, uint16_t level,
+                                          InternalResults& results) {
+  UniquePtr<SdpImageattrAttributeList> imageattrs(
+      new SdpImageattrAttributeList);
+
+  for (uint16_t i = 1; i < UINT16_MAX; ++i) {
+    const char* imageattrRaw =
+        sdp_attr_get_simple_string(sdp, SDP_ATTR_IMAGEATTR, level, 0, i);
+    if (!imageattrRaw) {
+      break;
+    }
+
+    std::string error;
+    size_t errorPos;
+    if (!imageattrs->PushEntry(imageattrRaw, &error, &errorPos)) {
+      std::ostringstream fullError;
+      fullError << error << " at column " << errorPos;
+      results.AddParseError(
+          sdp_attr_line_number(sdp, SDP_ATTR_IMAGEATTR, level, 0, i),
+          fullError.str());
+      return false;
+    }
+  }
+
+  if (!imageattrs->mImageattrs.empty()) {
+    SetAttribute(imageattrs.release());
+  }
+  return true;
+}
+
+bool SipccSdpAttributeList::LoadSimulcast(sdp_t* sdp, uint16_t level,
+                                          InternalResults& results) {
+  const char* simulcastRaw =
+      sdp_attr_get_simple_string(sdp, SDP_ATTR_SIMULCAST, level, 0, 1);
+  if (!simulcastRaw) {
+    return true;
+  }
+
+  UniquePtr<SdpSimulcastAttribute> simulcast(new SdpSimulcastAttribute);
+
+  std::istringstream is(simulcastRaw);
+  std::string error;
+  if (!simulcast->Parse(is, &error)) {
+    std::ostringstream fullError;
+    fullError << error << " at column " << is.tellg();
+    results.AddParseError(
+        sdp_attr_line_number(sdp, SDP_ATTR_SIMULCAST, level, 0, 1),
+        fullError.str());
+    return false;
+  }
+
+  SetAttribute(simulcast.release());
+  return true;
+}
+
+bool SipccSdpAttributeList::LoadGroups(sdp_t* sdp, uint16_t level,
+                                       InternalResults& results) {
+  uint16_t attrCount = 0;
+  if (sdp_attr_num_instances(sdp, level, 0, SDP_ATTR_GROUP, &attrCount) !=
+      SDP_SUCCESS) {
+    MOZ_ASSERT(false, "Could not get count of group attributes");
+    results.AddParseError(0, "Could not get count of group attributes");
+    return false;
+  }
+
+  UniquePtr<SdpGroupAttributeList> groups = MakeUnique<SdpGroupAttributeList>();
+  for (uint16_t attr = 1; attr <= attrCount; ++attr) {
+    SdpGroupAttributeList::Semantics semantics;
+    std::vector<std::string> tags;
+
+    switch (sdp_get_group_attr(sdp, level, 0, attr)) {
+      case SDP_GROUP_ATTR_FID:
+        semantics = SdpGroupAttributeList::kFid;
+        break;
+      case SDP_GROUP_ATTR_LS:
+        semantics = SdpGroupAttributeList::kLs;
+        break;
+      case SDP_GROUP_ATTR_ANAT:
+        semantics = SdpGroupAttributeList::kAnat;
+        break;
+      case SDP_GROUP_ATTR_BUNDLE:
+        semantics = SdpGroupAttributeList::kBundle;
+        break;
+      default:
+        continue;
+    }
+
+    uint16_t idCount = sdp_get_group_num_id(sdp, level, 0, attr);
+    for (uint16_t id = 1; id <= idCount; ++id) {
+      const char* idStr = sdp_get_group_id(sdp, level, 0, attr, id);
+      if (!idStr) {
+        std::ostringstream os;
+        os << "bad a=group identifier at " << (attr - 1) << ", " << (id - 1);
+        results.AddParseError(0, os.str());
+        return false;
+      }
+      tags.push_back(std::string(idStr));
+    }
+    groups->PushEntry(semantics, tags);
+  }
+
+  if (!groups->mGroups.empty()) {
+    SetAttribute(groups.release());
+  }
+
+  return true;
+}
+
+bool SipccSdpAttributeList::LoadMsidSemantics(sdp_t* sdp, uint16_t level,
+                                              InternalResults& results) {
+  auto msidSemantics = MakeUnique<SdpMsidSemanticAttributeList>();
+
+  for (uint16_t i = 1; i < UINT16_MAX; ++i) {
+    sdp_attr_t* attr = sdp_find_attr(sdp, level, 0, SDP_ATTR_MSID_SEMANTIC, i);
+
+    if (!attr) {
+      break;
+    }
+
+    sdp_msid_semantic_t* msid_semantic = &(attr->attr.msid_semantic);
+    std::vector<std::string> msids;
+    for (size_t i = 0; i < SDP_MAX_MEDIA_STREAMS; ++i) {
+      if (!msid_semantic->msids[i]) {
+        break;
+      }
+
+      msids.push_back(msid_semantic->msids[i]);
+    }
+
+    msidSemantics->PushEntry(msid_semantic->semantic, msids);
+  }
+
+  if (!msidSemantics->mMsidSemantics.empty()) {
+    SetAttribute(msidSemantics.release());
+  }
+  return true;
+}
+
+void SipccSdpAttributeList::LoadIdentity(sdp_t* sdp, uint16_t level) {
+  const char* val =
+      sdp_attr_get_long_string(sdp, SDP_ATTR_IDENTITY, level, 0, 1);
+  if (val) {
+    SetAttribute(new SdpStringAttribute(SdpAttribute::kIdentityAttribute,
+                                        std::string(val)));
+  }
+}
+
+void SipccSdpAttributeList::LoadDtlsMessage(sdp_t* sdp, uint16_t level) {
+  const char* val =
+      sdp_attr_get_long_string(sdp, SDP_ATTR_DTLS_MESSAGE, level, 0, 1);
+  if (val) {
+    // sipcc does not expose parse code for this, so we use a SDParta-provided
+    // parser
+    std::string strval(val);
+    SetAttribute(new SdpDtlsMessageAttribute(strval));
+  }
+}
+
+void SipccSdpAttributeList::LoadFmtp(sdp_t* sdp, uint16_t level) {
+  auto fmtps = MakeUnique<SdpFmtpAttributeList>();
+
+  for (uint16_t i = 1; i < UINT16_MAX; ++i) {
+    sdp_attr_t* attr = sdp_find_attr(sdp, level, 0, SDP_ATTR_FMTP, i);
+
+    if (!attr) {
+      break;
+    }
+
+    sdp_fmtp_t* fmtp = &(attr->attr.fmtp);
+
+    // Get the payload type
+    std::stringstream osPayloadType;
+    // payload_num is the number in the fmtp attribute, verbatim
+    osPayloadType << fmtp->payload_num;
+
+    // Get parsed form of parameters, if supported
+    UniquePtr<SdpFmtpAttributeList::Parameters> parameters;
+
+    rtp_ptype codec = sdp_get_known_payload_type(sdp, level, fmtp->payload_num);
+
+    switch (codec) {
+      case RTP_H264_P0:
+      case RTP_H264_P1: {
+        SdpFmtpAttributeList::H264Parameters* h264Parameters(
+            new SdpFmtpAttributeList::H264Parameters);
+
+        sstrncpy(h264Parameters->sprop_parameter_sets, fmtp->parameter_sets,
+                 sizeof(h264Parameters->sprop_parameter_sets));
+
+        h264Parameters->level_asymmetry_allowed =
+            !!(fmtp->level_asymmetry_allowed);
+
+        h264Parameters->packetization_mode = fmtp->packetization_mode;
+        sscanf(fmtp->profile_level_id, "%x", &h264Parameters->profile_level_id);
+        h264Parameters->max_mbps = fmtp->max_mbps;
+        h264Parameters->max_fs = fmtp->max_fs;
+        h264Parameters->max_cpb = fmtp->max_cpb;
+        h264Parameters->max_dpb = fmtp->max_dpb;
+        h264Parameters->max_br = fmtp->max_br;
+
+        parameters.reset(h264Parameters);
+      } break;
+      case RTP_VP9: {
+        SdpFmtpAttributeList::VP8Parameters* vp9Parameters(
+            new SdpFmtpAttributeList::VP8Parameters(
+                SdpRtpmapAttributeList::kVP9));
+
+        vp9Parameters->max_fs = fmtp->max_fs;
+        vp9Parameters->max_fr = fmtp->max_fr;
+
+        parameters.reset(vp9Parameters);
+      } break;
+      case RTP_VP8: {
+        SdpFmtpAttributeList::VP8Parameters* vp8Parameters(
+            new SdpFmtpAttributeList::VP8Parameters(
+                SdpRtpmapAttributeList::kVP8));
+
+        vp8Parameters->max_fs = fmtp->max_fs;
+        vp8Parameters->max_fr = fmtp->max_fr;
+
+        parameters.reset(vp8Parameters);
+      } break;
+      case RTP_RED: {
+        SdpFmtpAttributeList::RedParameters* redParameters(
+            new SdpFmtpAttributeList::RedParameters);
+        for (int i = 0; i < SDP_FMTP_MAX_REDUNDANT_ENCODINGS &&
+                        fmtp->redundant_encodings[i];
+             ++i) {
+          redParameters->encodings.push_back(fmtp->redundant_encodings[i]);
+        }
+
+        parameters.reset(redParameters);
+      } break;
+      case RTP_OPUS: {
+        SdpFmtpAttributeList::OpusParameters* opusParameters(
+            new SdpFmtpAttributeList::OpusParameters);
+        opusParameters->maxplaybackrate = fmtp->maxplaybackrate;
+        opusParameters->stereo = fmtp->stereo;
+        opusParameters->useInBandFec = fmtp->useinbandfec;
+        opusParameters->maxAverageBitrate = fmtp->maxaveragebitrate;
+        opusParameters->useDTX = fmtp->usedtx;
+        parameters.reset(opusParameters);
+      } break;
+      case RTP_TELEPHONE_EVENT: {
+        SdpFmtpAttributeList::TelephoneEventParameters* teParameters(
+            new SdpFmtpAttributeList::TelephoneEventParameters);
+        if (strlen(fmtp->dtmf_tones) > 0) {
+          teParameters->dtmfTones = fmtp->dtmf_tones;
+        }
+        parameters.reset(teParameters);
+      } break;
+      case RTP_RTX: {
+        SdpFmtpAttributeList::RtxParameters* rtxParameters(
+            new SdpFmtpAttributeList::RtxParameters);
+        rtxParameters->apt = fmtp->apt;
+        if (fmtp->has_rtx_time == TRUE) {
+          rtxParameters->rtx_time = Some(fmtp->rtx_time);
+        }
+        parameters.reset(rtxParameters);
+      } break;
+      default: {
+      }
+    }
+
+    if (parameters) {
+      fmtps->PushEntry(osPayloadType.str(), *parameters);
+    }
+  }
+
+  if (!fmtps->mFmtps.empty()) {
+    SetAttribute(fmtps.release());
+  }
+}
+
+void SipccSdpAttributeList::LoadMsids(sdp_t* sdp, uint16_t level,
+                                      InternalResults& results) {
+  uint16_t attrCount = 0;
+  if (sdp_attr_num_instances(sdp, level, 0, SDP_ATTR_MSID, &attrCount) !=
+      SDP_SUCCESS) {
+    MOZ_ASSERT(false, "Unable to get count of msid attributes");
+    results.AddParseError(0, "Unable to get count of msid attributes");
+    return;
+  }
+  auto msids = MakeUnique<SdpMsidAttributeList>();
+  for (uint16_t i = 1; i <= attrCount; ++i) {
+    uint32_t lineNumber = sdp_attr_line_number(sdp, SDP_ATTR_MSID, level, 0, i);
+
+    const char* identifier = sdp_attr_get_msid_identifier(sdp, level, 0, i);
+    if (!identifier) {
+      results.AddParseError(lineNumber, "msid attribute with bad identity");
+      continue;
+    }
+
+    const char* appdata = sdp_attr_get_msid_appdata(sdp, level, 0, i);
+    if (!appdata) {
+      results.AddParseError(lineNumber, "msid attribute with bad appdata");
+      continue;
+    }
+
+    msids->PushEntry(identifier, appdata);
+  }
+
+  if (!msids->mMsids.empty()) {
+    SetAttribute(msids.release());
+  }
+}
+
+bool SipccSdpAttributeList::LoadRid(sdp_t* sdp, uint16_t level,
+                                    InternalResults& results) {
+  UniquePtr<SdpRidAttributeList> rids(new SdpRidAttributeList);
+
+  for (uint16_t i = 1; i < UINT16_MAX; ++i) {
+    const char* ridRaw =
+        sdp_attr_get_simple_string(sdp, SDP_ATTR_RID, level, 0, i);
+    if (!ridRaw) {
+      break;
+    }
+
+    std::string error;
+    size_t errorPos;
+    if (!rids->PushEntry(ridRaw, &error, &errorPos)) {
+      std::ostringstream fullError;
+      fullError << error << " at column " << errorPos;
+      results.AddParseError(
+          sdp_attr_line_number(sdp, SDP_ATTR_RID, level, 0, i),
+          fullError.str());
+      return false;
+    }
+  }
+
+  if (!rids->mRids.empty()) {
+    SetAttribute(rids.release());
+  }
+  return true;
+}
+
+void SipccSdpAttributeList::LoadExtmap(sdp_t* sdp, uint16_t level,
+                                       InternalResults& results) {
+  auto extmaps = MakeUnique<SdpExtmapAttributeList>();
+
+  for (uint16_t i = 1; i < UINT16_MAX; ++i) {
+    sdp_attr_t* attr = sdp_find_attr(sdp, level, 0, SDP_ATTR_EXTMAP, i);
+
+    if (!attr) {
+      break;
+    }
+
+    sdp_extmap_t* extmap = &(attr->attr.extmap);
+
+    SdpDirectionAttribute::Direction dir = SdpDirectionAttribute::kSendrecv;
+
+    if (extmap->media_direction_specified) {
+      ConvertDirection(extmap->media_direction, &dir);
+    }
+
+    extmaps->PushEntry(extmap->id, dir, extmap->media_direction_specified,
+                       extmap->uri, extmap->extension_attributes);
+  }
+
+  if (!extmaps->mExtmaps.empty()) {
+    if (!AtSessionLevel() &&
+        mSessionLevel->HasAttribute(SdpAttribute::kExtmapAttribute)) {
+      uint32_t lineNumber =
+          sdp_attr_line_number(sdp, SDP_ATTR_EXTMAP, level, 0, 1);
+      results.AddParseError(
+          lineNumber, "extmap attributes in both session and media level");
+    }
+    SetAttribute(extmaps.release());
+  }
+}
+
+void SipccSdpAttributeList::LoadRtcpFb(sdp_t* sdp, uint16_t level,
+                                       InternalResults& results) {
+  auto rtcpfbs = MakeUnique<SdpRtcpFbAttributeList>();
+
+  for (uint16_t i = 1; i < UINT16_MAX; ++i) {
+    sdp_attr_t* attr = sdp_find_attr(sdp, level, 0, SDP_ATTR_RTCP_FB, i);
+
+    if (!attr) {
+      break;
+    }
+
+    sdp_fmtp_fb_t* rtcpfb = &attr->attr.rtcp_fb;
+
+    SdpRtcpFbAttributeList::Type type;
+    std::string parameter;
+
+    // Set type and parameter
+    switch (rtcpfb->feedback_type) {
+      case SDP_RTCP_FB_ACK:
+        type = SdpRtcpFbAttributeList::kAck;
+        switch (rtcpfb->param.ack) {
+          // TODO: sipcc doesn't seem to support ack with no following token.
+          // Issue 189.
+          case SDP_RTCP_FB_ACK_RPSI:
+            parameter = SdpRtcpFbAttributeList::rpsi;
+            break;
+          case SDP_RTCP_FB_ACK_APP:
+            parameter = SdpRtcpFbAttributeList::app;
+            break;
+          default:
+            // Type we don't care about, ignore.
+            continue;
+        }
+        break;
+      case SDP_RTCP_FB_CCM:
+        type = SdpRtcpFbAttributeList::kCcm;
+        switch (rtcpfb->param.ccm) {
+          case SDP_RTCP_FB_CCM_FIR:
+            parameter = SdpRtcpFbAttributeList::fir;
+            break;
+          case SDP_RTCP_FB_CCM_TMMBR:
+            parameter = SdpRtcpFbAttributeList::tmmbr;
+            break;
+          case SDP_RTCP_FB_CCM_TSTR:
+            parameter = SdpRtcpFbAttributeList::tstr;
+            break;
+          case SDP_RTCP_FB_CCM_VBCM:
+            parameter = SdpRtcpFbAttributeList::vbcm;
+            break;
+          default:
+            // Type we don't care about, ignore.
+            continue;
+        }
+        break;
+      case SDP_RTCP_FB_NACK:
+        type = SdpRtcpFbAttributeList::kNack;
+        switch (rtcpfb->param.nack) {
+          case SDP_RTCP_FB_NACK_BASIC:
+            break;
+          case SDP_RTCP_FB_NACK_SLI:
+            parameter = SdpRtcpFbAttributeList::sli;
+            break;
+          case SDP_RTCP_FB_NACK_PLI:
+            parameter = SdpRtcpFbAttributeList::pli;
+            break;
+          case SDP_RTCP_FB_NACK_RPSI:
+            parameter = SdpRtcpFbAttributeList::rpsi;
+            break;
+          case SDP_RTCP_FB_NACK_APP:
+            parameter = SdpRtcpFbAttributeList::app;
+            break;
+          default:
+            // Type we don't care about, ignore.
+            continue;
+        }
+        break;
+      case SDP_RTCP_FB_TRR_INT: {
+        type = SdpRtcpFbAttributeList::kTrrInt;
+        std::ostringstream os;
+        os << rtcpfb->param.trr_int;
+        parameter = os.str();
+      } break;
+      case SDP_RTCP_FB_REMB: {
+        type = SdpRtcpFbAttributeList::kRemb;
+      } break;
+      case SDP_RTCP_FB_TRANSPORT_CC: {
+        type = SdpRtcpFbAttributeList::kTransportCC;
+      } break;
+      default:
+        // Type we don't care about, ignore.
+        continue;
+    }
+
+    std::stringstream osPayloadType;
+    if (rtcpfb->payload_num == UINT16_MAX) {
+      osPayloadType << "*";
+    } else {
+      osPayloadType << rtcpfb->payload_num;
+    }
+
+    std::string pt(osPayloadType.str());
+    std::string extra(rtcpfb->extra);
+
+    rtcpfbs->PushEntry(pt, type, parameter, extra);
+  }
+
+  if (!rtcpfbs->mFeedbacks.empty()) {
+    SetAttribute(rtcpfbs.release());
+  }
+}
+
+void SipccSdpAttributeList::LoadRtcp(sdp_t* sdp, uint16_t level,
+                                     InternalResults& results) {
+  sdp_attr_t* attr = sdp_find_attr(sdp, level, 0, SDP_ATTR_RTCP, 1);
+
+  if (!attr) {
+    return;
+  }
+
+  sdp_rtcp_t* rtcp = &attr->attr.rtcp;
+
+  if (rtcp->nettype != SDP_NT_INTERNET) {
+    return;
+  }
+
+  if (rtcp->addrtype != SDP_AT_IP4 && rtcp->addrtype != SDP_AT_IP6) {
+    return;
+  }
+
+  if (!strlen(rtcp->addr)) {
+    SetAttribute(new SdpRtcpAttribute(rtcp->port));
+  } else {
+    SetAttribute(new SdpRtcpAttribute(
+        rtcp->port, sdp::kInternet,
+        rtcp->addrtype == SDP_AT_IP4 ? sdp::kIPv4 : sdp::kIPv6, rtcp->addr));
+  }
+}
+
+bool SipccSdpAttributeList::Load(sdp_t* sdp, uint16_t level,
+                                 InternalResults& results) {
+  LoadSimpleStrings(sdp, level, results);
+  LoadSimpleNumbers(sdp, level, results);
+  LoadFlags(sdp, level);
+  LoadDirection(sdp, level, results);
+
+  if (AtSessionLevel()) {
+    if (!LoadGroups(sdp, level, results)) {
+      return false;
+    }
+
+    if (!LoadMsidSemantics(sdp, level, results)) {
+      return false;
+    }
+
+    LoadIdentity(sdp, level);
+    LoadDtlsMessage(sdp, level);
+  } else {
+    sdp_media_e mtype = sdp_get_media_type(sdp, level);
+    if (mtype == SDP_MEDIA_APPLICATION) {
+      LoadSctpmap(sdp, level, results);
+    } else {
+      if (!LoadRtpmap(sdp, level, results)) {
+        return false;
+      }
+    }
+    LoadCandidate(sdp, level);
+    LoadFmtp(sdp, level);
+    LoadMsids(sdp, level, results);
+    LoadRtcpFb(sdp, level, results);
+    LoadRtcp(sdp, level, results);
+    LoadSsrc(sdp, level);
+    LoadSsrcGroup(sdp, level);
+    if (!LoadImageattr(sdp, level, results)) {
+      return false;
+    }
+    if (!LoadSimulcast(sdp, level, results)) {
+      return false;
+    }
+    if (!LoadRid(sdp, level, results)) {
+      return false;
+    }
+  }
+
+  LoadIceAttributes(sdp, level);
+  if (!LoadFingerprint(sdp, level, results)) {
+    return false;
+  }
+  LoadSetup(sdp, level);
+  LoadExtmap(sdp, level, results);
+
+  return true;
+}
+
+bool SipccSdpAttributeList::IsAllowedHere(
+    SdpAttribute::AttributeType type) const {
+  if (AtSessionLevel() && !SdpAttribute::IsAllowedAtSessionLevel(type)) {
+    return false;
+  }
+
+  if (!AtSessionLevel() && !SdpAttribute::IsAllowedAtMediaLevel(type)) {
+    return false;
+  }
+
+  return true;
+}
+
+void SipccSdpAttributeList::WarnAboutMisplacedAttribute(
+    SdpAttribute::AttributeType type, uint32_t lineNumber,
+    InternalResults& results) {
+  std::string warning = SdpAttribute::GetAttributeTypeString(type) +
+                        (AtSessionLevel() ? " at session level. Ignoring."
+                                          : " at media level. Ignoring.");
+  results.AddParseError(lineNumber, warning);
+}
+
+const std::vector<std::string>& SipccSdpAttributeList::GetCandidate() const {
+  if (!HasAttribute(SdpAttribute::kCandidateAttribute)) {
+    MOZ_CRASH();
+  }
+
+  return static_cast<const SdpMultiStringAttribute*>(
+             GetAttribute(SdpAttribute::kCandidateAttribute))
+      ->mValues;
+}
+
+const SdpConnectionAttribute& SipccSdpAttributeList::GetConnection() const {
+  if (!HasAttribute(SdpAttribute::kConnectionAttribute)) {
+    MOZ_CRASH();
+  }
+
+  return *static_cast<const SdpConnectionAttribute*>(
+      GetAttribute(SdpAttribute::kConnectionAttribute));
+}
+
+SdpDirectionAttribute::Direction SipccSdpAttributeList::GetDirection() const {
+  if (!HasAttribute(SdpAttribute::kDirectionAttribute)) {
+    MOZ_CRASH();
+  }
+
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kDirectionAttribute);
+  return static_cast<const SdpDirectionAttribute*>(attr)->mValue;
+}
+
+const SdpDtlsMessageAttribute& SipccSdpAttributeList::GetDtlsMessage() const {
+  if (!HasAttribute(SdpAttribute::kDtlsMessageAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kDtlsMessageAttribute);
+  return *static_cast<const SdpDtlsMessageAttribute*>(attr);
+}
+
+const SdpExtmapAttributeList& SipccSdpAttributeList::GetExtmap() const {
+  if (!HasAttribute(SdpAttribute::kExtmapAttribute)) {
+    MOZ_CRASH();
+  }
+
+  return *static_cast<const SdpExtmapAttributeList*>(
+      GetAttribute(SdpAttribute::kExtmapAttribute));
+}
+
+const SdpFingerprintAttributeList& SipccSdpAttributeList::GetFingerprint()
+    const {
+  if (!HasAttribute(SdpAttribute::kFingerprintAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kFingerprintAttribute);
+  return *static_cast<const SdpFingerprintAttributeList*>(attr);
+}
+
+const SdpFmtpAttributeList& SipccSdpAttributeList::GetFmtp() const {
+  if (!HasAttribute(SdpAttribute::kFmtpAttribute)) {
+    MOZ_CRASH();
+  }
+
+  return *static_cast<const SdpFmtpAttributeList*>(
+      GetAttribute(SdpAttribute::kFmtpAttribute));
+}
+
+const SdpGroupAttributeList& SipccSdpAttributeList::GetGroup() const {
+  if (!HasAttribute(SdpAttribute::kGroupAttribute)) {
+    MOZ_CRASH();
+  }
+
+  return *static_cast<const SdpGroupAttributeList*>(
+      GetAttribute(SdpAttribute::kGroupAttribute));
+}
+
+const SdpOptionsAttribute& SipccSdpAttributeList::GetIceOptions() const {
+  if (!HasAttribute(SdpAttribute::kIceOptionsAttribute)) {
+    MOZ_CRASH();
+  }
+
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kIceOptionsAttribute);
+  return *static_cast<const SdpOptionsAttribute*>(attr);
+}
+
+const std::string& SipccSdpAttributeList::GetIcePwd() const {
+  if (!HasAttribute(SdpAttribute::kIcePwdAttribute)) {
+    return kEmptyString;
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kIcePwdAttribute);
+  return static_cast<const SdpStringAttribute*>(attr)->mValue;
+}
+
+const std::string& SipccSdpAttributeList::GetIceUfrag() const {
+  if (!HasAttribute(SdpAttribute::kIceUfragAttribute)) {
+    return kEmptyString;
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kIceUfragAttribute);
+  return static_cast<const SdpStringAttribute*>(attr)->mValue;
+}
+
+const std::string& SipccSdpAttributeList::GetIdentity() const {
+  if (!HasAttribute(SdpAttribute::kIdentityAttribute)) {
+    return kEmptyString;
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kIdentityAttribute);
+  return static_cast<const SdpStringAttribute*>(attr)->mValue;
+}
+
+const SdpImageattrAttributeList& SipccSdpAttributeList::GetImageattr() const {
+  if (!HasAttribute(SdpAttribute::kImageattrAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kImageattrAttribute);
+  return *static_cast<const SdpImageattrAttributeList*>(attr);
+}
+
+const SdpSimulcastAttribute& SipccSdpAttributeList::GetSimulcast() const {
+  if (!HasAttribute(SdpAttribute::kSimulcastAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kSimulcastAttribute);
+  return *static_cast<const SdpSimulcastAttribute*>(attr);
+}
+
+const std::string& SipccSdpAttributeList::GetLabel() const {
+  if (!HasAttribute(SdpAttribute::kLabelAttribute)) {
+    return kEmptyString;
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kLabelAttribute);
+  return static_cast<const SdpStringAttribute*>(attr)->mValue;
+}
+
+uint32_t SipccSdpAttributeList::GetMaxptime() const {
+  if (!HasAttribute(SdpAttribute::kMaxptimeAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kMaxptimeAttribute);
+  return static_cast<const SdpNumberAttribute*>(attr)->mValue;
+}
+
+const std::string& SipccSdpAttributeList::GetMid() const {
+  if (!HasAttribute(SdpAttribute::kMidAttribute)) {
+    return kEmptyString;
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kMidAttribute);
+  return static_cast<const SdpStringAttribute*>(attr)->mValue;
+}
+
+const SdpMsidAttributeList& SipccSdpAttributeList::GetMsid() const {
+  if (!HasAttribute(SdpAttribute::kMsidAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kMsidAttribute);
+  return *static_cast<const SdpMsidAttributeList*>(attr);
+}
+
+const SdpMsidSemanticAttributeList& SipccSdpAttributeList::GetMsidSemantic()
+    const {
+  if (!HasAttribute(SdpAttribute::kMsidSemanticAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kMsidSemanticAttribute);
+  return *static_cast<const SdpMsidSemanticAttributeList*>(attr);
+}
+
+const SdpRidAttributeList& SipccSdpAttributeList::GetRid() const {
+  if (!HasAttribute(SdpAttribute::kRidAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kRidAttribute);
+  return *static_cast<const SdpRidAttributeList*>(attr);
+}
+
+uint32_t SipccSdpAttributeList::GetPtime() const {
+  if (!HasAttribute(SdpAttribute::kPtimeAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kPtimeAttribute);
+  return static_cast<const SdpNumberAttribute*>(attr)->mValue;
+}
+
+const SdpRtcpAttribute& SipccSdpAttributeList::GetRtcp() const {
+  if (!HasAttribute(SdpAttribute::kRtcpAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kRtcpAttribute);
+  return *static_cast<const SdpRtcpAttribute*>(attr);
+}
+
+const SdpRtcpFbAttributeList& SipccSdpAttributeList::GetRtcpFb() const {
+  if (!HasAttribute(SdpAttribute::kRtcpFbAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kRtcpFbAttribute);
+  return *static_cast<const SdpRtcpFbAttributeList*>(attr);
+}
+
+const SdpRemoteCandidatesAttribute& SipccSdpAttributeList::GetRemoteCandidates()
+    const {
+  MOZ_CRASH("Not yet implemented");
+}
+
+const SdpRtpmapAttributeList& SipccSdpAttributeList::GetRtpmap() const {
+  if (!HasAttribute(SdpAttribute::kRtpmapAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kRtpmapAttribute);
+  return *static_cast<const SdpRtpmapAttributeList*>(attr);
+}
+
+const SdpSctpmapAttributeList& SipccSdpAttributeList::GetSctpmap() const {
+  if (!HasAttribute(SdpAttribute::kSctpmapAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kSctpmapAttribute);
+  return *static_cast<const SdpSctpmapAttributeList*>(attr);
+}
+
+uint32_t SipccSdpAttributeList::GetSctpPort() const {
+  if (!HasAttribute(SdpAttribute::kSctpPortAttribute)) {
+    MOZ_CRASH();
+  }
+
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kSctpPortAttribute);
+  return static_cast<const SdpNumberAttribute*>(attr)->mValue;
+}
+
+uint32_t SipccSdpAttributeList::GetMaxMessageSize() const {
+  if (!HasAttribute(SdpAttribute::kMaxMessageSizeAttribute)) {
+    MOZ_CRASH();
+  }
+
+  const SdpAttribute* attr =
+      GetAttribute(SdpAttribute::kMaxMessageSizeAttribute);
+  return static_cast<const SdpNumberAttribute*>(attr)->mValue;
+}
+
+const SdpSetupAttribute& SipccSdpAttributeList::GetSetup() const {
+  if (!HasAttribute(SdpAttribute::kSetupAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kSetupAttribute);
+  return *static_cast<const SdpSetupAttribute*>(attr);
+}
+
+const SdpSsrcAttributeList& SipccSdpAttributeList::GetSsrc() const {
+  if (!HasAttribute(SdpAttribute::kSsrcAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kSsrcAttribute);
+  return *static_cast<const SdpSsrcAttributeList*>(attr);
+}
+
+const SdpSsrcGroupAttributeList& SipccSdpAttributeList::GetSsrcGroup() const {
+  if (!HasAttribute(SdpAttribute::kSsrcGroupAttribute)) {
+    MOZ_CRASH();
+  }
+  const SdpAttribute* attr = GetAttribute(SdpAttribute::kSsrcGroupAttribute);
+  return *static_cast<const SdpSsrcGroupAttributeList*>(attr);
+}
+
+void SipccSdpAttributeList::Serialize(std::ostream& os) const {
+  for (size_t i = 0; i < kNumAttributeTypes; ++i) {
+    if (mAttributes[i]) {
+      os << *mAttributes[i];
+    }
+  }
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/sdp/SipccSdpAttributeList.h b/dom/media/webrtc/sdp/SipccSdpAttributeList.h
new file mode 100644
index 0000000000..4499e84bb9
--- /dev/null
+++ b/dom/media/webrtc/sdp/SipccSdpAttributeList.h
@@ -0,0 +1,145 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _SIPCCSDPATTRIBUTELIST_H_
+#define _SIPCCSDPATTRIBUTELIST_H_
+
+#include "sdp/SdpParser.h"
+#include "sdp/SdpAttributeList.h"
+
+extern "C" {
+#include "sipcc_sdp.h"
+}
+
+namespace mozilla {
+
+class SipccSdp;
+class SipccSdpMediaSection;
+
+class SipccSdpAttributeList : public SdpAttributeList {
+  friend class SipccSdpMediaSection;
+  friend class SipccSdp;
+
+ public:
+  // Make sure we don't hide the default arg thunks
+  using SdpAttributeList::GetAttribute;
+  using SdpAttributeList::HasAttribute;
+
+  virtual bool HasAttribute(AttributeType type,
+                            bool sessionFallback) const override;
+  virtual const SdpAttribute* GetAttribute(AttributeType type,
+                                           bool sessionFallback) const override;
+  virtual void SetAttribute(SdpAttribute* attr) override;
+  virtual void RemoveAttribute(AttributeType type) override;
+  virtual void Clear() override;
+  virtual uint32_t Count() const override;
+
+  virtual const SdpConnectionAttribute& GetConnection() const override;
+  virtual const SdpFingerprintAttributeList& GetFingerprint() const override;
+  virtual const SdpGroupAttributeList& GetGroup() const override;
+  virtual const SdpOptionsAttribute& GetIceOptions() const override;
+  virtual const SdpRtcpAttribute& GetRtcp() const override;
+  virtual const SdpRemoteCandidatesAttribute& GetRemoteCandidates()
+      const override;
+  virtual const SdpSetupAttribute& GetSetup() const override;
+  virtual const SdpSsrcAttributeList& GetSsrc() const override;
+  virtual const SdpSsrcGroupAttributeList& GetSsrcGroup() const override;
+  virtual const SdpDtlsMessageAttribute& GetDtlsMessage() const override;
+
+  // These attributes can appear multiple times, so the returned
+  // classes actually represent a collection of values.
+  virtual const std::vector<std::string>& GetCandidate() const override;
+  virtual const SdpExtmapAttributeList& GetExtmap() const override;
+  virtual const SdpFmtpAttributeList& GetFmtp() const override;
+  virtual const SdpImageattrAttributeList& GetImageattr() const override;
+  const SdpSimulcastAttribute& GetSimulcast() const override;
+  virtual const SdpMsidAttributeList& GetMsid() const override;
+  virtual const SdpMsidSemanticAttributeList& GetMsidSemantic() const override;
+  const SdpRidAttributeList& GetRid() const override;
+  virtual const SdpRtcpFbAttributeList& GetRtcpFb() const override;
+  virtual const SdpRtpmapAttributeList& GetRtpmap() const override;
+  virtual const SdpSctpmapAttributeList& GetSctpmap() const override;
+  virtual uint32_t GetSctpPort() const override;
+  virtual uint32_t GetMaxMessageSize() const override;
+
+  // These attributes are effectively simple types, so we'll make life
+  // easy by just returning their value.
+  virtual const std::string& GetIcePwd() const override;
+  virtual const std::string& GetIceUfrag() const override;
+  virtual const std::string& GetIdentity() const override;
+  virtual const std::string& GetLabel() const override;
+  virtual unsigned int GetMaxptime() const override;
+  virtual const std::string& GetMid() const override;
+  virtual unsigned int GetPtime() const override;
+
+  virtual SdpDirectionAttribute::Direction GetDirection() const override;
+
+  virtual void Serialize(std::ostream&) const override;
+
+  virtual ~SipccSdpAttributeList();
+
+ private:
+  static const std::string kEmptyString;
+  static const size_t kNumAttributeTypes = SdpAttribute::kLastAttribute + 1;
+
+  // Pass a session-level attribute list if constructing a media-level one,
+  // otherwise pass nullptr
+  explicit SipccSdpAttributeList(const SipccSdpAttributeList* sessionLevel);
+
+  // Copy c'tor, sort of
+  SipccSdpAttributeList(const SipccSdpAttributeList& aOrig,
+                        const SipccSdpAttributeList* sessionLevel);
+
+  using InternalResults = SdpParser::InternalResults;
+
+  bool Load(sdp_t* sdp, uint16_t level, InternalResults& results);
+  void LoadSimpleStrings(sdp_t* sdp, uint16_t level, InternalResults& results);
+  void LoadSimpleString(sdp_t* sdp, uint16_t level, sdp_attr_e attr,
+                        AttributeType targetType, InternalResults& results);
+  void LoadSimpleNumbers(sdp_t* sdp, uint16_t level, InternalResults& results);
+  void LoadSimpleNumber(sdp_t* sdp, uint16_t level, sdp_attr_e attr,
+                        AttributeType targetType, InternalResults& results);
+  void LoadFlags(sdp_t* sdp, uint16_t level);
+  void LoadDirection(sdp_t* sdp, uint16_t level, InternalResults& results);
+  bool LoadRtpmap(sdp_t* sdp, uint16_t level, InternalResults& results);
+  bool LoadSctpmap(sdp_t* sdp, uint16_t level, InternalResults& results);
+  void LoadIceAttributes(sdp_t* sdp, uint16_t level);
+  bool LoadFingerprint(sdp_t* sdp, uint16_t level, InternalResults& results);
+  void LoadCandidate(sdp_t* sdp, uint16_t level);
+  void LoadSetup(sdp_t* sdp, uint16_t level);
+  void LoadSsrc(sdp_t* sdp, uint16_t level);
+  void LoadSsrcGroup(sdp_t* sdp, uint16_t level);
+  bool LoadImageattr(sdp_t* sdp, uint16_t level, InternalResults& results);
+  bool LoadSimulcast(sdp_t* sdp, uint16_t level, InternalResults& results);
+  bool LoadGroups(sdp_t* sdp, uint16_t level, InternalResults& results);
+  bool LoadMsidSemantics(sdp_t* sdp, uint16_t level, InternalResults& results);
+  void LoadIdentity(sdp_t* sdp, uint16_t level);
+  void LoadDtlsMessage(sdp_t* sdp, uint16_t level);
+  void LoadFmtp(sdp_t* sdp, uint16_t level);
+  void LoadMsids(sdp_t* sdp, uint16_t level, InternalResults& results);
+  bool LoadRid(sdp_t* sdp, uint16_t level, InternalResults& results);
+  void LoadExtmap(sdp_t* sdp, uint16_t level, InternalResults& results);
+  void LoadRtcpFb(sdp_t* sdp, uint16_t level, InternalResults& results);
+  void LoadRtcp(sdp_t* sdp, uint16_t level, InternalResults& results);
+  static SdpRtpmapAttributeList::CodecType GetCodecType(rtp_ptype type);
+
+  bool AtSessionLevel() const { return !mSessionLevel; }
+  bool IsAllowedHere(SdpAttribute::AttributeType type) const;
+  void WarnAboutMisplacedAttribute(SdpAttribute::AttributeType type,
+                                   uint32_t lineNumber,
+                                   InternalResults& results);
+
+  const SipccSdpAttributeList* mSessionLevel;
+
+  SdpAttribute* mAttributes[kNumAttributeTypes];
+
+  SipccSdpAttributeList(const SipccSdpAttributeList& orig) = delete;
+  SipccSdpAttributeList& operator=(const SipccSdpAttributeList& rhs) = delete;
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/sdp/SipccSdpMediaSection.cpp b/dom/media/webrtc/sdp/SipccSdpMediaSection.cpp
new file mode 100644
index 0000000000..4304ca541e
--- /dev/null
+++ b/dom/media/webrtc/sdp/SipccSdpMediaSection.cpp
@@ -0,0 +1,401 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "sdp/SipccSdpMediaSection.h"
+
+#include <ostream>
+#include "sdp/SdpParser.h"
+
+extern "C" {
+#include "sipcc_sdp.h"
+}
+
+#ifdef CRLF
+#  undef CRLF
+#endif
+#define CRLF "\r\n"
+
+namespace mozilla {
+
+SipccSdpMediaSection::SipccSdpMediaSection(
+    const SipccSdpMediaSection& aOrig,
+    const SipccSdpAttributeList* sessionLevel)
+    : SdpMediaSection(aOrig),
+      mMediaType(aOrig.mMediaType),
+      mPort(aOrig.mPort),
+      mPortCount(aOrig.mPortCount),
+      mProtocol(aOrig.mProtocol),
+      mFormats(aOrig.mFormats),
+      mConnection(new SdpConnection(*aOrig.mConnection)),
+      mBandwidths(aOrig.mBandwidths),
+      mAttributeList(aOrig.mAttributeList, sessionLevel) {}
+
+unsigned int SipccSdpMediaSection::GetPort() const { return mPort; }
+
+void SipccSdpMediaSection::SetPort(unsigned int port) { mPort = port; }
+
+unsigned int SipccSdpMediaSection::GetPortCount() const { return mPortCount; }
+
+SdpMediaSection::Protocol SipccSdpMediaSection::GetProtocol() const {
+  return mProtocol;
+}
+
+const SdpConnection& SipccSdpMediaSection::GetConnection() const {
+  return *mConnection;
+}
+
+SdpConnection& SipccSdpMediaSection::GetConnection() { return *mConnection; }
+
+uint32_t SipccSdpMediaSection::GetBandwidth(const std::string& type) const {
+  auto found = mBandwidths.find(type);
+  if (found == mBandwidths.end()) {
+    return 0;
+  }
+  return found->second;
+}
+
+const std::vector<std::string>& SipccSdpMediaSection::GetFormats() const {
+  return mFormats;
+}
+
+const SdpAttributeList& SipccSdpMediaSection::GetAttributeList() const {
+  return mAttributeList;
+}
+
+SdpAttributeList& SipccSdpMediaSection::GetAttributeList() {
+  return mAttributeList;
+}
+
+SdpDirectionAttribute SipccSdpMediaSection::GetDirectionAttribute() const {
+  return SdpDirectionAttribute(mAttributeList.GetDirection());
+}
+
+bool SipccSdpMediaSection::Load(sdp_t* sdp, uint16_t level,
+                                InternalResults& results) {
+  switch (sdp_get_media_type(sdp, level)) {
+    case SDP_MEDIA_AUDIO:
+      mMediaType = kAudio;
+      break;
+    case SDP_MEDIA_VIDEO:
+      mMediaType = kVideo;
+      break;
+    case SDP_MEDIA_APPLICATION:
+      mMediaType = kApplication;
+      break;
+    case SDP_MEDIA_TEXT:
+      mMediaType = kText;
+      break;
+
+    default:
+      results.AddParseError(sdp_get_media_line_number(sdp, level),
+                            "Unsupported media section type");
+      return false;
+  }
+
+  mPort = sdp_get_media_portnum(sdp, level);
+  int32_t pc = sdp_get_media_portcount(sdp, level);
+  if (pc == SDP_INVALID_VALUE) {
+    // SDP_INVALID_VALUE (ie; -2) is used when there is no port count. :(
+    mPortCount = 0;
+  } else if (pc > static_cast<int32_t>(UINT16_MAX) || pc < 0) {
+    results.AddParseError(sdp_get_media_line_number(sdp, level),
+                          "Invalid port count");
+    return false;
+  } else {
+    mPortCount = pc;
+  }
+
+  if (!LoadProtocol(sdp, level, results)) {
+    return false;
+  }
+
+  if (!LoadFormats(sdp, level, results)) {
+    return false;
+  }
+
+  if (!mAttributeList.Load(sdp, level, results)) {
+    return false;
+  }
+
+  if (!ValidateSimulcast(sdp, level, results)) {
+    return false;
+  }
+
+  if (!mBandwidths.Load(sdp, level, results)) {
+    return false;
+  }
+
+  return LoadConnection(sdp, level, results);
+}
+
+bool SipccSdpMediaSection::LoadProtocol(sdp_t* sdp, uint16_t level,
+                                        InternalResults& results) {
+  switch (sdp_get_media_transport(sdp, level)) {
+    case SDP_TRANSPORT_RTPAVP:
+      mProtocol = kRtpAvp;
+      break;
+    case SDP_TRANSPORT_RTPSAVP:
+      mProtocol = kRtpSavp;
+      break;
+    case SDP_TRANSPORT_RTPAVPF:
+      mProtocol = kRtpAvpf;
+      break;
+    case SDP_TRANSPORT_RTPSAVPF:
+      mProtocol = kRtpSavpf;
+      break;
+    case SDP_TRANSPORT_UDPTLSRTPSAVP:
+      mProtocol = kUdpTlsRtpSavp;
+      break;
+    case SDP_TRANSPORT_UDPTLSRTPSAVPF:
+      mProtocol = kUdpTlsRtpSavpf;
+      break;
+    case SDP_TRANSPORT_TCPDTLSRTPSAVP:
+      mProtocol = kTcpDtlsRtpSavp;
+      break;
+    case SDP_TRANSPORT_TCPDTLSRTPSAVPF:
+      mProtocol = kTcpDtlsRtpSavpf;
+      break;
+    case SDP_TRANSPORT_DTLSSCTP:
+      mProtocol = kDtlsSctp;
+      break;
+    case SDP_TRANSPORT_UDPDTLSSCTP:
+      mProtocol = kUdpDtlsSctp;
+      break;
+    case SDP_TRANSPORT_TCPDTLSSCTP:
+      mProtocol = kTcpDtlsSctp;
+      break;
+
+    default:
+      results.AddParseError(sdp_get_media_line_number(sdp, level),
+                            "Unsupported media transport type");
+      return false;
+  }
+  return true;
+}
+
+bool SipccSdpMediaSection::LoadFormats(sdp_t* sdp, uint16_t level,
+                                       InternalResults& results) {
+  sdp_media_e mtype = sdp_get_media_type(sdp, level);
+
+  if (mtype == SDP_MEDIA_APPLICATION) {
+    sdp_transport_e ttype = sdp_get_media_transport(sdp, level);
+    if ((ttype == SDP_TRANSPORT_UDPDTLSSCTP) ||
+        (ttype == SDP_TRANSPORT_TCPDTLSSCTP)) {
+      if (sdp_get_media_sctp_fmt(sdp, level) ==
+          SDP_SCTP_MEDIA_FMT_WEBRTC_DATACHANNEL) {
+        mFormats.push_back("webrtc-datachannel");
+      }
+    } else {
+      uint32_t ptype = sdp_get_media_sctp_port(sdp, level);
+      std::ostringstream osPayloadType;
+      osPayloadType << ptype;
+      mFormats.push_back(osPayloadType.str());
+    }
+  } else if (mtype == SDP_MEDIA_AUDIO || mtype == SDP_MEDIA_VIDEO) {
+    uint16_t count = sdp_get_media_num_payload_types(sdp, level);
+    for (uint16_t i = 0; i < count; ++i) {
+      sdp_payload_ind_e indicator;  // we ignore this, which is fine
+      uint32_t ptype =
+          sdp_get_media_payload_type(sdp, level, i + 1, &indicator);
+
+      if (GET_DYN_PAYLOAD_TYPE_VALUE(ptype) > UINT8_MAX) {
+        results.AddParseError(sdp_get_media_line_number(sdp, level),
+                              "Format is too large");
+        return false;
+      }
+
+      std::ostringstream osPayloadType;
+      // sipcc stores payload types in a funny way. When sipcc and the SDP it
+      // parsed differ on what payload type number should be used for a given
+      // codec, sipcc's value goes in the lower byte, and the SDP's value in
+      // the upper byte. When they do not differ, only the lower byte is used.
+      // We want what was in the SDP, verbatim.
+      osPayloadType << GET_DYN_PAYLOAD_TYPE_VALUE(ptype);
+      mFormats.push_back(osPayloadType.str());
+    }
+  }
+
+  return true;
+}
+
+bool SipccSdpMediaSection::ValidateSimulcast(sdp_t* sdp, uint16_t level,
+                                             InternalResults& results) const {
+  if (!GetAttributeList().HasAttribute(SdpAttribute::kSimulcastAttribute)) {
+    return true;
+  }
+
+  const SdpSimulcastAttribute& simulcast(GetAttributeList().GetSimulcast());
+  if (!ValidateSimulcastVersions(sdp, level, simulcast.sendVersions, sdp::kSend,
+                                 results)) {
+    return false;
+  }
+  if (!ValidateSimulcastVersions(sdp, level, simulcast.recvVersions, sdp::kRecv,
+                                 results)) {
+    return false;
+  }
+  return true;
+}
+
+bool SipccSdpMediaSection::ValidateSimulcastVersions(
+    sdp_t* sdp, uint16_t level, const SdpSimulcastAttribute::Versions& versions,
+    sdp::Direction direction, InternalResults& results) const {
+  for (const SdpSimulcastAttribute::Version& version : versions) {
+    for (const SdpSimulcastAttribute::Encoding& encoding : version.choices) {
+      const SdpRidAttributeList::Rid* ridAttr = FindRid(encoding.rid);
+      if (!ridAttr || (ridAttr->direction != direction)) {
+        std::ostringstream os;
+        os << "No rid attribute for \'" << encoding.rid << "\'";
+        results.AddParseError(sdp_get_media_line_number(sdp, level), os.str());
+        results.AddParseError(sdp_get_media_line_number(sdp, level), os.str());
+        return false;
+      }
+    }
+  }
+  return true;
+}
+
+bool SipccSdpMediaSection::LoadConnection(sdp_t* sdp, uint16_t level,
+                                          InternalResults& results) {
+  if (!sdp_connection_valid(sdp, level)) {
+    level = SDP_SESSION_LEVEL;
+    if (!sdp_connection_valid(sdp, level)) {
+      results.AddParseError(sdp_get_media_line_number(sdp, level),
+                            "Missing c= line");
+      return false;
+    }
+  }
+
+  sdp_nettype_e type = sdp_get_conn_nettype(sdp, level);
+  if (type != SDP_NT_INTERNET) {
+    results.AddParseError(sdp_get_media_line_number(sdp, level),
+                          "Unsupported network type");
+    return false;
+  }
+
+  sdp::AddrType addrType;
+  switch (sdp_get_conn_addrtype(sdp, level)) {
+    case SDP_AT_IP4:
+      addrType = sdp::kIPv4;
+      break;
+    case SDP_AT_IP6:
+      addrType = sdp::kIPv6;
+      break;
+    default:
+      results.AddParseError(sdp_get_media_line_number(sdp, level),
+                            "Unsupported address type");
+      return false;
+  }
+
+  std::string address = sdp_get_conn_address(sdp, level);
+  int16_t ttl = static_cast<uint16_t>(sdp_get_mcast_ttl(sdp, level));
+  if (ttl < 0) {
+    ttl = 0;
+  }
+  int32_t numAddr =
+      static_cast<uint32_t>(sdp_get_mcast_num_of_addresses(sdp, level));
+  if (numAddr < 0) {
+    numAddr = 0;
+  }
+  mConnection = MakeUnique<SdpConnection>(addrType, address, ttl, numAddr);
+  return true;
+}
+
+void SipccSdpMediaSection::AddCodec(const std::string& pt,
+                                    const std::string& name, uint32_t clockrate,
+                                    uint16_t channels) {
+  mFormats.push_back(pt);
+
+  SdpRtpmapAttributeList* rtpmap = new SdpRtpmapAttributeList();
+  if (mAttributeList.HasAttribute(SdpAttribute::kRtpmapAttribute)) {
+    const SdpRtpmapAttributeList& old = mAttributeList.GetRtpmap();
+    for (auto it = old.mRtpmaps.begin(); it != old.mRtpmaps.end(); ++it) {
+      rtpmap->mRtpmaps.push_back(*it);
+    }
+  }
+  SdpRtpmapAttributeList::CodecType codec = SdpRtpmapAttributeList::kOtherCodec;
+  if (name == "opus") {
+    codec = SdpRtpmapAttributeList::kOpus;
+  } else if (name == "G722") {
+    codec = SdpRtpmapAttributeList::kG722;
+  } else if (name == "PCMU") {
+    codec = SdpRtpmapAttributeList::kPCMU;
+  } else if (name == "PCMA") {
+    codec = SdpRtpmapAttributeList::kPCMA;
+  } else if (name == "VP8") {
+    codec = SdpRtpmapAttributeList::kVP8;
+  } else if (name == "VP9") {
+    codec = SdpRtpmapAttributeList::kVP9;
+  } else if (name == "H264") {
+    codec = SdpRtpmapAttributeList::kH264;
+  }
+
+  rtpmap->PushEntry(pt, codec, name, clockrate, channels);
+  mAttributeList.SetAttribute(rtpmap);
+}
+
+void SipccSdpMediaSection::ClearCodecs() {
+  mFormats.clear();
+  mAttributeList.RemoveAttribute(SdpAttribute::kRtpmapAttribute);
+  mAttributeList.RemoveAttribute(SdpAttribute::kFmtpAttribute);
+  mAttributeList.RemoveAttribute(SdpAttribute::kSctpmapAttribute);
+  mAttributeList.RemoveAttribute(SdpAttribute::kRtcpFbAttribute);
+}
+
+void SipccSdpMediaSection::AddDataChannel(const std::string& name,
+                                          uint16_t port, uint16_t streams,
+                                          uint32_t message_size) {
+  // Only one allowed, for now. This may change as the specs (and deployments)
+  // evolve.
+  mFormats.clear();
+  if ((mProtocol == kUdpDtlsSctp) || (mProtocol == kTcpDtlsSctp)) {
+    // new data channel format according to draft 21
+    mFormats.push_back(name);
+    mAttributeList.SetAttribute(
+        new SdpNumberAttribute(SdpAttribute::kSctpPortAttribute, port));
+    if (message_size) {
+      mAttributeList.SetAttribute(new SdpNumberAttribute(
+          SdpAttribute::kMaxMessageSizeAttribute, message_size));
+    }
+  } else {
+    // old data channels format according to draft 05
+    std::string port_str = std::to_string(port);
+    mFormats.push_back(port_str);
+    SdpSctpmapAttributeList* sctpmap = new SdpSctpmapAttributeList();
+    sctpmap->PushEntry(port_str, name, streams);
+    mAttributeList.SetAttribute(sctpmap);
+    if (message_size) {
+      // This is a workaround to allow detecting Firefox's w/o EOR support
+      mAttributeList.SetAttribute(new SdpNumberAttribute(
+          SdpAttribute::kMaxMessageSizeAttribute, message_size));
+    }
+  }
+}
+
+void SipccSdpMediaSection::Serialize(std::ostream& os) const {
+  os << "m=" << mMediaType << " " << mPort;
+  if (mPortCount) {
+    os << "/" << mPortCount;
+  }
+  os << " " << mProtocol;
+  for (auto i = mFormats.begin(); i != mFormats.end(); ++i) {
+    os << " " << (*i);
+  }
+  os << CRLF;
+
+  // We dont do i=
+
+  if (mConnection) {
+    os << *mConnection;
+  }
+
+  mBandwidths.Serialize(os);
+
+  // We dont do k= because they're evil
+
+  os << mAttributeList;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/sdp/SipccSdpMediaSection.h b/dom/media/webrtc/sdp/SipccSdpMediaSection.h
new file mode 100644
index 0000000000..c9abb4d33c
--- /dev/null
+++ b/dom/media/webrtc/sdp/SipccSdpMediaSection.h
@@ -0,0 +1,101 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _SIPCCSDPMEDIASECTION_H_
+#define _SIPCCSDPMEDIASECTION_H_
+
+#include "mozilla/Attributes.h"
+#include "mozilla/UniquePtr.h"
+#include "sdp/SdpMediaSection.h"
+#include "sdp/SipccSdpAttributeList.h"
+
+#include <map>
+
+extern "C" {
+#include "sipcc_sdp.h"
+}
+
+namespace mozilla {
+
+class SipccSdp;
+class SdpParser;
+
+using InternalResults = SdpParser::InternalResults;
+
+class SipccSdpBandwidths final : public std::map<std::string, uint32_t> {
+ public:
+  bool Load(sdp_t* sdp, uint16_t level, InternalResults& results);
+  void Serialize(std::ostream& os) const;
+};
+
+class SipccSdpMediaSection final : public SdpMediaSection {
+  friend class SipccSdp;
+
+ public:
+  ~SipccSdpMediaSection() {}
+
+  virtual MediaType GetMediaType() const override { return mMediaType; }
+
+  virtual unsigned int GetPort() const override;
+  virtual void SetPort(unsigned int port) override;
+  virtual unsigned int GetPortCount() const override;
+  virtual Protocol GetProtocol() const override;
+  virtual const SdpConnection& GetConnection() const override;
+  virtual SdpConnection& GetConnection() override;
+  virtual uint32_t GetBandwidth(const std::string& type) const override;
+  virtual const std::vector<std::string>& GetFormats() const override;
+
+  virtual const SdpAttributeList& GetAttributeList() const override;
+  virtual SdpAttributeList& GetAttributeList() override;
+  virtual SdpDirectionAttribute GetDirectionAttribute() const override;
+
+  virtual void AddCodec(const std::string& pt, const std::string& name,
+                        uint32_t clockrate, uint16_t channels) override;
+  virtual void ClearCodecs() override;
+
+  virtual void AddDataChannel(const std::string& name, uint16_t port,
+                              uint16_t streams, uint32_t message_size) override;
+
+  virtual void Serialize(std::ostream&) const override;
+
+ private:
+  SipccSdpMediaSection(size_t level, const SipccSdpAttributeList* sessionLevel)
+      : SdpMediaSection(level),
+        mMediaType(static_cast<MediaType>(0)),
+        mPort(0),
+        mPortCount(0),
+        mProtocol(static_cast<Protocol>(0)),
+        mAttributeList(sessionLevel) {}
+
+  SipccSdpMediaSection(const SipccSdpMediaSection& aOrig,
+                       const SipccSdpAttributeList* sessionLevel);
+
+  bool Load(sdp_t* sdp, uint16_t level, InternalResults& results);
+  bool LoadConnection(sdp_t* sdp, uint16_t level, InternalResults& results);
+  bool LoadProtocol(sdp_t* sdp, uint16_t level, InternalResults& results);
+  bool LoadFormats(sdp_t* sdp, uint16_t level, InternalResults& results);
+  bool ValidateSimulcast(sdp_t* sdp, uint16_t level,
+                         InternalResults& results) const;
+  bool ValidateSimulcastVersions(
+      sdp_t* sdp, uint16_t level,
+      const SdpSimulcastAttribute::Versions& versions, sdp::Direction direction,
+      InternalResults& results) const;
+
+  // the following values are cached on first get
+  MediaType mMediaType;
+  uint16_t mPort;
+  uint16_t mPortCount;
+  Protocol mProtocol;
+  std::vector<std::string> mFormats;
+
+  UniquePtr<SdpConnection> mConnection;
+  SipccSdpBandwidths mBandwidths;
+
+  SipccSdpAttributeList mAttributeList;
+};
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/sdp/SipccSdpParser.cpp b/dom/media/webrtc/sdp/SipccSdpParser.cpp
new file mode 100644
index 0000000000..901efd041f
--- /dev/null
+++ b/dom/media/webrtc/sdp/SipccSdpParser.cpp
@@ -0,0 +1,88 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "sdp/SipccSdpParser.h"
+#include "sdp/SipccSdp.h"
+
+#include <utility>
+extern "C" {
+#include "sipcc_sdp.h"
+}
+
+namespace mozilla {
+
+extern "C" {
+
+void sipcc_sdp_parser_results_handler(void* context, uint32_t line,
+                                      const char* message) {
+  auto* results = static_cast<UniquePtr<InternalResults>*>(context);
+  std::string err(message);
+  (*results)->AddParseError(line, err);
+}
+
+}  // extern "C"
+
+const std::string& SipccSdpParser::ParserName() {
+  static const std::string SIPCC_NAME = "SIPCC";
+  return SIPCC_NAME;
+}
+
+UniquePtr<SdpParser::Results> SipccSdpParser::Parse(const std::string& aText) {
+  UniquePtr<InternalResults> results(new InternalResults(Name()));
+  sdp_conf_options_t* sipcc_config = sdp_init_config();
+  if (!sipcc_config) {
+    return UniquePtr<SdpParser::Results>();
+  }
+
+  sdp_nettype_supported(sipcc_config, SDP_NT_INTERNET, true);
+  sdp_addrtype_supported(sipcc_config, SDP_AT_IP4, true);
+  sdp_addrtype_supported(sipcc_config, SDP_AT_IP6, true);
+  sdp_transport_supported(sipcc_config, SDP_TRANSPORT_RTPAVP, true);
+  sdp_transport_supported(sipcc_config, SDP_TRANSPORT_RTPAVPF, true);
+  sdp_transport_supported(sipcc_config, SDP_TRANSPORT_RTPSAVP, true);
+  sdp_transport_supported(sipcc_config, SDP_TRANSPORT_RTPSAVPF, true);
+  sdp_transport_supported(sipcc_config, SDP_TRANSPORT_UDPTLSRTPSAVP, true);
+  sdp_transport_supported(sipcc_config, SDP_TRANSPORT_UDPTLSRTPSAVPF, true);
+  sdp_transport_supported(sipcc_config, SDP_TRANSPORT_TCPDTLSRTPSAVP, true);
+  sdp_transport_supported(sipcc_config, SDP_TRANSPORT_TCPDTLSRTPSAVPF, true);
+  sdp_transport_supported(sipcc_config, SDP_TRANSPORT_DTLSSCTP, true);
+  sdp_transport_supported(sipcc_config, SDP_TRANSPORT_UDPDTLSSCTP, true);
+  sdp_transport_supported(sipcc_config, SDP_TRANSPORT_TCPDTLSSCTP, true);
+  sdp_require_session_name(sipcc_config, false);
+
+  sdp_config_set_error_handler(sipcc_config, &sipcc_sdp_parser_results_handler,
+                               &results);
+
+  // Takes ownership of |sipcc_config| iff it succeeds
+  sdp_t* sdp = sdp_init_description(sipcc_config);
+  if (!sdp) {
+    sdp_free_config(sipcc_config);
+    return results;
+  }
+
+  const char* rawString = aText.c_str();
+  sdp_result_e sdpres = sdp_parse(sdp, rawString, aText.length());
+  if (sdpres != SDP_SUCCESS) {
+    sdp_free_description(sdp);
+    return results;
+  }
+
+  UniquePtr<SipccSdp> sipccSdp(new SipccSdp);
+
+  bool success = sipccSdp->Load(sdp, *results);
+  sdp_free_description(sdp);
+  if (success) {
+    results->SetSdp(UniquePtr<mozilla::Sdp>(std::move(sipccSdp)));
+  }
+
+  return results;
+}
+
+bool SipccSdpParser::IsNamed(const std::string& aName) {
+  return aName == ParserName();
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/sdp/SipccSdpParser.h b/dom/media/webrtc/sdp/SipccSdpParser.h
new file mode 100644
index 0000000000..dc9c1ea02d
--- /dev/null
+++ b/dom/media/webrtc/sdp/SipccSdpParser.h
@@ -0,0 +1,35 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef _SIPCCSDPPARSER_H_
+#define _SIPCCSDPPARSER_H_
+
+#include <string>
+
+#include "mozilla/UniquePtr.h"
+
+#include "sdp/Sdp.h"
+#include "sdp/SdpParser.h"
+
+namespace mozilla {
+
+class SipccSdpParser final : public SdpParser {
+  static const std::string& ParserName();
+
+ public:
+  SipccSdpParser() = default;
+  virtual ~SipccSdpParser() = default;
+
+  const std::string& Name() const override { return ParserName(); }
+
+  UniquePtr<SdpParser::Results> Parse(const std::string& aText) override;
+
+  static bool IsNamed(const std::string& aName);
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/sdp/moz.build b/dom/media/webrtc/sdp/moz.build
new file mode 100644
index 0000000000..675009acd0
--- /dev/null
+++ b/dom/media/webrtc/sdp/moz.build
@@ -0,0 +1,48 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+if CONFIG["OS_TARGET"] == "WINNT":
+    DEFINES["SIP_OS_WINDOWS"] = True
+elif CONFIG["OS_TARGET"] == "Darwin":
+    DEFINES["SIP_OS_OSX"] = True
+else:
+    DEFINES["SIP_OS_LINUX"] = True
+
+# Add libFuzzer configuration directives
+include("/tools/fuzzing/libfuzzer-config.mozbuild")
+
+LOCAL_INCLUDES += [
+    "/dom/media/webrtc",
+    "/media/webrtc",
+    "/third_party/sipcc",
+]
+
+UNIFIED_SOURCES += [
+    "HybridSdpParser.cpp",
+    "ParsingResultComparer.cpp",
+    "SdpAttribute.cpp",
+    "SdpHelper.cpp",
+    "SdpLog.cpp",
+    "SdpMediaSection.cpp",
+    "SdpPref.cpp",
+    "SdpTelemetry.cpp",
+    "SipccSdp.cpp",
+    "SipccSdpAttributeList.cpp",
+    "SipccSdpMediaSection.cpp",
+    "SipccSdpParser.cpp",
+]
+
+SOURCES += [
+    # Building these as part of the unified build leads to multiply defined
+    # symbols on windows.
+    "RsdparsaSdp.cpp",
+    "RsdparsaSdpAttributeList.cpp",
+    "RsdparsaSdpGlue.cpp",
+    "RsdparsaSdpMediaSection.cpp",
+    "RsdparsaSdpParser.cpp",
+]
+
+FINAL_LIBRARY = "xul"
diff --git a/dom/media/webrtc/sdp/rsdparsa_capi/Cargo.toml b/dom/media/webrtc/sdp/rsdparsa_capi/Cargo.toml
new file mode 100644
index 0000000000..a0aeff086f
--- /dev/null
+++ b/dom/media/webrtc/sdp/rsdparsa_capi/Cargo.toml
@@ -0,0 +1,12 @@
+[package]
+name = "rsdparsa_capi"
+version = "0.1.0"
+authors = ["Paul Ellenbogen <pe5@cs.princeton.edu>",
+	   "Nils Ohlmeier <github@ohlmeier.org>"]
+license = "MPL-2.0"
+
+[dependencies]
+libc = "^0.2.0"
+log = "0.4"
+rsdparsa = {package = "webrtc-sdp", version = "0.3.10"}
+nserror = { path = "../../../../../xpcom/rust/nserror" }
diff --git a/dom/media/webrtc/sdp/rsdparsa_capi/src/attribute.rs b/dom/media/webrtc/sdp/rsdparsa_capi/src/attribute.rs
new file mode 100644
index 0000000000..82483f151f
--- /dev/null
+++ b/dom/media/webrtc/sdp/rsdparsa_capi/src/attribute.rs
@@ -0,0 +1,1472 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+use libc::{c_float, size_t};
+use std::ptr;
+use std::slice;
+
+use nserror::{nsresult, NS_ERROR_INVALID_ARG, NS_OK};
+use rsdparsa::attribute_type::*;
+use rsdparsa::SdpSession;
+
+use network::{RustAddress, RustExplicitlyTypedAddress};
+use types::StringView;
+
+#[no_mangle]
+pub unsafe extern "C" fn num_attributes(session: *const SdpSession) -> u32 {
+    (*session).attribute.len() as u32
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn get_attribute_ptr(
+    session: *const SdpSession,
+    index: u32,
+    ret: *mut *const SdpAttribute,
+) -> nsresult {
+    match (*session).attribute.get(index as usize) {
+        Some(attribute) => {
+            *ret = attribute as *const SdpAttribute;
+            NS_OK
+        }
+        None => NS_ERROR_INVALID_ARG,
+    }
+}
+
+fn count_attribute(attributes: &[SdpAttribute], search: SdpAttributeType) -> usize {
+    let mut count = 0;
+    for attribute in (*attributes).iter() {
+        if SdpAttributeType::from(attribute) == search {
+            count += 1;
+        }
+    }
+    count
+}
+
+fn argsearch(attributes: &[SdpAttribute], attribute_type: SdpAttributeType) -> Option<usize> {
+    for (i, attribute) in (*attributes).iter().enumerate() {
+        if SdpAttributeType::from(attribute) == attribute_type {
+            return Some(i);
+        }
+    }
+    None
+}
+
+pub unsafe fn has_attribute(
+    attributes: *const Vec<SdpAttribute>,
+    attribute_type: SdpAttributeType,
+) -> bool {
+    argsearch((*attributes).as_slice(), attribute_type).is_some()
+}
+
+fn get_attribute(
+    attributes: &[SdpAttribute],
+    attribute_type: SdpAttributeType,
+) -> Option<&SdpAttribute> {
+    argsearch(attributes, attribute_type).map(|i| &attributes[i])
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub enum RustSdpAttributeDtlsMessageType {
+    Client,
+    Server,
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustSdpAttributeDtlsMessage {
+    pub role: u8,
+    pub value: StringView,
+}
+
+impl<'a> From<&'a SdpAttributeDtlsMessage> for RustSdpAttributeDtlsMessage {
+    fn from(other: &SdpAttributeDtlsMessage) -> Self {
+        match other {
+            &SdpAttributeDtlsMessage::Client(ref x) => RustSdpAttributeDtlsMessage {
+                role: RustSdpAttributeDtlsMessageType::Client as u8,
+                value: StringView::from(x.as_str()),
+            },
+            &SdpAttributeDtlsMessage::Server(ref x) => RustSdpAttributeDtlsMessage {
+                role: RustSdpAttributeDtlsMessageType::Server as u8,
+                value: StringView::from(x.as_str()),
+            },
+        }
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_dtls_message(
+    attributes: *const Vec<SdpAttribute>,
+    ret: *mut RustSdpAttributeDtlsMessage,
+) -> nsresult {
+    let attr = get_attribute((*attributes).as_slice(), SdpAttributeType::DtlsMessage);
+    if let Some(&SdpAttribute::DtlsMessage(ref dtls_message)) = attr {
+        *ret = RustSdpAttributeDtlsMessage::from(dtls_message);
+        return NS_OK;
+    }
+    NS_ERROR_INVALID_ARG
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_iceufrag(
+    attributes: *const Vec<SdpAttribute>,
+    ret: *mut StringView,
+) -> nsresult {
+    let attr = get_attribute((*attributes).as_slice(), SdpAttributeType::IceUfrag);
+    if let Some(&SdpAttribute::IceUfrag(ref string)) = attr {
+        *ret = StringView::from(string.as_str());
+        return NS_OK;
+    }
+    NS_ERROR_INVALID_ARG
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_icepwd(
+    attributes: *const Vec<SdpAttribute>,
+    ret: *mut StringView,
+) -> nsresult {
+    let attr = get_attribute((*attributes).as_slice(), SdpAttributeType::IcePwd);
+    if let Some(&SdpAttribute::IcePwd(ref string)) = attr {
+        *ret = StringView::from(string.as_str());
+        return NS_OK;
+    }
+    NS_ERROR_INVALID_ARG
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_identity(
+    attributes: *const Vec<SdpAttribute>,
+    ret: *mut StringView,
+) -> nsresult {
+    let attr = get_attribute((*attributes).as_slice(), SdpAttributeType::Identity);
+    if let Some(&SdpAttribute::Identity(ref string)) = attr {
+        *ret = StringView::from(string.as_str());
+        return NS_OK;
+    }
+    NS_ERROR_INVALID_ARG
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_iceoptions(
+    attributes: *const Vec<SdpAttribute>,
+    ret: *mut *const Vec<String>,
+) -> nsresult {
+    let attr = get_attribute((*attributes).as_slice(), SdpAttributeType::IceOptions);
+    if let Some(&SdpAttribute::IceOptions(ref options)) = attr {
+        *ret = options;
+        return NS_OK;
+    }
+    NS_ERROR_INVALID_ARG
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_maxptime(
+    attributes: *const Vec<SdpAttribute>,
+    ret: *mut u64,
+) -> nsresult {
+    let attr = get_attribute((*attributes).as_slice(), SdpAttributeType::MaxPtime);
+    if let Some(&SdpAttribute::MaxPtime(ref max_ptime)) = attr {
+        *ret = *max_ptime;
+        return NS_OK;
+    }
+    NS_ERROR_INVALID_ARG
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustSdpAttributeFingerprint {
+    hash_algorithm: u16,
+    fingerprint: *const Vec<u8>,
+}
+
+impl<'a> From<&'a SdpAttributeFingerprint> for RustSdpAttributeFingerprint {
+    fn from(other: &SdpAttributeFingerprint) -> Self {
+        RustSdpAttributeFingerprint {
+            hash_algorithm: other.hash_algorithm as u16,
+            fingerprint: &other.fingerprint,
+        }
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_fingerprint_count(attributes: *const Vec<SdpAttribute>) -> size_t {
+    count_attribute((*attributes).as_slice(), SdpAttributeType::Fingerprint)
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_fingerprints(
+    attributes: *const Vec<SdpAttribute>,
+    ret_size: size_t,
+    ret_fingerprints: *mut RustSdpAttributeFingerprint,
+) {
+    let attrs: Vec<_> = (*attributes)
+        .iter()
+        .filter_map(|x| {
+            if let SdpAttribute::Fingerprint(ref data) = *x {
+                Some(RustSdpAttributeFingerprint::from(data))
+            } else {
+                None
+            }
+        })
+        .collect();
+    let fingerprints = slice::from_raw_parts_mut(ret_fingerprints, ret_size);
+    fingerprints.copy_from_slice(attrs.as_slice());
+}
+
+#[repr(C)]
+#[derive(Clone)]
+pub enum RustSdpAttributeSetup {
+    Active,
+    Actpass,
+    Holdconn,
+    Passive,
+}
+
+impl<'a> From<&'a SdpAttributeSetup> for RustSdpAttributeSetup {
+    fn from(other: &SdpAttributeSetup) -> Self {
+        match *other {
+            SdpAttributeSetup::Active => RustSdpAttributeSetup::Active,
+            SdpAttributeSetup::Actpass => RustSdpAttributeSetup::Actpass,
+            SdpAttributeSetup::Holdconn => RustSdpAttributeSetup::Holdconn,
+            SdpAttributeSetup::Passive => RustSdpAttributeSetup::Passive,
+        }
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_setup(
+    attributes: *const Vec<SdpAttribute>,
+    ret: *mut RustSdpAttributeSetup,
+) -> nsresult {
+    let attr = get_attribute((*attributes).as_slice(), SdpAttributeType::Setup);
+    if let Some(&SdpAttribute::Setup(ref setup)) = attr {
+        *ret = RustSdpAttributeSetup::from(setup);
+        return NS_OK;
+    }
+    NS_ERROR_INVALID_ARG
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustSdpAttributeSsrc {
+    pub id: u32,
+    pub attribute: StringView,
+    pub value: StringView,
+}
+
+impl<'a> From<&'a SdpAttributeSsrc> for RustSdpAttributeSsrc {
+    fn from(other: &SdpAttributeSsrc) -> Self {
+        RustSdpAttributeSsrc {
+            id: other.id,
+            attribute: StringView::from(&other.attribute),
+            value: StringView::from(&other.value),
+        }
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_ssrc_count(attributes: *const Vec<SdpAttribute>) -> size_t {
+    count_attribute((*attributes).as_slice(), SdpAttributeType::Ssrc)
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_ssrcs(
+    attributes: *const Vec<SdpAttribute>,
+    ret_size: size_t,
+    ret_ssrcs: *mut RustSdpAttributeSsrc,
+) {
+    let attrs: Vec<_> = (*attributes)
+        .iter()
+        .filter_map(|x| {
+            if let SdpAttribute::Ssrc(ref data) = *x {
+                Some(RustSdpAttributeSsrc::from(data))
+            } else {
+                None
+            }
+        })
+        .collect();
+    let ssrcs = slice::from_raw_parts_mut(ret_ssrcs, ret_size);
+    ssrcs.copy_from_slice(attrs.as_slice());
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub enum RustSdpSsrcGroupSemantic {
+    Duplication,
+    FlowIdentification,
+    ForwardErrorCorrection,
+    ForwardErrorCorrectionFr,
+    SIM,
+}
+
+impl<'a> From<&'a SdpSsrcGroupSemantic> for RustSdpSsrcGroupSemantic {
+    fn from(other: &SdpSsrcGroupSemantic) -> Self {
+        match *other {
+            SdpSsrcGroupSemantic::Duplication => RustSdpSsrcGroupSemantic::Duplication,
+            SdpSsrcGroupSemantic::FlowIdentification => {
+                RustSdpSsrcGroupSemantic::FlowIdentification
+            }
+            SdpSsrcGroupSemantic::ForwardErrorCorrection => {
+                RustSdpSsrcGroupSemantic::ForwardErrorCorrection
+            }
+            SdpSsrcGroupSemantic::ForwardErrorCorrectionFr => {
+                RustSdpSsrcGroupSemantic::ForwardErrorCorrectionFr
+            }
+            SdpSsrcGroupSemantic::Sim => RustSdpSsrcGroupSemantic::SIM,
+        }
+    }
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustSdpSsrcGroup {
+    pub semantic: RustSdpSsrcGroupSemantic,
+    pub ssrcs: *const Vec<SdpAttributeSsrc>,
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_ssrc_group_count(attributes: *const Vec<SdpAttribute>) -> size_t {
+    count_attribute((*attributes).as_slice(), SdpAttributeType::SsrcGroup)
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_ssrc_groups(
+    attributes: *const Vec<SdpAttribute>,
+    ret_size: size_t,
+    ret_ssrc_groups: *mut RustSdpSsrcGroup,
+) {
+    let attrs: Vec<_> = (*attributes)
+        .iter()
+        .filter_map(|x| {
+            if let SdpAttribute::SsrcGroup(ref semantic, ref ssrcs) = *x {
+                Some(RustSdpSsrcGroup {
+                    semantic: RustSdpSsrcGroupSemantic::from(semantic),
+                    ssrcs: ssrcs,
+                })
+            } else {
+                None
+            }
+        })
+        .collect();
+    let ssrc_groups = slice::from_raw_parts_mut(ret_ssrc_groups, ret_size);
+    ssrc_groups.copy_from_slice(attrs.as_slice());
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustSdpAttributeRtpmap {
+    pub payload_type: u8,
+    pub codec_name: StringView,
+    pub frequency: u32,
+    pub channels: u32,
+}
+
+impl<'a> From<&'a SdpAttributeRtpmap> for RustSdpAttributeRtpmap {
+    fn from(other: &SdpAttributeRtpmap) -> Self {
+        RustSdpAttributeRtpmap {
+            payload_type: other.payload_type as u8,
+            codec_name: StringView::from(other.codec_name.as_str()),
+            frequency: other.frequency as u32,
+            channels: other.channels.unwrap_or(0),
+        }
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_rtpmap_count(attributes: *const Vec<SdpAttribute>) -> size_t {
+    count_attribute((*attributes).as_slice(), SdpAttributeType::Rtpmap)
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_rtpmaps(
+    attributes: *const Vec<SdpAttribute>,
+    ret_size: size_t,
+    ret_rtpmaps: *mut RustSdpAttributeRtpmap,
+) {
+    let attrs: Vec<_> = (*attributes)
+        .iter()
+        .filter_map(|x| {
+            if let SdpAttribute::Rtpmap(ref data) = *x {
+                Some(RustSdpAttributeRtpmap::from(data))
+            } else {
+                None
+            }
+        })
+        .collect();
+    let rtpmaps = slice::from_raw_parts_mut(ret_rtpmaps, ret_size);
+    rtpmaps.copy_from_slice(attrs.as_slice());
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustRtxFmtpParameters {
+    pub apt: u8,
+    pub has_rtx_time: bool,
+    pub rtx_time: u32,
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustSdpAttributeFmtpParameters {
+    // H264
+    pub packetization_mode: u32,
+    pub level_asymmetry_allowed: bool,
+    pub profile_level_id: u32,
+    pub max_fs: u32,
+    pub max_cpb: u32,
+    pub max_dpb: u32,
+    pub max_br: u32,
+    pub max_mbps: u32,
+
+    // VP8 and VP9
+    // max_fs, already defined in H264
+    pub max_fr: u32,
+
+    // Opus
+    pub maxplaybackrate: u32,
+    pub maxaveragebitrate: u32,
+    pub usedtx: bool,
+    pub stereo: bool,
+    pub useinbandfec: bool,
+    pub cbr: bool,
+    pub ptime: u32,
+    pub minptime: u32,
+    pub maxptime: u32,
+
+    // telephone-event
+    pub dtmf_tones: StringView,
+
+    // RTX
+    pub rtx: RustRtxFmtpParameters,
+
+    // Red
+    pub encodings: *const Vec<u8>,
+
+    // Unknown
+    pub unknown_tokens: *const Vec<String>,
+}
+
+impl<'a> From<&'a SdpAttributeFmtpParameters> for RustSdpAttributeFmtpParameters {
+    fn from(other: &SdpAttributeFmtpParameters) -> Self {
+        let rtx = if let Some(rtx) = other.rtx {
+            RustRtxFmtpParameters {
+                apt: rtx.apt,
+                has_rtx_time: rtx.rtx_time.is_some(),
+                rtx_time: rtx.rtx_time.unwrap_or(0),
+            }
+        } else {
+            RustRtxFmtpParameters {
+                apt: 0,
+                has_rtx_time: false,
+                rtx_time: 0,
+            }
+        };
+
+        RustSdpAttributeFmtpParameters {
+            packetization_mode: other.packetization_mode,
+            level_asymmetry_allowed: other.level_asymmetry_allowed,
+            profile_level_id: other.profile_level_id,
+            max_fs: other.max_fs,
+            max_cpb: other.max_cpb,
+            max_dpb: other.max_dpb,
+            max_br: other.max_br,
+            max_mbps: other.max_mbps,
+            usedtx: other.usedtx,
+            stereo: other.stereo,
+            useinbandfec: other.useinbandfec,
+            cbr: other.cbr,
+            max_fr: other.max_fr,
+            maxplaybackrate: other.maxplaybackrate,
+            maxaveragebitrate: other.maxaveragebitrate,
+            ptime: other.ptime,
+            minptime: other.minptime,
+            maxptime: other.maxptime,
+            dtmf_tones: StringView::from(other.dtmf_tones.as_str()),
+            rtx,
+            encodings: &other.encodings,
+            unknown_tokens: &other.unknown_tokens,
+        }
+    }
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustSdpAttributeFmtp {
+    pub payload_type: u8,
+    pub codec_name: StringView,
+    pub parameters: RustSdpAttributeFmtpParameters,
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_fmtp_count(attributes: *const Vec<SdpAttribute>) -> size_t {
+    count_attribute((*attributes).as_slice(), SdpAttributeType::Fmtp)
+}
+
+fn find_payload_type(attributes: &[SdpAttribute], payload_type: u8) -> Option<&SdpAttributeRtpmap> {
+    attributes
+        .iter()
+        .filter_map(|x| {
+            if let SdpAttribute::Rtpmap(ref data) = *x {
+                if data.payload_type == payload_type {
+                    Some(data)
+                } else {
+                    None
+                }
+            } else {
+                None
+            }
+        })
+        .next()
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_fmtp(
+    attributes: *const Vec<SdpAttribute>,
+    ret_size: size_t,
+    ret_fmtp: *mut RustSdpAttributeFmtp,
+) -> size_t {
+    let fmtps = (*attributes).iter().filter_map(|x| {
+        if let SdpAttribute::Fmtp(ref data) = *x {
+            Some(data)
+        } else {
+            None
+        }
+    });
+    let mut rust_fmtps = Vec::new();
+    for fmtp in fmtps {
+        if let Some(rtpmap) = find_payload_type((*attributes).as_slice(), fmtp.payload_type) {
+            rust_fmtps.push(RustSdpAttributeFmtp {
+                payload_type: fmtp.payload_type as u8,
+                codec_name: StringView::from(rtpmap.codec_name.as_str()),
+                parameters: RustSdpAttributeFmtpParameters::from(&fmtp.parameters),
+            });
+        }
+    }
+    let fmtps = if ret_size <= rust_fmtps.len() {
+        slice::from_raw_parts_mut(ret_fmtp, ret_size)
+    } else {
+        slice::from_raw_parts_mut(ret_fmtp, rust_fmtps.len())
+    };
+    fmtps.copy_from_slice(rust_fmtps.as_slice());
+    fmtps.len()
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_ptime(attributes: *const Vec<SdpAttribute>) -> i64 {
+    for attribute in (*attributes).iter() {
+        if let SdpAttribute::Ptime(time) = *attribute {
+            return time as i64;
+        }
+    }
+    -1
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_max_msg_size(attributes: *const Vec<SdpAttribute>) -> i64 {
+    for attribute in (*attributes).iter() {
+        if let SdpAttribute::MaxMessageSize(max_msg_size) = *attribute {
+            return max_msg_size as i64;
+        }
+    }
+    -1
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_sctp_port(attributes: *const Vec<SdpAttribute>) -> i64 {
+    for attribute in (*attributes).iter() {
+        if let SdpAttribute::SctpPort(port) = *attribute {
+            return port as i64;
+        }
+    }
+    -1
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustSdpAttributeFlags {
+    pub ice_lite: bool,
+    pub rtcp_mux: bool,
+    pub rtcp_rsize: bool,
+    pub bundle_only: bool,
+    pub end_of_candidates: bool,
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_attribute_flags(
+    attributes: *const Vec<SdpAttribute>,
+) -> RustSdpAttributeFlags {
+    let mut ret = RustSdpAttributeFlags {
+        ice_lite: false,
+        rtcp_mux: false,
+        rtcp_rsize: false,
+        bundle_only: false,
+        end_of_candidates: false,
+    };
+    for attribute in (*attributes).iter() {
+        if let SdpAttribute::IceLite = *attribute {
+            ret.ice_lite = true;
+        } else if let SdpAttribute::RtcpMux = *attribute {
+            ret.rtcp_mux = true;
+        } else if let SdpAttribute::RtcpRsize = *attribute {
+            ret.rtcp_rsize = true;
+        } else if let SdpAttribute::BundleOnly = *attribute {
+            ret.bundle_only = true;
+        } else if let SdpAttribute::EndOfCandidates = *attribute {
+            ret.end_of_candidates = true;
+        }
+    }
+    ret
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_mid(
+    attributes: *const Vec<SdpAttribute>,
+    ret: *mut StringView,
+) -> nsresult {
+    for attribute in (*attributes).iter() {
+        if let SdpAttribute::Mid(ref data) = *attribute {
+            *ret = StringView::from(data.as_str());
+            return NS_OK;
+        }
+    }
+    NS_ERROR_INVALID_ARG
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustSdpAttributeMsid {
+    id: StringView,
+    appdata: StringView,
+}
+
+impl<'a> From<&'a SdpAttributeMsid> for RustSdpAttributeMsid {
+    fn from(other: &SdpAttributeMsid) -> Self {
+        RustSdpAttributeMsid {
+            id: StringView::from(other.id.as_str()),
+            appdata: StringView::from(&other.appdata),
+        }
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_msid_count(attributes: *const Vec<SdpAttribute>) -> size_t {
+    count_attribute((*attributes).as_slice(), SdpAttributeType::Msid)
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_msids(
+    attributes: *const Vec<SdpAttribute>,
+    ret_size: size_t,
+    ret_msids: *mut RustSdpAttributeMsid,
+) {
+    let attrs: Vec<_> = (*attributes)
+        .iter()
+        .filter_map(|x| {
+            if let SdpAttribute::Msid(ref data) = *x {
+                Some(RustSdpAttributeMsid::from(data))
+            } else {
+                None
+            }
+        })
+        .collect();
+    let msids = slice::from_raw_parts_mut(ret_msids, ret_size);
+    msids.copy_from_slice(attrs.as_slice());
+}
+
+// TODO: Finish msid attributes once parsing is changed upstream.
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustSdpAttributeMsidSemantic {
+    pub semantic: StringView,
+    pub msids: *const Vec<String>,
+}
+
+impl<'a> From<&'a SdpAttributeMsidSemantic> for RustSdpAttributeMsidSemantic {
+    fn from(other: &SdpAttributeMsidSemantic) -> Self {
+        RustSdpAttributeMsidSemantic {
+            semantic: StringView::from(other.semantic.as_str()),
+            msids: &other.msids,
+        }
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_msid_semantic_count(
+    attributes: *const Vec<SdpAttribute>,
+) -> size_t {
+    count_attribute((*attributes).as_slice(), SdpAttributeType::MsidSemantic)
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_msid_semantics(
+    attributes: *const Vec<SdpAttribute>,
+    ret_size: size_t,
+    ret_msid_semantics: *mut RustSdpAttributeMsidSemantic,
+) {
+    let attrs: Vec<_> = (*attributes)
+        .iter()
+        .filter_map(|x| {
+            if let SdpAttribute::MsidSemantic(ref data) = *x {
+                Some(RustSdpAttributeMsidSemantic::from(data))
+            } else {
+                None
+            }
+        })
+        .collect();
+    let msid_semantics = slice::from_raw_parts_mut(ret_msid_semantics, ret_size);
+    msid_semantics.copy_from_slice(attrs.as_slice());
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub enum RustSdpAttributeGroupSemantic {
+    LipSynchronization,
+    FlowIdentification,
+    SingleReservationFlow,
+    AlternateNetworkAddressType,
+    ForwardErrorCorrection,
+    DecodingDependency,
+    Bundle,
+}
+
+impl<'a> From<&'a SdpAttributeGroupSemantic> for RustSdpAttributeGroupSemantic {
+    fn from(other: &SdpAttributeGroupSemantic) -> Self {
+        match *other {
+            SdpAttributeGroupSemantic::LipSynchronization => {
+                RustSdpAttributeGroupSemantic::LipSynchronization
+            }
+            SdpAttributeGroupSemantic::FlowIdentification => {
+                RustSdpAttributeGroupSemantic::FlowIdentification
+            }
+            SdpAttributeGroupSemantic::SingleReservationFlow => {
+                RustSdpAttributeGroupSemantic::SingleReservationFlow
+            }
+            SdpAttributeGroupSemantic::AlternateNetworkAddressType => {
+                RustSdpAttributeGroupSemantic::AlternateNetworkAddressType
+            }
+            SdpAttributeGroupSemantic::ForwardErrorCorrection => {
+                RustSdpAttributeGroupSemantic::ForwardErrorCorrection
+            }
+            SdpAttributeGroupSemantic::DecodingDependency => {
+                RustSdpAttributeGroupSemantic::DecodingDependency
+            }
+            SdpAttributeGroupSemantic::Bundle => RustSdpAttributeGroupSemantic::Bundle,
+        }
+    }
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustSdpAttributeGroup {
+    pub semantic: RustSdpAttributeGroupSemantic,
+    pub tags: *const Vec<String>,
+}
+
+impl<'a> From<&'a SdpAttributeGroup> for RustSdpAttributeGroup {
+    fn from(other: &SdpAttributeGroup) -> Self {
+        RustSdpAttributeGroup {
+            semantic: RustSdpAttributeGroupSemantic::from(&other.semantics),
+            tags: &other.tags,
+        }
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_group_count(attributes: *const Vec<SdpAttribute>) -> size_t {
+    count_attribute((*attributes).as_slice(), SdpAttributeType::Group)
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_groups(
+    attributes: *const Vec<SdpAttribute>,
+    ret_size: size_t,
+    ret_groups: *mut RustSdpAttributeGroup,
+) {
+    let attrs: Vec<_> = (*attributes)
+        .iter()
+        .filter_map(|x| {
+            if let SdpAttribute::Group(ref data) = *x {
+                Some(RustSdpAttributeGroup::from(data))
+            } else {
+                None
+            }
+        })
+        .collect();
+    let groups = slice::from_raw_parts_mut(ret_groups, ret_size);
+    groups.copy_from_slice(attrs.as_slice());
+}
+
+#[repr(C)]
+pub struct RustSdpAttributeRtcp {
+    pub port: u32,
+    pub unicast_addr: RustExplicitlyTypedAddress,
+    pub has_address: bool,
+}
+
+impl<'a> From<&'a SdpAttributeRtcp> for RustSdpAttributeRtcp {
+    fn from(other: &SdpAttributeRtcp) -> Self {
+        match other.unicast_addr {
+            Some(ref address) => RustSdpAttributeRtcp {
+                port: other.port as u32,
+                unicast_addr: address.into(),
+                has_address: true,
+            },
+            None => RustSdpAttributeRtcp {
+                port: other.port as u32,
+                unicast_addr: RustExplicitlyTypedAddress::default(),
+                has_address: false,
+            },
+        }
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_rtcp(
+    attributes: *const Vec<SdpAttribute>,
+    ret: *mut RustSdpAttributeRtcp,
+) -> nsresult {
+    let attr = get_attribute((*attributes).as_slice(), SdpAttributeType::Rtcp);
+    if let Some(&SdpAttribute::Rtcp(ref data)) = attr {
+        *ret = RustSdpAttributeRtcp::from(data);
+        return NS_OK;
+    }
+    NS_ERROR_INVALID_ARG
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustSdpAttributeRtcpFb {
+    pub payload_type: u32,
+    pub feedback_type: u32,
+    pub parameter: StringView,
+    pub extra: StringView,
+}
+
+impl<'a> From<&'a SdpAttributeRtcpFb> for RustSdpAttributeRtcpFb {
+    fn from(other: &SdpAttributeRtcpFb) -> Self {
+        RustSdpAttributeRtcpFb {
+            payload_type: match other.payload_type {
+                SdpAttributePayloadType::Wildcard => u32::max_value(),
+                SdpAttributePayloadType::PayloadType(x) => x as u32,
+            },
+            feedback_type: other.feedback_type.clone() as u32,
+            parameter: StringView::from(other.parameter.as_str()),
+            extra: StringView::from(other.extra.as_str()),
+        }
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_rtcpfb_count(attributes: *const Vec<SdpAttribute>) -> size_t {
+    count_attribute((*attributes).as_slice(), SdpAttributeType::Rtcpfb)
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_rtcpfbs(
+    attributes: *const Vec<SdpAttribute>,
+    ret_size: size_t,
+    ret_rtcpfbs: *mut RustSdpAttributeRtcpFb,
+) {
+    let attrs: Vec<_> = (*attributes)
+        .iter()
+        .filter_map(|x| {
+            if let SdpAttribute::Rtcpfb(ref data) = *x {
+                Some(RustSdpAttributeRtcpFb::from(data))
+            } else {
+                None
+            }
+        })
+        .collect();
+    let rtcpfbs = slice::from_raw_parts_mut(ret_rtcpfbs, ret_size);
+    rtcpfbs.clone_from_slice(attrs.as_slice());
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustSdpAttributeImageAttrXyRange {
+    // range
+    pub min: u32,
+    pub max: u32,
+    pub step: u32,
+
+    // discrete values
+    pub discrete_values: *const Vec<u32>,
+}
+
+impl<'a> From<&'a SdpAttributeImageAttrXyRange> for RustSdpAttributeImageAttrXyRange {
+    fn from(other: &SdpAttributeImageAttrXyRange) -> Self {
+        match other {
+            &SdpAttributeImageAttrXyRange::Range(min, max, step) => {
+                RustSdpAttributeImageAttrXyRange {
+                    min,
+                    max,
+                    step: step.unwrap_or(1),
+                    discrete_values: ptr::null(),
+                }
+            }
+            &SdpAttributeImageAttrXyRange::DiscreteValues(ref discrete_values) => {
+                RustSdpAttributeImageAttrXyRange {
+                    min: 0,
+                    max: 1,
+                    step: 1,
+                    discrete_values,
+                }
+            }
+        }
+    }
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustSdpAttributeImageAttrSRange {
+    // range
+    pub min: c_float,
+    pub max: c_float,
+
+    // discrete values
+    pub discrete_values: *const Vec<c_float>,
+}
+
+impl<'a> From<&'a SdpAttributeImageAttrSRange> for RustSdpAttributeImageAttrSRange {
+    fn from(other: &SdpAttributeImageAttrSRange) -> Self {
+        match other {
+            &SdpAttributeImageAttrSRange::Range(min, max) => RustSdpAttributeImageAttrSRange {
+                min,
+                max,
+                discrete_values: ptr::null(),
+            },
+            &SdpAttributeImageAttrSRange::DiscreteValues(ref discrete_values) => {
+                RustSdpAttributeImageAttrSRange {
+                    min: 0.0,
+                    max: 1.0,
+                    discrete_values,
+                }
+            }
+        }
+    }
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustSdpAttributeImageAttrPRange {
+    pub min: c_float,
+    pub max: c_float,
+}
+
+impl<'a> From<&'a SdpAttributeImageAttrPRange> for RustSdpAttributeImageAttrPRange {
+    fn from(other: &SdpAttributeImageAttrPRange) -> Self {
+        RustSdpAttributeImageAttrPRange {
+            min: other.min,
+            max: other.max,
+        }
+    }
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustSdpAttributeImageAttrSet {
+    pub x: RustSdpAttributeImageAttrXyRange,
+    pub y: RustSdpAttributeImageAttrXyRange,
+
+    pub has_sar: bool,
+    pub sar: RustSdpAttributeImageAttrSRange,
+
+    pub has_par: bool,
+    pub par: RustSdpAttributeImageAttrPRange,
+
+    pub q: c_float,
+}
+
+impl<'a> From<&'a SdpAttributeImageAttrSet> for RustSdpAttributeImageAttrSet {
+    fn from(other: &SdpAttributeImageAttrSet) -> Self {
+        RustSdpAttributeImageAttrSet {
+            x: RustSdpAttributeImageAttrXyRange::from(&other.x),
+            y: RustSdpAttributeImageAttrXyRange::from(&other.y),
+
+            has_sar: other.sar.is_some(),
+            sar: match other.sar {
+                Some(ref x) => RustSdpAttributeImageAttrSRange::from(x),
+                // This is just any valid value accepted by rust,
+                // it might as well by uninitilized
+                None => RustSdpAttributeImageAttrSRange::from(
+                    &SdpAttributeImageAttrSRange::DiscreteValues(vec![]),
+                ),
+            },
+
+            has_par: other.par.is_some(),
+            par: match other.par {
+                Some(ref x) => RustSdpAttributeImageAttrPRange::from(x),
+                // This is just any valid value accepted by rust,
+                // it might as well by uninitilized
+                None => RustSdpAttributeImageAttrPRange { min: 0.0, max: 1.0 },
+            },
+
+            q: other.q.unwrap_or(0.5),
+        }
+    }
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustSdpAttributeImageAttrSetList {
+    pub sets: *const Vec<SdpAttributeImageAttrSet>,
+}
+
+impl<'a> From<&'a SdpAttributeImageAttrSetList> for RustSdpAttributeImageAttrSetList {
+    fn from(other: &SdpAttributeImageAttrSetList) -> Self {
+        match other {
+            &SdpAttributeImageAttrSetList::Wildcard => {
+                RustSdpAttributeImageAttrSetList { sets: ptr::null() }
+            }
+            &SdpAttributeImageAttrSetList::Sets(ref sets) => {
+                RustSdpAttributeImageAttrSetList { sets: sets }
+            }
+        }
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_imageattr_get_set_count(
+    sets: *const Vec<SdpAttributeImageAttrSet>,
+) -> size_t {
+    (*sets).len()
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_imageattr_get_sets(
+    sets: *const Vec<SdpAttributeImageAttrSet>,
+    ret_size: size_t,
+    ret: *mut RustSdpAttributeImageAttrSet,
+) {
+    let rust_sets: Vec<_> = (*sets)
+        .iter()
+        .map(RustSdpAttributeImageAttrSet::from)
+        .collect();
+    let sets = slice::from_raw_parts_mut(ret, ret_size);
+    sets.clone_from_slice(rust_sets.as_slice());
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustSdpAttributeImageAttr {
+    pub pt: u32,
+    pub send: RustSdpAttributeImageAttrSetList,
+    pub recv: RustSdpAttributeImageAttrSetList,
+}
+
+impl<'a> From<&'a SdpAttributeImageAttr> for RustSdpAttributeImageAttr {
+    fn from(other: &SdpAttributeImageAttr) -> Self {
+        RustSdpAttributeImageAttr {
+            pt: match other.pt {
+                SdpAttributePayloadType::Wildcard => u32::max_value(),
+                SdpAttributePayloadType::PayloadType(x) => x as u32,
+            },
+            send: RustSdpAttributeImageAttrSetList::from(&other.send),
+            recv: RustSdpAttributeImageAttrSetList::from(&other.recv),
+        }
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_imageattr_count(attributes: *const Vec<SdpAttribute>) -> size_t {
+    count_attribute((*attributes).as_slice(), SdpAttributeType::ImageAttr)
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_imageattrs(
+    attributes: *const Vec<SdpAttribute>,
+    ret_size: size_t,
+    ret_attrs: *mut RustSdpAttributeImageAttr,
+) {
+    let attrs: Vec<_> = (*attributes)
+        .iter()
+        .filter_map(|x| {
+            if let SdpAttribute::ImageAttr(ref data) = *x {
+                Some(RustSdpAttributeImageAttr::from(data))
+            } else {
+                None
+            }
+        })
+        .collect();
+    let imageattrs = slice::from_raw_parts_mut(ret_attrs, ret_size);
+    imageattrs.copy_from_slice(attrs.as_slice());
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustSdpAttributeSctpmap {
+    pub port: u32,
+    pub channels: u32,
+}
+
+impl<'a> From<&'a SdpAttributeSctpmap> for RustSdpAttributeSctpmap {
+    fn from(other: &SdpAttributeSctpmap) -> Self {
+        RustSdpAttributeSctpmap {
+            port: other.port as u32,
+            channels: other.channels,
+        }
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_sctpmap_count(attributes: *const Vec<SdpAttribute>) -> size_t {
+    count_attribute((*attributes).as_slice(), SdpAttributeType::Sctpmap)
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_sctpmaps(
+    attributes: *const Vec<SdpAttribute>,
+    ret_size: size_t,
+    ret_sctpmaps: *mut RustSdpAttributeSctpmap,
+) {
+    let attrs: Vec<_> = (*attributes)
+        .iter()
+        .filter_map(|x| {
+            if let SdpAttribute::Sctpmap(ref data) = *x {
+                Some(RustSdpAttributeSctpmap::from(data))
+            } else {
+                None
+            }
+        })
+        .collect();
+    let sctpmaps = slice::from_raw_parts_mut(ret_sctpmaps, ret_size);
+    sctpmaps.copy_from_slice(attrs.as_slice());
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustSdpAttributeSimulcastId {
+    pub id: StringView,
+    pub paused: bool,
+}
+
+impl<'a> From<&'a SdpAttributeSimulcastId> for RustSdpAttributeSimulcastId {
+    fn from(other: &SdpAttributeSimulcastId) -> Self {
+        RustSdpAttributeSimulcastId {
+            id: StringView::from(other.id.as_str()),
+            paused: other.paused,
+        }
+    }
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustSdpAttributeSimulcastVersion {
+    pub ids: *const Vec<SdpAttributeSimulcastId>,
+}
+
+impl<'a> From<&'a SdpAttributeSimulcastVersion> for RustSdpAttributeSimulcastVersion {
+    fn from(other: &SdpAttributeSimulcastVersion) -> Self {
+        RustSdpAttributeSimulcastVersion { ids: &other.ids }
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_simulcast_get_ids_count(
+    ids: *const Vec<SdpAttributeSimulcastId>,
+) -> size_t {
+    (*ids).len()
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_simulcast_get_ids(
+    ids: *const Vec<SdpAttributeSimulcastId>,
+    ret_size: size_t,
+    ret: *mut RustSdpAttributeSimulcastId,
+) {
+    let rust_ids: Vec<_> = (*ids)
+        .iter()
+        .map(RustSdpAttributeSimulcastId::from)
+        .collect();
+    let ids = slice::from_raw_parts_mut(ret, ret_size);
+    ids.clone_from_slice(rust_ids.as_slice());
+}
+
+#[repr(C)]
+pub struct RustSdpAttributeSimulcast {
+    pub send: *const Vec<SdpAttributeSimulcastVersion>,
+    pub receive: *const Vec<SdpAttributeSimulcastVersion>,
+}
+
+impl<'a> From<&'a SdpAttributeSimulcast> for RustSdpAttributeSimulcast {
+    fn from(other: &SdpAttributeSimulcast) -> Self {
+        RustSdpAttributeSimulcast {
+            send: &other.send,
+            receive: &other.receive,
+        }
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_simulcast_get_version_count(
+    version_list: *const Vec<SdpAttributeSimulcastVersion>,
+) -> size_t {
+    (*version_list).len()
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_simulcast_get_versions(
+    version_list: *const Vec<SdpAttributeSimulcastVersion>,
+    ret_size: size_t,
+    ret: *mut RustSdpAttributeSimulcastVersion,
+) {
+    let rust_versions_list: Vec<_> = (*version_list)
+        .iter()
+        .map(RustSdpAttributeSimulcastVersion::from)
+        .collect();
+    let versions = slice::from_raw_parts_mut(ret, ret_size);
+    versions.clone_from_slice(rust_versions_list.as_slice())
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_simulcast(
+    attributes: *const Vec<SdpAttribute>,
+    ret: *mut RustSdpAttributeSimulcast,
+) -> nsresult {
+    let attr = get_attribute((*attributes).as_slice(), SdpAttributeType::Simulcast);
+    if let Some(&SdpAttribute::Simulcast(ref data)) = attr {
+        *ret = RustSdpAttributeSimulcast::from(data);
+        return NS_OK;
+    }
+    NS_ERROR_INVALID_ARG
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub enum RustDirection {
+    Recvonly,
+    Sendonly,
+    Sendrecv,
+    Inactive,
+}
+
+impl<'a> From<&'a Option<SdpAttributeDirection>> for RustDirection {
+    fn from(other: &Option<SdpAttributeDirection>) -> Self {
+        match *other {
+            Some(ref direction) => match *direction {
+                SdpAttributeDirection::Recvonly => RustDirection::Recvonly,
+                SdpAttributeDirection::Sendonly => RustDirection::Sendonly,
+                SdpAttributeDirection::Sendrecv => RustDirection::Sendrecv,
+            },
+            None => RustDirection::Inactive,
+        }
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_direction(attributes: *const Vec<SdpAttribute>) -> RustDirection {
+    for attribute in (*attributes).iter() {
+        match *attribute {
+            SdpAttribute::Recvonly => {
+                return RustDirection::Recvonly;
+            }
+            SdpAttribute::Sendonly => {
+                return RustDirection::Sendonly;
+            }
+            SdpAttribute::Sendrecv => {
+                return RustDirection::Sendrecv;
+            }
+            SdpAttribute::Inactive => {
+                return RustDirection::Inactive;
+            }
+            _ => (),
+        }
+    }
+    RustDirection::Sendrecv
+}
+
+#[repr(C)]
+pub struct RustSdpAttributeRemoteCandidate {
+    pub component: u32,
+    pub address: RustAddress,
+    pub port: u32,
+}
+
+impl<'a> From<&'a SdpAttributeRemoteCandidate> for RustSdpAttributeRemoteCandidate {
+    fn from(other: &SdpAttributeRemoteCandidate) -> Self {
+        RustSdpAttributeRemoteCandidate {
+            component: other.component,
+            address: RustAddress::from(&other.address),
+            port: other.port,
+        }
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_remote_candidate_count(
+    attributes: *const Vec<SdpAttribute>,
+) -> size_t {
+    count_attribute((*attributes).as_slice(), SdpAttributeType::RemoteCandidate)
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_remote_candidates(
+    attributes: *const Vec<SdpAttribute>,
+    ret_size: size_t,
+    ret_candidates: *mut RustSdpAttributeRemoteCandidate,
+) {
+    let attrs = (*attributes).iter().filter_map(|x| {
+        if let SdpAttribute::RemoteCandidate(ref data) = *x {
+            Some(RustSdpAttributeRemoteCandidate::from(data))
+        } else {
+            None
+        }
+    });
+    let candidates = slice::from_raw_parts_mut(ret_candidates, ret_size);
+    for (source, destination) in attrs.zip(candidates) {
+        *destination = source
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_candidate_count(attributes: *const Vec<SdpAttribute>) -> size_t {
+    count_attribute((*attributes).as_slice(), SdpAttributeType::Candidate)
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_candidates(
+    attributes: *const Vec<SdpAttribute>,
+    _ret_size: size_t,
+    ret: *mut *const Vec<String>,
+) {
+    let attr_strings: Vec<String> = (*attributes)
+        .iter()
+        .filter_map(|x| {
+            if let SdpAttribute::Candidate(ref attr) = *x {
+                // The serialized attribute starts with "candidate:...", this needs to be removed
+                Some(attr.to_string())
+            } else {
+                None
+            }
+        })
+        .collect();
+
+    *ret = Box::into_raw(Box::from(attr_strings));
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustSdpAttributeRidParameters {
+    pub max_width: u32,
+    pub max_height: u32,
+    pub max_fps: u32,
+    pub max_fs: u32,
+    pub max_br: u32,
+    pub max_pps: u32,
+    pub unknown: *const Vec<String>,
+}
+
+impl<'a> From<&'a SdpAttributeRidParameters> for RustSdpAttributeRidParameters {
+    fn from(other: &SdpAttributeRidParameters) -> Self {
+        RustSdpAttributeRidParameters {
+            max_width: other.max_width,
+            max_height: other.max_height,
+            max_fps: other.max_fps,
+            max_fs: other.max_fs,
+            max_br: other.max_br,
+            max_pps: other.max_pps,
+
+            unknown: &other.unknown,
+        }
+    }
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustSdpAttributeRid {
+    pub id: StringView,
+    pub direction: u32,
+    pub formats: *const Vec<u16>,
+    pub params: RustSdpAttributeRidParameters,
+    pub depends: *const Vec<String>,
+}
+
+impl<'a> From<&'a SdpAttributeRid> for RustSdpAttributeRid {
+    fn from(other: &SdpAttributeRid) -> Self {
+        RustSdpAttributeRid {
+            id: StringView::from(other.id.as_str()),
+            direction: other.direction.clone() as u32,
+            formats: &other.formats,
+            params: RustSdpAttributeRidParameters::from(&other.params),
+            depends: &other.depends,
+        }
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_rid_count(attributes: *const Vec<SdpAttribute>) -> size_t {
+    count_attribute((*attributes).as_slice(), SdpAttributeType::Rid)
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_rids(
+    attributes: *const Vec<SdpAttribute>,
+    ret_size: size_t,
+    ret_rids: *mut RustSdpAttributeRid,
+) {
+    let attrs: Vec<_> = (*attributes)
+        .iter()
+        .filter_map(|x| {
+            if let SdpAttribute::Rid(ref data) = *x {
+                Some(RustSdpAttributeRid::from(data))
+            } else {
+                None
+            }
+        })
+        .collect();
+    let rids = slice::from_raw_parts_mut(ret_rids, ret_size);
+    rids.clone_from_slice(attrs.as_slice());
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct RustSdpAttributeExtmap {
+    pub id: u16,
+    pub direction_specified: bool,
+    pub direction: RustDirection,
+    pub url: StringView,
+    pub extension_attributes: StringView,
+}
+
+impl<'a> From<&'a SdpAttributeExtmap> for RustSdpAttributeExtmap {
+    fn from(other: &SdpAttributeExtmap) -> Self {
+        let dir = if other.direction.is_some() {
+            RustDirection::from(&other.direction)
+        } else {
+            RustDirection::from(&Some(SdpAttributeDirection::Sendrecv))
+        };
+        RustSdpAttributeExtmap {
+            id: other.id as u16,
+            direction_specified: other.direction.is_some(),
+            direction: dir,
+            url: StringView::from(other.url.as_str()),
+            extension_attributes: StringView::from(&other.extension_attributes),
+        }
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_extmap_count(attributes: *const Vec<SdpAttribute>) -> size_t {
+    count_attribute((*attributes).as_slice(), SdpAttributeType::Extmap)
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_extmaps(
+    attributes: *const Vec<SdpAttribute>,
+    ret_size: size_t,
+    ret_rids: *mut RustSdpAttributeExtmap,
+) {
+    let attrs: Vec<_> = (*attributes)
+        .iter()
+        .filter_map(|x| {
+            if let SdpAttribute::Extmap(ref data) = *x {
+                Some(RustSdpAttributeExtmap::from(data))
+            } else {
+                None
+            }
+        })
+        .collect();
+    let extmaps = slice::from_raw_parts_mut(ret_rids, ret_size);
+    extmaps.copy_from_slice(attrs.as_slice());
+}
diff --git a/dom/media/webrtc/sdp/rsdparsa_capi/src/lib.rs b/dom/media/webrtc/sdp/rsdparsa_capi/src/lib.rs
new file mode 100644
index 0000000000..20a13900a2
--- /dev/null
+++ b/dom/media/webrtc/sdp/rsdparsa_capi/src/lib.rs
@@ -0,0 +1,298 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+extern crate libc;
+extern crate nserror;
+extern crate rsdparsa;
+
+use std::ffi::CString;
+use std::os::raw::c_char;
+use std::ptr;
+
+use libc::size_t;
+
+use std::rc::Rc;
+
+use std::convert::{TryFrom, TryInto};
+
+use nserror::{nsresult, NS_ERROR_INVALID_ARG, NS_OK};
+use rsdparsa::address::ExplicitlyTypedAddress;
+use rsdparsa::anonymizer::{AnonymizingClone, StatefulSdpAnonymizer};
+use rsdparsa::attribute_type::SdpAttribute;
+use rsdparsa::error::SdpParserError;
+use rsdparsa::media_type::{SdpMediaValue, SdpProtocolValue};
+use rsdparsa::{SdpBandwidth, SdpSession, SdpTiming};
+
+#[macro_use]
+extern crate log;
+
+pub mod attribute;
+pub mod media_section;
+pub mod network;
+pub mod types;
+
+use network::{
+    get_bandwidth, origin_view_helper, RustAddressType, RustSdpConnection, RustSdpOrigin,
+};
+pub use types::{StringView, NULL_STRING};
+
+#[no_mangle]
+pub unsafe extern "C" fn parse_sdp(
+    sdp: StringView,
+    fail_on_warning: bool,
+    session: *mut *const SdpSession,
+    parser_error: *mut *const SdpParserError,
+) -> nsresult {
+    let sdp_str: String = match sdp.try_into() {
+        Ok(string) => string,
+        Err(boxed_error) => {
+            *session = ptr::null();
+            *parser_error = Box::into_raw(Box::new(SdpParserError::Sequence {
+                message: format!("{}", boxed_error),
+                line_number: 0,
+            }));
+            return NS_ERROR_INVALID_ARG;
+        }
+    };
+
+    let parser_result = rsdparsa::parse_sdp(&sdp_str, fail_on_warning);
+    match parser_result {
+        Ok(mut parsed) => {
+            *parser_error = match parsed.warnings.len() {
+                0 => ptr::null(),
+                _ => Box::into_raw(Box::new(parsed.warnings.remove(0))),
+            };
+            *session = Rc::into_raw(Rc::new(parsed));
+            NS_OK
+        }
+        Err(e) => {
+            *session = ptr::null();
+            error!("Error parsing SDP in rust: {}", e);
+            *parser_error = Box::into_raw(Box::new(e));
+            NS_ERROR_INVALID_ARG
+        }
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn create_anonymized_sdp_clone(
+    session: *const SdpSession,
+) -> *const SdpSession {
+    Rc::into_raw(Rc::new(
+        (*session).masked_clone(&mut StatefulSdpAnonymizer::new()),
+    ))
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn create_sdp_clone(session: *const SdpSession) -> *const SdpSession {
+    Rc::into_raw(Rc::new((*session).clone()))
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_free_session(sdp_ptr: *mut SdpSession) {
+    let sdp = Rc::from_raw(sdp_ptr);
+    drop(sdp);
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_new_reference(session: *mut SdpSession) -> *const SdpSession {
+    let original = Rc::from_raw(session);
+    let ret = Rc::into_raw(Rc::clone(&original));
+    Rc::into_raw(original); // So the original reference doesn't get dropped
+    ret
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_error_line_num(parser_error: *mut SdpParserError) -> size_t {
+    match *parser_error {
+        SdpParserError::Line { line_number, .. }
+        | SdpParserError::Unsupported { line_number, .. }
+        | SdpParserError::Sequence { line_number, .. } => line_number,
+    }
+}
+
+#[no_mangle]
+// Callee must check that a nullptr is not returned
+pub unsafe extern "C" fn sdp_get_error_message(parser_error: *mut SdpParserError) -> *mut c_char {
+    let message = format!("{}", *parser_error);
+    return match CString::new(message.as_str()) {
+        Ok(c_char_ptr) => c_char_ptr.into_raw(),
+        Err(_) => 0 as *mut c_char,
+    };
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_free_error_message(message: *mut c_char) {
+    if message != 0 as *mut c_char {
+        let _tmp = CString::from_raw(message);
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_free_error(parser_error: *mut SdpParserError) {
+    let e = Box::from_raw(parser_error);
+    drop(e);
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn get_version(session: *const SdpSession) -> u64 {
+    (*session).get_version()
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_origin(session: *const SdpSession) -> RustSdpOrigin {
+    origin_view_helper((*session).get_origin())
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn session_view(session: *const SdpSession) -> StringView {
+    StringView::from((*session).get_session())
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_session_has_connection(session: *const SdpSession) -> bool {
+    (*session).connection.is_some()
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_session_connection(
+    session: *const SdpSession,
+    connection: *mut RustSdpConnection,
+) -> nsresult {
+    match (*session).connection {
+        Some(ref c) => {
+            *connection = RustSdpConnection::from(c);
+            NS_OK
+        }
+        None => NS_ERROR_INVALID_ARG,
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_add_media_section(
+    session: *mut SdpSession,
+    media_type: u32,
+    direction: u32,
+    port: u16,
+    protocol: u32,
+    addr_type: u32,
+    address: StringView,
+) -> nsresult {
+    let addr_type = match RustAddressType::try_from(addr_type) {
+        Ok(a) => a.into(),
+        Err(e) => {
+            return e;
+        }
+    };
+    let address_string: String = match address.try_into() {
+        Ok(x) => x,
+        Err(boxed_error) => {
+            error!("Error while parsing string, description: {}", boxed_error);
+            return NS_ERROR_INVALID_ARG;
+        }
+    };
+    let address = match ExplicitlyTypedAddress::try_from((addr_type, address_string.as_str())) {
+        Ok(a) => a,
+        Err(_) => {
+            return NS_ERROR_INVALID_ARG;
+        }
+    };
+
+    let media_type = match media_type {
+        0 => SdpMediaValue::Audio,       // MediaType::kAudio
+        1 => SdpMediaValue::Video,       // MediaType::kVideo
+        3 => SdpMediaValue::Application, // MediaType::kApplication
+        _ => {
+            return NS_ERROR_INVALID_ARG;
+        }
+    };
+    let protocol = match protocol {
+        20 => SdpProtocolValue::RtpSavpf,        // Protocol::kRtpSavpf
+        21 => SdpProtocolValue::UdpTlsRtpSavp,   // Protocol::kUdpTlsRtpSavp
+        22 => SdpProtocolValue::TcpDtlsRtpSavp,  // Protocol::kTcpDtlsRtpSavp
+        24 => SdpProtocolValue::UdpTlsRtpSavpf,  // Protocol::kUdpTlsRtpSavpf
+        25 => SdpProtocolValue::TcpDtlsRtpSavpf, // Protocol::kTcpTlsRtpSavpf
+        37 => SdpProtocolValue::DtlsSctp,        // Protocol::kDtlsSctp
+        38 => SdpProtocolValue::UdpDtlsSctp,     // Protocol::kUdpDtlsSctp
+        39 => SdpProtocolValue::TcpDtlsSctp,     // Protocol::kTcpDtlsSctp
+        _ => {
+            return NS_ERROR_INVALID_ARG;
+        }
+    };
+    let direction = match direction {
+        1 => SdpAttribute::Sendonly,
+        2 => SdpAttribute::Recvonly,
+        3 => SdpAttribute::Sendrecv,
+        _ => {
+            return NS_ERROR_INVALID_ARG;
+        }
+    };
+
+    match (*session).add_media(media_type, direction, port as u32, protocol, address) {
+        Ok(_) => NS_OK,
+        Err(_) => NS_ERROR_INVALID_ARG,
+    }
+}
+
+#[repr(C)]
+#[derive(Clone)]
+pub struct RustSdpTiming {
+    pub start: u64,
+    pub stop: u64,
+}
+
+impl<'a> From<&'a SdpTiming> for RustSdpTiming {
+    fn from(timing: &SdpTiming) -> Self {
+        RustSdpTiming {
+            start: timing.start,
+            stop: timing.stop,
+        }
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_session_has_timing(session: *const SdpSession) -> bool {
+    (*session).timing.is_some()
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_session_timing(
+    session: *const SdpSession,
+    timing: *mut RustSdpTiming,
+) -> nsresult {
+    match (*session).timing {
+        Some(ref t) => {
+            *timing = RustSdpTiming::from(t);
+            NS_OK
+        }
+        None => NS_ERROR_INVALID_ARG,
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_media_section_count(session: *const SdpSession) -> size_t {
+    (*session).media.len()
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn get_sdp_bandwidth(
+    session: *const SdpSession,
+    bandwidth_type: *const c_char,
+) -> u32 {
+    get_bandwidth(&(*session).bandwidth, bandwidth_type)
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_session_bandwidth_vec(
+    session: *const SdpSession,
+) -> *const Vec<SdpBandwidth> {
+    &(*session).bandwidth
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn get_sdp_session_attributes(
+    session: *const SdpSession,
+) -> *const Vec<SdpAttribute> {
+    &(*session).attribute
+}
diff --git a/dom/media/webrtc/sdp/rsdparsa_capi/src/media_section.rs b/dom/media/webrtc/sdp/rsdparsa_capi/src/media_section.rs
new file mode 100644
index 0000000000..8429ab2815
--- /dev/null
+++ b/dom/media/webrtc/sdp/rsdparsa_capi/src/media_section.rs
@@ -0,0 +1,233 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+use std::convert::TryInto;
+use std::os::raw::c_char;
+use std::ptr;
+
+use libc::size_t;
+
+use nserror::{nsresult, NS_ERROR_INVALID_ARG, NS_OK};
+use rsdparsa::attribute_type::{SdpAttribute, SdpAttributeRtpmap};
+use rsdparsa::media_type::{SdpFormatList, SdpMedia, SdpMediaValue, SdpProtocolValue};
+use rsdparsa::{SdpBandwidth, SdpSession};
+
+use network::{get_bandwidth, RustSdpConnection};
+use types::StringView;
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_media_section(
+    session: *const SdpSession,
+    index: size_t,
+) -> *const SdpMedia {
+    return match (*session).media.get(index) {
+        Some(m) => m,
+        None => ptr::null(),
+    };
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub enum RustSdpMediaValue {
+    Audio,
+    Video,
+    Application,
+}
+
+impl<'a> From<&'a SdpMediaValue> for RustSdpMediaValue {
+    fn from(val: &SdpMediaValue) -> Self {
+        match *val {
+            SdpMediaValue::Audio => RustSdpMediaValue::Audio,
+            SdpMediaValue::Video => RustSdpMediaValue::Video,
+            SdpMediaValue::Application => RustSdpMediaValue::Application,
+        }
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_rust_get_media_type(sdp_media: *const SdpMedia) -> RustSdpMediaValue {
+    RustSdpMediaValue::from((*sdp_media).get_type())
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub enum RustSdpProtocolValue {
+    RtpSavpf,
+    UdpTlsRtpSavp,
+    TcpDtlsRtpSavp,
+    UdpTlsRtpSavpf,
+    TcpDtlsRtpSavpf,
+    DtlsSctp,
+    UdpDtlsSctp,
+    TcpDtlsSctp,
+    RtpAvp,
+    RtpAvpf,
+    RtpSavp,
+}
+
+impl<'a> From<&'a SdpProtocolValue> for RustSdpProtocolValue {
+    fn from(val: &SdpProtocolValue) -> Self {
+        match *val {
+            SdpProtocolValue::RtpSavpf => RustSdpProtocolValue::RtpSavpf,
+            SdpProtocolValue::UdpTlsRtpSavp => RustSdpProtocolValue::UdpTlsRtpSavp,
+            SdpProtocolValue::TcpDtlsRtpSavp => RustSdpProtocolValue::TcpDtlsRtpSavp,
+            SdpProtocolValue::UdpTlsRtpSavpf => RustSdpProtocolValue::UdpTlsRtpSavpf,
+            SdpProtocolValue::TcpDtlsRtpSavpf => RustSdpProtocolValue::TcpDtlsRtpSavpf,
+            SdpProtocolValue::DtlsSctp => RustSdpProtocolValue::DtlsSctp,
+            SdpProtocolValue::UdpDtlsSctp => RustSdpProtocolValue::UdpDtlsSctp,
+            SdpProtocolValue::TcpDtlsSctp => RustSdpProtocolValue::TcpDtlsSctp,
+            SdpProtocolValue::RtpAvp => RustSdpProtocolValue::RtpAvp,
+            SdpProtocolValue::RtpAvpf => RustSdpProtocolValue::RtpAvpf,
+            SdpProtocolValue::RtpSavp => RustSdpProtocolValue::RtpSavp,
+        }
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_media_protocol(
+    sdp_media: *const SdpMedia,
+) -> RustSdpProtocolValue {
+    RustSdpProtocolValue::from((*sdp_media).get_proto())
+}
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub enum RustSdpFormatType {
+    Integers,
+    Strings,
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_format_type(sdp_media: *const SdpMedia) -> RustSdpFormatType {
+    match *(*sdp_media).get_formats() {
+        SdpFormatList::Integers(_) => RustSdpFormatType::Integers,
+        SdpFormatList::Strings(_) => RustSdpFormatType::Strings,
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_format_string_vec(
+    sdp_media: *const SdpMedia,
+) -> *const Vec<String> {
+    if let SdpFormatList::Strings(ref formats) = *(*sdp_media).get_formats() {
+        formats
+    } else {
+        ptr::null()
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_format_u32_vec(sdp_media: *const SdpMedia) -> *const Vec<u32> {
+    if let SdpFormatList::Integers(ref formats) = *(*sdp_media).get_formats() {
+        formats
+    } else {
+        ptr::null()
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_set_media_port(sdp_media: *mut SdpMedia, port: u32) {
+    (*sdp_media).set_port(port);
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_media_port(sdp_media: *const SdpMedia) -> u32 {
+    (*sdp_media).get_port()
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_media_port_count(sdp_media: *const SdpMedia) -> u32 {
+    (*sdp_media).get_port_count()
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_media_bandwidth(
+    sdp_media: *const SdpMedia,
+    bandwidth_type: *const c_char,
+) -> u32 {
+    get_bandwidth((*sdp_media).get_bandwidth(), bandwidth_type)
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_media_bandwidth_vec(
+    sdp_media: *const SdpMedia,
+) -> *const Vec<SdpBandwidth> {
+    (*sdp_media).get_bandwidth()
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_media_has_connection(sdp_media: *const SdpMedia) -> bool {
+    (*sdp_media).get_connection().is_some()
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_media_connection(
+    sdp_media: *const SdpMedia,
+    ret: *mut RustSdpConnection,
+) -> nsresult {
+    if let &Some(ref connection) = (*sdp_media).get_connection() {
+        *ret = RustSdpConnection::from(connection);
+        return NS_OK;
+    }
+    NS_ERROR_INVALID_ARG
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_get_media_attribute_list(
+    sdp_media: *const SdpMedia,
+) -> *const Vec<SdpAttribute> {
+    (*sdp_media).get_attributes()
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_media_clear_codecs(sdp_media: *mut SdpMedia) {
+    (*sdp_media).remove_codecs()
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_media_add_codec(
+    sdp_media: *mut SdpMedia,
+    pt: u8,
+    codec_name: StringView,
+    clockrate: u32,
+    channels: u16,
+) -> nsresult {
+    let rtpmap = SdpAttributeRtpmap {
+        payload_type: pt,
+        codec_name: match codec_name.try_into() {
+            Ok(x) => x,
+            Err(boxed_error) => {
+                error!("Error while parsing string, description: {}", boxed_error);
+                return NS_ERROR_INVALID_ARG;
+            }
+        },
+        frequency: clockrate,
+        channels: Some(channels as u32),
+    };
+
+    match (*sdp_media).add_codec(rtpmap) {
+        Ok(_) => NS_OK,
+        Err(_) => NS_ERROR_INVALID_ARG,
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_media_add_datachannel(
+    sdp_media: *mut SdpMedia,
+    name: StringView,
+    port: u16,
+    streams: u16,
+    message_size: u32,
+) -> nsresult {
+    let name_str = match name.try_into() {
+        Ok(x) => x,
+        Err(_) => {
+            return NS_ERROR_INVALID_ARG;
+        }
+    };
+    match (*sdp_media).add_datachannel(name_str, port, streams, message_size) {
+        Ok(_) => NS_OK,
+        Err(_) => NS_ERROR_INVALID_ARG,
+    }
+}
diff --git a/dom/media/webrtc/sdp/rsdparsa_capi/src/network.rs b/dom/media/webrtc/sdp/rsdparsa_capi/src/network.rs
new file mode 100644
index 0000000000..970f7c8dec
--- /dev/null
+++ b/dom/media/webrtc/sdp/rsdparsa_capi/src/network.rs
@@ -0,0 +1,266 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+extern crate nserror;
+
+use std::ffi::{CStr, CString};
+use std::net::IpAddr;
+use std::os::raw::c_char;
+
+use rsdparsa::address::{Address, AddressType, AddressTyped, ExplicitlyTypedAddress};
+use rsdparsa::{SdpBandwidth, SdpConnection, SdpOrigin};
+use std::convert::TryFrom;
+use types::{StringView, NULL_STRING};
+
+#[repr(C)]
+#[derive(Clone, Copy, PartialEq)]
+pub enum RustAddressType {
+    IP4,
+    IP6,
+}
+
+impl TryFrom<u32> for RustAddressType {
+    type Error = nserror::nsresult;
+    fn try_from(address_type: u32) -> Result<Self, Self::Error> {
+        match address_type {
+            1 => Ok(RustAddressType::IP4),
+            2 => Ok(RustAddressType::IP6),
+            _ => Err(nserror::NS_ERROR_INVALID_ARG),
+        }
+    }
+}
+
+impl From<AddressType> for RustAddressType {
+    fn from(address_type: AddressType) -> Self {
+        match address_type {
+            AddressType::IpV4 => RustAddressType::IP4,
+            AddressType::IpV6 => RustAddressType::IP6,
+        }
+    }
+}
+
+impl Into<AddressType> for RustAddressType {
+    fn into(self) -> AddressType {
+        match self {
+            RustAddressType::IP4 => AddressType::IpV4,
+            RustAddressType::IP6 => AddressType::IpV6,
+        }
+    }
+}
+
+impl<'a> From<&'a IpAddr> for RustAddressType {
+    fn from(addr: &IpAddr) -> RustAddressType {
+        addr.address_type().into()
+    }
+}
+
+pub fn get_octets(addr: &IpAddr) -> [u8; 16] {
+    let mut octets = [0; 16];
+    match *addr {
+        IpAddr::V4(v4_addr) => {
+            let v4_octets = v4_addr.octets();
+            (&mut octets[0..4]).copy_from_slice(&v4_octets);
+        }
+        IpAddr::V6(v6_addr) => {
+            let v6_octets = v6_addr.octets();
+            octets.copy_from_slice(&v6_octets);
+        }
+    }
+    octets
+}
+
+#[repr(C)]
+pub struct RustAddress {
+    ip_address: [u8; 50],
+    fqdn: StringView,
+    is_fqdn: bool,
+}
+
+impl<'a> From<&'a Address> for RustAddress {
+    fn from(address: &Address) -> Self {
+        match address {
+            Address::Ip(ip) => Self::from(ip),
+            Address::Fqdn(fqdn) => Self {
+                ip_address: [0; 50],
+                fqdn: fqdn.as_str().into(),
+                is_fqdn: true,
+            },
+        }
+    }
+}
+
+impl<'a> From<&'a IpAddr> for RustAddress {
+    fn from(addr: &IpAddr) -> Self {
+        let mut c_addr = [0; 50];
+        let str_addr = format!("{}", addr);
+        let str_bytes = str_addr.as_bytes();
+        if str_bytes.len() < 50 {
+            c_addr[..str_bytes.len()].copy_from_slice(&str_bytes);
+        }
+        Self {
+            ip_address: c_addr,
+            fqdn: NULL_STRING,
+            is_fqdn: false,
+        }
+    }
+}
+
+#[repr(C)]
+pub struct RustExplicitlyTypedAddress {
+    address_type: RustAddressType,
+    address: RustAddress,
+}
+
+impl Default for RustExplicitlyTypedAddress {
+    fn default() -> Self {
+        Self {
+            address_type: RustAddressType::IP4,
+            address: RustAddress {
+                ip_address: [0; 50],
+                fqdn: NULL_STRING,
+                is_fqdn: false,
+            },
+        }
+    }
+}
+
+impl<'a> From<&'a ExplicitlyTypedAddress> for RustExplicitlyTypedAddress {
+    fn from(address: &ExplicitlyTypedAddress) -> Self {
+        match address {
+            ExplicitlyTypedAddress::Fqdn { domain, .. } => Self {
+                address_type: address.address_type().into(),
+                address: RustAddress {
+                    ip_address: [0; 50],
+                    fqdn: StringView::from(domain.as_str()),
+                    is_fqdn: true,
+                },
+            },
+            ExplicitlyTypedAddress::Ip(ip_address) => Self {
+                address_type: ip_address.address_type().into(),
+                address: ip_address.into(),
+            },
+        }
+    }
+}
+
+// TODO @@NG remove
+impl<'a> From<&'a Option<IpAddr>> for RustExplicitlyTypedAddress {
+    fn from(addr: &Option<IpAddr>) -> Self {
+        match *addr {
+            Some(ref x) => Self {
+                address_type: RustAddressType::from(x.address_type()),
+                address: RustAddress::from(x),
+            },
+            None => Self::default(),
+        }
+    }
+}
+
+#[repr(C)]
+pub struct RustSdpConnection {
+    pub addr: RustExplicitlyTypedAddress,
+    pub ttl: u8,
+    pub amount: u64,
+}
+
+impl<'a> From<&'a SdpConnection> for RustSdpConnection {
+    fn from(sdp_connection: &SdpConnection) -> Self {
+        let ttl = match sdp_connection.ttl {
+            Some(x) => x as u8,
+            None => 0,
+        };
+        let amount = match sdp_connection.amount {
+            Some(x) => x as u64,
+            None => 0,
+        };
+        RustSdpConnection {
+            addr: RustExplicitlyTypedAddress::from(&sdp_connection.address),
+            ttl: ttl,
+            amount: amount,
+        }
+    }
+}
+
+#[repr(C)]
+pub struct RustSdpOrigin {
+    username: StringView,
+    session_id: u64,
+    session_version: u64,
+    addr: RustExplicitlyTypedAddress,
+}
+
+fn bandwidth_match(str_bw: &str, enum_bw: &SdpBandwidth) -> bool {
+    match *enum_bw {
+        SdpBandwidth::As(_) => str_bw == "AS",
+        SdpBandwidth::Ct(_) => str_bw == "CT",
+        SdpBandwidth::Tias(_) => str_bw == "TIAS",
+        SdpBandwidth::Unknown(ref type_name, _) => str_bw == type_name,
+    }
+}
+
+fn bandwidth_value(bandwidth: &SdpBandwidth) -> u32 {
+    match *bandwidth {
+        SdpBandwidth::As(x) | SdpBandwidth::Ct(x) | SdpBandwidth::Tias(x) => x,
+        SdpBandwidth::Unknown(_, _) => 0,
+    }
+}
+
+pub unsafe fn get_bandwidth(bandwidths: &Vec<SdpBandwidth>, bandwidth_type: *const c_char) -> u32 {
+    let bw_type = match CStr::from_ptr(bandwidth_type).to_str() {
+        Ok(string) => string,
+        Err(_) => return 0,
+    };
+    for bandwidth in bandwidths.iter() {
+        if bandwidth_match(bw_type, bandwidth) {
+            return bandwidth_value(bandwidth);
+        }
+    }
+    0
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_serialize_bandwidth(bw: *const Vec<SdpBandwidth>) -> *mut c_char {
+    let mut builder = String::new();
+    for bandwidth in (*bw).iter() {
+        match *bandwidth {
+            SdpBandwidth::As(val) => {
+                builder.push_str("b=AS:");
+                builder.push_str(&val.to_string());
+                builder.push_str("\r\n");
+            }
+            SdpBandwidth::Ct(val) => {
+                builder.push_str("b=CT:");
+                builder.push_str(&val.to_string());
+                builder.push_str("\r\n");
+            }
+            SdpBandwidth::Tias(val) => {
+                builder.push_str("b=TIAS:");
+                builder.push_str(&val.to_string());
+                builder.push_str("\r\n");
+            }
+            SdpBandwidth::Unknown(ref name, val) => {
+                builder.push_str("b=");
+                builder.push_str(name.as_str());
+                builder.push(':');
+                builder.push_str(&val.to_string());
+                builder.push_str("\r\n");
+            }
+        }
+    }
+    CString::from_vec_unchecked(builder.into_bytes()).into_raw()
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn sdp_free_string(s: *mut c_char) {
+    drop(CString::from_raw(s));
+}
+
+pub unsafe fn origin_view_helper(origin: &SdpOrigin) -> RustSdpOrigin {
+    RustSdpOrigin {
+        username: StringView::from(origin.username.as_str()),
+        session_id: origin.session_id,
+        session_version: origin.session_version,
+        addr: RustExplicitlyTypedAddress::from(&origin.unicast_addr),
+    }
+}
diff --git a/dom/media/webrtc/sdp/rsdparsa_capi/src/types.rs b/dom/media/webrtc/sdp/rsdparsa_capi/src/types.rs
new file mode 100644
index 0000000000..2522c8333d
--- /dev/null
+++ b/dom/media/webrtc/sdp/rsdparsa_capi/src/types.rs
@@ -0,0 +1,199 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+use libc::size_t;
+use std::boxed::Box;
+use std::convert::TryInto;
+use std::error::Error;
+use std::ffi::CStr;
+use std::{slice, str};
+
+use nserror::{nsresult, NS_ERROR_INVALID_ARG, NS_OK};
+
+use rsdparsa::attribute_type::SdpAttributeSsrc;
+
+#[repr(C)]
+#[derive(Clone, Copy)]
+pub struct StringView {
+    buffer: *const u8,
+    len: size_t,
+}
+
+pub const NULL_STRING: StringView = StringView {
+    buffer: 0 as *const u8,
+    len: 0,
+};
+
+impl<'a> From<&'a str> for StringView {
+    fn from(input: &str) -> StringView {
+        StringView {
+            buffer: input.as_ptr(),
+            len: input.len(),
+        }
+    }
+}
+
+impl TryInto<String> for StringView {
+    type Error = Box<dyn Error>;
+    fn try_into(self) -> Result<String, Box<dyn Error>> {
+        // This block must be unsafe as it converts a StringView, most likly provided from the
+        // C++ code, into a rust String and thus needs to operate with raw pointers.
+        let string_slice: &[u8];
+        unsafe {
+            // Add one to the length as the length passed in the StringView is the length of
+            // the string and is missing the null terminator
+            string_slice = slice::from_raw_parts(self.buffer, self.len + 1 as usize);
+        }
+
+        let c_str = match CStr::from_bytes_with_nul(string_slice) {
+            Ok(string) => string,
+            Err(x) => {
+                return Err(Box::new(x));
+            }
+        };
+
+        let str_slice: &str = match str::from_utf8(c_str.to_bytes()) {
+            Ok(string) => string,
+            Err(x) => {
+                return Err(Box::new(x));
+            }
+        };
+
+        Ok(str_slice.to_string())
+    }
+}
+
+impl<'a, T: AsRef<str>> From<&'a Option<T>> for StringView {
+    fn from(input: &Option<T>) -> StringView {
+        match *input {
+            Some(ref x) => StringView {
+                buffer: x.as_ref().as_ptr(),
+                len: x.as_ref().len(),
+            },
+            None => NULL_STRING,
+        }
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn string_vec_len(vec: *const Vec<String>) -> size_t {
+    (*vec).len() as size_t
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn string_vec_get_view(
+    vec: *const Vec<String>,
+    index: size_t,
+    ret: *mut StringView,
+) -> nsresult {
+    match (*vec).get(index) {
+        Some(ref string) => {
+            *ret = StringView::from(string.as_str());
+            NS_OK
+        }
+        None => NS_ERROR_INVALID_ARG,
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn free_boxed_string_vec(ptr: *mut Vec<String>) -> nsresult {
+    drop(Box::from_raw(ptr));
+    NS_OK
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn f32_vec_len(vec: *const Vec<f32>) -> size_t {
+    (*vec).len()
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn f32_vec_get(
+    vec: *const Vec<f32>,
+    index: size_t,
+    ret: *mut f32,
+) -> nsresult {
+    match (*vec).get(index) {
+        Some(val) => {
+            *ret = *val;
+            NS_OK
+        }
+        None => NS_ERROR_INVALID_ARG,
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn u32_vec_len(vec: *const Vec<u32>) -> size_t {
+    (*vec).len()
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn u32_vec_get(
+    vec: *const Vec<u32>,
+    index: size_t,
+    ret: *mut u32,
+) -> nsresult {
+    match (*vec).get(index) {
+        Some(val) => {
+            *ret = *val;
+            NS_OK
+        }
+        None => NS_ERROR_INVALID_ARG,
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn u16_vec_len(vec: *const Vec<u16>) -> size_t {
+    (*vec).len()
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn u16_vec_get(
+    vec: *const Vec<u16>,
+    index: size_t,
+    ret: *mut u16,
+) -> nsresult {
+    match (*vec).get(index) {
+        Some(val) => {
+            *ret = *val;
+            NS_OK
+        }
+        None => NS_ERROR_INVALID_ARG,
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn u8_vec_len(vec: *const Vec<u8>) -> size_t {
+    (*vec).len()
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn u8_vec_get(vec: *const Vec<u8>, index: size_t, ret: *mut u8) -> nsresult {
+    match (*vec).get(index) {
+        Some(val) => {
+            *ret = *val;
+            NS_OK
+        }
+        None => NS_ERROR_INVALID_ARG,
+    }
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn ssrc_vec_len(vec: *const Vec<SdpAttributeSsrc>) -> size_t {
+    (*vec).len()
+}
+
+#[no_mangle]
+pub unsafe extern "C" fn ssrc_vec_get_id(
+    vec: *const Vec<SdpAttributeSsrc>,
+    index: size_t,
+    ret: *mut u32,
+) -> nsresult {
+    match (*vec).get(index) {
+        Some(val) => {
+            *ret = val.id;
+            NS_OK
+        }
+        None => NS_ERROR_INVALID_ARG,
+    }
+}
diff --git a/dom/media/webrtc/tests/crashtests/1770075.html b/dom/media/webrtc/tests/crashtests/1770075.html
new file mode 100644
index 0000000000..4d451216bd
--- /dev/null
+++ b/dom/media/webrtc/tests/crashtests/1770075.html
@@ -0,0 +1,8 @@
+<script>
+window.addEventListener('load', () => {
+  let a = new RTCPeerConnection({}, {})
+  a.createOffer({'offerToReceiveVideo': true})
+  let b = new WeakRef(a.getTransceivers()[0])
+  setTimeout("self.close()", 200)
+})
+</script>
diff --git a/dom/media/webrtc/tests/crashtests/1789908.html b/dom/media/webrtc/tests/crashtests/1789908.html
new file mode 100644
index 0000000000..3d58d3dc6b
--- /dev/null
+++ b/dom/media/webrtc/tests/crashtests/1789908.html
@@ -0,0 +1,25 @@
+<script>
+window.addEventListener('load', () => {
+  const sdp = `v=0
+o=mozilla...THIS_IS_SDPARTA-99.0 4978061689314146455 0 IN IP4 0.0.0.0
+s=-
+t=0 0
+a=fingerprint:sha-256 1D:E5:0C:97:18:43:38:3D:FF:7D:6A:BF:E3:AC:CA:70:AB:53:5A:35:95:92:4F:98:86:61:CA:5D:D5:9D:5E:41
+a=group:BUNDLE 0
+a=ice-options:trickle
+a=msid-semantic:WMS *
+m=video 9 UDP/TLS/RTP/SAVPF 120
+c=IN IP4 0.0.0.0
+a=fmtp:120 max-fs=12288;max-fr=60
+a=ice-pwd:c3a5e05023a8c38f671aef91ed1802d6
+a=ice-ufrag:91e4526d
+a=setup:actpass
+
+a=rtpmap:120 VP8/90000
+`;
+
+  let a = new RTCPeerConnection()
+  a.setRemoteDescription({sdp, type: "offer"});
+  setTimeout("self.close()", 200)
+})
+</script>
diff --git a/dom/media/webrtc/tests/crashtests/1799168.html b/dom/media/webrtc/tests/crashtests/1799168.html
new file mode 100644
index 0000000000..6c5c9db237
--- /dev/null
+++ b/dom/media/webrtc/tests/crashtests/1799168.html
@@ -0,0 +1,16 @@
+<script>
+window.addEventListener('load', async () => {
+  const offerer = new RTCPeerConnection();
+  const answerer = new RTCPeerConnection();
+  offerer.addTransceiver('audio');
+  await offerer.setLocalDescription();
+  await answerer.setRemoteDescription(offerer.localDescription);
+  const answer = await answerer.createAnswer();
+  await offerer.setRemoteDescription(answer);
+  // relay candidate with TCP!
+  const candidate = 'candidate:3 1 tcp 18087935 20.253.151.225 3478 typ relay raddr 10.0.48.153 rport 3478 tcptype passive';
+  await offerer.addIceCandidate({candidate, sdpMLineIndex: 0});
+  await new Promise(r => setTimeout(r, 2000));
+  self.close();
+})
+</script>
diff --git a/dom/media/webrtc/tests/crashtests/1816708.html b/dom/media/webrtc/tests/crashtests/1816708.html
new file mode 100644
index 0000000000..c7ba824041
--- /dev/null
+++ b/dom/media/webrtc/tests/crashtests/1816708.html
@@ -0,0 +1,21 @@
+<!DOCTYPE html>
+<html class="reftest-wait">
+<head>
+<script>
+document.addEventListener('DOMContentLoaded', async () => {
+  const peer = new RTCPeerConnection()
+  const stream = await navigator.mediaDevices.getUserMedia({
+    video: true,
+    audio: true,
+    fake: true,
+    peerIdentity: 'name',
+  })
+  stream.getTracks().forEach((track) => peer.addTrack(track, stream))
+  const offer = await peer.createOffer({})
+  await peer.setLocalDescription(offer)
+  await peer.setRemoteDescription(offer)
+  document.documentElement.removeAttribute("class");
+})
+</script>
+</head>
+</html>
diff --git a/dom/media/webrtc/tests/crashtests/1821477.html b/dom/media/webrtc/tests/crashtests/1821477.html
new file mode 100644
index 0000000000..c37bd6bd02
--- /dev/null
+++ b/dom/media/webrtc/tests/crashtests/1821477.html
@@ -0,0 +1,16 @@
+<html class="reftest-wait">
+<script>
+document.addEventListener("DOMContentLoaded", async () => {
+  SpecialPowers.wrap(document).notifyUserGestureActivation();
+  try {
+    (await navigator.mediaDevices.getDisplayMedia({
+      "video": {
+        "frameRate": 2147483647,
+      },
+    })).stop();
+  } finally {
+    document.documentElement.removeAttribute("class");
+  }
+});
+</script>
+</html>
diff --git a/dom/media/webrtc/tests/crashtests/crashtests.list b/dom/media/webrtc/tests/crashtests/crashtests.list
new file mode 100644
index 0000000000..64434b28e2
--- /dev/null
+++ b/dom/media/webrtc/tests/crashtests/crashtests.list
@@ -0,0 +1,7 @@
+defaults pref(media.navigator.permission.disabled,true) pref(media.devices.insecure.enabled,true) pref(media.getusermedia.insecure.enabled,true)
+
+load 1770075.html
+load 1789908.html
+load 1799168.html
+load 1816708.html
+skip-if(/^Windows\x20NT\x206\.1/.test(http.oscpu)) load 1821477.html
diff --git a/dom/media/webrtc/tests/fuzztests/moz.build b/dom/media/webrtc/tests/fuzztests/moz.build
new file mode 100644
index 0000000000..fef388e6c9
--- /dev/null
+++ b/dom/media/webrtc/tests/fuzztests/moz.build
@@ -0,0 +1,22 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+Library("FuzzingSdp")
+
+LOCAL_INCLUDES += [
+    "/dom/media/webrtc",
+    "/ipc/chromium/src",
+    "/media/webrtc",
+]
+
+# Add libFuzzer configuration directives
+include("/tools/fuzzing/libfuzzer-config.mozbuild")
+
+SOURCES += [
+    "sdp_parser_libfuzz.cpp",
+]
+
+FINAL_LIBRARY = "xul-gtest"
diff --git a/dom/media/webrtc/tests/fuzztests/sdp_parser_libfuzz.cpp b/dom/media/webrtc/tests/fuzztests/sdp_parser_libfuzz.cpp
new file mode 100644
index 0000000000..3451d6fd21
--- /dev/null
+++ b/dom/media/webrtc/tests/fuzztests/sdp_parser_libfuzz.cpp
@@ -0,0 +1,30 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include <string>
+
+#include "gtest/gtest.h"
+
+#include "FuzzingInterface.h"
+
+#include "sdp/SipccSdpParser.h"
+
+using namespace mozilla;
+
+static mozilla::UniquePtr<SdpParser::Results> sdpPtr;
+static SipccSdpParser mParser;
+
+int FuzzingInitSdpParser(int* argc, char*** argv) { return 0; }
+
+static int RunSdpParserFuzzing(const uint8_t* data, size_t size) {
+  std::string message(reinterpret_cast<const char*>(data), size);
+
+  sdpPtr = mParser.Parse(message);
+
+  return 0;
+}
+
+MOZ_FUZZING_INTERFACE_RAW(FuzzingInitSdpParser, RunSdpParserFuzzing, SdpParser);
diff --git a/dom/media/webrtc/tests/mochitests/NetworkPreparationChromeScript.js b/dom/media/webrtc/tests/mochitests/NetworkPreparationChromeScript.js
new file mode 100644
index 0000000000..d3872f1519
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/NetworkPreparationChromeScript.js
@@ -0,0 +1,43 @@
+/* Any copyright is dedicated to the Public Domain.
+   http://creativecommons.org/publicdomain/zero/1.0/ */
+
+var browser = Services.wm.getMostRecentWindow("navigator:browser");
+var connection = browser.navigator.mozMobileConnections[0];
+
+// provide a fake APN and enable data connection.
+// enable 3G radio
+function enableRadio() {
+  if (connection.radioState !== "enabled") {
+    connection.setRadioEnabled(true);
+  }
+}
+
+// disable 3G radio
+function disableRadio() {
+  if (connection.radioState === "enabled") {
+    connection.setRadioEnabled(false);
+  }
+}
+
+addMessageListener("prepare-network", function (message) {
+  connection.addEventListener("datachange", function onDataChange() {
+    if (connection.data.connected) {
+      connection.removeEventListener("datachange", onDataChange);
+      Services.prefs.setIntPref("network.proxy.type", 2);
+      sendAsyncMessage("network-ready", true);
+    }
+  });
+
+  enableRadio();
+});
+
+addMessageListener("network-cleanup", function (message) {
+  connection.addEventListener("datachange", function onDataChange() {
+    if (!connection.data.connected) {
+      connection.removeEventListener("datachange", onDataChange);
+      Services.prefs.setIntPref("network.proxy.type", 2);
+      sendAsyncMessage("network-disabled", true);
+    }
+  });
+  disableRadio();
+});
diff --git a/dom/media/webrtc/tests/mochitests/addTurnsSelfsignedCert.js b/dom/media/webrtc/tests/mochitests/addTurnsSelfsignedCert.js
new file mode 100644
index 0000000000..1e8be3a397
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/addTurnsSelfsignedCert.js
@@ -0,0 +1,32 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+/* eslint-env mozilla/chrome-script */
+
+"use strict";
+
+// This is only usable from the parent process, even for doing simple stuff like
+// serializing a cert.
+var gCertMaker = Cc["@mozilla.org/security/x509certdb;1"].getService(
+  Ci.nsIX509CertDB
+);
+
+var gCertOverrides = Cc["@mozilla.org/security/certoverride;1"].getService(
+  Ci.nsICertOverrideService
+);
+
+addMessageListener("add-turns-certs", certs => {
+  var port = 5349;
+  certs.forEach(certDescription => {
+    var cert = gCertMaker.constructX509FromBase64(certDescription.cert);
+    gCertOverrides.rememberValidityOverride(
+      certDescription.hostname,
+      port,
+      {},
+      cert,
+      false
+    );
+  });
+  sendAsyncMessage("certs-added");
+});
diff --git a/dom/media/webrtc/tests/mochitests/blacksilence.js b/dom/media/webrtc/tests/mochitests/blacksilence.js
new file mode 100644
index 0000000000..5ea35f8a7f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/blacksilence.js
@@ -0,0 +1,134 @@
+(function (global) {
+  "use strict";
+
+  // an invertible check on the condition.
+  // if the constraint is applied, then the check is direct
+  // if not applied, then the result should be reversed
+  function check(constraintApplied, condition, message) {
+    var good = constraintApplied ? condition : !condition;
+    message =
+      (constraintApplied ? "with" : "without") +
+      " constraint: should " +
+      (constraintApplied ? "" : "not ") +
+      message +
+      " = " +
+      (good ? "OK" : "waiting...");
+    info(message);
+    return good;
+  }
+
+  function mkElement(type) {
+    // This makes an unattached element.
+    // It's not rendered to save the cycles that costs on b2g emulator
+    // and it gets dropped (and GC'd) when the test is done.
+    var e = document.createElement(type);
+    e.width = 32;
+    e.height = 24;
+    document.getElementById("display").appendChild(e);
+    return e;
+  }
+
+  // Runs checkFunc until it reports success.
+  // This is kludgy, but you have to wait for media to start flowing, and it
+  // can't be any old media, it has to include real data, for which we have no
+  // reliable signals to use as a trigger.
+  function periodicCheck(checkFunc) {
+    var resolve;
+    var done = false;
+    // This returns a function so that we create 10 closures in the loop, not
+    // one; and so that the timers don't all start straight away
+    var waitAndCheck = counter => () => {
+      if (done) {
+        return Promise.resolve();
+      }
+      return new Promise(r => setTimeout(r, 200 << counter)).then(() => {
+        if (checkFunc()) {
+          done = true;
+          resolve();
+        }
+      });
+    };
+
+    var chain = Promise.resolve();
+    for (var i = 0; i < 10; ++i) {
+      chain = chain.then(waitAndCheck(i));
+    }
+    return new Promise(r => (resolve = r));
+  }
+
+  function isSilence(audioData) {
+    var silence = true;
+    for (var i = 0; i < audioData.length; ++i) {
+      if (audioData[i] !== 128) {
+        silence = false;
+      }
+    }
+    return silence;
+  }
+
+  function checkAudio(constraintApplied, stream) {
+    var audio = mkElement("audio");
+    audio.srcObject = stream;
+    audio.play();
+
+    var context = new AudioContext();
+    var source = context.createMediaStreamSource(stream);
+    var analyser = context.createAnalyser();
+    source.connect(analyser);
+    analyser.connect(context.destination);
+
+    return periodicCheck(() => {
+      var sampleCount = analyser.frequencyBinCount;
+      info("got some audio samples: " + sampleCount);
+      var buffer = new Uint8Array(sampleCount);
+      analyser.getByteTimeDomainData(buffer);
+
+      var silent = check(
+        constraintApplied,
+        isSilence(buffer),
+        "be silence for audio"
+      );
+      return sampleCount > 0 && silent;
+    }).then(() => {
+      source.disconnect();
+      analyser.disconnect();
+      audio.pause();
+      ok(true, "audio is " + (constraintApplied ? "" : "not ") + "silent");
+    });
+  }
+
+  function checkVideo(constraintApplied, stream) {
+    var video = mkElement("video");
+    video.srcObject = stream;
+    video.play();
+
+    return periodicCheck(() => {
+      try {
+        var canvas = mkElement("canvas");
+        var ctx = canvas.getContext("2d");
+        // Have to guard drawImage with the try as well, due to bug 879717. If
+        // we get an error, this round fails, but that failure is usually just
+        // transitory.
+        ctx.drawImage(video, 0, 0);
+        ctx.getImageData(0, 0, 1, 1);
+        return check(
+          constraintApplied,
+          false,
+          "throw on getImageData for video"
+        );
+      } catch (e) {
+        return check(
+          constraintApplied,
+          e.name === "SecurityError",
+          "get a security error: " + e.name
+        );
+      }
+    }).then(() => {
+      video.pause();
+      ok(true, "video is " + (constraintApplied ? "" : "not ") + "protected");
+    });
+  }
+
+  global.audioIsSilence = checkAudio;
+  global.videoIsBlack = checkVideo;
+})(this);
diff --git a/dom/media/webrtc/tests/mochitests/dataChannel.js b/dom/media/webrtc/tests/mochitests/dataChannel.js
new file mode 100644
index 0000000000..eac52f96ab
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/dataChannel.js
@@ -0,0 +1,352 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+/**
+ * Returns the contents of a blob as text
+ *
+ * @param {Blob} blob
+          The blob to retrieve the contents from
+ */
+function getBlobContent(blob) {
+  return new Promise(resolve => {
+    var reader = new FileReader();
+    // Listen for 'onloadend' which will always be called after a success or failure
+    reader.onloadend = event => resolve(event.target.result);
+    reader.readAsText(blob);
+  });
+}
+
+var commandsCreateDataChannel = [
+  function PC_REMOTE_EXPECT_DATA_CHANNEL(test) {
+    test.pcRemote.expectDataChannel();
+  },
+
+  function PC_LOCAL_CREATE_DATA_CHANNEL(test) {
+    var channel = test.pcLocal.createDataChannel({});
+    is(channel.binaryType, "blob", channel + " is of binary type 'blob'");
+
+    is(
+      test.pcLocal.signalingState,
+      STABLE,
+      "Create datachannel does not change signaling state"
+    );
+    return test.pcLocal.observedNegotiationNeeded;
+  },
+];
+
+var commandsWaitForDataChannel = [
+  function PC_LOCAL_VERIFY_DATA_CHANNEL_STATE(test) {
+    return test.pcLocal.dataChannels[0].opened;
+  },
+
+  function PC_REMOTE_VERIFY_DATA_CHANNEL_STATE(test) {
+    return test.pcRemote.nextDataChannel.then(channel => channel.opened);
+  },
+];
+
+var commandsCheckDataChannel = [
+  function SEND_MESSAGE(test) {
+    var message = "Lorem ipsum dolor sit amet";
+
+    info("Sending message:" + message);
+    return test.send(message).then(result => {
+      is(
+        result.data,
+        message,
+        "Message correctly transmitted from pcLocal to pcRemote."
+      );
+    });
+  },
+
+  function SEND_BLOB(test) {
+    var contents = "At vero eos et accusam et justo duo dolores et ea rebum.";
+    var blob = new Blob([contents], { type: "text/plain" });
+
+    info("Sending blob");
+    return test
+      .send(blob)
+      .then(result => {
+        ok(result.data instanceof Blob, "Received data is of instance Blob");
+        is(result.data.size, blob.size, "Received data has the correct size.");
+
+        return getBlobContent(result.data);
+      })
+      .then(recv_contents =>
+        is(recv_contents, contents, "Received data has the correct content.")
+      );
+  },
+
+  function CREATE_SECOND_DATA_CHANNEL(test) {
+    return test.createDataChannel({}).then(result => {
+      is(
+        result.remote.binaryType,
+        "blob",
+        "remote data channel is of binary type 'blob'"
+      );
+    });
+  },
+
+  function SEND_MESSAGE_THROUGH_LAST_OPENED_CHANNEL(test) {
+    var channels = test.pcRemote.dataChannels;
+    var message = "I am the Omega";
+
+    info("Sending message:" + message);
+    return test.send(message).then(result => {
+      is(
+        channels.indexOf(result.channel),
+        channels.length - 1,
+        "Last channel used"
+      );
+      is(result.data, message, "Received message has the correct content.");
+    });
+  },
+
+  function SEND_MESSAGE_THROUGH_FIRST_CHANNEL(test) {
+    var message = "Message through 1st channel";
+    var options = {
+      sourceChannel: test.pcLocal.dataChannels[0],
+      targetChannel: test.pcRemote.dataChannels[0],
+    };
+
+    info("Sending message:" + message);
+    return test.send(message, options).then(result => {
+      is(
+        test.pcRemote.dataChannels.indexOf(result.channel),
+        0,
+        "1st channel used"
+      );
+      is(result.data, message, "Received message has the correct content.");
+    });
+  },
+
+  function SEND_MESSAGE_BACK_THROUGH_FIRST_CHANNEL(test) {
+    var message = "Return a message also through 1st channel";
+    var options = {
+      sourceChannel: test.pcRemote.dataChannels[0],
+      targetChannel: test.pcLocal.dataChannels[0],
+    };
+
+    info("Sending message:" + message);
+    return test.send(message, options).then(result => {
+      is(
+        test.pcLocal.dataChannels.indexOf(result.channel),
+        0,
+        "1st channel used"
+      );
+      is(result.data, message, "Return message has the correct content.");
+    });
+  },
+
+  function CREATE_NEGOTIATED_DATA_CHANNEL_MAX_RETRANSMITS(test) {
+    var options = {
+      negotiated: true,
+      id: 5,
+      protocol: "foo/bar",
+      ordered: false,
+      maxRetransmits: 500,
+    };
+    return test.createDataChannel(options).then(result => {
+      is(
+        result.local.binaryType,
+        "blob",
+        result.remote + " is of binary type 'blob'"
+      );
+      is(
+        result.local.id,
+        options.id,
+        result.local + " id is:" + result.local.id
+      );
+      is(
+        result.local.protocol,
+        options.protocol,
+        result.local + " protocol is:" + result.local.protocol
+      );
+      is(
+        result.local.reliable,
+        false,
+        result.local + " reliable is:" + result.local.reliable
+      );
+      is(
+        result.local.ordered,
+        options.ordered,
+        result.local + " ordered is:" + result.local.ordered
+      );
+      is(
+        result.local.maxRetransmits,
+        options.maxRetransmits,
+        result.local + " maxRetransmits is:" + result.local.maxRetransmits
+      );
+      is(
+        result.local.maxPacketLifeTime,
+        null,
+        result.local + " maxPacketLifeTime is:" + result.local.maxPacketLifeTime
+      );
+
+      is(
+        result.remote.binaryType,
+        "blob",
+        result.remote + " is of binary type 'blob'"
+      );
+      is(
+        result.remote.id,
+        options.id,
+        result.remote + " id is:" + result.remote.id
+      );
+      is(
+        result.remote.protocol,
+        options.protocol,
+        result.remote + " protocol is:" + result.remote.protocol
+      );
+      is(
+        result.remote.reliable,
+        false,
+        result.remote + " reliable is:" + result.remote.reliable
+      );
+      is(
+        result.remote.ordered,
+        options.ordered,
+        result.remote + " ordered is:" + result.remote.ordered
+      );
+      is(
+        result.remote.maxRetransmits,
+        options.maxRetransmits,
+        result.remote + " maxRetransmits is:" + result.remote.maxRetransmits
+      );
+      is(
+        result.remote.maxPacketLifeTime,
+        null,
+        result.remote +
+          " maxPacketLifeTime is:" +
+          result.remote.maxPacketLifeTime
+      );
+    });
+  },
+
+  function SEND_MESSAGE_THROUGH_LAST_OPENED_CHANNEL2(test) {
+    var channels = test.pcRemote.dataChannels;
+    var message = "I am the walrus; Goo goo g'joob";
+
+    info("Sending message:" + message);
+    return test.send(message).then(result => {
+      is(
+        channels.indexOf(result.channel),
+        channels.length - 1,
+        "Last channel used"
+      );
+      is(result.data, message, "Received message has the correct content.");
+    });
+  },
+
+  function CREATE_NEGOTIATED_DATA_CHANNEL_MAX_PACKET_LIFE_TIME(test) {
+    var options = {
+      ordered: false,
+      maxPacketLifeTime: 10,
+    };
+    return test.createDataChannel(options).then(result => {
+      is(
+        result.local.binaryType,
+        "blob",
+        result.local + " is of binary type 'blob'"
+      );
+      is(
+        result.local.protocol,
+        "",
+        result.local + " protocol is:" + result.local.protocol
+      );
+      is(
+        result.local.reliable,
+        false,
+        result.local + " reliable is:" + result.local.reliable
+      );
+      is(
+        result.local.ordered,
+        options.ordered,
+        result.local + " ordered is:" + result.local.ordered
+      );
+      is(
+        result.local.maxRetransmits,
+        null,
+        result.local + " maxRetransmits is:" + result.local.maxRetransmits
+      );
+      is(
+        result.local.maxPacketLifeTime,
+        options.maxPacketLifeTime,
+        result.local + " maxPacketLifeTime is:" + result.local.maxPacketLifeTime
+      );
+
+      is(
+        result.remote.binaryType,
+        "blob",
+        result.remote + " is of binary type 'blob'"
+      );
+      is(
+        result.remote.protocol,
+        "",
+        result.remote + " protocol is:" + result.remote.protocol
+      );
+      is(
+        result.remote.reliable,
+        false,
+        result.remote + " reliable is:" + result.remote.reliable
+      );
+      is(
+        result.remote.ordered,
+        options.ordered,
+        result.remote + " ordered is:" + result.remote.ordered
+      );
+      is(
+        result.remote.maxRetransmits,
+        null,
+        result.remote + " maxRetransmits is:" + result.remote.maxRetransmits
+      );
+      is(
+        result.remote.maxPacketLifeTime,
+        options.maxPacketLifeTime,
+        result.remote +
+          " maxPacketLifeTime is:" +
+          result.remote.maxPacketLifeTime
+      );
+    });
+  },
+
+  function SEND_MESSAGE_THROUGH_LAST_OPENED_CHANNEL3(test) {
+    var channels = test.pcRemote.dataChannels;
+    var message = "Nice to see you working maxPacketLifeTime";
+
+    info("Sending message:" + message);
+    return test.send(message).then(result => {
+      is(
+        channels.indexOf(result.channel),
+        channels.length - 1,
+        "Last channel used"
+      );
+      is(result.data, message, "Received message has the correct content.");
+    });
+  },
+];
+
+var commandsCheckLargeXfer = [
+  function SEND_BIG_BUFFER(test) {
+    var size = 2 * 1024 * 1024; // SCTP internal buffer is now 1MB, so use 2MB to ensure the buffer gets full
+    var buffer = new ArrayBuffer(size);
+    // note: type received is always blob for binary data
+    var options = {};
+    options.bufferedAmountLowThreshold = 64 * 1024;
+    info("Sending arraybuffer");
+    return test.send(buffer, options).then(result => {
+      ok(result.data instanceof Blob, "Received data is of instance Blob");
+      is(result.data.size, size, "Received data has the correct size.");
+    });
+  },
+];
+
+function addInitialDataChannel(chain) {
+  chain.insertBefore("PC_LOCAL_CREATE_OFFER", commandsCreateDataChannel);
+  chain.insertBefore(
+    "PC_LOCAL_WAIT_FOR_MEDIA_FLOW",
+    commandsWaitForDataChannel
+  );
+  chain.removeAfter("PC_REMOTE_CHECK_ICE_CONNECTIONS");
+  chain.append(commandsCheckDataChannel);
+}
diff --git a/dom/media/webrtc/tests/mochitests/head.js b/dom/media/webrtc/tests/mochitests/head.js
new file mode 100644
index 0000000000..7c9f6d52de
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/head.js
@@ -0,0 +1,1445 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+"use strict";
+
+var Cc = SpecialPowers.Cc;
+var Ci = SpecialPowers.Ci;
+
+// Specifies if we want fake audio streams for this run
+let WANT_FAKE_AUDIO = true;
+// Specifies if we want fake video streams for this run
+let WANT_FAKE_VIDEO = true;
+let TEST_AUDIO_FREQ = 1000;
+
+/**
+ * Reads the current values of preferences affecting fake and loopback devices
+ * and sets the WANT_FAKE_AUDIO and WANT_FAKE_VIDEO gloabals appropriately.
+ */
+function updateConfigFromFakeAndLoopbackPrefs() {
+  let audioDevice = SpecialPowers.getCharPref("media.audio_loopback_dev", "");
+  if (audioDevice) {
+    WANT_FAKE_AUDIO = false;
+    dump("TEST DEVICES: Got loopback audio: " + audioDevice + "\n");
+  } else {
+    WANT_FAKE_AUDIO = true;
+    dump(
+      "TEST DEVICES: No test device found in media.audio_loopback_dev, using fake audio streams.\n"
+    );
+  }
+  let videoDevice = SpecialPowers.getCharPref("media.video_loopback_dev", "");
+  if (videoDevice) {
+    WANT_FAKE_VIDEO = false;
+    dump("TEST DEVICES: Got loopback video: " + videoDevice + "\n");
+  } else {
+    WANT_FAKE_VIDEO = true;
+    dump(
+      "TEST DEVICES: No test device found in media.video_loopback_dev, using fake video streams.\n"
+    );
+  }
+}
+
+updateConfigFromFakeAndLoopbackPrefs();
+
+/**
+ *  Global flag to skip LoopbackTone
+ */
+let DISABLE_LOOPBACK_TONE = false;
+/**
+ * Helper class to setup a sine tone of a given frequency.
+ */
+class LoopbackTone {
+  constructor(audioContext, frequency) {
+    if (!audioContext) {
+      throw new Error("You must provide a valid AudioContext");
+    }
+    this.oscNode = audioContext.createOscillator();
+    var gainNode = audioContext.createGain();
+    gainNode.gain.value = 0.5;
+    this.oscNode.connect(gainNode);
+    gainNode.connect(audioContext.destination);
+    this.changeFrequency(frequency);
+  }
+
+  // Method should be used when WANT_FAKE_AUDIO is false.
+  start() {
+    if (!this.oscNode) {
+      throw new Error("Attempt to start a stopped LoopbackTone");
+    }
+    info(`Start loopback tone at ${this.oscNode.frequency.value}`);
+    this.oscNode.start();
+  }
+
+  // Change the frequency of the tone. It can be used after start.
+  // Frequency will change on the fly. No need to stop and create a new instance.
+  changeFrequency(frequency) {
+    if (!this.oscNode) {
+      throw new Error("Attempt to change frequency on a stopped LoopbackTone");
+    }
+    this.oscNode.frequency.value = frequency;
+  }
+
+  stop() {
+    if (!this.oscNode) {
+      throw new Error("Attempt to stop a stopped LoopbackTone");
+    }
+    this.oscNode.stop();
+    this.oscNode = null;
+  }
+}
+// Object that holds the default loopback tone.
+var DefaultLoopbackTone = null;
+
+/**
+ * This class provides helpers around analysing the audio content in a stream
+ * using WebAudio AnalyserNodes.
+ *
+ * @constructor
+ * @param {object} stream
+ *                 A MediaStream object whose audio track we shall analyse.
+ */
+function AudioStreamAnalyser(ac, stream) {
+  this.audioContext = ac;
+  this.stream = stream;
+  this.sourceNodes = [];
+  this.analyser = this.audioContext.createAnalyser();
+  // Setting values lower than default for speedier testing on emulators
+  this.analyser.smoothingTimeConstant = 0.2;
+  this.analyser.fftSize = 1024;
+  this.connectTrack = t => {
+    let source = this.audioContext.createMediaStreamSource(
+      new MediaStream([t])
+    );
+    this.sourceNodes.push(source);
+    source.connect(this.analyser);
+  };
+  this.stream.getAudioTracks().forEach(t => this.connectTrack(t));
+  this.onaddtrack = ev => this.connectTrack(ev.track);
+  this.stream.addEventListener("addtrack", this.onaddtrack);
+  this.data = new Uint8Array(this.analyser.frequencyBinCount);
+}
+
+AudioStreamAnalyser.prototype = {
+  /**
+   * Get an array of frequency domain data for our stream's audio track.
+   *
+   * @returns {array} A Uint8Array containing the frequency domain data.
+   */
+  getByteFrequencyData() {
+    this.analyser.getByteFrequencyData(this.data);
+    return this.data;
+  },
+
+  /**
+   * Append a canvas to the DOM where the frequency data are drawn.
+   * Useful to debug tests.
+   */
+  enableDebugCanvas() {
+    var cvs = (this.debugCanvas = document.createElement("canvas"));
+    const content = document.getElementById("content");
+    content.insertBefore(cvs, content.children[0]);
+
+    // Easy: 1px per bin
+    cvs.width = this.analyser.frequencyBinCount;
+    cvs.height = 128;
+    cvs.style.border = "1px solid red";
+
+    var c = cvs.getContext("2d");
+    c.fillStyle = "black";
+
+    var self = this;
+    function render() {
+      c.clearRect(0, 0, cvs.width, cvs.height);
+      var array = self.getByteFrequencyData();
+      for (var i = 0; i < array.length; i++) {
+        c.fillRect(i, cvs.height - array[i] / 2, 1, cvs.height);
+      }
+      if (!cvs.stopDrawing) {
+        requestAnimationFrame(render);
+      }
+    }
+    requestAnimationFrame(render);
+  },
+
+  /**
+   * Stop drawing of and remove the debug canvas from the DOM if it was
+   * previously added.
+   */
+  disableDebugCanvas() {
+    if (!this.debugCanvas || !this.debugCanvas.parentElement) {
+      return;
+    }
+
+    this.debugCanvas.stopDrawing = true;
+    this.debugCanvas.parentElement.removeChild(this.debugCanvas);
+  },
+
+  /**
+   * Disconnects the input stream from our internal analyser node.
+   * Call this to reduce main thread processing, mostly necessary on slow
+   * devices.
+   */
+  disconnect() {
+    this.disableDebugCanvas();
+    this.sourceNodes.forEach(n => n.disconnect());
+    this.sourceNodes = [];
+    this.stream.removeEventListener("addtrack", this.onaddtrack);
+  },
+
+  /**
+   * Return a Promise, that will be resolved when the function passed as
+   * argument, when called, returns true (meaning the analysis was a
+   * success). The promise is rejected if the cancel promise resolves first.
+   *
+   * @param {function} analysisFunction
+   *        A function that performs an analysis, and resolves with true if the
+   *        analysis was a success (i.e. it found what it was looking for)
+   * @param {promise} cancel
+   *        A promise that on resolving will reject the promise we returned.
+   */
+  async waitForAnalysisSuccess(
+    analysisFunction,
+    cancel = wait(60000, new Error("Audio analysis timed out"))
+  ) {
+    let aborted = false;
+    cancel.then(() => (aborted = true));
+
+    // We need to give the Analyser some time to start gathering data.
+    await wait(200);
+
+    do {
+      await new Promise(resolve => requestAnimationFrame(resolve));
+      if (aborted) {
+        throw await cancel;
+      }
+    } while (!analysisFunction(this.getByteFrequencyData()));
+  },
+
+  /**
+   * Return the FFT bin index for a given frequency.
+   *
+   * @param {double} frequency
+   *        The frequency for whicht to return the bin number.
+   * @returns {integer} the index of the bin in the FFT array.
+   */
+  binIndexForFrequency(frequency) {
+    return (
+      1 +
+      Math.round(
+        (frequency * this.analyser.fftSize) / this.audioContext.sampleRate
+      )
+    );
+  },
+
+  /**
+   * Reverse operation, get the frequency for a bin index.
+   *
+   * @param {integer} index an index in an FFT array
+   * @returns {double} the frequency for this bin
+   */
+  frequencyForBinIndex(index) {
+    return ((index - 1) * this.audioContext.sampleRate) / this.analyser.fftSize;
+  },
+};
+
+/**
+ * Creates a MediaStream with an audio track containing a sine tone at the
+ * given frequency.
+ *
+ * @param {AudioContext} ac
+ *        AudioContext in which to create the OscillatorNode backing the stream
+ * @param {double} frequency
+ *        The frequency in Hz of the generated sine tone
+ * @returns {MediaStream} the MediaStream containing sine tone audio track
+ */
+function createOscillatorStream(ac, frequency) {
+  var osc = ac.createOscillator();
+  osc.frequency.value = frequency;
+
+  var oscDest = ac.createMediaStreamDestination();
+  osc.connect(oscDest);
+  osc.start();
+  return oscDest.stream;
+}
+
+/**
+ * Create the necessary HTML elements for head and body as used by Mochitests
+ *
+ * @param {object} meta
+ *        Meta information of the test
+ * @param {string} meta.title
+ *        Description of the test
+ * @param {string} [meta.bug]
+ *        Bug the test was created for
+ * @param {boolean} [meta.visible=false]
+ *        Visibility of the media elements
+ */
+function realCreateHTML(meta) {
+  var test = document.getElementById("test");
+
+  // Create the head content
+  var elem = document.createElement("meta");
+  elem.setAttribute("charset", "utf-8");
+  document.head.appendChild(elem);
+
+  var title = document.createElement("title");
+  title.textContent = meta.title;
+  document.head.appendChild(title);
+
+  // Create the body content
+  var anchor = document.createElement("a");
+  anchor.textContent = meta.title;
+  if (meta.bug) {
+    anchor.setAttribute(
+      "href",
+      "https://bugzilla.mozilla.org/show_bug.cgi?id=" + meta.bug
+    );
+  } else {
+    anchor.setAttribute("target", "_blank");
+  }
+
+  document.body.insertBefore(anchor, test);
+
+  var display = document.createElement("p");
+  display.setAttribute("id", "display");
+  document.body.insertBefore(display, test);
+
+  var content = document.createElement("div");
+  content.setAttribute("id", "content");
+  content.style.display = meta.visible ? "block" : "none";
+  document.body.appendChild(content);
+}
+
+/**
+ * Creates an element of the given type, assigns the given id, sets the controls
+ * and autoplay attributes and adds it to the content node.
+ *
+ * @param {string} type
+ *        Defining if we should create an "audio" or "video" element
+ * @param {string} id
+ *        A string to use as the element id.
+ */
+function createMediaElement(type, id) {
+  const element = document.createElement(type);
+  element.setAttribute("id", id);
+  element.setAttribute("height", 100);
+  element.setAttribute("width", 150);
+  element.setAttribute("controls", "controls");
+  element.setAttribute("autoplay", "autoplay");
+  element.setAttribute("muted", "muted");
+  element.muted = true;
+  document.getElementById("content").appendChild(element);
+
+  return element;
+}
+
+/**
+ * Returns an existing element for the given track with the given idPrefix,
+ * as it was added by createMediaElementForTrack().
+ *
+ * @param {MediaStreamTrack} track
+ *        Track used as the element's source.
+ * @param {string} idPrefix
+ *        A string to use as the element id. The track id will also be appended.
+ */
+function getMediaElementForTrack(track, idPrefix) {
+  return document.getElementById(idPrefix + "_" + track.id);
+}
+
+/**
+ * Create a media element with a track as source and attach it to the content
+ * node.
+ *
+ * @param {MediaStreamTrack} track
+ *        Track for use as source.
+ * @param {string} idPrefix
+ *        A string to use as the element id. The track id will also be appended.
+ * @return {HTMLMediaElement} The created HTML media element
+ */
+function createMediaElementForTrack(track, idPrefix) {
+  const id = idPrefix + "_" + track.id;
+  const element = createMediaElement(track.kind, id);
+  element.srcObject = new MediaStream([track]);
+
+  return element;
+}
+
+/**
+ * Wrapper function for mediaDevices.getUserMedia used by some tests. Whether
+ * to use fake devices or not is now determined in pref further below instead.
+ *
+ * @param {Dictionary} constraints
+ *        The constraints for this mozGetUserMedia callback
+ */
+function getUserMedia(constraints) {
+  // Tests may have changed the values of prefs, so recheck
+  updateConfigFromFakeAndLoopbackPrefs();
+  if (
+    !WANT_FAKE_AUDIO &&
+    !constraints.fake &&
+    constraints.audio &&
+    !DISABLE_LOOPBACK_TONE
+  ) {
+    // Loopback device is configured, start the default loopback tone
+    if (!DefaultLoopbackTone) {
+      TEST_AUDIO_FREQ = 440;
+      DefaultLoopbackTone = new LoopbackTone(
+        new AudioContext(),
+        TEST_AUDIO_FREQ
+      );
+      DefaultLoopbackTone.start();
+    }
+    // Disable input processing mode when it's not explicity enabled.
+    // This is to avoid distortion of the loopback tone
+    constraints.audio = Object.assign(
+      {},
+      { autoGainControl: false },
+      { echoCancellation: false },
+      { noiseSuppression: false },
+      constraints.audio
+    );
+  } else {
+    // Fake device configured, ensure our test freq is correct.
+    TEST_AUDIO_FREQ = 1000;
+  }
+  info("Call getUserMedia for " + JSON.stringify(constraints));
+  return navigator.mediaDevices
+    .getUserMedia(constraints)
+    .then(stream => (checkMediaStreamTracks(constraints, stream), stream));
+}
+
+// These are the promises we use to track that the prerequisites for the test
+// are in place before running it.
+var setTestOptions;
+var testConfigured = new Promise(r => (setTestOptions = r));
+
+function pushPrefs(...p) {
+  return SpecialPowers.pushPrefEnv({ set: p });
+}
+
+async function withPrefs(prefs, func) {
+  await SpecialPowers.pushPrefEnv({ set: prefs });
+  try {
+    return await func();
+  } finally {
+    await SpecialPowers.popPrefEnv();
+  }
+}
+
+function setupEnvironment() {
+  var defaultMochitestPrefs = {
+    set: [
+      ["media.peerconnection.enabled", true],
+      ["media.peerconnection.identity.enabled", true],
+      ["media.peerconnection.identity.timeout", 120000],
+      ["media.peerconnection.ice.stun_client_maximum_transmits", 14],
+      ["media.peerconnection.ice.trickle_grace_period", 30000],
+      ["media.navigator.permission.disabled", true],
+      // If either fake audio or video is desired we enable fake streams.
+      // If loopback devices are set they will be chosen instead of fakes in gecko.
+      ["media.navigator.streams.fake", WANT_FAKE_AUDIO || WANT_FAKE_VIDEO],
+      ["media.getusermedia.audiocapture.enabled", true],
+      ["media.getusermedia.screensharing.enabled", true],
+      ["media.getusermedia.window.focus_source.enabled", false],
+      ["media.recorder.audio_node.enabled", true],
+      ["media.peerconnection.ice.obfuscate_host_addresses", false],
+      ["media.peerconnection.nat_simulator.filtering_type", ""],
+      ["media.peerconnection.nat_simulator.mapping_type", ""],
+      ["media.peerconnection.nat_simulator.block_tcp", false],
+      ["media.peerconnection.nat_simulator.block_udp", false],
+      ["media.peerconnection.nat_simulator.redirect_address", ""],
+      ["media.peerconnection.nat_simulator.redirect_targets", ""],
+    ],
+  };
+
+  if (navigator.userAgent.includes("Android")) {
+    defaultMochitestPrefs.set.push(
+      ["media.navigator.video.default_width", 320],
+      ["media.navigator.video.default_height", 240],
+      ["media.navigator.video.max_fr", 10],
+      ["media.autoplay.default", Ci.nsIAutoplay.ALLOWED]
+    );
+  }
+
+  // Platform codec prefs should be matched because fake H.264 GMP codec doesn't
+  // produce/consume real bitstreams. [TODO] remove after bug 1509012 is fixed.
+  const platformEncoderEnabled = SpecialPowers.getBoolPref(
+    "media.webrtc.platformencoder"
+  );
+  defaultMochitestPrefs.set.push([
+    "media.navigator.mediadatadecoder_h264_enabled",
+    platformEncoderEnabled,
+  ]);
+
+  // Running as a Mochitest.
+  SimpleTest.requestFlakyTimeout("WebRTC inherently depends on timeouts");
+  window.finish = () => SimpleTest.finish();
+  SpecialPowers.pushPrefEnv(defaultMochitestPrefs, setTestOptions);
+
+  // We don't care about waiting for this to complete, we just want to ensure
+  // that we don't build up a huge backlog of GC work.
+  SpecialPowers.exactGC();
+}
+
+// [TODO] remove after bug 1509012 is fixed.
+async function matchPlatformH264CodecPrefs() {
+  const hasHW264 =
+    SpecialPowers.getBoolPref("media.webrtc.platformencoder") &&
+    !SpecialPowers.getBoolPref("media.webrtc.platformencoder.sw_only") &&
+    (navigator.userAgent.includes("Android") ||
+      navigator.userAgent.includes("Mac OS X"));
+
+  await pushPrefs(
+    ["media.webrtc.platformencoder", hasHW264],
+    ["media.navigator.mediadatadecoder_h264_enabled", hasHW264]
+  );
+}
+
+async function runTestWhenReady(testFunc) {
+  setupEnvironment();
+  const options = await testConfigured;
+  try {
+    await testFunc(options);
+  } catch (e) {
+    ok(
+      false,
+      `Error executing test: ${e}
+${e.stack ? e.stack : ""}`
+    );
+  } finally {
+    SimpleTest.finish();
+  }
+}
+
+/**
+ * Checks that the media stream tracks have the expected amount of tracks
+ * with the correct attributes based on the type and constraints given.
+ *
+ * @param {Object} constraints specifies whether the stream should have
+ *                             audio, video, or both
+ * @param {String} type the type of media stream tracks being checked
+ * @param {sequence<MediaStreamTrack>} mediaStreamTracks the media stream
+ *                                     tracks being checked
+ */
+function checkMediaStreamTracksByType(constraints, type, mediaStreamTracks) {
+  if (constraints[type]) {
+    is(mediaStreamTracks.length, 1, "One " + type + " track shall be present");
+
+    if (mediaStreamTracks.length) {
+      is(mediaStreamTracks[0].kind, type, "Track kind should be " + type);
+      ok(mediaStreamTracks[0].id, "Track id should be defined");
+      ok(!mediaStreamTracks[0].muted, "Track should not be muted");
+    }
+  } else {
+    is(mediaStreamTracks.length, 0, "No " + type + " tracks shall be present");
+  }
+}
+
+/**
+ * Check that the given media stream contains the expected media stream
+ * tracks given the associated audio & video constraints provided.
+ *
+ * @param {Object} constraints specifies whether the stream should have
+ *                             audio, video, or both
+ * @param {MediaStream} mediaStream the media stream being checked
+ */
+function checkMediaStreamTracks(constraints, mediaStream) {
+  checkMediaStreamTracksByType(
+    constraints,
+    "audio",
+    mediaStream.getAudioTracks()
+  );
+  checkMediaStreamTracksByType(
+    constraints,
+    "video",
+    mediaStream.getVideoTracks()
+  );
+}
+
+/**
+ * Check that a media stream contains exactly a set of media stream tracks.
+ *
+ * @param {MediaStream} mediaStream the media stream being checked
+ * @param {Array} tracks the tracks that should exist in mediaStream
+ * @param {String} [message] an optional message to pass to asserts
+ */
+function checkMediaStreamContains(mediaStream, tracks, message) {
+  message = message ? message + ": " : "";
+  tracks.forEach(t =>
+    ok(
+      mediaStream.getTrackById(t.id),
+      message + "MediaStream " + mediaStream.id + " contains track " + t.id
+    )
+  );
+  is(
+    mediaStream.getTracks().length,
+    tracks.length,
+    message + "MediaStream " + mediaStream.id + " contains no extra tracks"
+  );
+}
+
+function checkMediaStreamCloneAgainstOriginal(clone, original) {
+  isnot(clone.id.length, 0, "Stream clone should have an id string");
+  isnot(clone, original, "Stream clone should be different from the original");
+  isnot(
+    clone.id,
+    original.id,
+    "Stream clone's id should be different from the original's"
+  );
+  is(
+    clone.getAudioTracks().length,
+    original.getAudioTracks().length,
+    "All audio tracks should get cloned"
+  );
+  is(
+    clone.getVideoTracks().length,
+    original.getVideoTracks().length,
+    "All video tracks should get cloned"
+  );
+  is(clone.active, original.active, "Active state should be preserved");
+  original
+    .getTracks()
+    .forEach(t =>
+      ok(!clone.getTrackById(t.id), "The clone's tracks should be originals")
+    );
+}
+
+function checkMediaStreamTrackCloneAgainstOriginal(clone, original) {
+  isnot(clone.id.length, 0, "Track clone should have an id string");
+  isnot(clone, original, "Track clone should be different from the original");
+  isnot(
+    clone.id,
+    original.id,
+    "Track clone's id should be different from the original's"
+  );
+  is(
+    clone.kind,
+    original.kind,
+    "Track clone's kind should be same as the original's"
+  );
+  is(
+    clone.enabled,
+    original.enabled,
+    "Track clone's kind should be same as the original's"
+  );
+  is(
+    clone.readyState,
+    original.readyState,
+    "Track clone's readyState should be same as the original's"
+  );
+  is(
+    clone.muted,
+    original.muted,
+    "Track clone's muted state should be same as the original's"
+  );
+}
+
+/*** Utility methods */
+
+/** The dreadful setTimeout, use sparingly */
+function wait(time, message) {
+  return new Promise(r => setTimeout(() => r(message), time));
+}
+
+/** The even more dreadful setInterval, use even more sparingly */
+function waitUntil(func, time) {
+  return new Promise(resolve => {
+    var interval = setInterval(() => {
+      if (func()) {
+        clearInterval(interval);
+        resolve();
+      }
+    }, time || 200);
+  });
+}
+
+/** Time out while waiting for a promise to get resolved or rejected. */
+var timeout = (promise, time, msg) =>
+  Promise.race([
+    promise,
+    wait(time).then(() => Promise.reject(new Error(msg))),
+  ]);
+
+/** Adds a |finally| function to a promise whose argument is invoked whether the
+ * promise is resolved or rejected, and that does not interfere with chaining.*/
+var addFinallyToPromise = promise => {
+  promise.finally = func => {
+    return promise.then(
+      result => {
+        func();
+        return Promise.resolve(result);
+      },
+      error => {
+        func();
+        return Promise.reject(error);
+      }
+    );
+  };
+  return promise;
+};
+
+/** Use event listener to call passed-in function on fire until it returns true */
+var listenUntil = (target, eventName, onFire) => {
+  return new Promise(resolve =>
+    target.addEventListener(eventName, function callback(event) {
+      var result = onFire(event);
+      if (result) {
+        target.removeEventListener(eventName, callback);
+        resolve(result);
+      }
+    })
+  );
+};
+
+/* Test that a function throws the right error */
+function mustThrowWith(msg, reason, f) {
+  try {
+    f();
+    ok(false, msg + " must throw");
+  } catch (e) {
+    is(e.name, reason, msg + " must throw: " + e.message);
+  }
+}
+
+/* Get a dummy audio track */
+function getSilentTrack() {
+  let ctx = new AudioContext(),
+    oscillator = ctx.createOscillator();
+  let dst = oscillator.connect(ctx.createMediaStreamDestination());
+  oscillator.start();
+  return Object.assign(dst.stream.getAudioTracks()[0], { enabled: false });
+}
+
+function getBlackTrack({ width = 640, height = 480 } = {}) {
+  let canvas = Object.assign(document.createElement("canvas"), {
+    width,
+    height,
+  });
+  canvas.getContext("2d").fillRect(0, 0, width, height);
+  let stream = canvas.captureStream();
+  return Object.assign(stream.getVideoTracks()[0], { enabled: false });
+}
+
+/*** Test control flow methods */
+
+/**
+ * Generates a callback function fired only under unexpected circumstances
+ * while running the tests. The generated function kills off the test as well
+ * gracefully.
+ *
+ * @param {String} [message]
+ *        An optional message to show if no object gets passed into the
+ *        generated callback method.
+ */
+function generateErrorCallback(message) {
+  var stack = new Error().stack.split("\n");
+  stack.shift(); // Don't include this instantiation frame
+
+  /**
+   * @param {object} aObj
+   *        The object fired back from the callback
+   */
+  return aObj => {
+    if (aObj) {
+      if (aObj.name && aObj.message) {
+        ok(
+          false,
+          "Unexpected callback for '" +
+            aObj.name +
+            "' with message = '" +
+            aObj.message +
+            "' at " +
+            JSON.stringify(stack)
+        );
+      } else {
+        ok(
+          false,
+          "Unexpected callback with = '" +
+            aObj +
+            "' at: " +
+            JSON.stringify(stack)
+        );
+      }
+    } else {
+      ok(
+        false,
+        "Unexpected callback with message = '" +
+          message +
+          "' at: " +
+          JSON.stringify(stack)
+      );
+    }
+    throw new Error("Unexpected callback");
+  };
+}
+
+var unexpectedEventArrived;
+var rejectOnUnexpectedEvent = new Promise((x, reject) => {
+  unexpectedEventArrived = reject;
+});
+
+/**
+ * Generates a callback function fired only for unexpected events happening.
+ *
+ * @param {String} description
+          Description of the object for which the event has been fired
+ * @param {String} eventName
+          Name of the unexpected event
+ */
+function unexpectedEvent(message, eventName) {
+  var stack = new Error().stack.split("\n");
+  stack.shift(); // Don't include this instantiation frame
+
+  return e => {
+    var details =
+      "Unexpected event '" +
+      eventName +
+      "' fired with message = '" +
+      message +
+      "' at: " +
+      JSON.stringify(stack);
+    ok(false, details);
+    unexpectedEventArrived(new Error(details));
+  };
+}
+
+/**
+ * Implements the one-shot event pattern used throughout.  Each of the 'onxxx'
+ * attributes on the wrappers can be set with a custom handler.  Prior to the
+ * handler being set, if the event fires, it causes the test execution to halt.
+ * That handler is used exactly once, after which the original, error-generating
+ * handler is re-installed.  Thus, each event handler is used at most once.
+ *
+ * @param {object} wrapper
+ *        The wrapper on which the psuedo-handler is installed
+ * @param {object} obj
+ *        The real source of events
+ * @param {string} event
+ *        The name of the event
+ */
+function createOneShotEventWrapper(wrapper, obj, event) {
+  var onx = "on" + event;
+  var unexpected = unexpectedEvent(wrapper, event);
+  wrapper[onx] = unexpected;
+  obj[onx] = e => {
+    info(wrapper + ': "on' + event + '" event fired');
+    e.wrapper = wrapper;
+    wrapper[onx](e);
+    wrapper[onx] = unexpected;
+  };
+}
+
+/**
+ * Returns a promise that resolves when `target` has raised an event with the
+ * given name the given number of times. Cancel the returned promise by passing
+ * in a `cancel` promise and resolving it.
+ *
+ * @param {object} target
+ *        The target on which the event should occur.
+ * @param {string} name
+ *        The name of the event that should occur.
+ * @param {integer} count
+ *        Optional number of times the event should be raised before resolving.
+ * @param {promise} cancel
+ *        Optional promise that on resolving rejects the returned promise,
+ *        so we can avoid logging results after a test has finished.
+ * @returns {promise} A promise that resolves to the last of the seen events.
+ */
+function haveEvents(target, name, count, cancel) {
+  var listener;
+  var counter = count || 1;
+  return Promise.race([
+    (cancel || new Promise(() => {})).then(e => Promise.reject(e)),
+    new Promise(resolve =>
+      target.addEventListener(
+        name,
+        (listener = e => --counter < 1 && resolve(e))
+      )
+    ),
+  ]).then(e => (target.removeEventListener(name, listener), e));
+}
+
+/**
+ * Returns a promise that resolves when `target` has raised an event with the
+ * given name. Cancel the returned promise by passing in a `cancel` promise and
+ * resolving it.
+ *
+ * @param {object} target
+ *        The target on which the event should occur.
+ * @param {string} name
+ *        The name of the event that should occur.
+ * @param {promise} cancel
+ *        Optional promise that on resolving rejects the returned promise,
+ *        so we can avoid logging results after a test has finished.
+ * @returns {promise} A promise that resolves to the seen event.
+ */
+function haveEvent(target, name, cancel) {
+  return haveEvents(target, name, 1, cancel);
+}
+
+/**
+ * Returns a promise that resolves if the target has not seen the given event
+ * after one crank (or until the given timeoutPromise resolves) of the event
+ * loop.
+ *
+ * @param {object} target
+ *        The target on which the event should not occur.
+ * @param {string} name
+ *        The name of the event that should not occur.
+ * @param {promise} timeoutPromise
+ *        Optional promise defining how long we should wait before resolving.
+ * @returns {promise} A promise that is rejected if we see the given event, or
+ *                    resolves after a timeout otherwise.
+ */
+function haveNoEvent(target, name, timeoutPromise) {
+  return haveEvent(target, name, timeoutPromise || wait(0)).then(
+    () => Promise.reject(new Error("Too many " + name + " events")),
+    () => {}
+  );
+}
+
+/**
+ * Returns a promise that resolves after the target has seen the given number
+ * of events but no such event in a following crank of the event loop.
+ *
+ * @param {object} target
+ *        The target on which the events should occur.
+ * @param {string} name
+ *        The name of the event that should occur.
+ * @param {integer} count
+ *        Optional number of times the event should be raised before resolving.
+ * @param {promise} cancel
+ *        Optional promise that on resolving rejects the returned promise,
+ *        so we can avoid logging results after a test has finished.
+ * @returns {promise} A promise that resolves to the last of the seen events.
+ */
+function haveEventsButNoMore(target, name, count, cancel) {
+  return haveEvents(target, name, count, cancel).then(e =>
+    haveNoEvent(target, name).then(() => e)
+  );
+}
+
+/*
+ * Resolves the returned promise with an object with usage and reportCount
+ * properties.  `usage` is in the same units as reported by the reporter for
+ * `path`.
+ */
+const collectMemoryUsage = async path => {
+  const MemoryReporterManager = Cc[
+    "@mozilla.org/memory-reporter-manager;1"
+  ].getService(Ci.nsIMemoryReporterManager);
+
+  let usage = 0;
+  let reportCount = 0;
+  await new Promise(resolve =>
+    MemoryReporterManager.getReports(
+      (aProcess, aPath, aKind, aUnits, aAmount, aDesc) => {
+        if (aPath != path) {
+          return;
+        }
+        ++reportCount;
+        usage += aAmount;
+      },
+      null,
+      resolve,
+      null,
+      /* anonymized = */ false
+    )
+  );
+  return { usage, reportCount };
+};
+
+// Some DNS helper functions
+const dnsLookup = async hostname => {
+  // Convenience API for various networking related stuff. _Almost_ convenient
+  // enough.
+  const neckoDashboard = SpecialPowers.Cc[
+    "@mozilla.org/network/dashboard;1"
+  ].getService(Ci.nsIDashboard);
+
+  const results = await new Promise(r => {
+    neckoDashboard.requestDNSLookup(hostname, results => {
+      r(SpecialPowers.wrap(results));
+    });
+  });
+
+  // |address| is an array-like dictionary (ie; keys are all integers).
+  // We convert to an array to make it less unwieldy.
+  const addresses = [...results.address];
+  info(`DNS results for ${hostname}: ${JSON.stringify(addresses)}`);
+  return addresses;
+};
+
+const dnsLookupV4 = async hostname => {
+  const addresses = await dnsLookup(hostname);
+  return addresses.filter(address => !address.includes(":"));
+};
+
+const dnsLookupV6 = async hostname => {
+  const addresses = await dnsLookup(hostname);
+  return addresses.filter(address => address.includes(":"));
+};
+
+const getTurnHostname = turnUrl => {
+  const urlNoParams = turnUrl.split("?")[0];
+  // Strip off scheme
+  const hostAndMaybePort = urlNoParams.split(":", 2)[1];
+  if (hostAndMaybePort[0] == "[") {
+    // IPV6 literal, strip out '[', and split at closing ']'
+    return hostAndMaybePort.substring(1).split("]")[0];
+  }
+  return hostAndMaybePort.split(":")[0];
+};
+
+// Yo dawg I heard you like yo dawg I heard you like Proxies
+// Example: let value = await GleanTest.category.metric.testGetValue();
+// For labeled metrics:
+//    let value = await GleanTest.category.metric["label"].testGetValue();
+// Please don't try to use the string "testGetValue" as a label.
+const GleanTest = new Proxy(
+  {},
+  {
+    get(target, categoryName, receiver) {
+      return new Proxy(
+        {},
+        {
+          get(target, metricName, receiver) {
+            return new Proxy(
+              {
+                async testGetValue() {
+                  return SpecialPowers.spawnChrome(
+                    [categoryName, metricName],
+                    async (categoryName, metricName) => {
+                      await Services.fog.testFlushAllChildren();
+                      const window = this.browsingContext.topChromeWindow;
+                      return window.Glean[categoryName][
+                        metricName
+                      ].testGetValue();
+                    }
+                  );
+                },
+              },
+              {
+                get(target, prop, receiver) {
+                  // The only prop that will be there is testGetValue, but we
+                  // might add more later.
+                  if (prop in target) {
+                    return target[prop];
+                  }
+
+                  // |prop| must be a label?
+                  const label = prop;
+                  return {
+                    async testGetValue() {
+                      return SpecialPowers.spawnChrome(
+                        [categoryName, metricName, label],
+                        async (categoryName, metricName, label) => {
+                          await Services.fog.testFlushAllChildren();
+                          const window = this.browsingContext.topChromeWindow;
+                          return window.Glean[categoryName][metricName][
+                            label
+                          ].testGetValue();
+                        }
+                      );
+                    },
+                  };
+                },
+              }
+            );
+          },
+        }
+      );
+    },
+  }
+);
+
+/**
+ * This class executes a series of functions in a continuous sequence.
+ * Promise-bearing functions are executed after the previous promise completes.
+ *
+ * @constructor
+ * @param {object} framework
+ *        A back reference to the framework which makes use of the class. It is
+ *        passed to each command callback.
+ * @param {function[]} commandList
+ *        Commands to set during initialization
+ */
+function CommandChain(framework, commandList) {
+  this._framework = framework;
+  this.commands = commandList || [];
+}
+
+CommandChain.prototype = {
+  /**
+   * Start the command chain.  This returns a promise that always resolves
+   * cleanly (this catches errors and fails the test case).
+   */
+  execute() {
+    return this.commands
+      .reduce((prev, next, i) => {
+        if (typeof next !== "function" || !next.name) {
+          throw new Error("registered non-function" + next);
+        }
+
+        return prev.then(() => {
+          info("Run step " + (i + 1) + ": " + next.name);
+          return Promise.race([next(this._framework), rejectOnUnexpectedEvent]);
+        });
+      }, Promise.resolve())
+      .catch(e =>
+        ok(
+          false,
+          "Error in test execution: " +
+            e +
+            (typeof e.stack === "string"
+              ? " " + e.stack.split("\n").join(" ... ")
+              : "")
+        )
+      );
+  },
+
+  /**
+   * Add new commands to the end of the chain
+   */
+  append(commands) {
+    this.commands = this.commands.concat(commands);
+  },
+
+  /**
+   * Returns the index of the specified command in the chain.
+   * @param {occurrence} Optional param specifying which occurrence to match,
+   * with 0 representing the first occurrence.
+   */
+  indexOf(functionOrName, occurrence) {
+    occurrence = occurrence || 0;
+    return this.commands.findIndex(func => {
+      if (typeof functionOrName === "string") {
+        if (func.name !== functionOrName) {
+          return false;
+        }
+      } else if (func !== functionOrName) {
+        return false;
+      }
+      if (occurrence) {
+        --occurrence;
+        return false;
+      }
+      return true;
+    });
+  },
+
+  mustHaveIndexOf(functionOrName, occurrence) {
+    var index = this.indexOf(functionOrName, occurrence);
+    if (index == -1) {
+      throw new Error("Unknown test: " + functionOrName);
+    }
+    return index;
+  },
+
+  /**
+   * Inserts the new commands after the specified command.
+   */
+  insertAfter(functionOrName, commands, all, occurrence) {
+    this._insertHelper(functionOrName, commands, 1, all, occurrence);
+  },
+
+  /**
+   * Inserts the new commands after every occurrence of the specified command
+   */
+  insertAfterEach(functionOrName, commands) {
+    this._insertHelper(functionOrName, commands, 1, true);
+  },
+
+  /**
+   * Inserts the new commands before the specified command.
+   */
+  insertBefore(functionOrName, commands, all, occurrence) {
+    this._insertHelper(functionOrName, commands, 0, all, occurrence);
+  },
+
+  _insertHelper(functionOrName, commands, delta, all, occurrence) {
+    occurrence = occurrence || 0;
+    for (
+      var index = this.mustHaveIndexOf(functionOrName, occurrence);
+      index !== -1;
+      index = this.indexOf(functionOrName, ++occurrence)
+    ) {
+      this.commands = [].concat(
+        this.commands.slice(0, index + delta),
+        commands,
+        this.commands.slice(index + delta)
+      );
+      if (!all) {
+        break;
+      }
+    }
+  },
+
+  /**
+   * Removes the specified command, returns what was removed.
+   */
+  remove(functionOrName, occurrence) {
+    return this.commands.splice(
+      this.mustHaveIndexOf(functionOrName, occurrence),
+      1
+    );
+  },
+
+  /**
+   * Removes all commands after the specified one, returns what was removed.
+   */
+  removeAfter(functionOrName, occurrence) {
+    return this.commands.splice(
+      this.mustHaveIndexOf(functionOrName, occurrence) + 1
+    );
+  },
+
+  /**
+   * Removes all commands before the specified one, returns what was removed.
+   */
+  removeBefore(functionOrName, occurrence) {
+    return this.commands.splice(
+      0,
+      this.mustHaveIndexOf(functionOrName, occurrence)
+    );
+  },
+
+  /**
+   * Replaces a single command, returns what was removed.
+   */
+  replace(functionOrName, commands) {
+    this.insertBefore(functionOrName, commands);
+    return this.remove(functionOrName);
+  },
+
+  /**
+   * Replaces all commands after the specified one, returns what was removed.
+   */
+  replaceAfter(functionOrName, commands, occurrence) {
+    var oldCommands = this.removeAfter(functionOrName, occurrence);
+    this.append(commands);
+    return oldCommands;
+  },
+
+  /**
+   * Replaces all commands before the specified one, returns what was removed.
+   */
+  replaceBefore(functionOrName, commands) {
+    var oldCommands = this.removeBefore(functionOrName);
+    this.insertBefore(functionOrName, commands);
+    return oldCommands;
+  },
+
+  /**
+   * Remove all commands whose name match the specified regex.
+   */
+  filterOut(id_match) {
+    this.commands = this.commands.filter(c => !id_match.test(c.name));
+  },
+};
+
+function AudioStreamHelper() {
+  this._context = new AudioContext();
+}
+
+AudioStreamHelper.prototype = {
+  checkAudio(stream, analyser, fun) {
+    /*
+    analyser.enableDebugCanvas();
+    return analyser.waitForAnalysisSuccess(fun)
+      .then(() => analyser.disableDebugCanvas());
+    */
+    return analyser.waitForAnalysisSuccess(fun);
+  },
+
+  checkAudioFlowing(stream) {
+    var analyser = new AudioStreamAnalyser(this._context, stream);
+    var freq = analyser.binIndexForFrequency(TEST_AUDIO_FREQ);
+    return this.checkAudio(stream, analyser, array => array[freq] > 200);
+  },
+
+  checkAudioNotFlowing(stream) {
+    var analyser = new AudioStreamAnalyser(this._context, stream);
+    var freq = analyser.binIndexForFrequency(TEST_AUDIO_FREQ);
+    return this.checkAudio(stream, analyser, array => array[freq] < 50);
+  },
+};
+
+class VideoFrameEmitter {
+  constructor(color1, color2, width, height) {
+    if (!width) {
+      width = 50;
+    }
+    if (!height) {
+      height = width;
+    }
+    this._helper = new CaptureStreamTestHelper2D(width, height);
+    this._canvas = this._helper.createAndAppendElement(
+      "canvas",
+      "source_canvas"
+    );
+    this._canvas.width = width;
+    this._canvas.height = height;
+    this._color1 = color1 ? color1 : this._helper.green;
+    this._color2 = color2 ? color2 : this._helper.red;
+    // Make sure this is initted
+    this._helper.drawColor(this._canvas, this._color1);
+    this._stream = this._canvas.captureStream();
+    this._started = false;
+  }
+
+  stream() {
+    return this._stream;
+  }
+
+  helper() {
+    return this._helper;
+  }
+
+  colors(color1, color2) {
+    this._color1 = color1 ? color1 : this._helper.green;
+    this._color2 = color2 ? color2 : this._helper.red;
+    try {
+      this._helper.drawColor(this._canvas, this._color1);
+    } catch (e) {
+      // ignore; stream might have shut down
+    }
+  }
+
+  size(width, height) {
+    this._canvas.width = width;
+    this._canvas.height = height;
+  }
+
+  start() {
+    if (this._started) {
+      info("*** emitter already started");
+      return;
+    }
+
+    let i = 0;
+    this._started = true;
+    this._intervalId = setInterval(() => {
+      try {
+        this._helper.drawColor(this._canvas, i ? this._color1 : this._color2);
+        i = 1 - i;
+      } catch (e) {
+        // ignore; stream might have shut down, and we don't bother clearing
+        // the setInterval.
+      }
+    }, 500);
+  }
+
+  stop() {
+    if (this._started) {
+      clearInterval(this._intervalId);
+      this._started = false;
+    }
+  }
+}
+
+class VideoStreamHelper {
+  constructor() {
+    this._helper = new CaptureStreamTestHelper2D(50, 50);
+  }
+
+  async checkHasFrame(video, { offsetX, offsetY, threshold } = {}) {
+    const h = this._helper;
+    await h.waitForPixel(
+      video,
+      px => {
+        let result = h.isOpaquePixelNot(px, h.black, threshold);
+        info(
+          "Checking that we have a frame, got [" +
+            Array.from(px) +
+            "]. Ref=[" +
+            Array.from(h.black.data) +
+            "]. Threshold=" +
+            threshold +
+            ". Pass=" +
+            result
+        );
+        return result;
+      },
+      { offsetX, offsetY }
+    );
+  }
+
+  async checkVideoPlaying(
+    video,
+    { offsetX = 10, offsetY = 10, threshold = 16 } = {}
+  ) {
+    const h = this._helper;
+    await this.checkHasFrame(video, { offsetX, offsetY, threshold });
+    let startPixel = {
+      data: h.getPixel(video, offsetX, offsetY),
+      name: "startcolor",
+    };
+    await h.waitForPixel(
+      video,
+      px => {
+        let result = h.isPixelNot(px, startPixel, threshold);
+        info(
+          "Checking playing, [" +
+            Array.from(px) +
+            "] vs [" +
+            Array.from(startPixel.data) +
+            "]. Threshold=" +
+            threshold +
+            " Pass=" +
+            result
+        );
+        return result;
+      },
+      { offsetX, offsetY }
+    );
+  }
+
+  async checkVideoPaused(
+    video,
+    { offsetX = 10, offsetY = 10, threshold = 16, time = 5000 } = {}
+  ) {
+    const h = this._helper;
+    await this.checkHasFrame(video, { offsetX, offsetY, threshold });
+    let startPixel = {
+      data: h.getPixel(video, offsetX, offsetY),
+      name: "startcolor",
+    };
+    try {
+      await h.waitForPixel(
+        video,
+        px => {
+          let result = h.isOpaquePixelNot(px, startPixel, threshold);
+          info(
+            "Checking paused, [" +
+              Array.from(px) +
+              "] vs [" +
+              Array.from(startPixel.data) +
+              "]. Threshold=" +
+              threshold +
+              " Pass=" +
+              result
+          );
+          return result;
+        },
+        { offsetX, offsetY, cancel: wait(time, "timeout") }
+      );
+      ok(false, "Frame changed within " + time / 1000 + " seconds");
+    } catch (e) {
+      is(
+        e,
+        "timeout",
+        "Frame shouldn't change for " + time / 1000 + " seconds"
+      );
+    }
+  }
+}
+
+(function () {
+  var el = document.createElement("link");
+  el.rel = "stylesheet";
+  el.type = "text/css";
+  el.href = "/tests/SimpleTest/test.css";
+  document.head.appendChild(el);
+})();
diff --git a/dom/media/webrtc/tests/mochitests/helpers_from_wpt/sdp.js b/dom/media/webrtc/tests/mochitests/helpers_from_wpt/sdp.js
new file mode 100644
index 0000000000..6460f64a44
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/helpers_from_wpt/sdp.js
@@ -0,0 +1,889 @@
+/* eslint-env node */
+"use strict";
+
+// SDP helpers.
+var SDPUtils = {};
+
+// Generate an alphanumeric identifier for cname or mids.
+// TODO: use UUIDs instead? https://gist.github.com/jed/982883
+SDPUtils.generateIdentifier = function() {
+  return Math.random()
+    .toString(36)
+    .substr(2, 10);
+};
+
+// The RTCP CNAME used by all peerconnections from the same JS.
+SDPUtils.localCName = SDPUtils.generateIdentifier();
+
+// Splits SDP into lines, dealing with both CRLF and LF.
+SDPUtils.splitLines = function(blob) {
+  return blob
+    .trim()
+    .split("\n")
+    .map(function(line) {
+      return line.trim();
+    });
+};
+// Splits SDP into sessionpart and mediasections. Ensures CRLF.
+SDPUtils.splitSections = function(blob) {
+  var parts = blob.split("\nm=");
+  return parts.map(function(part, index) {
+    return (index > 0 ? "m=" + part : part).trim() + "\r\n";
+  });
+};
+
+// returns the session description.
+SDPUtils.getDescription = function(blob) {
+  var sections = SDPUtils.splitSections(blob);
+  return sections && sections[0];
+};
+
+// returns the individual media sections.
+SDPUtils.getMediaSections = function(blob) {
+  var sections = SDPUtils.splitSections(blob);
+  sections.shift();
+  return sections;
+};
+
+// Returns lines that start with a certain prefix.
+SDPUtils.matchPrefix = function(blob, prefix) {
+  return SDPUtils.splitLines(blob).filter(function(line) {
+    return line.indexOf(prefix) === 0;
+  });
+};
+
+// Parses an ICE candidate line. Sample input:
+// candidate:702786350 2 udp 41819902 8.8.8.8 60769 typ relay raddr 8.8.8.8
+// rport 55996"
+SDPUtils.parseCandidate = function(line) {
+  var parts;
+  // Parse both variants.
+  if (line.indexOf("a=candidate:") === 0) {
+    parts = line.substring(12).split(" ");
+  } else {
+    parts = line.substring(10).split(" ");
+  }
+
+  var candidate = {
+    foundation: parts[0],
+    component: parseInt(parts[1], 10),
+    protocol: parts[2].toLowerCase(),
+    priority: parseInt(parts[3], 10),
+    ip: parts[4],
+    address: parts[4], // address is an alias for ip.
+    port: parseInt(parts[5], 10),
+    // skip parts[6] == 'typ'
+    type: parts[7],
+  };
+
+  for (var i = 8; i < parts.length; i += 2) {
+    switch (parts[i]) {
+      case "raddr":
+        candidate.relatedAddress = parts[i + 1];
+        break;
+      case "rport":
+        candidate.relatedPort = parseInt(parts[i + 1], 10);
+        break;
+      case "tcptype":
+        candidate.tcpType = parts[i + 1];
+        break;
+      case "ufrag":
+        candidate.ufrag = parts[i + 1]; // for backward compability.
+        candidate.usernameFragment = parts[i + 1];
+        break;
+      default:
+        // extension handling, in particular ufrag
+        candidate[parts[i]] = parts[i + 1];
+        break;
+    }
+  }
+  return candidate;
+};
+
+// Translates a candidate object into SDP candidate attribute.
+SDPUtils.writeCandidate = function(candidate) {
+  var sdp = [];
+  sdp.push(candidate.foundation);
+  sdp.push(candidate.component);
+  sdp.push(candidate.protocol.toUpperCase());
+  sdp.push(candidate.priority);
+  sdp.push(candidate.address || candidate.ip);
+  sdp.push(candidate.port);
+
+  var type = candidate.type;
+  sdp.push("typ");
+  sdp.push(type);
+  if (type !== "host" && candidate.relatedAddress && candidate.relatedPort) {
+    sdp.push("raddr");
+    sdp.push(candidate.relatedAddress);
+    sdp.push("rport");
+    sdp.push(candidate.relatedPort);
+  }
+  if (candidate.tcpType && candidate.protocol.toLowerCase() === "tcp") {
+    sdp.push("tcptype");
+    sdp.push(candidate.tcpType);
+  }
+  if (candidate.usernameFragment || candidate.ufrag) {
+    sdp.push("ufrag");
+    sdp.push(candidate.usernameFragment || candidate.ufrag);
+  }
+  return "candidate:" + sdp.join(" ");
+};
+
+// Parses an ice-options line, returns an array of option tags.
+// a=ice-options:foo bar
+SDPUtils.parseIceOptions = function(line) {
+  return line.substr(14).split(" ");
+};
+
+// Parses an rtpmap line, returns RTCRtpCoddecParameters. Sample input:
+// a=rtpmap:111 opus/48000/2
+SDPUtils.parseRtpMap = function(line) {
+  var parts = line.substr(9).split(" ");
+  var parsed = {
+    payloadType: parseInt(parts.shift(), 10), // was: id
+  };
+
+  parts = parts[0].split("/");
+
+  parsed.name = parts[0];
+  parsed.clockRate = parseInt(parts[1], 10); // was: clockrate
+  parsed.channels = parts.length === 3 ? parseInt(parts[2], 10) : 1;
+  // legacy alias, got renamed back to channels in ORTC.
+  parsed.numChannels = parsed.channels;
+  return parsed;
+};
+
+// Generate an a=rtpmap line from RTCRtpCodecCapability or
+// RTCRtpCodecParameters.
+SDPUtils.writeRtpMap = function(codec) {
+  var pt = codec.payloadType;
+  if (codec.preferredPayloadType !== undefined) {
+    pt = codec.preferredPayloadType;
+  }
+  var channels = codec.channels || codec.numChannels || 1;
+  return (
+    "a=rtpmap:" +
+    pt +
+    " " +
+    codec.name +
+    "/" +
+    codec.clockRate +
+    (channels !== 1 ? "/" + channels : "") +
+    "\r\n"
+  );
+};
+
+// Parses an a=extmap line (headerextension from RFC 5285). Sample input:
+// a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
+// a=extmap:2/sendonly urn:ietf:params:rtp-hdrext:toffset
+SDPUtils.parseExtmap = function(line) {
+  var parts = line.substr(9).split(" ");
+  return {
+    id: parseInt(parts[0], 10),
+    direction: parts[0].indexOf("/") > 0 ? parts[0].split("/")[1] : "sendrecv",
+    uri: parts[1],
+  };
+};
+
+// Generates a=extmap line from RTCRtpHeaderExtensionParameters or
+// RTCRtpHeaderExtension.
+SDPUtils.writeExtmap = function(headerExtension) {
+  return (
+    "a=extmap:" +
+    (headerExtension.id || headerExtension.preferredId) +
+    (headerExtension.direction && headerExtension.direction !== "sendrecv"
+      ? "/" + headerExtension.direction
+      : "") +
+    " " +
+    headerExtension.uri +
+    "\r\n"
+  );
+};
+
+// Parses an ftmp line, returns dictionary. Sample input:
+// a=fmtp:96 vbr=on;cng=on
+// Also deals with vbr=on; cng=on
+SDPUtils.parseFmtp = function(line) {
+  var parsed = {};
+  var kv;
+  var parts = line.substr(line.indexOf(" ") + 1).split(";");
+  for (var j = 0; j < parts.length; j++) {
+    kv = parts[j].trim().split("=");
+    parsed[kv[0].trim()] = kv[1];
+  }
+  return parsed;
+};
+
+// Generates an a=ftmp line from RTCRtpCodecCapability or RTCRtpCodecParameters.
+SDPUtils.writeFmtp = function(codec) {
+  var line = "";
+  var pt = codec.payloadType;
+  if (codec.preferredPayloadType !== undefined) {
+    pt = codec.preferredPayloadType;
+  }
+  if (codec.parameters && Object.keys(codec.parameters).length) {
+    var params = [];
+    Object.keys(codec.parameters).forEach(function(param) {
+      if (codec.parameters[param]) {
+        params.push(param + "=" + codec.parameters[param]);
+      } else {
+        params.push(param);
+      }
+    });
+    line += "a=fmtp:" + pt + " " + params.join(";") + "\r\n";
+  }
+  return line;
+};
+
+// Parses an rtcp-fb line, returns RTCPRtcpFeedback object. Sample input:
+// a=rtcp-fb:98 nack rpsi
+SDPUtils.parseRtcpFb = function(line) {
+  var parts = line.substr(line.indexOf(" ") + 1).split(" ");
+  return {
+    type: parts.shift(),
+    parameter: parts.join(" "),
+  };
+};
+// Generate a=rtcp-fb lines from RTCRtpCodecCapability or RTCRtpCodecParameters.
+SDPUtils.writeRtcpFb = function(codec) {
+  var lines = "";
+  var pt = codec.payloadType;
+  if (codec.preferredPayloadType !== undefined) {
+    pt = codec.preferredPayloadType;
+  }
+  if (codec.rtcpFeedback && codec.rtcpFeedback.length) {
+    // FIXME: special handling for trr-int?
+    codec.rtcpFeedback.forEach(function(fb) {
+      lines +=
+        "a=rtcp-fb:" +
+        pt +
+        " " +
+        fb.type +
+        (fb.parameter && fb.parameter.length ? " " + fb.parameter : "") +
+        "\r\n";
+    });
+  }
+  return lines;
+};
+
+// Parses an RFC 5576 ssrc media attribute. Sample input:
+// a=ssrc:3735928559 cname:something
+SDPUtils.parseSsrcMedia = function(line) {
+  var sp = line.indexOf(" ");
+  var parts = {
+    ssrc: parseInt(line.substr(7, sp - 7), 10),
+  };
+  var colon = line.indexOf(":", sp);
+  if (colon > -1) {
+    parts.attribute = line.substr(sp + 1, colon - sp - 1);
+    parts.value = line.substr(colon + 1);
+  } else {
+    parts.attribute = line.substr(sp + 1);
+  }
+  return parts;
+};
+
+SDPUtils.parseSsrcGroup = function(line) {
+  var parts = line.substr(13).split(" ");
+  return {
+    semantics: parts.shift(),
+    ssrcs: parts.map(function(ssrc) {
+      return parseInt(ssrc, 10);
+    }),
+  };
+};
+
+// Extracts the MID (RFC 5888) from a media section.
+// returns the MID or undefined if no mid line was found.
+SDPUtils.getMid = function(mediaSection) {
+  var mid = SDPUtils.matchPrefix(mediaSection, "a=mid:")[0];
+  if (mid) {
+    return mid.substr(6);
+  }
+};
+
+SDPUtils.parseFingerprint = function(line) {
+  var parts = line.substr(14).split(" ");
+  return {
+    algorithm: parts[0].toLowerCase(), // algorithm is case-sensitive in Edge.
+    value: parts[1],
+  };
+};
+
+// Extracts DTLS parameters from SDP media section or sessionpart.
+// FIXME: for consistency with other functions this should only
+//   get the fingerprint line as input. See also getIceParameters.
+SDPUtils.getDtlsParameters = function(mediaSection, sessionpart) {
+  var lines = SDPUtils.matchPrefix(
+    mediaSection + sessionpart,
+    "a=fingerprint:"
+  );
+  // Note: a=setup line is ignored since we use the 'auto' role.
+  // Note2: 'algorithm' is not case sensitive except in Edge.
+  return {
+    role: "auto",
+    fingerprints: lines.map(SDPUtils.parseFingerprint),
+  };
+};
+
+// Serializes DTLS parameters to SDP.
+SDPUtils.writeDtlsParameters = function(params, setupType) {
+  var sdp = "a=setup:" + setupType + "\r\n";
+  params.fingerprints.forEach(function(fp) {
+    sdp += "a=fingerprint:" + fp.algorithm + " " + fp.value + "\r\n";
+  });
+  return sdp;
+};
+
+// Parses a=crypto lines into
+//   https://rawgit.com/aboba/edgertc/master/msortc-rs4.html#dictionary-rtcsrtpsdesparameters-members
+SDPUtils.parseCryptoLine = function(line) {
+  var parts = line.substr(9).split(" ");
+  return {
+    tag: parseInt(parts[0], 10),
+    cryptoSuite: parts[1],
+    keyParams: parts[2],
+    sessionParams: parts.slice(3),
+  };
+};
+
+SDPUtils.writeCryptoLine = function(parameters) {
+  return (
+    "a=crypto:" +
+    parameters.tag +
+    " " +
+    parameters.cryptoSuite +
+    " " +
+    (typeof parameters.keyParams === "object"
+      ? SDPUtils.writeCryptoKeyParams(parameters.keyParams)
+      : parameters.keyParams) +
+    (parameters.sessionParams ? " " + parameters.sessionParams.join(" ") : "") +
+    "\r\n"
+  );
+};
+
+// Parses the crypto key parameters into
+//   https://rawgit.com/aboba/edgertc/master/msortc-rs4.html#rtcsrtpkeyparam*
+SDPUtils.parseCryptoKeyParams = function(keyParams) {
+  if (keyParams.indexOf("inline:") !== 0) {
+    return null;
+  }
+  var parts = keyParams.substr(7).split("|");
+  return {
+    keyMethod: "inline",
+    keySalt: parts[0],
+    lifeTime: parts[1],
+    mkiValue: parts[2] ? parts[2].split(":")[0] : undefined,
+    mkiLength: parts[2] ? parts[2].split(":")[1] : undefined,
+  };
+};
+
+SDPUtils.writeCryptoKeyParams = function(keyParams) {
+  return (
+    keyParams.keyMethod +
+    ":" +
+    keyParams.keySalt +
+    (keyParams.lifeTime ? "|" + keyParams.lifeTime : "") +
+    (keyParams.mkiValue && keyParams.mkiLength
+      ? "|" + keyParams.mkiValue + ":" + keyParams.mkiLength
+      : "")
+  );
+};
+
+// Extracts all SDES paramters.
+SDPUtils.getCryptoParameters = function(mediaSection, sessionpart) {
+  var lines = SDPUtils.matchPrefix(mediaSection + sessionpart, "a=crypto:");
+  return lines.map(SDPUtils.parseCryptoLine);
+};
+
+// Parses ICE information from SDP media section or sessionpart.
+// FIXME: for consistency with other functions this should only
+//   get the ice-ufrag and ice-pwd lines as input.
+SDPUtils.getIceParameters = function(mediaSection, sessionpart) {
+  var ufrag = SDPUtils.matchPrefix(
+    mediaSection + sessionpart,
+    "a=ice-ufrag:"
+  )[0];
+  var pwd = SDPUtils.matchPrefix(mediaSection + sessionpart, "a=ice-pwd:")[0];
+  if (!(ufrag && pwd)) {
+    return null;
+  }
+  return {
+    usernameFragment: ufrag.substr(12),
+    password: pwd.substr(10),
+  };
+};
+
+// Serializes ICE parameters to SDP.
+SDPUtils.writeIceParameters = function(params) {
+  return (
+    "a=ice-ufrag:" +
+    params.usernameFragment +
+    "\r\n" +
+    "a=ice-pwd:" +
+    params.password +
+    "\r\n"
+  );
+};
+
+// Parses the SDP media section and returns RTCRtpParameters.
+SDPUtils.parseRtpParameters = function(mediaSection) {
+  var description = {
+    codecs: [],
+    headerExtensions: [],
+    fecMechanisms: [],
+    rtcp: [],
+  };
+  var lines = SDPUtils.splitLines(mediaSection);
+  var mline = lines[0].split(" ");
+  for (var i = 3; i < mline.length; i++) {
+    // find all codecs from mline[3..]
+    var pt = mline[i];
+    var rtpmapline = SDPUtils.matchPrefix(
+      mediaSection,
+      "a=rtpmap:" + pt + " "
+    )[0];
+    if (rtpmapline) {
+      var codec = SDPUtils.parseRtpMap(rtpmapline);
+      var fmtps = SDPUtils.matchPrefix(mediaSection, "a=fmtp:" + pt + " ");
+      // Only the first a=fmtp:<pt> is considered.
+      codec.parameters = fmtps.length ? SDPUtils.parseFmtp(fmtps[0]) : {};
+      codec.rtcpFeedback = SDPUtils.matchPrefix(
+        mediaSection,
+        "a=rtcp-fb:" + pt + " "
+      ).map(SDPUtils.parseRtcpFb);
+      description.codecs.push(codec);
+      // parse FEC mechanisms from rtpmap lines.
+      switch (codec.name.toUpperCase()) {
+        case "RED":
+        case "ULPFEC":
+          description.fecMechanisms.push(codec.name.toUpperCase());
+          break;
+        default:
+          // only RED and ULPFEC are recognized as FEC mechanisms.
+          break;
+      }
+    }
+  }
+  SDPUtils.matchPrefix(mediaSection, "a=extmap:").forEach(function(line) {
+    description.headerExtensions.push(SDPUtils.parseExtmap(line));
+  });
+  // FIXME: parse rtcp.
+  return description;
+};
+
+// Generates parts of the SDP media section describing the capabilities /
+// parameters.
+SDPUtils.writeRtpDescription = function(kind, caps) {
+  var sdp = "";
+
+  // Build the mline.
+  sdp += "m=" + kind + " ";
+  sdp += caps.codecs.length > 0 ? "9" : "0"; // reject if no codecs.
+  sdp += " UDP/TLS/RTP/SAVPF ";
+  sdp +=
+    caps.codecs
+      .map(function(codec) {
+        if (codec.preferredPayloadType !== undefined) {
+          return codec.preferredPayloadType;
+        }
+        return codec.payloadType;
+      })
+      .join(" ") + "\r\n";
+
+  sdp += "c=IN IP4 0.0.0.0\r\n";
+  sdp += "a=rtcp:9 IN IP4 0.0.0.0\r\n";
+
+  // Add a=rtpmap lines for each codec. Also fmtp and rtcp-fb.
+  caps.codecs.forEach(function(codec) {
+    sdp += SDPUtils.writeRtpMap(codec);
+    sdp += SDPUtils.writeFmtp(codec);
+    sdp += SDPUtils.writeRtcpFb(codec);
+  });
+  var maxptime = 0;
+  caps.codecs.forEach(function(codec) {
+    if (codec.maxptime > maxptime) {
+      maxptime = codec.maxptime;
+    }
+  });
+  if (maxptime > 0) {
+    sdp += "a=maxptime:" + maxptime + "\r\n";
+  }
+  sdp += "a=rtcp-mux\r\n";
+
+  if (caps.headerExtensions) {
+    caps.headerExtensions.forEach(function(extension) {
+      sdp += SDPUtils.writeExtmap(extension);
+    });
+  }
+  // FIXME: write fecMechanisms.
+  return sdp;
+};
+
+// Parses the SDP media section and returns an array of
+// RTCRtpEncodingParameters.
+SDPUtils.parseRtpEncodingParameters = function(mediaSection) {
+  var encodingParameters = [];
+  var description = SDPUtils.parseRtpParameters(mediaSection);
+  var hasRed = description.fecMechanisms.indexOf("RED") !== -1;
+  var hasUlpfec = description.fecMechanisms.indexOf("ULPFEC") !== -1;
+
+  // filter a=ssrc:... cname:, ignore PlanB-msid
+  var ssrcs = SDPUtils.matchPrefix(mediaSection, "a=ssrc:")
+    .map(function(line) {
+      return SDPUtils.parseSsrcMedia(line);
+    })
+    .filter(function(parts) {
+      return parts.attribute === "cname";
+    });
+  var primarySsrc = ssrcs.length > 0 && ssrcs[0].ssrc;
+  var secondarySsrc;
+
+  var flows = SDPUtils.matchPrefix(mediaSection, "a=ssrc-group:FID").map(
+    function(line) {
+      var parts = line.substr(17).split(" ");
+      return parts.map(function(part) {
+        return parseInt(part, 10);
+      });
+    }
+  );
+  if (flows.length > 0 && flows[0].length > 1 && flows[0][0] === primarySsrc) {
+    secondarySsrc = flows[0][1];
+  }
+
+  description.codecs.forEach(function(codec) {
+    if (codec.name.toUpperCase() === "RTX" && codec.parameters.apt) {
+      var encParam = {
+        ssrc: primarySsrc,
+        codecPayloadType: parseInt(codec.parameters.apt, 10),
+      };
+      if (primarySsrc && secondarySsrc) {
+        encParam.rtx = { ssrc: secondarySsrc };
+      }
+      encodingParameters.push(encParam);
+      if (hasRed) {
+        encParam = JSON.parse(JSON.stringify(encParam));
+        encParam.fec = {
+          ssrc: primarySsrc,
+          mechanism: hasUlpfec ? "red+ulpfec" : "red",
+        };
+        encodingParameters.push(encParam);
+      }
+    }
+  });
+  if (encodingParameters.length === 0 && primarySsrc) {
+    encodingParameters.push({
+      ssrc: primarySsrc,
+    });
+  }
+
+  // we support both b=AS and b=TIAS but interpret AS as TIAS.
+  var bandwidth = SDPUtils.matchPrefix(mediaSection, "b=");
+  if (bandwidth.length) {
+    if (bandwidth[0].indexOf("b=TIAS:") === 0) {
+      bandwidth = parseInt(bandwidth[0].substr(7), 10);
+    } else if (bandwidth[0].indexOf("b=AS:") === 0) {
+      // use formula from JSEP to convert b=AS to TIAS value.
+      bandwidth =
+        parseInt(bandwidth[0].substr(5), 10) * 1000 * 0.95 - 50 * 40 * 8;
+    } else {
+      bandwidth = undefined;
+    }
+    encodingParameters.forEach(function(params) {
+      params.maxBitrate = bandwidth;
+    });
+  }
+  return encodingParameters;
+};
+
+// parses http://draft.ortc.org/#rtcrtcpparameters*
+SDPUtils.parseRtcpParameters = function(mediaSection) {
+  var rtcpParameters = {};
+
+  // Gets the first SSRC. Note tha with RTX there might be multiple
+  // SSRCs.
+  var remoteSsrc = SDPUtils.matchPrefix(mediaSection, "a=ssrc:")
+    .map(function(line) {
+      return SDPUtils.parseSsrcMedia(line);
+    })
+    .filter(function(obj) {
+      return obj.attribute === "cname";
+    })[0];
+  if (remoteSsrc) {
+    rtcpParameters.cname = remoteSsrc.value;
+    rtcpParameters.ssrc = remoteSsrc.ssrc;
+  }
+
+  // Edge uses the compound attribute instead of reducedSize
+  // compound is !reducedSize
+  var rsize = SDPUtils.matchPrefix(mediaSection, "a=rtcp-rsize");
+  rtcpParameters.reducedSize = rsize.length > 0;
+  rtcpParameters.compound = rsize.length === 0;
+
+  // parses the rtcp-mux attrіbute.
+  // Note that Edge does not support unmuxed RTCP.
+  var mux = SDPUtils.matchPrefix(mediaSection, "a=rtcp-mux");
+  rtcpParameters.mux = mux.length > 0;
+
+  return rtcpParameters;
+};
+
+// parses either a=msid: or a=ssrc:... msid lines and returns
+// the id of the MediaStream and MediaStreamTrack.
+SDPUtils.parseMsid = function(mediaSection) {
+  var parts;
+  var spec = SDPUtils.matchPrefix(mediaSection, "a=msid:");
+  if (spec.length === 1) {
+    parts = spec[0].substr(7).split(" ");
+    return { stream: parts[0], track: parts[1] };
+  }
+  var planB = SDPUtils.matchPrefix(mediaSection, "a=ssrc:")
+    .map(function(line) {
+      return SDPUtils.parseSsrcMedia(line);
+    })
+    .filter(function(msidParts) {
+      return msidParts.attribute === "msid";
+    });
+  if (planB.length > 0) {
+    parts = planB[0].value.split(" ");
+    return { stream: parts[0], track: parts[1] };
+  }
+};
+
+// SCTP
+// parses draft-ietf-mmusic-sctp-sdp-26 first and falls back
+// to draft-ietf-mmusic-sctp-sdp-05
+SDPUtils.parseSctpDescription = function(mediaSection) {
+  var mline = SDPUtils.parseMLine(mediaSection);
+  var maxSizeLine = SDPUtils.matchPrefix(mediaSection, "a=max-message-size:");
+  var maxMessageSize;
+  if (maxSizeLine.length > 0) {
+    maxMessageSize = parseInt(maxSizeLine[0].substr(19), 10);
+  }
+  if (isNaN(maxMessageSize)) {
+    maxMessageSize = 65536;
+  }
+  var sctpPort = SDPUtils.matchPrefix(mediaSection, "a=sctp-port:");
+  if (sctpPort.length > 0) {
+    return {
+      port: parseInt(sctpPort[0].substr(12), 10),
+      protocol: mline.fmt,
+      maxMessageSize,
+    };
+  }
+  var sctpMapLines = SDPUtils.matchPrefix(mediaSection, "a=sctpmap:");
+  if (sctpMapLines.length > 0) {
+    var parts = SDPUtils.matchPrefix(mediaSection, "a=sctpmap:")[0]
+      .substr(10)
+      .split(" ");
+    return {
+      port: parseInt(parts[0], 10),
+      protocol: parts[1],
+      maxMessageSize,
+    };
+  }
+};
+
+// SCTP
+// outputs the draft-ietf-mmusic-sctp-sdp-26 version that all browsers
+// support by now receiving in this format, unless we originally parsed
+// as the draft-ietf-mmusic-sctp-sdp-05 format (indicated by the m-line
+// protocol of DTLS/SCTP -- without UDP/ or TCP/)
+SDPUtils.writeSctpDescription = function(media, sctp) {
+  var output = [];
+  if (media.protocol !== "DTLS/SCTP") {
+    output = [
+      "m=" + media.kind + " 9 " + media.protocol + " " + sctp.protocol + "\r\n",
+      "c=IN IP4 0.0.0.0\r\n",
+      "a=sctp-port:" + sctp.port + "\r\n",
+    ];
+  } else {
+    output = [
+      "m=" + media.kind + " 9 " + media.protocol + " " + sctp.port + "\r\n",
+      "c=IN IP4 0.0.0.0\r\n",
+      "a=sctpmap:" + sctp.port + " " + sctp.protocol + " 65535\r\n",
+    ];
+  }
+  if (sctp.maxMessageSize !== undefined) {
+    output.push("a=max-message-size:" + sctp.maxMessageSize + "\r\n");
+  }
+  return output.join("");
+};
+
+// Generate a session ID for SDP.
+// https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-20#section-5.2.1
+// recommends using a cryptographically random +ve 64-bit value
+// but right now this should be acceptable and within the right range
+SDPUtils.generateSessionId = function() {
+  return Math.floor((Math.random() * 4294967296) + 1);
+};
+
+// Write boilder plate for start of SDP
+// sessId argument is optional - if not supplied it will
+// be generated randomly
+// sessVersion is optional and defaults to 2
+// sessUser is optional and defaults to 'thisisadapterortc'
+SDPUtils.writeSessionBoilerplate = function(sessId, sessVer, sessUser) {
+  var sessionId;
+  var version = sessVer !== undefined ? sessVer : 2;
+  if (sessId) {
+    sessionId = sessId;
+  } else {
+    sessionId = SDPUtils.generateSessionId();
+  }
+  var user = sessUser || "thisisadapterortc";
+  // FIXME: sess-id should be an NTP timestamp.
+  return (
+    "v=0\r\n" +
+    "o=" +
+    user +
+    " " +
+    sessionId +
+    " " +
+    version +
+    " IN IP4 127.0.0.1\r\n" +
+    "s=-\r\n" +
+    "t=0 0\r\n"
+  );
+};
+
+SDPUtils.writeMediaSection = function(transceiver, caps, type, stream) {
+  var sdp = SDPUtils.writeRtpDescription(transceiver.kind, caps);
+
+  // Map ICE parameters (ufrag, pwd) to SDP.
+  sdp += SDPUtils.writeIceParameters(
+    transceiver.iceGatherer.getLocalParameters()
+  );
+
+  // Map DTLS parameters to SDP.
+  sdp += SDPUtils.writeDtlsParameters(
+    transceiver.dtlsTransport.getLocalParameters(),
+    type === "offer" ? "actpass" : "active"
+  );
+
+  sdp += "a=mid:" + transceiver.mid + "\r\n";
+
+  if (transceiver.direction) {
+    sdp += "a=" + transceiver.direction + "\r\n";
+  } else if (transceiver.rtpSender && transceiver.rtpReceiver) {
+    sdp += "a=sendrecv\r\n";
+  } else if (transceiver.rtpSender) {
+    sdp += "a=sendonly\r\n";
+  } else if (transceiver.rtpReceiver) {
+    sdp += "a=recvonly\r\n";
+  } else {
+    sdp += "a=inactive\r\n";
+  }
+
+  if (transceiver.rtpSender) {
+    // spec.
+    var msid =
+      "msid:" + stream.id + " " + transceiver.rtpSender.track.id + "\r\n";
+    sdp += "a=" + msid;
+
+    // for Chrome.
+    sdp += "a=ssrc:" + transceiver.sendEncodingParameters[0].ssrc + " " + msid;
+    if (transceiver.sendEncodingParameters[0].rtx) {
+      sdp +=
+        "a=ssrc:" + transceiver.sendEncodingParameters[0].rtx.ssrc + " " + msid;
+      sdp +=
+        "a=ssrc-group:FID " +
+        transceiver.sendEncodingParameters[0].ssrc +
+        " " +
+        transceiver.sendEncodingParameters[0].rtx.ssrc +
+        "\r\n";
+    }
+  }
+  // FIXME: this should be written by writeRtpDescription.
+  sdp +=
+    "a=ssrc:" +
+    transceiver.sendEncodingParameters[0].ssrc +
+    " cname:" +
+    SDPUtils.localCName +
+    "\r\n";
+  if (transceiver.rtpSender && transceiver.sendEncodingParameters[0].rtx) {
+    sdp +=
+      "a=ssrc:" +
+      transceiver.sendEncodingParameters[0].rtx.ssrc +
+      " cname:" +
+      SDPUtils.localCName +
+      "\r\n";
+  }
+  return sdp;
+};
+
+// Gets the direction from the mediaSection or the sessionpart.
+SDPUtils.getDirection = function(mediaSection, sessionpart) {
+  // Look for sendrecv, sendonly, recvonly, inactive, default to sendrecv.
+  var lines = SDPUtils.splitLines(mediaSection);
+  for (var i = 0; i < lines.length; i++) {
+    switch (lines[i]) {
+      case "a=sendrecv":
+      case "a=sendonly":
+      case "a=recvonly":
+      case "a=inactive":
+        return lines[i].substr(2);
+      default:
+      // FIXME: What should happen here?
+    }
+  }
+  if (sessionpart) {
+    return SDPUtils.getDirection(sessionpart);
+  }
+  return "sendrecv";
+};
+
+SDPUtils.getKind = function(mediaSection) {
+  var lines = SDPUtils.splitLines(mediaSection);
+  var mline = lines[0].split(" ");
+  return mline[0].substr(2);
+};
+
+SDPUtils.isRejected = function(mediaSection) {
+  return mediaSection.split(" ", 2)[1] === "0";
+};
+
+SDPUtils.parseMLine = function(mediaSection) {
+  var lines = SDPUtils.splitLines(mediaSection);
+  var parts = lines[0].substr(2).split(" ");
+  return {
+    kind: parts[0],
+    port: parseInt(parts[1], 10),
+    protocol: parts[2],
+    fmt: parts.slice(3).join(" "),
+  };
+};
+
+SDPUtils.parseOLine = function(mediaSection) {
+  var line = SDPUtils.matchPrefix(mediaSection, "o=")[0];
+  var parts = line.substr(2).split(" ");
+  return {
+    username: parts[0],
+    sessionId: parts[1],
+    sessionVersion: parseInt(parts[2], 10),
+    netType: parts[3],
+    addressType: parts[4],
+    address: parts[5],
+  };
+};
+
+// a very naive interpretation of a valid SDP.
+SDPUtils.isValidSDP = function(blob) {
+  if (typeof blob !== "string" || blob.length === 0) {
+    return false;
+  }
+  var lines = SDPUtils.splitLines(blob);
+  for (var i = 0; i < lines.length; i++) {
+    if (lines[i].length < 2 || lines[i].charAt(1) !== "=") {
+      return false;
+    }
+    // TODO: check the modifier a bit more.
+  }
+  return true;
+};
+
+// Expose public methods.
+if (typeof module === "object") {
+  module.exports = SDPUtils;
+}
diff --git a/dom/media/webrtc/tests/mochitests/iceTestUtils.js b/dom/media/webrtc/tests/mochitests/iceTestUtils.js
new file mode 100644
index 0000000000..d4d1f5c4b4
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/iceTestUtils.js
@@ -0,0 +1,302 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+"use strict";
+
+// This is mostly so test_peerConnection_gatherWithStun300.html and
+// test_peerConnection_gatherWithStun300IPv6 can share this code. I would have
+// put the ipv6 test code in the same file, but our ipv6 tester support is
+// inconsistent enough that we need to be able to track the ipv6 test
+// separately.
+
+async function findStatsRelayCandidates(pc, protocol) {
+  const stats = await pc.getStats();
+  return [...stats.values()].filter(
+    v =>
+      v.type == "local-candidate" &&
+      v.candidateType == "relay" &&
+      v.relayProtocol == protocol
+  );
+}
+
+// Trickles candidates if pcDst is set, and resolves the candidate list
+async function trickleIce(pc, pcDst) {
+  const candidates = [],
+    addCandidatePromises = [];
+  while (true) {
+    const { candidate } = await new Promise(r =>
+      pc.addEventListener("icecandidate", r, { once: true })
+    );
+    if (!candidate) {
+      break;
+    }
+    candidates.push(candidate);
+    if (pcDst) {
+      addCandidatePromises.push(pcDst.addIceCandidate(candidate));
+    }
+  }
+  await Promise.all(addCandidatePromises);
+  return candidates;
+}
+
+async function gather(pc) {
+  if (pc.signalingState == "stable") {
+    await pc.setLocalDescription(
+      await pc.createOffer({ offerToReceiveAudio: true })
+    );
+  } else if (pc.signalingState == "have-remote-offer") {
+    await pc.setLocalDescription();
+  }
+
+  return trickleIce(pc);
+}
+
+async function gatherWithTimeout(pc, timeout, context) {
+  const throwOnTimeout = async () => {
+    await wait(timeout);
+    throw new Error(
+      `Gathering did not complete within ${timeout} ms with ${context}`
+    );
+  };
+
+  return Promise.race([gather(pc), throwOnTimeout()]);
+}
+
+async function iceConnected(pc) {
+  return new Promise((resolve, reject) => {
+    pc.addEventListener("iceconnectionstatechange", () => {
+      if (["connected", "completed"].includes(pc.iceConnectionState)) {
+        resolve();
+      } else if (pc.iceConnectionState == "failed") {
+        reject(new Error(`ICE failed`));
+      }
+    });
+  });
+}
+
+// Set up trickle, but does not wait for it to complete. Can be used by itself
+// in cases where we do not expect any new candidates, but want to still set up
+// the signal handling in case new candidates _do_ show up.
+async function connectNoTrickleWait(offerer, answerer, timeout, context) {
+  return connect(offerer, answerer, timeout, context, true);
+}
+
+async function connect(
+  offerer,
+  answerer,
+  timeout,
+  context,
+  noTrickleWait = false
+) {
+  const trickle1 = trickleIce(offerer, answerer);
+  const trickle2 = trickleIce(answerer, offerer);
+  try {
+    const offer = await offerer.createOffer({ offerToReceiveAudio: true });
+    await offerer.setLocalDescription(offer);
+    await answerer.setRemoteDescription(offer);
+    const answer = await answerer.createAnswer();
+    await Promise.all([
+      offerer.setRemoteDescription(answer),
+      answerer.setLocalDescription(answer),
+    ]);
+
+    const throwOnTimeout = async () => {
+      if (timeout) {
+        await wait(timeout);
+        throw new Error(
+          `ICE did not complete within ${timeout} ms with ${context}`
+        );
+      }
+    };
+
+    await Promise.race([
+      Promise.all([iceConnected(offerer), iceConnected(answerer)]),
+      throwOnTimeout(timeout, context),
+    ]);
+  } finally {
+    if (!noTrickleWait) {
+      // TODO(bug 1751509): For now, we need to let gathering finish before we
+      // proceed, because there are races in ICE restart wrt gathering state.
+      await Promise.all([trickle1, trickle2]);
+    }
+  }
+}
+
+function isV6HostCandidate(candidate) {
+  const fields = candidate.candidate.split(" ");
+  const type = fields[7];
+  const ipAddress = fields[4];
+  return type == "host" && ipAddress.includes(":");
+}
+
+async function ipv6Supported() {
+  const pc = new RTCPeerConnection();
+  const candidates = await gatherWithTimeout(pc, 8000);
+  info(`baseline candidates: ${JSON.stringify(candidates)}`);
+  pc.close();
+  return candidates.some(isV6HostCandidate);
+}
+
+function makeContextString(iceServers) {
+  const currentRedirectAddress = SpecialPowers.getCharPref(
+    "media.peerconnection.nat_simulator.redirect_address",
+    ""
+  );
+  const currentRedirectTargets = SpecialPowers.getCharPref(
+    "media.peerconnection.nat_simulator.redirect_targets",
+    ""
+  );
+  return `redirect rule: ${currentRedirectAddress}=>${currentRedirectTargets} iceServers: ${JSON.stringify(
+    iceServers
+  )}`;
+}
+
+async function checkSrflx(iceServers) {
+  const context = makeContextString(iceServers);
+  info(`checkSrflx ${context}`);
+  const pc = new RTCPeerConnection({
+    iceServers,
+    bundlePolicy: "max-bundle", // Avoids extra candidates
+  });
+  const candidates = await gatherWithTimeout(pc, 8000, context);
+  const srflxCandidates = candidates.filter(c => c.candidate.includes("srflx"));
+  info(`candidates: ${JSON.stringify(srflxCandidates)}`);
+  // TODO(bug 1339203): Once we support rtcpMuxPolicy, set it to "require" to
+  // result in a single srflx candidate
+  is(
+    srflxCandidates.length,
+    2,
+    `Should have two srflx candidates with ${context}`
+  );
+  pc.close();
+}
+
+async function checkNoSrflx(iceServers) {
+  const context = makeContextString(iceServers);
+  info(`checkNoSrflx ${context}`);
+  const pc = new RTCPeerConnection({
+    iceServers,
+    bundlePolicy: "max-bundle", // Avoids extra candidates
+  });
+  const candidates = await gatherWithTimeout(pc, 8000, context);
+  const srflxCandidates = candidates.filter(c => c.candidate.includes("srflx"));
+  info(`candidates: ${JSON.stringify(srflxCandidates)}`);
+  is(
+    srflxCandidates.length,
+    0,
+    `Should have no srflx candidates with ${context}`
+  );
+  pc.close();
+}
+
+async function checkRelayUdp(iceServers) {
+  const context = makeContextString(iceServers);
+  info(`checkRelayUdp ${context}`);
+  const pc = new RTCPeerConnection({
+    iceServers,
+    bundlePolicy: "max-bundle", // Avoids extra candidates
+  });
+  const candidates = await gatherWithTimeout(pc, 8000, context);
+  const relayCandidates = candidates.filter(c => c.candidate.includes("relay"));
+  info(`candidates: ${JSON.stringify(relayCandidates)}`);
+  // TODO(bug 1339203): Once we support rtcpMuxPolicy, set it to "require" to
+  // result in a single relay candidate
+  is(
+    relayCandidates.length,
+    2,
+    `Should have two relay candidates with ${context}`
+  );
+  // It would be nice if RTCIceCandidate had a field telling us what the
+  // "related protocol" is (similar to relatedAddress and relatedPort).
+  // Because there is no such thing, we need to go through the stats API,
+  // which _does_ have that information.
+  is(
+    (await findStatsRelayCandidates(pc, "tcp")).length,
+    0,
+    `No TCP relay candidates should be present with ${context}`
+  );
+  pc.close();
+}
+
+async function checkRelayTcp(iceServers) {
+  const context = makeContextString(iceServers);
+  info(`checkRelayTcp ${context}`);
+  const pc = new RTCPeerConnection({
+    iceServers,
+    bundlePolicy: "max-bundle", // Avoids extra candidates
+  });
+  const candidates = await gatherWithTimeout(pc, 8000, context);
+  const relayCandidates = candidates.filter(c => c.candidate.includes("relay"));
+  info(`candidates: ${JSON.stringify(relayCandidates)}`);
+  // TODO(bug 1339203): Once we support rtcpMuxPolicy, set it to "require" to
+  // result in a single relay candidate
+  is(
+    relayCandidates.length,
+    2,
+    `Should have two relay candidates with ${context}`
+  );
+  // It would be nice if RTCIceCandidate had a field telling us what the
+  // "related protocol" is (similar to relatedAddress and relatedPort).
+  // Because there is no such thing, we need to go through the stats API,
+  // which _does_ have that information.
+  is(
+    (await findStatsRelayCandidates(pc, "udp")).length,
+    0,
+    `No UDP relay candidates should be present with ${context}`
+  );
+  pc.close();
+}
+
+async function checkRelayUdpTcp(iceServers) {
+  const context = makeContextString(iceServers);
+  info(`checkRelayUdpTcp ${context}`);
+  const pc = new RTCPeerConnection({
+    iceServers,
+    bundlePolicy: "max-bundle", // Avoids extra candidates
+  });
+  const candidates = await gatherWithTimeout(pc, 8000, context);
+  const relayCandidates = candidates.filter(c => c.candidate.includes("relay"));
+  info(`candidates: ${JSON.stringify(relayCandidates)}`);
+  // TODO(bug 1339203): Once we support rtcpMuxPolicy, set it to "require" to
+  // result in a single relay candidate each for UDP and TCP
+  is(
+    relayCandidates.length,
+    4,
+    `Should have two relay candidates for each protocol with ${context}`
+  );
+  // It would be nice if RTCIceCandidate had a field telling us what the
+  // "related protocol" is (similar to relatedAddress and relatedPort).
+  // Because there is no such thing, we need to go through the stats API,
+  // which _does_ have that information.
+  is(
+    (await findStatsRelayCandidates(pc, "udp")).length,
+    2,
+    `Two UDP relay candidates should be present with ${context}`
+  );
+  // TODO(bug 1705563): This is 1 because of bug 1705563
+  is(
+    (await findStatsRelayCandidates(pc, "tcp")).length,
+    1,
+    `One TCP relay candidates should be present with ${context}`
+  );
+  pc.close();
+}
+
+async function checkNoRelay(iceServers) {
+  const context = makeContextString(iceServers);
+  info(`checkNoRelay ${context}`);
+  const pc = new RTCPeerConnection({
+    iceServers,
+    bundlePolicy: "max-bundle", // Avoids extra candidates
+  });
+  const candidates = await gatherWithTimeout(pc, 8000, context);
+  const relayCandidates = candidates.filter(c => c.candidate.includes("relay"));
+  info(`candidates: ${JSON.stringify(relayCandidates)}`);
+  is(
+    relayCandidates.length,
+    0,
+    `Should have no relay candidates with ${context}`
+  );
+  pc.close();
+}
diff --git a/dom/media/webrtc/tests/mochitests/identity/identityPcTest.js b/dom/media/webrtc/tests/mochitests/identity/identityPcTest.js
new file mode 100644
index 0000000000..1381873f9d
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/identityPcTest.js
@@ -0,0 +1,79 @@
+function identityPcTest(remoteOptions) {
+  var user = "someone";
+  var domain1 = "test1.example.com";
+  var domain2 = "test2.example.com";
+  var id1 = user + "@" + domain1;
+  var id2 = user + "@" + domain2;
+
+  test = new PeerConnectionTest({
+    config_local: {
+      peerIdentity: id2,
+    },
+    config_remote: {
+      peerIdentity: id1,
+    },
+  });
+  test.setMediaConstraints(
+    [
+      {
+        audio: true,
+        video: true,
+        peerIdentity: id2,
+      },
+    ],
+    [
+      remoteOptions || {
+        audio: true,
+        video: true,
+        peerIdentity: id1,
+      },
+    ]
+  );
+  test.pcLocal.setIdentityProvider("test1.example.com", { protocol: "idp.js" });
+  test.pcRemote.setIdentityProvider("test2.example.com", {
+    protocol: "idp.js",
+  });
+  test.chain.append([
+    function PEER_IDENTITY_IS_SET_CORRECTLY(test) {
+      // no need to wait to check identity in this case,
+      // setRemoteDescription should wait for the IdP to complete
+      function checkIdentity(pc, pfx, idp, name) {
+        return pc.peerIdentity.then(peerInfo => {
+          is(peerInfo.idp, idp, pfx + "IdP check");
+          is(peerInfo.name, name + "@" + idp, pfx + "identity check");
+        });
+      }
+
+      return Promise.all([
+        checkIdentity(
+          test.pcLocal._pc,
+          "local: ",
+          "test2.example.com",
+          "someone"
+        ),
+        checkIdentity(
+          test.pcRemote._pc,
+          "remote: ",
+          "test1.example.com",
+          "someone"
+        ),
+      ]);
+    },
+
+    function REMOTE_STREAMS_ARE_RESTRICTED(test) {
+      var remoteStream = test.pcLocal._pc.getRemoteStreams()[0];
+      for (const track of remoteStream.getTracks()) {
+        mustThrowWith(
+          `Freshly received ${track.kind} track with peerIdentity`,
+          "SecurityError",
+          () => new MediaRecorder(new MediaStream([track])).start()
+        );
+      }
+      return Promise.all([
+        audioIsSilence(true, remoteStream),
+        videoIsBlack(true, remoteStream),
+      ]);
+    },
+  ]);
+  return test.run();
+}
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp-bad.js b/dom/media/webrtc/tests/mochitests/identity/idp-bad.js
new file mode 100644
index 0000000000..86e1cb7a34
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp-bad.js
@@ -0,0 +1 @@
+<This isn't valid JS>
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp-min.js b/dom/media/webrtc/tests/mochitests/identity/idp-min.js
new file mode 100644
index 0000000000..a4b2c55cee
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp-min.js
@@ -0,0 +1,24 @@
+(function (global) {
+  "use strict";
+  // A minimal implementation of the interface.
+  // Though this isn't particularly functional.
+  // This is needed so that we can have a "working" IdP served
+  // from two different locations in the tree.
+  global.rtcIdentityProvider.register({
+    generateAssertion(payload, origin, usernameHint) {
+      dump("idp: generateAssertion(" + payload + ")\n");
+      return Promise.resolve({
+        idp: { domain: "example.com", protocol: "idp.js" },
+        assertion: "bogus",
+      });
+    },
+
+    validateAssertion(assertion, origin) {
+      dump("idp: validateAssertion(" + assertion + ")\n");
+      return Promise.resolve({
+        identity: "user@example.com",
+        contents: "bogus",
+      });
+    },
+  });
+})(this);
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp-redirect-http-trick.js b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-http-trick.js
new file mode 100644
index 0000000000..75390cbf4f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-http-trick.js
@@ -0,0 +1,3 @@
+(function () {
+  dump("ERROR\n");
+})();
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp-redirect-http-trick.js^headers^ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-http-trick.js^headers^
new file mode 100644
index 0000000000..b3a2afd90a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-http-trick.js^headers^
@@ -0,0 +1,2 @@
+HTTP 301 Moved Permanently
+Location: http://example.com/.well-known/idp-proxy/idp-redirect-https.js
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp-redirect-http.js b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-http.js
new file mode 100644
index 0000000000..75390cbf4f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-http.js
@@ -0,0 +1,3 @@
+(function () {
+  dump("ERROR\n");
+})();
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp-redirect-http.js^headers^ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-http.js^headers^
new file mode 100644
index 0000000000..d2380984e7
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-http.js^headers^
@@ -0,0 +1,2 @@
+HTTP 301 Moved Permanently
+Location: http://example.com/.well-known/idp-proxy/idp.js
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https-double.js b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https-double.js
new file mode 100644
index 0000000000..75390cbf4f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https-double.js
@@ -0,0 +1,3 @@
+(function () {
+  dump("ERROR\n");
+})();
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https-double.js^headers^ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https-double.js^headers^
new file mode 100644
index 0000000000..3fb8a35ae7
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https-double.js^headers^
@@ -0,0 +1,2 @@
+HTTP 301 Moved Permanently
+Location: https://example.com/.well-known/idp-proxy/idp-redirect-https.js
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https-odd-path.js b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https-odd-path.js
new file mode 100644
index 0000000000..75390cbf4f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https-odd-path.js
@@ -0,0 +1,3 @@
+(function () {
+  dump("ERROR\n");
+})();
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https-odd-path.js^headers^ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https-odd-path.js^headers^
new file mode 100644
index 0000000000..6e2931eda9
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https-odd-path.js^headers^
@@ -0,0 +1,2 @@
+HTTP 301 Moved Permanently
+Location: https://example.com/.well-known/idp-min.js
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https.js b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https.js
new file mode 100644
index 0000000000..75390cbf4f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https.js
@@ -0,0 +1,3 @@
+(function () {
+  dump("ERROR\n");
+})();
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https.js^headers^ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https.js^headers^
new file mode 100644
index 0000000000..77d56ac442
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp-redirect-https.js^headers^
@@ -0,0 +1,2 @@
+HTTP 301 Moved Permanently
+Location: https://example.com/.well-known/idp-proxy/idp.js
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp.js b/dom/media/webrtc/tests/mochitests/identity/idp.js
new file mode 100644
index 0000000000..557740657f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp.js
@@ -0,0 +1,119 @@
+(function (global) {
+  "use strict";
+
+  // rather than create a million different IdP configurations and litter the
+  // world with files all containing near-identical code, let's use the hash/URL
+  // fragment as a way of generating instructions for the IdP
+  var instructions = global.location.hash.replace("#", "").split(":");
+  function is(target) {
+    return function (instruction) {
+      return instruction === target;
+    };
+  }
+
+  function IDPJS() {
+    this.domain = global.location.host;
+    var path = global.location.pathname;
+    this.protocol =
+      path.substring(path.lastIndexOf("/") + 1) + global.location.hash;
+    this.id = crypto.getRandomValues(new Uint8Array(10)).join(".");
+  }
+
+  IDPJS.prototype = {
+    getLogin() {
+      return fetch(
+        "https://example.com/.well-known/idp-proxy/idp.sjs?" + this.id
+      ).then(response => response.status === 200);
+    },
+    checkLogin(result) {
+      return this.getLogin().then(loggedIn => {
+        if (loggedIn) {
+          return result;
+        }
+        return Promise.reject({
+          name: "IdpLoginError",
+          loginUrl:
+            "https://example.com/.well-known/idp-proxy/login.html#" + this.id,
+        });
+      });
+    },
+
+    borkResult(result) {
+      if (instructions.some(is("throw"))) {
+        throw new Error("Throwing!");
+      }
+      if (instructions.some(is("fail"))) {
+        return Promise.reject(new Error("Failing!"));
+      }
+      if (instructions.some(is("login"))) {
+        return this.checkLogin(result);
+      }
+      if (instructions.some(is("hang"))) {
+        return new Promise(r => {});
+      }
+      dump("idp: result=" + JSON.stringify(result) + "\n");
+      return Promise.resolve(result);
+    },
+
+    _selectUsername(usernameHint) {
+      dump("_selectUsername: usernameHint(" + usernameHint + ")\n");
+      var username = "someone@" + this.domain;
+      if (usernameHint) {
+        var at = usernameHint.indexOf("@");
+        if (at < 0) {
+          username = usernameHint + "@" + this.domain;
+        } else if (usernameHint.substring(at + 1) === this.domain) {
+          username = usernameHint;
+        }
+      }
+      return username;
+    },
+
+    generateAssertion(payload, origin, options) {
+      dump(
+        "idp: generateAssertion(" +
+          payload +
+          ", " +
+          origin +
+          ", " +
+          JSON.stringify(options) +
+          ")\n"
+      );
+      var idpDetails = {
+        domain: this.domain,
+        protocol: this.protocol,
+      };
+      if (instructions.some(is("bad-assert"))) {
+        idpDetails = {};
+      }
+      return this.borkResult({
+        idp: idpDetails,
+        assertion: JSON.stringify({
+          username: this._selectUsername(options.usernameHint),
+          contents: payload,
+        }),
+      });
+    },
+
+    validateAssertion(assertion, origin) {
+      dump("idp: validateAssertion(" + assertion + ")\n");
+      var assertion = JSON.parse(assertion);
+      if (instructions.some(is("bad-validate"))) {
+        assertion.contents = {};
+      }
+      return this.borkResult({
+        identity: assertion.username,
+        contents: assertion.contents,
+      });
+    },
+  };
+
+  if (!instructions.some(is("not_ready"))) {
+    dump("registering idp.js" + global.location.hash + "\n");
+    var idp = new IDPJS();
+    global.rtcIdentityProvider.register({
+      generateAssertion: idp.generateAssertion.bind(idp),
+      validateAssertion: idp.validateAssertion.bind(idp),
+    });
+  }
+})(this);
diff --git a/dom/media/webrtc/tests/mochitests/identity/idp.sjs b/dom/media/webrtc/tests/mochitests/identity/idp.sjs
new file mode 100644
index 0000000000..e1a245be78
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/idp.sjs
@@ -0,0 +1,18 @@
+function handleRequest(request, response) {
+  var key = "/.well-known/idp-proxy/" + request.queryString;
+  dump(getState(key) + "\n");
+  if (request.method === "GET") {
+    if (getState(key)) {
+      response.setStatusLine(request.httpVersion, 200, "OK");
+    } else {
+      response.setStatusLine(request.httpVersion, 404, "Not Found");
+    }
+  } else if (request.method === "PUT") {
+    setState(key, "OK");
+    response.setStatusLine(request.httpVersion, 200, "OK");
+  } else {
+    response.setStatusLine(request.httpVersion, 406, "Method Not Allowed");
+  }
+  response.setHeader("Content-Type", "text/plain;charset=UTF-8");
+  response.write("OK");
+}
diff --git a/dom/media/webrtc/tests/mochitests/identity/login.html b/dom/media/webrtc/tests/mochitests/identity/login.html
new file mode 100644
index 0000000000..eafba22f2d
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/login.html
@@ -0,0 +1,31 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Identity Provider Login</title>
+  <script type="application/javascript">
+  window.onload = () => {
+    var xhr = new XMLHttpRequest();
+    xhr.open("PUT", "https://example.com/.well-known/idp-proxy/idp.sjs?" +
+             window.location.hash.replace('#', ''));
+    xhr.onload = () => {
+      var isFramed = (window !== window.top);
+      var parent = isFramed ? window.parent : window.opener;
+      // Using '*' is cheating, but that's OK.
+      parent.postMessage('LOGINDONE', '*');
+      var done = document.createElement('div');
+
+      done.textContent = 'Done';
+      document.body.appendChild(done);
+
+      if (!isFramed) {
+        window.close();
+      }
+    };
+    xhr.send();
+  };
+  </script>
+</head>
+<body>
+  <div>Logging in...</div>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/identity/mochitest.ini b/dom/media/webrtc/tests/mochitests/identity/mochitest.ini
new file mode 100644
index 0000000000..7a60cc6c6e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/mochitest.ini
@@ -0,0 +1,47 @@
+[DEFAULT]
+subsuite = media
+skip-if = (os == 'linux' && !debug)
+support-files =
+  /.well-known/idp-proxy/idp.js
+  identityPcTest.js
+  !/dom/media/webrtc/tests/mochitests/blacksilence.js
+  !/dom/media/webrtc/tests/mochitests/dataChannel.js
+  !/dom/media/webrtc/tests/mochitests/head.js
+  !/dom/media/webrtc/tests/mochitests/network.js
+  !/dom/media/webrtc/tests/mochitests/pc.js
+  !/dom/media/webrtc/tests/mochitests/sdpUtils.js
+  !/dom/media/webrtc/tests/mochitests/templates.js
+  !/dom/media/webrtc/tests/mochitests/turnConfig.js
+tags = mtg
+
+[test_fingerprints.html]
+scheme=https
+[test_getIdentityAssertion.html]
+[test_idpproxy.html]
+support-files =
+  /.well-known/idp-proxy/idp-redirect-http.js
+  /.well-known/idp-proxy/idp-redirect-http.js^headers^
+  /.well-known/idp-proxy/idp-redirect-http-trick.js
+  /.well-known/idp-proxy/idp-redirect-http-trick.js^headers^
+  /.well-known/idp-proxy/idp-redirect-https.js
+  /.well-known/idp-proxy/idp-redirect-https.js^headers^
+  /.well-known/idp-proxy/idp-redirect-https-double.js
+  /.well-known/idp-proxy/idp-redirect-https-double.js^headers^
+  /.well-known/idp-proxy/idp-redirect-https-odd-path.js
+  /.well-known/idp-proxy/idp-redirect-https-odd-path.js^headers^
+  /.well-known/idp-min.js
+  /.well-known/idp-proxy/idp-bad.js
+[test_loginNeeded.html]
+support-files =
+  /.well-known/idp-proxy/login.html
+  /.well-known/idp-proxy/idp.sjs
+[test_peerConnection_asymmetricIsolation.html]
+scheme=https
+skip-if = os == 'android'
+[test_peerConnection_peerIdentity.html]
+scheme=https
+skip-if = os == 'android'
+[test_setIdentityProvider.html]
+scheme=https
+[test_setIdentityProviderWithErrors.html]
+scheme=https
diff --git a/dom/media/webrtc/tests/mochitests/identity/test_fingerprints.html b/dom/media/webrtc/tests/mochitests/identity/test_fingerprints.html
new file mode 100644
index 0000000000..0a7f0a2033
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/test_fingerprints.html
@@ -0,0 +1,91 @@
+<html>
+<head>
+<meta charset="utf-8" />
+<script type="application/javascript">var scriptRelativePath = "../";</script>
+<script type="application/javascript" src="../pc.js"></script>
+</head>
+<body>
+<script class="testbody" type="application/javascript">
+createHTML({ title: "Test multiple identity fingerprints",  bug: "1005152" });
+
+// here we call the identity provider directly
+async function getIdentityAssertion(fingerprint) {
+  const { IdpSandbox } = SpecialPowers.ChromeUtils.import(
+    'resource://gre/modules/media/IdpSandbox.jsm'
+  );
+  const sandbox = new IdpSandbox('example.com', 'idp.js', window);
+  const idp = SpecialPowers.wrap(await sandbox.start());
+  const assertion = SpecialPowers.wrap(await
+      idp.generateAssertion(JSON.stringify({ fingerprint }),
+                            'https://example.com',
+                            {}));
+  const assertionString = btoa(JSON.stringify(assertion));
+  sandbox.stop();
+  return assertionString;
+}
+
+// This takes a real fingerprint and makes some extra bad ones.
+function makeFingerprints(algorithm, digest) {
+  const fingerprints = [];
+  fingerprints.push({ algorithm, digest });
+  for (var i = 0; i < 3; ++i) {
+    fingerprints.push({
+      algorithm,
+      digest: digest.replace(/:./g, ':' + i.toString(16))
+    });
+  }
+  return fingerprints;
+}
+
+const fingerprintRegex = /^a=fingerprint:(\S+) (\S+)/m;
+const identityRegex = /^a=identity:(\S+)/m;
+
+function fingerprintSdp(fingerprints) {
+  return fingerprints.map(fp => 'a=fInGeRpRiNt:' + fp.algorithm +
+                                ' ' + fp.digest + '\n').join('');
+}
+
+// Firefox only uses a single fingerprint.
+// That doesn't mean we have it create SDP that describes two.
+// This function synthesizes that SDP and tries to set it.
+
+runNetworkTest(async () => {
+  // this one fails setRemoteDescription if the identity is not good
+  const pcStrict = new RTCPeerConnection({ peerIdentity: 'someone@example.com'});
+  // this one will be manually tweaked to have two fingerprints
+  const pcDouble = new RTCPeerConnection({});
+
+  const stream = await getUserMedia({ video: true });
+  ok(stream, 'Got test stream');
+  const [track] = stream.getTracks();
+  pcDouble.addTrack(track, stream);
+  try {
+    const offer = await pcDouble.createOffer();
+    ok(offer, 'Got offer');
+    const match = offer.sdp.match(fingerprintRegex);
+    if (!match) {
+      throw new Error('No fingerprint in offer SDP');
+    }
+    const fingerprints = makeFingerprints(match[1], match[2]);
+    const assertion = await getIdentityAssertion(fingerprints);
+    ok(assertion, 'Should have assertion');
+
+    const sdp = offer.sdp.slice(0, match.index) +
+        'a=identity:' + assertion + '\n' +
+        fingerprintSdp(fingerprints.slice(1)) +
+        offer.sdp.slice(match.index);
+
+    await pcStrict.setRemoteDescription({ type: 'offer', sdp });
+    ok(true, 'Modified fingerprints were accepted');
+  } catch (error) {
+    const e = SpecialPowers.wrap(error);
+    ok(false, 'error in test: ' +
+       (e.message ? (e.message + '\n' + e.stack) : e));
+  }
+  pcStrict.close();
+  pcDouble.close();
+  track.stop();
+});
+</script>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/identity/test_getIdentityAssertion.html b/dom/media/webrtc/tests/mochitests/identity/test_getIdentityAssertion.html
new file mode 100644
index 0000000000..47e1cb1df6
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/test_getIdentityAssertion.html
@@ -0,0 +1,101 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript">var scriptRelativePath = "../";</script>
+  <script type="application/javascript" src="../pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "getIdentityAssertion Tests",
+    bug: "942367"
+  });
+
+function checkIdentity(assertion, identity) {
+  // here we dig into the payload, which means we need to know something
+  // about how the IdP actually works (not good in general, but OK here)
+  var assertion = JSON.parse(atob(assertion)).assertion;
+  var user = JSON.parse(assertion).username;
+  is(user, identity, 'id should be "' + identity + '" is "' + user + '"');
+}
+
+function getAssertion(t, instructions, userHint) {
+  dump('instructions: ' + instructions + '\n');
+  dump('userHint: ' + userHint + '\n');
+  t.pcLocal.setIdentityProvider('example.com',
+                                { protocol: 'idp.js' + instructions,
+                                  usernameHint: userHint });
+  return t.pcLocal._pc.getIdentityAssertion();
+}
+
+var test;
+function theTest() {
+  test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.chain.removeAfter('PC_REMOTE_CHECK_INITIAL_SIGNALINGSTATE');
+  test.chain.append([
+    function PC_LOCAL_IDENTITY_ASSERTION_FAILS_WITHOUT_PROVIDER(t) {
+      return t.pcLocal._pc.getIdentityAssertion()
+        .then(a => ok(false, 'should fail without provider'),
+              e => ok(e, 'should fail without provider'));
+    },
+
+    function PC_LOCAL_IDENTITY_ASSERTION_FAILS_WITH_BAD_PROVIDER(t) {
+      t.pcLocal._pc.setIdentityProvider('example.com',
+                                        { protocol: 'idp-bad.js',
+                                          usernameHint: '' });
+      return t.pcLocal._pc.getIdentityAssertion()
+        .then(a => ok(false, 'should fail with bad provider'),
+              e => {
+                is(e.name, 'IdpError', 'should fail with bad provider');
+                ok(e.message, 'should include a nice message');
+              });
+    },
+
+    function PC_LOCAL_GET_TWO_ASSERTIONS(t) {
+      return Promise.all([
+        getAssertion(t, ''),
+        getAssertion(t, '')
+      ]).then(assertions => {
+        is(assertions.length, 2, "Two assertions generated");
+        assertions.forEach(a => checkIdentity(a, 'someone@example.com'));
+      });
+    },
+
+    function PC_LOCAL_IDP_FAILS(t) {
+      return getAssertion(t, '#fail')
+        .then(a => ok(false, '#fail should not get an identity result'),
+              e => is(e.name, 'IdpError', '#fail should cause rejection'));
+    },
+
+    function PC_LOCAL_IDP_LOGIN_ERROR(t) {
+      return getAssertion(t, '#login')
+        .then(a => ok(false, '#login should not work'),
+              e => {
+                is(e.name, 'IdpLoginError', 'name is IdpLoginError');
+                is(t.pcLocal._pc.idpLoginUrl.split('#')[0],
+                   'https://example.com/.well-known/idp-proxy/login.html',
+                   'got the right login URL from the IdP');
+              });
+    },
+
+    function PC_LOCAL_IDP_NOT_READY(t) {
+      return getAssertion(t, '#not_ready')
+        .then(a => ok(false, '#not_ready should not get an identity result'),
+              e => is(e.name, 'IdpError', '#not_ready should cause rejection'));
+    },
+
+    function PC_LOCAL_ASSERTION_WITH_SPECIFIC_NAME(t) {
+      return getAssertion(t, '', 'user@example.com')
+        .then(a => checkIdentity(a, 'user@example.com'));
+    }
+  ]);
+  return test.run();
+}
+runNetworkTest(theTest);
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/identity/test_idpproxy.html b/dom/media/webrtc/tests/mochitests/identity/test_idpproxy.html
new file mode 100644
index 0000000000..065501b8a4
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/test_idpproxy.html
@@ -0,0 +1,178 @@
+<html>
+<head>
+<meta charset="utf-8" />
+<link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+<script src="/tests/SimpleTest/SimpleTest.js"></script>
+</head>
+<body>
+  <script class="testbody" type="application/javascript">
+"use strict";
+var { IdpSandbox } = SpecialPowers.ChromeUtils.import(
+  "resource://gre/modules/media/IdpSandbox.jsm"
+);
+var dummyPayload = JSON.stringify({
+  this: 'is',
+  a: ['stu', 6],
+  obj: null
+});
+
+function test_domain_sandbox() {
+  var diabolical = {
+    toString() {
+      return 'example.com/path';
+    }
+  };
+  var domains = [ 'ex/foo', 'user@ex', 'user:pass@ex', 'ex#foo', 'ex?foo',
+                  '', 12, null, diabolical, true ];
+  domains.forEach(function(domain) {
+    try {
+      var idp = new IdpSandbox(domain, undefined, window);
+      ok(false, 'IdpSandbox allowed a bad domain: ' + domain);
+    } catch (e) {
+      var str = (typeof domain === 'string') ? domain : typeof domain;
+      ok(true, 'Evil domain "' + str + '" raises exception');
+    }
+  });
+}
+
+function test_protocol_sandbox() {
+  var protos = [ '../evil/proto', '..%2Fevil%2Fproto',
+                 '\\evil', '%5cevil', 12, true, {} ];
+  protos.forEach(function(proto) {
+    try {
+      var idp = new IdpSandbox('example.com', proto, window);
+      ok(false, 'IdpSandbox allowed a bad protocol: ' + proto);
+    } catch (e) {
+      var str = (typeof proto === 'string') ? proto : typeof proto;
+      ok(true, 'Evil protocol "' + proto + '" raises exception');
+    }
+  });
+}
+
+function idpName(hash) {
+  return 'idp.js' + (hash ? ('#' + hash) : '');
+}
+
+function makeSandbox(js) {
+  var name = js || idpName();
+  info('Creating a sandbox for the protocol: ' + name);
+  var sandbox = new IdpSandbox('example.com', name, window);
+  return sandbox.start().then(idp => SpecialPowers.wrap(idp));
+}
+
+function test_generate_assertion() {
+  return makeSandbox()
+    .then(idp => idp.generateAssertion(dummyPayload,
+                                       'https://example.net',
+                                        {}))
+    .then(response => {
+      response = SpecialPowers.wrap(response);
+      is(response.idp.domain, 'example.com', 'domain is correct');
+      is(response.idp.protocol, 'idp.js', 'protocol is correct');
+      ok(typeof response.assertion === 'string', 'assertion is present');
+    });
+}
+
+// test that the test IdP can eat its own dogfood; which is the only way to test
+// validateAssertion, since that consumes the output of generateAssertion (in
+// theory, generateAssertion could identify a different IdP domain).
+
+function test_validate_assertion() {
+  return makeSandbox()
+    .then(idp => idp.generateAssertion(dummyPayload,
+                                       'https://example.net',
+                                       { usernameHint: 'user' }))
+    .then(assertion => {
+      var wrapped = SpecialPowers.wrap(assertion);
+      return makeSandbox()
+        .then(idp => idp.validateAssertion(wrapped.assertion,
+                                           'https://example.net'));
+    }).then(response => {
+      response = SpecialPowers.wrap(response);
+      is(response.identity, 'user@example.com');
+      is(response.contents, dummyPayload);
+    });
+}
+
+// We don't want to test the #bad or the #hang instructions,
+// errors of the sort those generate aren't handled by the sandbox code.
+function test_assertion_failure(reason) {
+  return () => {
+    return makeSandbox(idpName(reason))
+      .then(idp => idp.generateAssertion('hello', 'example.net', {}))
+      .then(r => ok(false, 'should not succeed on ' + reason),
+            e => ok(true, 'failed correctly on ' + reason));
+  };
+}
+
+function test_load_failure() {
+  return makeSandbox('non-existent-file')
+    .then(() => ok(false, 'Should fail to load non-existent file'),
+          e => ok(e, 'Should fail to load non-existent file'));
+}
+
+function test_redirect_ok(from) {
+  return () => {
+    return makeSandbox(from)
+      .then(idp => idp.generateAssertion('hello', 'example.net'))
+      .then(r => ok(SpecialPowers.wrap(r).assertion,
+                    'Redirect to https should be OK'));
+  };
+}
+
+function test_redirect_fail(from) {
+  return () => {
+    return makeSandbox(from)
+      .then(() => ok(false, 'Redirect to https should fail'),
+            e => ok(e, 'Redirect to https should fail'));
+  };
+}
+
+function test_bad_js() {
+  return makeSandbox('idp-bad.js')
+    .then(() => ok(false, 'Bad JS should not load'),
+          e => ok(e, 'Bad JS should not load'));
+}
+
+function run_all_tests() {
+  [
+    test_domain_sandbox,
+    test_protocol_sandbox,
+    test_generate_assertion,
+    test_validate_assertion,
+
+    // fail of the IdP fails
+    test_assertion_failure('fail'),
+    // fail if the IdP throws
+    test_assertion_failure('throw'),
+    // fail if the IdP is not ready
+    test_assertion_failure('not_ready'),
+
+    test_load_failure(),
+    // Test a redirect to an HTTPS origin, which should be OK
+    test_redirect_ok('idp-redirect-https.js'),
+    // Two redirects is fine too
+    test_redirect_ok('idp-redirect-https-double.js'),
+    // A secure redirect to a path other than /.well-known/idp-proxy/* should
+    // also work fine.
+    test_redirect_ok('idp-redirect-https-odd-path.js'),
+    // A redirect to HTTP is not-cool
+    test_redirect_fail('idp-redirect-http.js'),
+    // Also catch tricks like https->http->https
+    test_redirect_fail('idp-redirect-http-trick.js'),
+
+    test_bad_js
+  ].reduce((p, test) => {
+    return p.then(test)
+      .catch(e => ok(false, test.name + ' failed: ' +
+                     SpecialPowers.wrap(e).message + '\n' +
+                     SpecialPowers.wrap(e).stack));
+  }, Promise.resolve())
+    .then(() => SimpleTest.finish());
+}
+
+SimpleTest.waitForExplicitFinish();
+run_all_tests();
+</script>
+  </body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/identity/test_loginNeeded.html b/dom/media/webrtc/tests/mochitests/identity/test_loginNeeded.html
new file mode 100644
index 0000000000..550dc20d92
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/test_loginNeeded.html
@@ -0,0 +1,72 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript">var scriptRelativePath = "../";</script>
+  <script type="application/javascript" src="../pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: 'RTCPeerConnection identity with login',
+    bug: '1153314'
+  });
+
+function waitForLoginDone() {
+  return new Promise(resolve => {
+    window.addEventListener('message', function listener(e) {
+      is(e.origin, 'https://example.com', 'got the right message origin');
+      is(e.data, 'LOGINDONE', 'got the right message');
+      window.removeEventListener('message', listener);
+      resolve();
+    });
+  });
+}
+
+function checkLogin(t, name, onLoginNeeded) {
+  t.pcLocal.setIdentityProvider('example.com',
+                                { protocol: 'idp.js#login:' + name });
+  return t.pcLocal._pc.getIdentityAssertion()
+    .then(a => ok(false, 'should request login'),
+          e => {
+            is(e.name, 'IdpLoginError', 'name is IdpLoginError');
+            is(t.pcLocal._pc.idpLoginUrl.split('#')[0],
+               'https://example.com/.well-known/idp-proxy/login.html',
+               'got the right login URL from the IdP');
+            return t.pcLocal._pc.idpLoginUrl;
+          })
+    .then(onLoginNeeded)
+    .then(waitForLoginDone)
+    .then(() => t.pcLocal._pc.getIdentityAssertion())
+    .then(a => ok(a, 'got assertion'));
+}
+
+function theTest() {
+  var test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.chain.removeAfter('PC_REMOTE_CHECK_INITIAL_SIGNALINGSTATE');
+  test.chain.append([
+    function PC_LOCAL_IDENTITY_ASSERTION_WITH_IFRAME_LOGIN(t) {
+      return checkLogin(t, 'iframe', loginUrl => {
+        var iframe = document.createElement('iframe');
+        iframe.setAttribute('src', loginUrl);
+        iframe.frameBorder = 0;
+        iframe.width = 400;
+        iframe.height = 60;
+        document.getElementById('display').appendChild(iframe);
+      });
+    },
+    function PC_LOCAL_IDENTITY_ASSERTION_WITH_WINDOW_LOGIN(t) {
+      return checkLogin(t, 'openwin', loginUrl => {
+        window.open(loginUrl, 'login', 'width=400,height=60');
+      });
+    }
+  ]);
+  return test.run();
+}
+runNetworkTest(theTest);
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/identity/test_peerConnection_asymmetricIsolation.html b/dom/media/webrtc/tests/mochitests/identity/test_peerConnection_asymmetricIsolation.html
new file mode 100644
index 0000000000..65a2fc5392
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/test_peerConnection_asymmetricIsolation.html
@@ -0,0 +1,31 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript">var scriptRelativePath = "../";</script>
+  <script type="application/javascript" src="../pc.js"></script>
+  <script type="application/javascript" src="../blacksilence.js"></script>
+  <script type="application/javascript" src="identityPcTest.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  title: "Non-isolated media entering an isolated session becomes isolated",
+  bug: "996238"
+});
+
+function theTest() {
+  // Override the remote media capture options to remove isolation for the
+  // remote party; the test verifies that the media it receives on the local
+  // side is isolated anyway.
+  return identityPcTest({
+    audio: true,
+    video: true
+  });
+}
+runNetworkTest(theTest);
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/identity/test_peerConnection_peerIdentity.html b/dom/media/webrtc/tests/mochitests/identity/test_peerConnection_peerIdentity.html
new file mode 100644
index 0000000000..a8116cc451
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/test_peerConnection_peerIdentity.html
@@ -0,0 +1,21 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript">var scriptRelativePath = "../";</script>
+  <script type="application/javascript" src="../pc.js"></script>
+  <script type="application/javascript" src="../blacksilence.js"></script>
+  <script type="application/javascript" src="identityPcTest.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  title: "setIdentityProvider leads to peerIdentity and assertions in SDP",
+  bug: "942367"
+});
+
+runNetworkTest(identityPcTest);
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/identity/test_setIdentityProvider.html b/dom/media/webrtc/tests/mochitests/identity/test_setIdentityProvider.html
new file mode 100644
index 0000000000..ac7cba6a5e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/test_setIdentityProvider.html
@@ -0,0 +1,67 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript">var scriptRelativePath = "../";</script>
+  <script type="application/javascript" src="../pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "setIdentityProvider leads to peerIdentity and assertions in SDP",
+    bug: "942367"
+  });
+
+function checkIdentity(peer, prefix, idp, name) {
+  prefix = prefix + ": ";
+  return peer._pc.peerIdentity.then(peerIdentity => {
+    ok(peerIdentity, prefix + "peerIdentity is set");
+    is(peerIdentity.idp, idp, prefix + "IdP is correct");
+    is(peerIdentity.name, name + "@" + idp, prefix + "identity is correct");
+  });
+}
+
+function theTest() {
+  var test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.pcLocal.setIdentityProvider("test1.example.com",
+                                   { protocol: "idp.js",
+                                     usernameHint: "someone" });
+  test.pcRemote.setIdentityProvider("test2.example.com",
+                                    { protocol: "idp.js",
+                                      usernameHinte: "someone"});
+
+  test.chain.append([
+    function PC_LOCAL_PEER_IDENTITY_IS_SET_CORRECTLY(test) {
+      return checkIdentity(test.pcLocal, "local", "test2.example.com", "someone");
+    },
+    function PC_REMOTE_PEER_IDENTITY_IS_SET_CORRECTLY(test) {
+      return checkIdentity(test.pcRemote, "remote", "test1.example.com", "someone");
+    },
+
+    function OFFER_AND_ANSWER_INCLUDES_IDENTITY(test) {
+      ok(test.originalOffer.sdp.includes("a=identity"), "a=identity is in the offer SDP");
+      ok(test.originalAnswer.sdp.includes("a=identity"), "a=identity is in the answer SDP");
+    },
+
+    function PC_LOCAL_DESCRIPTIONS_CONTAIN_IDENTITY(test) {
+      ok(test.pcLocal.localDescription.sdp.includes("a=identity"),
+         "a=identity is in the local copy of the offer");
+      ok(test.pcLocal.remoteDescription.sdp.includes("a=identity"),
+         "a=identity is in the local copy of the answer");
+    },
+    function PC_REMOTE_DESCRIPTIONS_CONTAIN_IDENTITY(test) {
+      ok(test.pcRemote.localDescription.sdp.includes("a=identity"),
+                         "a=identity is in the remote copy of the offer");
+      ok(test.pcRemote.remoteDescription.sdp.includes("a=identity"),
+                         "a=identity is in the remote copy of the answer");
+    }
+  ]);
+  return test.run();
+}
+runNetworkTest(theTest);
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/identity/test_setIdentityProviderWithErrors.html b/dom/media/webrtc/tests/mochitests/identity/test_setIdentityProviderWithErrors.html
new file mode 100644
index 0000000000..ce6832d1e6
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/identity/test_setIdentityProviderWithErrors.html
@@ -0,0 +1,57 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript">var scriptRelativePath = "../";</script>
+  <script type="application/javascript" src="../pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+'use strict';
+  createHTML({
+    title: "Identity Provider returning errors is handled correctly",
+    bug: "942367"
+  });
+
+runNetworkTest(function () {
+  var test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  // No IdP for local.
+  // Remote generates a bad assertion, but that only fails to validate
+  test.pcRemote.setIdentityProvider('example.com',
+                                    { protocol: 'idp.js#bad-validate',
+                                      usernameHint: 'nobody' });
+
+  // Save the peerIdentity promises now, since when they reject they are
+  // replaced and we expect them to be rejected this time
+  var peerIdentityLocal = test.pcLocal._pc.peerIdentity;
+  var peerIdentityRemote = test.pcRemote._pc.peerIdentity;
+
+  test.chain.append([
+    function ONLY_REMOTE_SDP_INCLUDES_IDENTITY_ASSERTION(t) {
+      ok(!t.originalOffer.sdp.includes('a=identity'),
+         'a=identity not contained in the offer SDP');
+      ok(t.originalAnswer.sdp.includes('a=identity'),
+         'a=identity is contained in the answer SDP');
+    },
+    function PEER_IDENTITY_IS_EMPTY(t) {
+      // we are only waiting for the local side to complete
+      // an error on the remote side is immediately fatal though
+      return Promise.race([
+        peerIdentityLocal.then(
+          () => ok(false, t.pcLocal + ' incorrectly received valid peer identity'),
+          e => ok(e, t.pcLocal + ' correctly failed to validate peer identity')),
+        peerIdentityRemote.then(
+          () => ok(false, t.pcRemote + ' incorrecly received a valid peer identity'),
+          e => ok(false, t.pcRemote + ' incorrectly rejected peer identity'))
+      ]);
+    }
+  ]);
+
+  return test.run();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/mediaStreamPlayback.js b/dom/media/webrtc/tests/mochitests/mediaStreamPlayback.js
new file mode 100644
index 0000000000..44c1c78ea0
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/mediaStreamPlayback.js
@@ -0,0 +1,241 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+const ENDED_TIMEOUT_LENGTH = 30000;
+
+/* The time we wait depends primarily on the canplaythrough event firing
+ * Note: this needs to be at least 30s because the
+ *       B2G emulator in VMs is really slow. */
+const VERIFYPLAYING_TIMEOUT_LENGTH = 60000;
+
+/**
+ * This class manages playback of a HTMLMediaElement with a MediaStream.
+ * When constructed by a caller, an object instance is created with
+ * a media element and a media stream object.
+ *
+ * @param {HTMLMediaElement} mediaElement the media element for playback
+ * @param {MediaStream} mediaStream the media stream used in
+ *                                  the mediaElement for playback
+ */
+function MediaStreamPlayback(mediaElement, mediaStream) {
+  this.mediaElement = mediaElement;
+  this.mediaStream = mediaStream;
+}
+
+MediaStreamPlayback.prototype = {
+  /**
+   * Starts media element with a media stream, runs it until a canplaythrough
+   * and timeupdate event fires, and calls stop() on all its tracks.
+   *
+   * @param {Boolean} isResume specifies if this media element is being resumed
+   *                           from a previous run
+   */
+  playMedia(isResume) {
+    this.startMedia(isResume);
+    return this.verifyPlaying()
+      .then(() => this.stopTracksForStreamInMediaPlayback())
+      .then(() => this.detachFromMediaElement());
+  },
+
+  /**
+   * Stops the local media stream's tracks while it's currently in playback in
+   * a media element.
+   *
+   * Precondition: The media stream and element should both be actively
+   *               being played. All the stream's tracks must be local.
+   */
+  stopTracksForStreamInMediaPlayback() {
+    var elem = this.mediaElement;
+    return Promise.all([
+      haveEvent(
+        elem,
+        "ended",
+        wait(ENDED_TIMEOUT_LENGTH, new Error("Timeout"))
+      ),
+      ...this.mediaStream
+        .getTracks()
+        .map(t => (t.stop(), haveNoEvent(t, "ended"))),
+    ]);
+  },
+
+  /**
+   * Starts media with a media stream, runs it until a canplaythrough and
+   * timeupdate event fires, and detaches from the element without stopping media.
+   *
+   * @param {Boolean} isResume specifies if this media element is being resumed
+   *                           from a previous run
+   */
+  playMediaWithoutStoppingTracks(isResume) {
+    this.startMedia(isResume);
+    return this.verifyPlaying().then(() => this.detachFromMediaElement());
+  },
+
+  /**
+   * Starts the media with the associated stream.
+   *
+   * @param {Boolean} isResume specifies if the media element playback
+   *                           is being resumed from a previous run
+   */
+  startMedia(isResume) {
+    // If we're playing media element for the first time, check that time is zero.
+    if (!isResume) {
+      is(
+        this.mediaElement.currentTime,
+        0,
+        "Before starting the media element, currentTime = 0"
+      );
+    }
+    this.canPlayThroughFired = listenUntil(
+      this.mediaElement,
+      "canplaythrough",
+      () => true
+    );
+
+    // Hooks up the media stream to the media element and starts playing it
+    this.mediaElement.srcObject = this.mediaStream;
+    this.mediaElement.play();
+  },
+
+  /**
+   * Verifies that media is playing.
+   */
+  verifyPlaying() {
+    var lastElementTime = this.mediaElement.currentTime;
+
+    var mediaTimeProgressed = listenUntil(
+      this.mediaElement,
+      "timeupdate",
+      () => this.mediaElement.currentTime > lastElementTime
+    );
+
+    return timeout(
+      Promise.all([this.canPlayThroughFired, mediaTimeProgressed]),
+      VERIFYPLAYING_TIMEOUT_LENGTH,
+      "verifyPlaying timed out"
+    ).then(() => {
+      is(this.mediaElement.paused, false, "Media element should be playing");
+      is(
+        this.mediaElement.duration,
+        Number.POSITIVE_INFINITY,
+        "Duration should be infinity"
+      );
+
+      // When the media element is playing with a real-time stream, we
+      // constantly switch between having data to play vs. queuing up data,
+      // so we can only check that the ready state is one of those two values
+      ok(
+        this.mediaElement.readyState === HTMLMediaElement.HAVE_ENOUGH_DATA ||
+          this.mediaElement.readyState === HTMLMediaElement.HAVE_CURRENT_DATA,
+        "Ready state shall be HAVE_ENOUGH_DATA or HAVE_CURRENT_DATA"
+      );
+
+      is(this.mediaElement.seekable.length, 0, "Seekable length shall be zero");
+      is(this.mediaElement.buffered.length, 0, "Buffered length shall be zero");
+
+      is(
+        this.mediaElement.seeking,
+        false,
+        "MediaElement is not seekable with MediaStream"
+      );
+      ok(
+        isNaN(this.mediaElement.startOffsetTime),
+        "Start offset time shall not be a number"
+      );
+      is(
+        this.mediaElement.defaultPlaybackRate,
+        1,
+        "DefaultPlaybackRate should be 1"
+      );
+      is(this.mediaElement.playbackRate, 1, "PlaybackRate should be 1");
+      is(this.mediaElement.preload, "none", 'Preload should be "none"');
+      is(this.mediaElement.src, "", "No src should be defined");
+      is(
+        this.mediaElement.currentSrc,
+        "",
+        "Current src should still be an empty string"
+      );
+    });
+  },
+
+  /**
+   * Detaches from the element without stopping the media.
+   *
+   * Precondition: The media stream and element should both be actively
+   *               being played.
+   */
+  detachFromMediaElement() {
+    this.mediaElement.pause();
+    this.mediaElement.srcObject = null;
+  },
+};
+
+// haxx to prevent SimpleTest from failing at window.onload
+function addLoadEvent() {}
+
+/* import-globals-from /testing/mochitest/tests/SimpleTest/SimpleTest.js */
+/* import-globals-from head.js */
+const scriptsReady = Promise.all(
+  ["/tests/SimpleTest/SimpleTest.js", "head.js"].map(script => {
+    const el = document.createElement("script");
+    el.src = script;
+    document.head.appendChild(el);
+    return new Promise(r => (el.onload = r));
+  })
+);
+
+function createHTML(options) {
+  return scriptsReady.then(() => realCreateHTML(options));
+}
+
+async function runTest(testFunction) {
+  await Promise.all([
+    scriptsReady,
+    SpecialPowers.pushPrefEnv({
+      set: [["media.navigator.permission.fake", true]],
+    }),
+  ]);
+  await runTestWhenReady(async (...args) => {
+    await testFunction(...args);
+    await noGum();
+  });
+}
+
+// noGum - Helper to detect whether active guM tracks still exist.
+//
+// Note it relies on the permissions system to detect active tracks, so it won't
+// catch getUserMedia use while media.navigator.permission.disabled is true
+// (which is common in automation), UNLESS we set
+// media.navigator.permission.fake to true also, like runTest() does above.
+async function noGum() {
+  if (!navigator.mediaDevices) {
+    // No mediaDevices, then gUM cannot have been called either.
+    return;
+  }
+  const mediaManagerService = Cc[
+    "@mozilla.org/mediaManagerService;1"
+  ].getService(Ci.nsIMediaManagerService);
+
+  const hasCamera = {};
+  const hasMicrophone = {};
+  mediaManagerService.mediaCaptureWindowState(
+    window,
+    hasCamera,
+    hasMicrophone,
+    {},
+    {},
+    {},
+    {},
+    false
+  );
+  is(
+    hasCamera.value,
+    mediaManagerService.STATE_NOCAPTURE,
+    "Test must leave no active camera gUM tracks behind."
+  );
+  is(
+    hasMicrophone.value,
+    mediaManagerService.STATE_NOCAPTURE,
+    "Test must leave no active microphone gUM tracks behind."
+  );
+}
diff --git a/dom/media/webrtc/tests/mochitests/mochitest.ini b/dom/media/webrtc/tests/mochitests/mochitest.ini
new file mode 100644
index 0000000000..d1a9800984
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/mochitest.ini
@@ -0,0 +1,65 @@
+[DEFAULT]
+tags = mtg webrtc
+subsuite = media
+scheme = https
+support-files =
+  head.js
+  dataChannel.js
+  mediaStreamPlayback.js
+  network.js
+  nonTrickleIce.js
+  pc.js
+  stats.js
+  templates.js
+  test_enumerateDevices_iframe.html
+  test_enumerateDevices_iframe_pre_gum.html
+  test_getUserMedia_permission_iframe.html
+  NetworkPreparationChromeScript.js
+  blacksilence.js
+  turnConfig.js
+  sdpUtils.js
+  addTurnsSelfsignedCert.js
+  parser_rtp.js
+  peerconnection_audio_forced_sample_rate.js
+  iceTestUtils.js
+  simulcast.js
+  helpers_from_wpt/sdp.js
+  !/dom/canvas/test/captureStream_common.js
+  !/dom/canvas/test/webgl-mochitest/webgl-util.js
+  !/dom/media/test/manifest.js
+  !/dom/media/test/seek.webm
+  !/dom/media/test/gizmo.mp4
+  !/docshell/test/navigation/blank.html
+prefs =
+  focusmanager.testmode=true # emulate focus
+  privacy.partition.network_state=false
+  network.proxy.allow_hijacking_localhost=true
+  media.devices.enumerate.legacy.enabled=false
+
+[test_1488832.html]
+skip-if =
+  os == 'linux' # Bug 1714410
+[test_1717318.html]
+[test_a_noOp.html]
+scheme=http
+[test_enumerateDevices.html]
+[test_enumerateDevices_getUserMediaFake.html]
+[test_enumerateDevices_legacy.html]
+[test_enumerateDevices_navigation.html]
+skip-if = true # Disabled because it is a racy test and causes timeouts, see bug 1650932
+[test_fingerprinting_resistance.html]
+skip-if =
+  os == "linux" && asan # Bug 1646309 - low frequency intermittent
+[test_forceSampleRate.html]
+scheme=http
+[test_groupId.html]
+[test_multi_mics.html]
+skip-if = os == 'android'
+[test_ondevicechange.html]
+run-sequentially = sets prefs that may disrupt other tests
+[test_setSinkId.html]
+skip-if =
+  os != 'linux' # the only platform with real devices
+[test_setSinkId_default_addTrack.html]
+[test_setSinkId_preMutedElement.html]
+[test_unfocused_pref.html]
diff --git a/dom/media/webrtc/tests/mochitests/mochitest_datachannel.ini b/dom/media/webrtc/tests/mochitests/mochitest_datachannel.ini
new file mode 100644
index 0000000000..881421af0e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/mochitest_datachannel.ini
@@ -0,0 +1,52 @@
+[DEFAULT]
+tags = mtg webrtc
+subsuite = media
+scheme = https
+support-files =
+  head.js
+  dataChannel.js
+  mediaStreamPlayback.js
+  network.js
+  nonTrickleIce.js
+  pc.js
+  stats.js
+  templates.js
+  test_enumerateDevices_iframe.html
+  test_getUserMedia_permission_iframe.html
+  NetworkPreparationChromeScript.js
+  blacksilence.js
+  turnConfig.js
+  sdpUtils.js
+  addTurnsSelfsignedCert.js
+  parser_rtp.js
+  peerconnection_audio_forced_sample_rate.js
+  iceTestUtils.js
+  simulcast.js
+  helpers_from_wpt/sdp.js
+  !/dom/canvas/test/captureStream_common.js
+  !/dom/canvas/test/webgl-mochitest/webgl-util.js
+  !/dom/media/test/manifest.js
+  !/dom/media/test/seek.webm
+  !/dom/media/test/gizmo.mp4
+  !/docshell/test/navigation/blank.html
+prefs =
+  focusmanager.testmode=true # emulate focus
+  privacy.partition.network_state=false
+  network.proxy.allow_hijacking_localhost=true
+
+[test_dataChannel_basicAudio.html]
+[test_dataChannel_basicAudioVideo.html]
+[test_dataChannel_basicAudioVideoCombined.html]
+[test_dataChannel_basicAudioVideoNoBundle.html]
+[test_dataChannel_basicDataOnly.html]
+[test_dataChannel_basicVideo.html]
+[test_dataChannel_bug1013809.html]
+[test_dataChannel_dataOnlyBufferedAmountLow.html]
+scheme=http
+[test_dataChannel_dtlsVersions.html]
+[test_dataChannel_hostnameObfuscation.html]
+scheme=http
+[test_dataChannel_noOffer.html]
+scheme=http
+[test_dataChannel_stats.html]
+scheme=http
diff --git a/dom/media/webrtc/tests/mochitests/mochitest_getusermedia.ini b/dom/media/webrtc/tests/mochitests/mochitest_getusermedia.ini
new file mode 100644
index 0000000000..02c8272b5b
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/mochitest_getusermedia.ini
@@ -0,0 +1,105 @@
+[DEFAULT]
+tags = mtg webrtc
+subsuite = media
+scheme = https
+support-files =
+  head.js
+  dataChannel.js
+  mediaStreamPlayback.js
+  network.js
+  nonTrickleIce.js
+  pc.js
+  stats.js
+  templates.js
+  test_enumerateDevices_iframe.html
+  test_getUserMedia_permission_iframe.html
+  NetworkPreparationChromeScript.js
+  blacksilence.js
+  turnConfig.js
+  sdpUtils.js
+  addTurnsSelfsignedCert.js
+  parser_rtp.js
+  peerconnection_audio_forced_sample_rate.js
+  iceTestUtils.js
+  simulcast.js
+  helpers_from_wpt/sdp.js
+  !/dom/canvas/test/captureStream_common.js
+  !/dom/canvas/test/webgl-mochitest/webgl-util.js
+  !/dom/media/test/manifest.js
+  !/dom/media/test/seek.webm
+  !/dom/media/test/gizmo.mp4
+  !/docshell/test/navigation/blank.html
+prefs =
+  focusmanager.testmode=true # emulate focus
+  privacy.partition.network_state=false
+  network.proxy.allow_hijacking_localhost=true
+  media.devices.enumerate.legacy.enabled=false
+
+[test_defaultAudioConstraints.html]
+skip-if = os == 'mac'
+  os == 'win'
+  toolkit == 'android' # Bug 1404995, no loopback devices on some platforms
+[test_getUserMedia_GC_MediaStream.html]
+[test_getUserMedia_active_autoplay.html]
+[test_getUserMedia_addTrackRemoveTrack.html]
+[test_getUserMedia_addtrack_removetrack_events.html]
+[test_getUserMedia_audioCapture.html]
+skip-if = toolkit == 'android'
+  (os == "win" && processor == "aarch64") # android(Bug 1189784, timeouts on 4.3 emulator), android(Bug 1264333), aarch64 due to 1538359
+[test_getUserMedia_audioConstraints.html]
+skip-if = os == 'mac'
+  os == 'win'
+  toolkit == 'android' # Bug 1404995, no loopback devices on some platforms
+[test_getUserMedia_audioConstraints_concurrentIframes.html]
+skip-if = os == 'mac'
+  os == 'win'
+  toolkit == 'android'
+  (os == 'linux' && debug)  # Bug 1404995, no loopback devices on some platforms # Bug 1481101
+  os == "linux" && !debug && !fission # bug 1645930, lower frequency intermittent
+[test_getUserMedia_audioConstraints_concurrentStreams.html]
+skip-if = os == 'mac'
+  os == 'win'
+  toolkit == 'android' # Bug 1404995, no loopback devices on some platforms
+[test_getUserMedia_basicAudio.html]
+[test_getUserMedia_basicAudio_loopback.html]
+skip-if = os == 'mac'
+  os == 'win'
+  toolkit == 'android' # Bug 1404995, no loopback devices on some platforms
+[test_getUserMedia_basicScreenshare.html]
+skip-if =
+  toolkit == 'android'     # no screenshare on android
+  apple_silicon            # bug 1707742
+[test_getUserMedia_basicTabshare.html]
+skip-if = 
+  toolkit == 'android' # no windowshare on android
+[test_getUserMedia_basicVideo.html]
+[test_getUserMedia_basicVideoAudio.html]
+[test_getUserMedia_basicVideo_playAfterLoadedmetadata.html]
+[test_getUserMedia_basicWindowshare.html]
+skip-if = toolkit == 'android' # no windowshare on android
+[test_getUserMedia_bug1223696.html]
+[test_getUserMedia_callbacks.html]
+[test_getUserMedia_constraints.html]
+[test_getUserMedia_cubebDisabled.html]
+[test_getUserMedia_cubebDisabledFakeStreams.html]
+[test_getUserMedia_getTrackById.html]
+[test_getUserMedia_gumWithinGum.html]
+[test_getUserMedia_loadedmetadata.html]
+[test_getUserMedia_mediaElementCapture_audio.html]
+[test_getUserMedia_mediaElementCapture_tracks.html]
+[test_getUserMedia_mediaElementCapture_video.html]
+[test_getUserMedia_mediaStreamClone.html]
+[test_getUserMedia_mediaStreamConstructors.html]
+[test_getUserMedia_mediaStreamTrackClone.html]
+[test_getUserMedia_nonDefaultRate.html]
+[test_getUserMedia_peerIdentity.html]
+[test_getUserMedia_permission.html]
+[test_getUserMedia_playAudioTwice.html]
+[test_getUserMedia_playVideoAudioTwice.html]
+[test_getUserMedia_playVideoTwice.html]
+[test_getUserMedia_scarySources.html]
+skip-if = toolkit == 'android' # no screenshare or windowshare on android
+[test_getUserMedia_spinEventLoop.html]
+[test_getUserMedia_trackCloneCleanup.html]
+[test_getUserMedia_trackEnded.html]
+
diff --git a/dom/media/webrtc/tests/mochitests/mochitest_peerconnection.ini b/dom/media/webrtc/tests/mochitests/mochitest_peerconnection.ini
new file mode 100644
index 0000000000..37fb551435
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/mochitest_peerconnection.ini
@@ -0,0 +1,311 @@
+[DEFAULT]
+tags = mtg webrtc
+subsuite = media
+scheme = https
+support-files =
+  head.js
+  dataChannel.js
+  mediaStreamPlayback.js
+  network.js
+  nonTrickleIce.js
+  pc.js
+  stats.js
+  templates.js
+  test_enumerateDevices_iframe.html
+  test_getUserMedia_permission_iframe.html
+  NetworkPreparationChromeScript.js
+  blacksilence.js
+  turnConfig.js
+  sdpUtils.js
+  addTurnsSelfsignedCert.js
+  parser_rtp.js
+  peerconnection_audio_forced_sample_rate.js
+  iceTestUtils.js
+  simulcast.js
+  helpers_from_wpt/sdp.js
+  !/dom/canvas/test/captureStream_common.js
+  !/dom/canvas/test/webgl-mochitest/webgl-util.js
+  !/dom/media/test/manifest.js
+  !/dom/media/test/seek.webm
+  !/dom/media/test/gizmo.mp4
+  !/docshell/test/navigation/blank.html
+prefs =
+  focusmanager.testmode=true # emulate focus
+  privacy.partition.network_state=false
+  network.proxy.allow_hijacking_localhost=true
+  media.devices.enumerate.legacy.enabled=false
+
+[test_peerConnection_addAudioTrackToExistingVideoStream.html]
+[test_peerConnection_addDataChannel.html]
+[test_peerConnection_addDataChannelNoBundle.html]
+[test_peerConnection_addSecondAudioStream.html]
+[test_peerConnection_addSecondAudioStreamNoBundle.html]
+[test_peerConnection_addSecondVideoStream.html]
+[test_peerConnection_addSecondVideoStreamNoBundle.html]
+[test_peerConnection_addtrack_removetrack_events.html]
+[test_peerConnection_answererAddSecondAudioStream.html]
+[test_peerConnection_audioChannels.html]
+[test_peerConnection_audioCodecs.html]
+[test_peerConnection_audioContributingSources.html]
+[test_peerConnection_audioRenegotiationInactiveAnswer.html]
+[test_peerConnection_audioSynchronizationSources.html]
+[test_peerConnection_audioSynchronizationSourcesUnidirectional.html]
+[test_peerConnection_basicAudio.html]
+[test_peerConnection_basicAudioDynamicPtMissingRtpmap.html]
+[test_peerConnection_basicAudioNATRelay.html]
+skip-if =
+    toolkit == 'android' # websockets don't work on android (bug 1266217)
+scheme=http
+[test_peerConnection_basicAudioNATRelayTCP.html]
+skip-if =
+    toolkit == 'android' # websockets don't work on android (bug 1266217)
+    (os == 'win' && os_version == '6.1') # WinError 10048
+scheme=http
+[test_peerConnection_basicAudioNATRelayTCPWithStun300.html]
+skip-if =
+    toolkit == 'android' # websockets don't work on android (bug 1266217)
+    (os == 'win' && os_version == '6.1') # WinError 10048
+scheme=http
+[test_peerConnection_basicAudioNATRelayTLS.html]
+skip-if =
+    toolkit == 'android' # websockets don't work on android (bug 1266217)
+    (os == 'win' && os_version == '6.1') # WinError 10048
+scheme=http
+[test_peerConnection_basicAudioNATRelayWithStun300.html]
+skip-if =
+    toolkit == 'android' # websockets don't work on android (bug 1266217)
+    (os == 'win' && os_version == '6.1') # WinError 10048
+scheme=http
+[test_peerConnection_basicAudioNATSrflx.html]
+skip-if =
+    toolkit == 'android' # websockets don't work on android (bug 1266217)
+    (os == 'win' && os_version == '6.1') # WinError 10048
+scheme=http
+[test_peerConnection_basicAudioNoisyUDPBlock.html]
+skip-if =
+    toolkit == 'android' # websockets don't work on android (bug 1266217)
+    (os == 'win' && os_version == '6.1') # WinError 10048
+scheme=http
+[test_peerConnection_basicAudioPcmaPcmuOnly.html]
+[test_peerConnection_basicAudioRelayPolicy.html]
+skip-if =
+    toolkit == 'android' # websockets don't work on android (bug 1266217)
+    (os == 'win' && os_version == '6.1') # WinError 10048
+scheme=http
+[test_peerConnection_basicAudioRequireEOC.html]
+[test_peerConnection_basicAudioVerifyRtpHeaderExtensions.html]
+[test_peerConnection_basicAudioVideo.html]
+[test_peerConnection_basicAudioVideoCombined.html]
+[test_peerConnection_basicAudioVideoNoBundle.html]
+[test_peerConnection_basicAudioVideoNoBundleNoRtcpMux.html]
+[test_peerConnection_basicAudioVideoNoRtcpMux.html]
+[test_peerConnection_basicAudioVideoTransceivers.html]
+[test_peerConnection_basicAudioVideoVerifyExtmap.html]
+[test_peerConnection_basicAudioVideoVerifyExtmapSendonly.html]
+[test_peerConnection_basicAudioVideoVerifyTooLongMidFails.html]
+[test_peerConnection_basicAudio_forced_higher_rate.html]
+[test_peerConnection_basicAudio_forced_lower_rate.html]
+[test_peerConnection_basicH264Video.html]
+skip-if =
+    toolkit == 'android' && is_emulator  # Bug 1355786, No h264 support on android emulator
+[test_peerConnection_basicScreenshare.html]
+skip-if = toolkit == 'android' # no screenshare on android
+[test_peerConnection_basicVideo.html]
+[test_peerConnection_basicVideoVerifyRtpHeaderExtensions.html]
+[test_peerConnection_basicWindowshare.html]
+skip-if = toolkit == 'android' # no screenshare on android
+[test_peerConnection_bug1013809.html]
+[test_peerConnection_bug1042791.html]
+skip-if = (toolkit == 'android' && is_emulator) # Bug 1355786, No h264 support on android emulator
+[test_peerConnection_bug1227781.html]
+scheme=http
+[test_peerConnection_bug1512281.html]
+fail-if = 1
+[test_peerConnection_bug1773067.html]
+[test_peerConnection_bug822674.html]
+scheme=http
+[test_peerConnection_bug825703.html]
+scheme=http
+[test_peerConnection_bug827843.html]
+[test_peerConnection_bug834153.html]
+scheme=http
+[test_peerConnection_callbacks.html]
+[test_peerConnection_captureStream_canvas_2d.html]
+scheme=http
+[test_peerConnection_captureStream_canvas_2d_noSSRC.html]
+scheme=http
+[test_peerConnection_captureStream_canvas_webgl.html]
+scheme=http
+[test_peerConnection_capturedVideo.html]
+tags=capturestream
+skip-if = toolkit == 'android' # android(Bug 1189784, timeouts on 4.3 emulator), Bug 1264340
+[test_peerConnection_certificates.html]
+[test_peerConnection_checkPacketDumpHook.html]
+[test_peerConnection_close.html]
+scheme=http
+[test_peerConnection_closeDuringIce.html]
+[test_peerConnection_codecNegotiationFailure.html]
+[test_peerConnection_constructedStream.html]
+[test_peerConnection_disabledVideoPreNegotiation.html]
+[test_peerConnection_encodingsNegotiation.html]
+[test_peerConnection_errorCallbacks.html]
+scheme=http
+[test_peerConnection_extmapRenegotiation.html]
+[test_peerConnection_forwarding_basicAudioVideoCombined.html]
+skip-if = toolkit == 'android'  # Bug 1189784
+[test_peerConnection_gatherWithSetConfiguration.html]
+skip-if =
+    toolkit == 'android' # websockets don't work on android (bug 1266217)
+    (os == 'win' && os_version == '6.1') # WinError 10048
+scheme=http
+[test_peerConnection_gatherWithStun300.html]
+skip-if =
+    toolkit == 'android' # websockets don't work on android (bug 1266217)
+    (os == 'win' && os_version == '6.1') # WinError 10048
+scheme=http
+[test_peerConnection_gatherWithStun300IPv6.html]
+skip-if =
+    toolkit == 'android' # websockets don't work on android (bug 1266217)
+    os == 'mac' # no ipv6 support on OS X testers (bug 1710706)
+    os == 'win' # no ipv6 support on windows testers (bug 1710706)
+scheme=http
+[test_peerConnection_glean.html]
+[test_peerConnection_iceFailure.html]
+skip-if = true # (Bug 1180388 for win, mac and linux), android(Bug 1189784), Bug 1180388
+scheme=http
+[test_peerConnection_insertDTMF.html]
+[test_peerConnection_localReofferRollback.html]
+[test_peerConnection_localRollback.html]
+[test_peerConnection_maxFsConstraint.html]
+[test_peerConnection_multiple_captureStream_canvas_2d.html]
+scheme=http
+[test_peerConnection_noTrickleAnswer.html]
+[test_peerConnection_noTrickleOffer.html]
+[test_peerConnection_noTrickleOfferAnswer.html]
+[test_peerConnection_nonDefaultRate.html]
+[test_peerConnection_offerRequiresReceiveAudio.html]
+[test_peerConnection_offerRequiresReceiveVideo.html]
+[test_peerConnection_offerRequiresReceiveVideoAudio.html]
+[test_peerConnection_portRestrictions.html]
+[test_peerConnection_promiseSendOnly.html]
+[test_peerConnection_recordReceiveTrack.html]
+[test_peerConnection_relayOnly.html]
+disabled=bug 1612063 # test is racy
+[test_peerConnection_remoteReofferRollback.html]
+[test_peerConnection_remoteRollback.html]
+[test_peerConnection_removeAudioTrack.html]
+[test_peerConnection_removeThenAddAudioTrack.html]
+[test_peerConnection_removeThenAddAudioTrackNoBundle.html]
+[test_peerConnection_removeThenAddVideoTrack.html]
+[test_peerConnection_removeThenAddVideoTrackNoBundle.html]
+[test_peerConnection_removeVideoTrack.html]
+[test_peerConnection_renderAfterRenegotiation.html]
+scheme=http
+[test_peerConnection_replaceNullTrackThenRenegotiateAudio.html]
+[test_peerConnection_replaceNullTrackThenRenegotiateVideo.html]
+[test_peerConnection_replaceTrack.html]
+[test_peerConnection_replaceTrack_camera.html]
+skip-if = toolkit == 'android' # Bug 1614460
+[test_peerConnection_replaceTrack_disabled.html]
+skip-if =
+  toolkit == 'android'        # Bug 1614460
+[test_peerConnection_replaceTrack_microphone.html]
+[test_peerConnection_replaceVideoThenRenegotiate.html]
+[test_peerConnection_restartIce.html]
+[test_peerConnection_restartIceBadAnswer.html]
+[test_peerConnection_restartIceLocalAndRemoteRollback.html]
+[test_peerConnection_restartIceLocalAndRemoteRollbackNoSubsequentRestart.html]
+[test_peerConnection_restartIceLocalRollback.html]
+[test_peerConnection_restartIceLocalRollbackNoSubsequentRestart.html]
+[test_peerConnection_restartIceNoBundle.html]
+[test_peerConnection_restartIceNoBundleNoRtcpMux.html]
+[test_peerConnection_restartIceNoRtcpMux.html]
+[test_peerConnection_restrictBandwidthTargetBitrate.html]
+[test_peerConnection_restrictBandwidthWithTias.html]
+[test_peerConnection_rtcp_rsize.html]
+[test_peerConnection_scaleResolution.html]
+[test_peerConnection_scaleResolution_oldSetParameters.html]
+[test_peerConnection_sender_and_receiver_stats.html]
+[test_peerConnection_setLocalAnswerInHaveLocalOffer.html]
+[test_peerConnection_setLocalAnswerInStable.html]
+[test_peerConnection_setLocalOfferInHaveRemoteOffer.html]
+[test_peerConnection_setParameters.html]
+[test_peerConnection_setParameters_maxFramerate.html]
+[test_peerConnection_setParameters_maxFramerate_oldSetParameters.html]
+[test_peerConnection_setParameters_oldSetParameters.html]
+[test_peerConnection_setParameters_scaleResolutionDownBy.html]
+skip-if = (os == 'win' && processor == 'aarch64') # aarch64 due to bug 1537567
+[test_peerConnection_setParameters_scaleResolutionDownBy_oldSetParameters.html]
+skip-if = (os == 'win' && processor == 'aarch64') # aarch64 due to bug 1537567
+[test_peerConnection_setRemoteAnswerInHaveRemoteOffer.html]
+[test_peerConnection_setRemoteAnswerInStable.html]
+[test_peerConnection_setRemoteOfferInHaveLocalOffer.html]
+[test_peerConnection_simulcastAnswer.html]
+skip-if = toolkit == 'android' # no simulcast support on android
+[test_peerConnection_simulcastAnswer_lowResFirst.html]
+skip-if = toolkit == 'android' # no simulcast support on android
+[test_peerConnection_simulcastAnswer_lowResFirst_oldSetParameters.html]
+skip-if = toolkit == 'android' # no simulcast support on android
+[test_peerConnection_simulcastAnswer_oldSetParameters.html]
+skip-if = toolkit == 'android' # no simulcast support on android
+[test_peerConnection_simulcastOddResolution.html]
+skip-if = toolkit == 'android' # no simulcast support on android
+[test_peerConnection_simulcastOddResolution_oldSetParameters.html]
+skip-if = toolkit == 'android' # no simulcast support on android
+[test_peerConnection_simulcastOffer.html]
+skip-if = toolkit == 'android' # no simulcast support on android
+[test_peerConnection_simulcastOffer_lowResFirst.html]
+skip-if = toolkit == 'android' # no simulcast support on android
+[test_peerConnection_simulcastOffer_lowResFirst_oldSetParameters.html]
+skip-if = toolkit == 'android' # no simulcast support on android
+[test_peerConnection_simulcastOffer_oldSetParameters.html]
+skip-if = toolkit == 'android' # no simulcast support on android
+[test_peerConnection_stats.html]
+[test_peerConnection_stats_jitter.html]
+skip-if = tsan # Bug 1672590, TSan is just too slow to pass this test
+[test_peerConnection_stats_oneway.html]
+[test_peerConnection_stats_relayProtocol.html]
+skip-if =
+    toolkit == 'android' # android(Bug 1189784, timeouts on 4.3 emulator, Bug 1373858, Bug 1521117)
+    socketprocess_e10s
+    (os == 'win' && os_version == '6.1') # WinError 10048
+scheme=http
+[test_peerConnection_stereoFmtpPref.html]
+[test_peerConnection_syncSetDescription.html]
+[test_peerConnection_telephoneEventFirst.html]
+[test_peerConnection_threeUnbundledConnections.html]
+[test_peerConnection_throwInCallbacks.html]
+[test_peerConnection_toJSON.html]
+scheme=http
+[test_peerConnection_trackDisabling.html]
+skip-if = toolkit == 'android' # Bug 1614460
+[test_peerConnection_trackDisabling_clones.html]
+[test_peerConnection_trackless_sender_stats.html]
+[test_peerConnection_twoAudioStreams.html]
+[test_peerConnection_twoAudioTracksInOneStream.html]
+[test_peerConnection_twoAudioVideoStreams.html]
+[test_peerConnection_twoAudioVideoStreamsCombined.html]
+skip-if = (toolkit == 'android') || (os == 'linux' && asan) # android(Bug 1189784), Bug 1480942 for Linux asan
+[test_peerConnection_twoAudioVideoStreamsCombinedNoBundle.html]
+skip-if =
+  (toolkit == 'android') # Bug 1189784
+  (os == 'linux' && asan) # Bug 1480942
+  (os == 'win' && processor == 'aarch64') # Bug 1777081
+[test_peerConnection_twoVideoStreams.html]
+[test_peerConnection_twoVideoTracksInOneStream.html]
+[test_peerConnection_verifyAudioAfterRenegotiation.html]
+skip-if =
+  os == "android" && processor == "x86_64" && !debug # Bug 1783287
+[test_peerConnection_verifyDescriptions.html]
+[test_peerConnection_verifyVideoAfterRenegotiation.html]
+[test_peerConnection_videoCodecs.html]
+skip-if =
+  toolkit == 'android' # android(Bug 1614460)
+  win10_2004 && !debug # Bug 1777082
+[test_peerConnection_videoRenegotiationInactiveAnswer.html]
+[test_peerConnection_webAudio.html]
+tags = webaudio webrtc
+scheme=http
+[test_selftest.html]
+# Bug 1227781: Crash with bogus TURN server.
+scheme=http
diff --git a/dom/media/webrtc/tests/mochitests/network.js b/dom/media/webrtc/tests/mochitests/network.js
new file mode 100644
index 0000000000..223721b111
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/network.js
@@ -0,0 +1,16 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+"use strict";
+
+/**
+ * A stub function for preparing the network if needed
+ *
+ */
+async function startNetworkAndTest() {}
+
+/**
+ * A stub function to shutdown the network if needed
+ */
+async function networkTestFinished() {}
diff --git a/dom/media/webrtc/tests/mochitests/nonTrickleIce.js b/dom/media/webrtc/tests/mochitests/nonTrickleIce.js
new file mode 100644
index 0000000000..9361944791
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/nonTrickleIce.js
@@ -0,0 +1,97 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+function removeTrickleOption(desc) {
+  var sdp = desc.sdp.replace(/\r\na=ice-options:trickle\r\n/, "\r\n");
+  return new RTCSessionDescription({ type: desc.type, sdp });
+}
+
+function makeOffererNonTrickle(chain) {
+  chain.replace("PC_LOCAL_SETUP_ICE_HANDLER", [
+    function PC_LOCAL_SETUP_NOTRICKLE_ICE_HANDLER(test) {
+      // We need to install this callback before calling setLocalDescription
+      // otherwise we might miss callbacks
+      test.pcLocal.setupIceCandidateHandler(test, () => {});
+      // We ignore ICE candidates because we want the full offer
+    },
+  ]);
+  chain.replace("PC_REMOTE_GET_OFFER", [
+    function PC_REMOTE_GET_FULL_OFFER(test) {
+      return test.pcLocal.endOfTrickleIce.then(() => {
+        test._local_offer = removeTrickleOption(test.pcLocal.localDescription);
+        test._offer_constraints = test.pcLocal.constraints;
+        test._offer_options = test.pcLocal.offerOptions;
+      });
+    },
+  ]);
+  chain.insertAfter("PC_REMOTE_SANE_REMOTE_SDP", [
+    function PC_REMOTE_REQUIRE_REMOTE_SDP_CANDIDATES(test) {
+      info(
+        "test.pcLocal.localDescription.sdp: " +
+          JSON.stringify(test.pcLocal.localDescription.sdp)
+      );
+      info("test._local_offer.sdp" + JSON.stringify(test._local_offer.sdp));
+      is(
+        test.pcRemote._pc.canTrickleIceCandidates,
+        false,
+        "Remote thinks that trickle isn't supported"
+      );
+      ok(!test.localRequiresTrickleIce, "Local does NOT require trickle");
+      ok(
+        test._local_offer.sdp.includes("a=candidate"),
+        "offer has ICE candidates"
+      );
+      ok(
+        test._local_offer.sdp.includes("a=end-of-candidates"),
+        "offer has end-of-candidates"
+      );
+    },
+  ]);
+  chain.remove("PC_REMOTE_CHECK_CAN_TRICKLE_SYNC");
+}
+
+function makeAnswererNonTrickle(chain) {
+  chain.replace("PC_REMOTE_SETUP_ICE_HANDLER", [
+    function PC_REMOTE_SETUP_NOTRICKLE_ICE_HANDLER(test) {
+      // We need to install this callback before calling setLocalDescription
+      // otherwise we might miss callbacks
+      test.pcRemote.setupIceCandidateHandler(test, () => {});
+      // We ignore ICE candidates because we want the full offer
+    },
+  ]);
+  chain.replace("PC_LOCAL_GET_ANSWER", [
+    function PC_LOCAL_GET_FULL_ANSWER(test) {
+      return test.pcRemote.endOfTrickleIce.then(() => {
+        test._remote_answer = removeTrickleOption(
+          test.pcRemote.localDescription
+        );
+        test._answer_constraints = test.pcRemote.constraints;
+      });
+    },
+  ]);
+  chain.insertAfter("PC_LOCAL_SANE_REMOTE_SDP", [
+    function PC_LOCAL_REQUIRE_REMOTE_SDP_CANDIDATES(test) {
+      info(
+        "test.pcRemote.localDescription.sdp: " +
+          JSON.stringify(test.pcRemote.localDescription.sdp)
+      );
+      info("test._remote_answer.sdp" + JSON.stringify(test._remote_answer.sdp));
+      is(
+        test.pcLocal._pc.canTrickleIceCandidates,
+        false,
+        "Local thinks that trickle isn't supported"
+      );
+      ok(!test.remoteRequiresTrickleIce, "Remote does NOT require trickle");
+      ok(
+        test._remote_answer.sdp.includes("a=candidate"),
+        "answer has ICE candidates"
+      );
+      ok(
+        test._remote_answer.sdp.includes("a=end-of-candidates"),
+        "answer has end-of-candidates"
+      );
+    },
+  ]);
+  chain.remove("PC_LOCAL_CHECK_CAN_TRICKLE_SYNC");
+}
diff --git a/dom/media/webrtc/tests/mochitests/parser_rtp.js b/dom/media/webrtc/tests/mochitests/parser_rtp.js
new file mode 100644
index 0000000000..2275c1f787
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/parser_rtp.js
@@ -0,0 +1,131 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+"use strict";
+
+/*
+ * Parses an RTP packet
+ * @param buffer an ArrayBuffer that contains the packet
+ * @return { type: "rtp", header: {...}, payload: a DataView }
+ */
+var ParseRtpPacket = buffer => {
+  // DataView.getFooInt returns big endian numbers by default
+  let view = new DataView(buffer);
+
+  // Standard Header Fields
+  // https://tools.ietf.org/html/rfc3550#section-5.1
+  //  0                   1                   2                   3
+  //  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  // |V=2|P|X|  CC   |M|     PT      |       sequence number         |
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  // |                           timestamp                           |
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  // |           synchronization source (SSRC) identifier            |
+  // +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+  // |            contributing source (CSRC) identifiers             |
+  // |                             ....                              |
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+  let header = {};
+  let offset = 0;
+  // Note that incrementing the offset happens as close to reading the data as
+  // possible. This simplifies ensuring that the number of read bytes and the
+  // offset increment match. Data may be manipulated between when the offset is
+  // incremented and before the next read.
+  let byte = view.getUint8(offset);
+  offset++;
+  // Version            2 Bit
+  header.version = (0xc0 & byte) >> 6;
+  // Padding            1 Bit
+  header.padding = (0x30 & byte) >> 5;
+  // Extension          1 Bit
+  header.extensionsPresent = (0x10 & byte) >> 4 == 1;
+  // CSRC count         4 Bit
+  header.csrcCount = 0xf & byte;
+
+  byte = view.getUint8(offset);
+  offset++;
+  // Marker             1 Bit
+  header.marker = (0x80 & byte) >> 7;
+  // Payload Type       7 Bit
+  header.payloadType = 0x7f & byte;
+  // Sequence Number   16 Bit
+  header.sequenceNumber = view.getUint16(offset);
+  offset += 2;
+  // Timestamp         32 Bit
+  header.timestamp = view.getUint32(offset);
+  offset += 4;
+  // SSRC              32 Bit
+  header.ssrc = view.getUint32(offset);
+  offset += 4;
+
+  // CSRC              32 Bit
+  header.csrcs = [];
+  for (let c = 0; c < header.csrcCount; c++) {
+    header.csrcs.push(view.getUint32(offset));
+    offset += 4;
+  }
+
+  // Extensions
+  header.extensions = [];
+  header.extensionPaddingBytes = 0;
+  header.extensionsTotalLength = 0;
+  if (header.extensionsPresent) {
+    // https://tools.ietf.org/html/rfc3550#section-5.3.1
+    //  0                   1                   2                   3
+    //  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    // |      defined by profile       |           length              |
+    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    // |                        header extension                       |
+    // |                             ....                              |
+    let addExtension = (id, len) =>
+      header.extensions.push({
+        id,
+        data: new DataView(buffer, offset, len),
+      });
+    let extensionId = view.getUint16(offset);
+    offset += 2;
+    // len is in 32 bit units, not bytes
+    header.extensionsTotalLength = view.getUint16(offset) * 4;
+    offset += 2;
+    // Check for https://tools.ietf.org/html/rfc5285
+    if (extensionId != 0xbede) {
+      // No rfc5285
+      addExtension(extensionId, header.extensionsTotalLength);
+      offset += header.extensionsTotalLength;
+    } else {
+      let expectedEnd = offset + header.extensionsTotalLength;
+      while (offset < expectedEnd) {
+        // We only support "one-byte" extension headers ATM
+        // https://tools.ietf.org/html/rfc5285#section-4.2
+        //  0
+        //  0 1 2 3 4 5 6 7
+        // +-+-+-+-+-+-+-+-+
+        // |  ID   |  len  |
+        // +-+-+-+-+-+-+-+-+
+        byte = view.getUint8(offset);
+        offset++;
+        // Check for padding which can occur between extensions or at the end
+        if (byte == 0) {
+          header.extensionPaddingBytes++;
+          continue;
+        }
+        let id = (byte & 0xf0) >> 4;
+        // Check for the FORBIDDEN id (15), dun dun dun
+        if (id == 15) {
+          // Ignore bytes until until the end of extensions
+          offset = expectedEnd;
+          break;
+        }
+        // the length of the extention is len + 1
+        let len = (byte & 0x0f) + 1;
+        addExtension(id, len);
+        offset += len;
+      }
+    }
+  }
+  return { type: "rtp", header, payload: new DataView(buffer, offset) };
+};
diff --git a/dom/media/webrtc/tests/mochitests/pc.js b/dom/media/webrtc/tests/mochitests/pc.js
new file mode 100644
index 0000000000..36a923fbed
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/pc.js
@@ -0,0 +1,2495 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+"use strict";
+
+const LOOPBACK_ADDR = "127.0.0.";
+
+const iceStateTransitions = {
+  new: ["checking", "closed"], //Note: 'failed' might need to added here
+  //      even though it is not in the standard
+  checking: ["new", "connected", "failed", "closed"], //Note: do we need to
+  // allow 'completed' in
+  // here as well?
+  connected: ["new", "completed", "disconnected", "closed"],
+  completed: ["new", "disconnected", "closed"],
+  disconnected: ["new", "connected", "completed", "failed", "closed"],
+  failed: ["new", "disconnected", "closed"],
+  closed: [],
+};
+
+const signalingStateTransitions = {
+  stable: ["have-local-offer", "have-remote-offer", "closed"],
+  "have-local-offer": [
+    "have-remote-pranswer",
+    "stable",
+    "closed",
+    "have-local-offer",
+  ],
+  "have-remote-pranswer": ["stable", "closed", "have-remote-pranswer"],
+  "have-remote-offer": [
+    "have-local-pranswer",
+    "stable",
+    "closed",
+    "have-remote-offer",
+  ],
+  "have-local-pranswer": ["stable", "closed", "have-local-pranswer"],
+  closed: [],
+};
+
+var makeDefaultCommands = () => {
+  return [].concat(
+    commandsPeerConnectionInitial,
+    commandsGetUserMedia,
+    commandsPeerConnectionOfferAnswer
+  );
+};
+
+/**
+ * This class handles tests for peer connections.
+ *
+ * @constructor
+ * @param {object} [options={}]
+ *        Optional options for the peer connection test
+ * @param {object} [options.commands=commandsPeerConnection]
+ *        Commands to run for the test
+ * @param {bool}   [options.is_local=true]
+ *        true if this test should run the tests for the "local" side.
+ * @param {bool}   [options.is_remote=true]
+ *        true if this test should run the tests for the "remote" side.
+ * @param {object} [options.config_local=undefined]
+ *        Configuration for the local peer connection instance
+ * @param {object} [options.config_remote=undefined]
+ *        Configuration for the remote peer connection instance. If not defined
+ *        the configuration from the local instance will be used
+ */
+function PeerConnectionTest(options) {
+  // If no options are specified make it an empty object
+  options = options || {};
+  options.commands = options.commands || makeDefaultCommands();
+  options.is_local = "is_local" in options ? options.is_local : true;
+  options.is_remote = "is_remote" in options ? options.is_remote : true;
+
+  options.h264 = "h264" in options ? options.h264 : false;
+  options.bundle = "bundle" in options ? options.bundle : true;
+  options.rtcpmux = "rtcpmux" in options ? options.rtcpmux : true;
+  options.opus = "opus" in options ? options.opus : true;
+  options.ssrc = "ssrc" in options ? options.ssrc : true;
+
+  options.config_local = options.config_local || {};
+  options.config_remote = options.config_remote || {};
+
+  if (!options.bundle) {
+    // Make sure neither end tries to use bundle-only!
+    options.config_local.bundlePolicy = "max-compat";
+    options.config_remote.bundlePolicy = "max-compat";
+  }
+
+  if (iceServersArray.length) {
+    if (!options.turn_disabled_local && !options.config_local.iceServers) {
+      options.config_local.iceServers = iceServersArray;
+    }
+    if (!options.turn_disabled_remote && !options.config_remote.iceServers) {
+      options.config_remote.iceServers = iceServersArray;
+    }
+  } else if (typeof turnServers !== "undefined") {
+    if (!options.turn_disabled_local && turnServers.local) {
+      if (!options.config_local.hasOwnProperty("iceServers")) {
+        options.config_local.iceServers = turnServers.local.iceServers;
+      }
+    }
+    if (!options.turn_disabled_remote && turnServers.remote) {
+      if (!options.config_remote.hasOwnProperty("iceServers")) {
+        options.config_remote.iceServers = turnServers.remote.iceServers;
+      }
+    }
+  }
+
+  if (options.is_local) {
+    this.pcLocal = new PeerConnectionWrapper("pcLocal", options.config_local);
+  } else {
+    this.pcLocal = null;
+  }
+
+  if (options.is_remote) {
+    this.pcRemote = new PeerConnectionWrapper(
+      "pcRemote",
+      options.config_remote || options.config_local
+    );
+  } else {
+    this.pcRemote = null;
+  }
+
+  // Create command chain instance and assign default commands
+  this.chain = new CommandChain(this, options.commands);
+
+  this.testOptions = options;
+}
+
+/** TODO: consider removing this dependency on timeouts */
+function timerGuard(p, time, message) {
+  return Promise.race([
+    p,
+    wait(time).then(() => {
+      throw new Error("timeout after " + time / 1000 + "s: " + message);
+    }),
+  ]);
+}
+
+/**
+ * Closes the peer connection if it is active
+ */
+PeerConnectionTest.prototype.closePC = function () {
+  info("Closing peer connections");
+
+  var closeIt = pc => {
+    if (!pc || pc.signalingState === "closed") {
+      return Promise.resolve();
+    }
+
+    var promise = Promise.all([
+      Promise.all(
+        pc._pc
+          .getReceivers()
+          .filter(receiver => receiver.track.readyState == "live")
+          .map(receiver => {
+            info(
+              "Waiting for track " +
+                receiver.track.id +
+                " (" +
+                receiver.track.kind +
+                ") to end."
+            );
+            return haveEvent(receiver.track, "ended", wait(50000)).then(
+              event => {
+                is(
+                  event.target,
+                  receiver.track,
+                  "Event target should be the correct track"
+                );
+                info(pc + " ended fired for track " + receiver.track.id);
+              },
+              e =>
+                e
+                  ? Promise.reject(e)
+                  : ok(
+                      false,
+                      "ended never fired for track " + receiver.track.id
+                    )
+            );
+          })
+      ),
+    ]);
+    pc.close();
+    return promise;
+  };
+
+  return timerGuard(
+    Promise.all([closeIt(this.pcLocal), closeIt(this.pcRemote)]),
+    60000,
+    "failed to close peer connection"
+  );
+};
+
+/**
+ * Close the open data channels, followed by the underlying peer connection
+ */
+PeerConnectionTest.prototype.close = function () {
+  var allChannels = (this.pcLocal || this.pcRemote).dataChannels;
+  return timerGuard(
+    Promise.all(allChannels.map((channel, i) => this.closeDataChannels(i))),
+    120000,
+    "failed to close data channels"
+  ).then(() => this.closePC());
+};
+
+/**
+ * Close the specified data channels
+ *
+ * @param {Number} index
+ *        Index of the data channels to close on both sides
+ */
+PeerConnectionTest.prototype.closeDataChannels = function (index) {
+  info("closeDataChannels called with index: " + index);
+  var localChannel = null;
+  if (this.pcLocal) {
+    localChannel = this.pcLocal.dataChannels[index];
+  }
+  var remoteChannel = null;
+  if (this.pcRemote) {
+    remoteChannel = this.pcRemote.dataChannels[index];
+  }
+
+  // We need to setup all the close listeners before calling close
+  var setupClosePromise = channel => {
+    if (!channel) {
+      return Promise.resolve();
+    }
+    return new Promise(resolve => {
+      channel.onclose = () => {
+        is(
+          channel.readyState,
+          "closed",
+          name + " channel " + index + " closed"
+        );
+        resolve();
+      };
+    });
+  };
+
+  // make sure to setup close listeners before triggering any actions
+  var allClosed = Promise.all([
+    setupClosePromise(localChannel),
+    setupClosePromise(remoteChannel),
+  ]);
+  var complete = timerGuard(
+    allClosed,
+    120000,
+    "failed to close data channel pair"
+  );
+
+  // triggering close on one side should suffice
+  if (remoteChannel) {
+    remoteChannel.close();
+  } else if (localChannel) {
+    localChannel.close();
+  }
+
+  return complete;
+};
+
+/**
+ * Send data (message or blob) to the other peer
+ *
+ * @param {String|Blob} data
+ *        Data to send to the other peer. For Blobs the MIME type will be lost.
+ * @param {Object} [options={ }]
+ *        Options to specify the data channels to be used
+ * @param {DataChannelWrapper} [options.sourceChannel=pcLocal.dataChannels[length - 1]]
+ *        Data channel to use for sending the message
+ * @param {DataChannelWrapper} [options.targetChannel=pcRemote.dataChannels[length - 1]]
+ *        Data channel to use for receiving the message
+ */
+PeerConnectionTest.prototype.send = async function (data, options) {
+  options = options || {};
+  const source =
+    options.sourceChannel ||
+    this.pcLocal.dataChannels[this.pcLocal.dataChannels.length - 1];
+  const target =
+    options.targetChannel ||
+    this.pcRemote.dataChannels[this.pcRemote.dataChannels.length - 1];
+  source.bufferedAmountLowThreshold = options.bufferedAmountLowThreshold || 0;
+
+  const getSizeInBytes = d => {
+    if (d instanceof Blob) {
+      return d.size;
+    } else if (d instanceof ArrayBuffer) {
+      return d.byteLength;
+    } else if (d instanceof String || typeof d === "string") {
+      return new TextEncoder().encode(d).length;
+    } else {
+      ok(false);
+    }
+  };
+
+  const expectedSizeInBytes = getSizeInBytes(data);
+  const bufferedAmount = source.bufferedAmount;
+
+  source.send(data);
+  is(
+    source.bufferedAmount,
+    expectedSizeInBytes + bufferedAmount,
+    `Buffered amount should be ${expectedSizeInBytes}`
+  );
+
+  await new Promise(resolve => (source.onbufferedamountlow = resolve));
+
+  return new Promise(resolve => {
+    // Register event handler for the target channel
+    target.onmessage = e => {
+      is(
+        getSizeInBytes(e.data),
+        expectedSizeInBytes,
+        `Expected to receive the same number of bytes as we sent (${expectedSizeInBytes})`
+      );
+      resolve({ channel: target, data: e.data });
+    };
+  });
+};
+
+/**
+ * Create a data channel
+ *
+ * @param {Dict} options
+ *        Options for the data channel (see nsIPeerConnection)
+ */
+PeerConnectionTest.prototype.createDataChannel = function (options) {
+  var remotePromise;
+  if (!options.negotiated) {
+    this.pcRemote.expectDataChannel("pcRemote expected data channel");
+    remotePromise = this.pcRemote.nextDataChannel;
+  }
+
+  // Create the datachannel
+  var localChannel = this.pcLocal.createDataChannel(options);
+  var localPromise = localChannel.opened;
+
+  if (options.negotiated) {
+    remotePromise = localPromise.then(localChannel => {
+      // externally negotiated - we need to open from both ends
+      options.id = options.id || channel.id; // allow for no id on options
+      var remoteChannel = this.pcRemote.createDataChannel(options);
+      return remoteChannel.opened;
+    });
+  }
+
+  // pcRemote.observedNegotiationNeeded might be undefined if
+  // !options.negotiated, which means we just wait on pcLocal
+  return Promise.all([
+    this.pcLocal.observedNegotiationNeeded,
+    this.pcRemote.observedNegotiationNeeded,
+  ]).then(() => {
+    return Promise.all([localPromise, remotePromise]).then(result => {
+      return { local: result[0], remote: result[1] };
+    });
+  });
+};
+
+/**
+ * Creates an answer for the specified peer connection instance
+ * and automatically handles the failure case.
+ *
+ * @param {PeerConnectionWrapper} peer
+ *        The peer connection wrapper to run the command on
+ */
+PeerConnectionTest.prototype.createAnswer = function (peer) {
+  return peer.createAnswer().then(answer => {
+    // make a copy so this does not get updated with ICE candidates
+    this.originalAnswer = new RTCSessionDescription(
+      JSON.parse(JSON.stringify(answer))
+    );
+    return answer;
+  });
+};
+
+/**
+ * Creates an offer for the specified peer connection instance
+ * and automatically handles the failure case.
+ *
+ * @param {PeerConnectionWrapper} peer
+ *        The peer connection wrapper to run the command on
+ */
+PeerConnectionTest.prototype.createOffer = function (peer) {
+  return peer.createOffer().then(offer => {
+    // make a copy so this does not get updated with ICE candidates
+    this.originalOffer = new RTCSessionDescription(
+      JSON.parse(JSON.stringify(offer))
+    );
+    return offer;
+  });
+};
+
+/**
+ * Sets the local description for the specified peer connection instance
+ * and automatically handles the failure case.
+ *
+ * @param {PeerConnectionWrapper} peer
+          The peer connection wrapper to run the command on
+ * @param {RTCSessionDescriptionInit} desc
+ *        Session description for the local description request
+ */
+PeerConnectionTest.prototype.setLocalDescription = function (
+  peer,
+  desc,
+  stateExpected
+) {
+  var eventFired = new Promise(resolve => {
+    peer.onsignalingstatechange = e => {
+      info(peer + ": 'signalingstatechange' event received");
+      var state = e.target.signalingState;
+      if (stateExpected === state) {
+        peer.setLocalDescStableEventDate = new Date();
+        resolve();
+      } else {
+        ok(
+          false,
+          "This event has either already fired or there has been a " +
+            "mismatch between event received " +
+            state +
+            " and event expected " +
+            stateExpected
+        );
+      }
+    };
+  });
+
+  var stateChanged = peer.setLocalDescription(desc).then(() => {
+    peer.setLocalDescDate = new Date();
+  });
+
+  peer.endOfTrickleSdp = peer.endOfTrickleIce
+    .then(() => {
+      return peer._pc.localDescription;
+    })
+    .catch(e => ok(false, "Sending EOC message failed: " + e));
+
+  return Promise.all([eventFired, stateChanged]);
+};
+
+/**
+ * Sets the media constraints for both peer connection instances.
+ *
+ * @param {object} constraintsLocal
+ *        Media constrains for the local peer connection instance
+ * @param constraintsRemote
+ */
+PeerConnectionTest.prototype.setMediaConstraints = function (
+  constraintsLocal,
+  constraintsRemote
+) {
+  if (this.pcLocal) {
+    this.pcLocal.constraints = constraintsLocal;
+  }
+  if (this.pcRemote) {
+    this.pcRemote.constraints = constraintsRemote;
+  }
+};
+
+/**
+ * Sets the media options used on a createOffer call in the test.
+ *
+ * @param {object} options the media constraints to use on createOffer
+ */
+PeerConnectionTest.prototype.setOfferOptions = function (options) {
+  if (this.pcLocal) {
+    this.pcLocal.offerOptions = options;
+  }
+};
+
+/**
+ * Sets the remote description for the specified peer connection instance
+ * and automatically handles the failure case.
+ *
+ * @param {PeerConnectionWrapper} peer
+          The peer connection wrapper to run the command on
+ * @param {RTCSessionDescriptionInit} desc
+ *        Session description for the remote description request
+ */
+PeerConnectionTest.prototype.setRemoteDescription = function (
+  peer,
+  desc,
+  stateExpected
+) {
+  var eventFired = new Promise(resolve => {
+    peer.onsignalingstatechange = e => {
+      info(peer + ": 'signalingstatechange' event received");
+      var state = e.target.signalingState;
+      if (stateExpected === state) {
+        peer.setRemoteDescStableEventDate = new Date();
+        resolve();
+      } else {
+        ok(
+          false,
+          "This event has either already fired or there has been a " +
+            "mismatch between event received " +
+            state +
+            " and event expected " +
+            stateExpected
+        );
+      }
+    };
+  });
+
+  var stateChanged = peer.setRemoteDescription(desc).then(() => {
+    peer.setRemoteDescDate = new Date();
+    peer.checkMediaTracks();
+  });
+
+  return Promise.all([eventFired, stateChanged]);
+};
+
+/**
+ * Adds and removes steps to/from the execution chain based on the configured
+ * testOptions.
+ */
+PeerConnectionTest.prototype.updateChainSteps = function () {
+  if (this.testOptions.h264) {
+    this.chain.insertAfterEach("PC_LOCAL_CREATE_OFFER", [
+      PC_LOCAL_REMOVE_ALL_BUT_H264_FROM_OFFER,
+    ]);
+  }
+  if (!this.testOptions.bundle) {
+    this.chain.insertAfterEach("PC_LOCAL_CREATE_OFFER", [
+      PC_LOCAL_REMOVE_BUNDLE_FROM_OFFER,
+    ]);
+  }
+  if (!this.testOptions.rtcpmux) {
+    this.chain.insertAfterEach("PC_LOCAL_CREATE_OFFER", [
+      PC_LOCAL_REMOVE_RTCPMUX_FROM_OFFER,
+    ]);
+  }
+  if (!this.testOptions.ssrc) {
+    this.chain.insertAfterEach("PC_LOCAL_CREATE_OFFER", [
+      PC_LOCAL_REMOVE_SSRC_FROM_OFFER,
+    ]);
+    this.chain.insertAfterEach("PC_REMOTE_CREATE_ANSWER", [
+      PC_REMOTE_REMOVE_SSRC_FROM_ANSWER,
+    ]);
+  }
+  if (!this.testOptions.is_local) {
+    this.chain.filterOut(/^PC_LOCAL/);
+  }
+  if (!this.testOptions.is_remote) {
+    this.chain.filterOut(/^PC_REMOTE/);
+  }
+};
+
+/**
+ * Start running the tests as assigned to the command chain.
+ */
+PeerConnectionTest.prototype.run = async function () {
+  /* We have to modify the chain here to allow tests which modify the default
+   * test chain instantiating a PeerConnectionTest() */
+  this.updateChainSteps();
+  try {
+    await this.chain.execute();
+    await this.close();
+  } catch (e) {
+    const stack =
+      typeof e.stack === "string"
+        ? ` ${e.stack.split("\n").join(" ... ")}`
+        : "";
+    ok(false, `Error in test execution: ${e} (${stack})`);
+  }
+};
+
+/**
+ * Routes ice candidates from one PCW to the other PCW
+ */
+PeerConnectionTest.prototype.iceCandidateHandler = function (
+  caller,
+  candidate
+) {
+  info("Received: " + JSON.stringify(candidate) + " from " + caller);
+
+  var target = null;
+  if (caller.includes("pcLocal")) {
+    if (this.pcRemote) {
+      target = this.pcRemote;
+    }
+  } else if (caller.includes("pcRemote")) {
+    if (this.pcLocal) {
+      target = this.pcLocal;
+    }
+  } else {
+    ok(false, "received event from unknown caller: " + caller);
+    return;
+  }
+
+  if (target) {
+    target.storeOrAddIceCandidate(candidate);
+  } else {
+    info("sending ice candidate to signaling server");
+    send_message({ type: "ice_candidate", ice_candidate: candidate });
+  }
+};
+
+/**
+ * Installs a polling function for the socket.io client to read
+ * all messages from the chat room into a message queue.
+ */
+PeerConnectionTest.prototype.setupSignalingClient = function () {
+  this.signalingMessageQueue = [];
+  this.signalingCallbacks = {};
+  this.signalingLoopRun = true;
+
+  var queueMessage = message => {
+    info("Received signaling message: " + JSON.stringify(message));
+    var fired = false;
+    Object.keys(this.signalingCallbacks).forEach(name => {
+      if (name === message.type) {
+        info("Invoking callback for message type: " + name);
+        this.signalingCallbacks[name](message);
+        fired = true;
+      }
+    });
+    if (!fired) {
+      this.signalingMessageQueue.push(message);
+      info(
+        "signalingMessageQueue.length: " + this.signalingMessageQueue.length
+      );
+    }
+    if (this.signalingLoopRun) {
+      wait_for_message().then(queueMessage);
+    } else {
+      info("Exiting signaling message event loop");
+    }
+  };
+  wait_for_message().then(queueMessage);
+};
+
+/**
+ * Sets a flag to stop reading further messages from the chat room.
+ */
+PeerConnectionTest.prototype.signalingMessagesFinished = function () {
+  this.signalingLoopRun = false;
+};
+
+/**
+ * Register a callback function to deliver messages from the chat room
+ * directly instead of storing them in the message queue.
+ *
+ * @param {string} messageType
+ *        For which message types should the callback get invoked.
+ *
+ * @param {function} onMessage
+ *        The function which gets invoked if a message of the messageType
+ *        has been received from the chat room.
+ */
+PeerConnectionTest.prototype.registerSignalingCallback = function (
+  messageType,
+  onMessage
+) {
+  this.signalingCallbacks[messageType] = onMessage;
+};
+
+/**
+ * Searches the message queue for the first message of a given type
+ * and invokes the given callback function, or registers the callback
+ * function for future messages if the queue contains no such message.
+ *
+ * @param {string} messageType
+ *        The type of message to search and register for.
+ */
+PeerConnectionTest.prototype.getSignalingMessage = function (messageType) {
+  var i = this.signalingMessageQueue.findIndex(m => m.type === messageType);
+  if (i >= 0) {
+    info(
+      "invoking callback on message " +
+        i +
+        " from message queue, for message type:" +
+        messageType
+    );
+    return Promise.resolve(this.signalingMessageQueue.splice(i, 1)[0]);
+  }
+  return new Promise(resolve =>
+    this.registerSignalingCallback(messageType, resolve)
+  );
+};
+
+/**
+ * This class acts as a wrapper around a DataChannel instance.
+ *
+ * @param dataChannel
+ * @param peerConnectionWrapper
+ * @constructor
+ */
+function DataChannelWrapper(dataChannel, peerConnectionWrapper) {
+  this._channel = dataChannel;
+  this._pc = peerConnectionWrapper;
+
+  info("Creating " + this);
+
+  /**
+   * Setup appropriate callbacks
+   */
+  createOneShotEventWrapper(this, this._channel, "close");
+  createOneShotEventWrapper(this, this._channel, "error");
+  createOneShotEventWrapper(this, this._channel, "message");
+  createOneShotEventWrapper(this, this._channel, "bufferedamountlow");
+
+  this.opened = timerGuard(
+    new Promise(resolve => {
+      this._channel.onopen = () => {
+        this._channel.onopen = unexpectedEvent(this, "onopen");
+        is(this.readyState, "open", "data channel is 'open' after 'onopen'");
+        resolve(this);
+      };
+    }),
+    180000,
+    "channel didn't open in time"
+  );
+}
+
+DataChannelWrapper.prototype = {
+  /**
+   * Returns the binary type of the channel
+   *
+   * @returns {String} The binary type
+   */
+  get binaryType() {
+    return this._channel.binaryType;
+  },
+
+  /**
+   * Sets the binary type of the channel
+   *
+   * @param {String} type
+   *        The new binary type of the channel
+   */
+  set binaryType(type) {
+    this._channel.binaryType = type;
+  },
+
+  /**
+   * Returns the label of the underlying data channel
+   *
+   * @returns {String} The label
+   */
+  get label() {
+    return this._channel.label;
+  },
+
+  /**
+   * Returns the protocol of the underlying data channel
+   *
+   * @returns {String} The protocol
+   */
+  get protocol() {
+    return this._channel.protocol;
+  },
+
+  /**
+   * Returns the id of the underlying data channel
+   *
+   * @returns {number} The stream id
+   */
+  get id() {
+    return this._channel.id;
+  },
+
+  /**
+   * Returns the reliable state of the underlying data channel
+   *
+   * @returns {bool} The stream's reliable state
+   */
+  get reliable() {
+    return this._channel.reliable;
+  },
+
+  /**
+   * Returns the ordered attribute of the data channel
+   *
+   * @returns {bool} The ordered attribute
+   */
+  get ordered() {
+    return this._channel.ordered;
+  },
+
+  /**
+   * Returns the maxPacketLifeTime attribute of the data channel
+   *
+   * @returns {number} The maxPacketLifeTime attribute
+   */
+  get maxPacketLifeTime() {
+    return this._channel.maxPacketLifeTime;
+  },
+
+  /**
+   * Returns the maxRetransmits attribute of the data channel
+   *
+   * @returns {number} The maxRetransmits attribute
+   */
+  get maxRetransmits() {
+    return this._channel.maxRetransmits;
+  },
+
+  /**
+   * Returns the readyState bit of the data channel
+   *
+   * @returns {String} The state of the channel
+   */
+  get readyState() {
+    return this._channel.readyState;
+  },
+
+  get bufferedAmount() {
+    return this._channel.bufferedAmount;
+  },
+
+  /**
+   * Sets the bufferlowthreshold of the channel
+   *
+   * @param {integer} amoutn
+   *        The new threshold for the chanel
+   */
+  set bufferedAmountLowThreshold(amount) {
+    this._channel.bufferedAmountLowThreshold = amount;
+  },
+
+  /**
+   * Close the data channel
+   */
+  close() {
+    info(this + ": Closing channel");
+    this._channel.close();
+  },
+
+  /**
+   * Send data through the data channel
+   *
+   * @param {String|Object} data
+   *        Data which has to be sent through the data channel
+   */
+  send(data) {
+    info(this + ": Sending data '" + data + "'");
+    this._channel.send(data);
+  },
+
+  /**
+   * Returns the string representation of the class
+   *
+   * @returns {String} The string representation
+   */
+  toString() {
+    return (
+      "DataChannelWrapper (" + this._pc.label + "_" + this._channel.label + ")"
+    );
+  },
+};
+
+/**
+ * This class acts as a wrapper around a PeerConnection instance.
+ *
+ * @constructor
+ * @param {string} label
+ *        Description for the peer connection instance
+ * @param {object} configuration
+ *        Configuration for the peer connection instance
+ */
+function PeerConnectionWrapper(label, configuration) {
+  this.configuration = configuration;
+  if (configuration && configuration.label_suffix) {
+    label = label + "_" + configuration.label_suffix;
+  }
+  this.label = label;
+
+  this.constraints = [];
+  this.offerOptions = {};
+
+  this.dataChannels = [];
+
+  this._local_ice_candidates = [];
+  this._remote_ice_candidates = [];
+  this.localRequiresTrickleIce = false;
+  this.remoteRequiresTrickleIce = false;
+  this.localMediaElements = [];
+  this.remoteMediaElements = [];
+  this.audioElementsOnly = false;
+
+  this._sendStreams = [];
+
+  this.expectedLocalTrackInfo = [];
+  this.remoteStreamsByTrackId = new Map();
+
+  this.disableRtpCountChecking = false;
+
+  this.iceConnectedResolve;
+  this.iceConnectedReject;
+  this.iceConnected = new Promise((resolve, reject) => {
+    this.iceConnectedResolve = resolve;
+    this.iceConnectedReject = reject;
+  });
+  this.iceCheckingRestartExpected = false;
+  this.iceCheckingIceRollbackExpected = false;
+
+  info("Creating " + this);
+  this._pc = new RTCPeerConnection(this.configuration);
+
+  /**
+   * Setup callback handlers
+   */
+  // This allows test to register their own callbacks for ICE connection state changes
+  this.ice_connection_callbacks = {};
+
+  this._pc.oniceconnectionstatechange = e => {
+    isnot(
+      typeof this._pc.iceConnectionState,
+      "undefined",
+      "iceConnectionState should not be undefined"
+    );
+    var iceState = this._pc.iceConnectionState;
+    info(
+      this + ": oniceconnectionstatechange fired, new state is: " + iceState
+    );
+    Object.keys(this.ice_connection_callbacks).forEach(name => {
+      this.ice_connection_callbacks[name]();
+    });
+    if (iceState === "connected") {
+      this.iceConnectedResolve();
+    } else if (iceState === "failed") {
+      this.iceConnectedReject(new Error("ICE failed"));
+    }
+  };
+
+  this._pc.onicegatheringstatechange = e => {
+    isnot(
+      typeof this._pc.iceGatheringState,
+      "undefined",
+      "iceGetheringState should not be undefined"
+    );
+    var gatheringState = this._pc.iceGatheringState;
+    info(
+      this +
+        ": onicegatheringstatechange fired, new state is: " +
+        gatheringState
+    );
+  };
+
+  createOneShotEventWrapper(this, this._pc, "datachannel");
+  this._pc.addEventListener("datachannel", e => {
+    var wrapper = new DataChannelWrapper(e.channel, this);
+    this.dataChannels.push(wrapper);
+  });
+
+  createOneShotEventWrapper(this, this._pc, "signalingstatechange");
+  createOneShotEventWrapper(this, this._pc, "negotiationneeded");
+}
+
+PeerConnectionWrapper.prototype = {
+  /**
+   * Returns the senders
+   *
+   * @returns {sequence<RTCRtpSender>} the senders
+   */
+  getSenders() {
+    return this._pc.getSenders();
+  },
+
+  /**
+   * Returns the getters
+   *
+   * @returns {sequence<RTCRtpReceiver>} the receivers
+   */
+  getReceivers() {
+    return this._pc.getReceivers();
+  },
+
+  /**
+   * Returns the local description.
+   *
+   * @returns {object} The local description
+   */
+  get localDescription() {
+    return this._pc.localDescription;
+  },
+
+  /**
+   * Returns the remote description.
+   *
+   * @returns {object} The remote description
+   */
+  get remoteDescription() {
+    return this._pc.remoteDescription;
+  },
+
+  /**
+   * Returns the signaling state.
+   *
+   * @returns {object} The local description
+   */
+  get signalingState() {
+    return this._pc.signalingState;
+  },
+  /**
+   * Returns the ICE connection state.
+   *
+   * @returns {object} The local description
+   */
+  get iceConnectionState() {
+    return this._pc.iceConnectionState;
+  },
+
+  setIdentityProvider(provider, options) {
+    this._pc.setIdentityProvider(provider, options);
+  },
+
+  elementPrefix: direction => {
+    return [this.label, direction].join("_");
+  },
+
+  getMediaElementForTrack(track, direction) {
+    var prefix = this.elementPrefix(direction);
+    return getMediaElementForTrack(track, prefix);
+  },
+
+  createMediaElementForTrack(track, direction) {
+    var prefix = this.elementPrefix(direction);
+    return createMediaElementForTrack(track, prefix);
+  },
+
+  ensureMediaElement(track, direction) {
+    var prefix = this.elementPrefix(direction);
+    var element = this.getMediaElementForTrack(track, direction);
+    if (!element) {
+      element = this.createMediaElementForTrack(track, direction);
+      if (direction == "local") {
+        this.localMediaElements.push(element);
+      } else if (direction == "remote") {
+        this.remoteMediaElements.push(element);
+      }
+    }
+
+    // We do this regardless, because sometimes we end up with a new stream with
+    // an old id (ie; the rollback tests cause the same stream to be added
+    // twice)
+    element.srcObject = new MediaStream([track]);
+    element.play();
+  },
+
+  addSendStream(stream) {
+    // The PeerConnection will not necessarily know about this stream
+    // automatically, because replaceTrack is not told about any streams the
+    // new track might be associated with. Only content really knows.
+    this._sendStreams.push(stream);
+  },
+
+  getStreamForSendTrack(track) {
+    return this._sendStreams.find(str => str.getTrackById(track.id));
+  },
+
+  getStreamForRecvTrack(track) {
+    return this._pc.getRemoteStreams().find(s => !!s.getTrackById(track.id));
+  },
+
+  /**
+   * Attaches a local track to this RTCPeerConnection using
+   * RTCPeerConnection.addTrack().
+   *
+   * Also creates a media element playing a MediaStream containing all
+   * tracks that have been added to `stream` using `attachLocalTrack()`.
+   *
+   * @param {MediaStreamTrack} track
+   *        MediaStreamTrack to handle
+   * @param {MediaStream} stream
+   *        MediaStream to use as container for `track` on remote side
+   */
+  attachLocalTrack(track, stream) {
+    info("Got a local " + track.kind + " track");
+
+    this.expectNegotiationNeeded();
+    var sender = this._pc.addTrack(track, stream);
+    is(sender.track, track, "addTrack returns sender");
+    is(
+      this._pc.getSenders().pop(),
+      sender,
+      "Sender should be the last element in getSenders()"
+    );
+
+    ok(track.id, "track has id");
+    ok(track.kind, "track has kind");
+    ok(stream.id, "stream has id");
+    this.expectedLocalTrackInfo.push({ track, sender, streamId: stream.id });
+    this.addSendStream(stream);
+
+    // This will create one media element per track, which might not be how
+    // we set up things with the RTCPeerConnection. It's the only way
+    // we can ensure all sent tracks are flowing however.
+    this.ensureMediaElement(track, "local");
+
+    return this.observedNegotiationNeeded;
+  },
+
+  /**
+   * Callback when we get local media. Also an appropriate HTML media element
+   * will be created and added to the content node.
+   *
+   * @param {MediaStream} stream
+   *        Media stream to handle
+   */
+  attachLocalStream(stream, useAddTransceiver) {
+    info("Got local media stream: (" + stream.id + ")");
+
+    this.expectNegotiationNeeded();
+    if (useAddTransceiver) {
+      info("Using addTransceiver (on PC).");
+      stream.getTracks().forEach(track => {
+        var transceiver = this._pc.addTransceiver(track, { streams: [stream] });
+        is(transceiver.sender.track, track, "addTransceiver returns sender");
+      });
+    }
+    // In order to test both the addStream and addTrack APIs, we do half one
+    // way, half the other, at random.
+    else if (Math.random() < 0.5) {
+      info("Using addStream.");
+      this._pc.addStream(stream);
+      ok(
+        this._pc
+          .getSenders()
+          .find(sender => sender.track == stream.getTracks()[0]),
+        "addStream returns sender"
+      );
+    } else {
+      info("Using addTrack (on PC).");
+      stream.getTracks().forEach(track => {
+        var sender = this._pc.addTrack(track, stream);
+        is(sender.track, track, "addTrack returns sender");
+      });
+    }
+
+    this.addSendStream(stream);
+
+    stream.getTracks().forEach(track => {
+      ok(track.id, "track has id");
+      ok(track.kind, "track has kind");
+      const sender = this._pc.getSenders().find(s => s.track == track);
+      ok(sender, "track has a sender");
+      this.expectedLocalTrackInfo.push({ track, sender, streamId: stream.id });
+      this.ensureMediaElement(track, "local");
+    });
+
+    return this.observedNegotiationNeeded;
+  },
+
+  removeSender(index) {
+    var sender = this._pc.getSenders()[index];
+    this.expectedLocalTrackInfo = this.expectedLocalTrackInfo.filter(
+      i => i.sender != sender
+    );
+    this.expectNegotiationNeeded();
+    this._pc.removeTrack(sender);
+    return this.observedNegotiationNeeded;
+  },
+
+  senderReplaceTrack(sender, withTrack, stream) {
+    const info = this.expectedLocalTrackInfo.find(i => i.sender == sender);
+    if (!info) {
+      return; // replaceTrack on a null track, probably
+    }
+    info.track = withTrack;
+    this.addSendStream(stream);
+    this.ensureMediaElement(withTrack, "local");
+    return sender.replaceTrack(withTrack);
+  },
+
+  async getUserMedia(constraints) {
+    SpecialPowers.wrap(document).notifyUserGestureActivation();
+    var stream = await getUserMedia(constraints);
+    if (constraints.audio) {
+      stream.getAudioTracks().forEach(track => {
+        info(
+          this +
+            " gUM local stream " +
+            stream.id +
+            " with audio track " +
+            track.id
+        );
+      });
+    }
+    if (constraints.video) {
+      stream.getVideoTracks().forEach(track => {
+        info(
+          this +
+            " gUM local stream " +
+            stream.id +
+            " with video track " +
+            track.id
+        );
+      });
+    }
+    return stream;
+  },
+
+  /**
+   * Requests all the media streams as specified in the constrains property.
+   *
+   * @param {array} constraintsList
+   *        Array of constraints for GUM calls
+   */
+  getAllUserMedia(constraintsList) {
+    if (constraintsList.length === 0) {
+      info("Skipping GUM: no UserMedia requested");
+      return Promise.resolve();
+    }
+
+    info("Get " + constraintsList.length + " local streams");
+    return Promise.all(
+      constraintsList.map(constraints => this.getUserMedia(constraints))
+    );
+  },
+
+  async getAllUserMediaAndAddStreams(constraintsList) {
+    var streams = await this.getAllUserMedia(constraintsList);
+    if (!streams) {
+      return;
+    }
+    return Promise.all(streams.map(stream => this.attachLocalStream(stream)));
+  },
+
+  async getAllUserMediaAndAddTransceivers(constraintsList) {
+    var streams = await this.getAllUserMedia(constraintsList);
+    if (!streams) {
+      return;
+    }
+    return Promise.all(
+      streams.map(stream => this.attachLocalStream(stream, true))
+    );
+  },
+
+  /**
+   * Create a new data channel instance.  Also creates a promise called
+   * `this.nextDataChannel` that resolves when the next data channel arrives.
+   */
+  expectDataChannel(message) {
+    this.nextDataChannel = new Promise(resolve => {
+      this.ondatachannel = e => {
+        ok(e.channel, message);
+        is(
+          e.channel.readyState,
+          "open",
+          "data channel in 'open' after 'ondatachannel'"
+        );
+        resolve(e.channel);
+      };
+    });
+  },
+
+  /**
+   * Create a new data channel instance
+   *
+   * @param {Object} options
+   *        Options which get forwarded to nsIPeerConnection.createDataChannel
+   * @returns {DataChannelWrapper} The created data channel
+   */
+  createDataChannel(options) {
+    var label = "channel_" + this.dataChannels.length;
+    info(this + ": Create data channel '" + label);
+
+    if (!this.dataChannels.length) {
+      this.expectNegotiationNeeded();
+    }
+    var channel = this._pc.createDataChannel(label, options);
+    is(channel.readyState, "connecting", "initial readyState is 'connecting'");
+    var wrapper = new DataChannelWrapper(channel, this);
+    this.dataChannels.push(wrapper);
+    return wrapper;
+  },
+
+  /**
+   * Creates an offer and automatically handles the failure case.
+   */
+  createOffer() {
+    return this._pc.createOffer(this.offerOptions).then(offer => {
+      info("Got offer: " + JSON.stringify(offer));
+      // note: this might get updated through ICE gathering
+      this._latest_offer = offer;
+      return offer;
+    });
+  },
+
+  /**
+   * Creates an answer and automatically handles the failure case.
+   */
+  createAnswer() {
+    return this._pc.createAnswer().then(answer => {
+      info(this + ": Got answer: " + JSON.stringify(answer));
+      this._last_answer = answer;
+      return answer;
+    });
+  },
+
+  /**
+   * Sets the local description and automatically handles the failure case.
+   *
+   * @param {object} desc
+   *        RTCSessionDescriptionInit for the local description request
+   */
+  setLocalDescription(desc) {
+    this.observedNegotiationNeeded = undefined;
+    return this._pc.setLocalDescription(desc).then(() => {
+      info(this + ": Successfully set the local description");
+    });
+  },
+
+  /**
+   * Tries to set the local description and expect failure. Automatically
+   * causes the test case to fail if the call succeeds.
+   *
+   * @param {object} desc
+   *        RTCSessionDescriptionInit for the local description request
+   * @returns {Promise}
+   *        A promise that resolves to the expected error
+   */
+  setLocalDescriptionAndFail(desc) {
+    return this._pc
+      .setLocalDescription(desc)
+      .then(
+        generateErrorCallback("setLocalDescription should have failed."),
+        err => {
+          info(this + ": As expected, failed to set the local description");
+          return err;
+        }
+      );
+  },
+
+  /**
+   * Sets the remote description and automatically handles the failure case.
+   *
+   * @param {object} desc
+   *        RTCSessionDescriptionInit for the remote description request
+   */
+  setRemoteDescription(desc) {
+    this.observedNegotiationNeeded = undefined;
+    // This has to be done before calling sRD, otherwise a candidate in flight
+    // could end up in the PC's operations queue before sRD resolves.
+    if (desc.type == "rollback") {
+      this.holdIceCandidates = new Promise(
+        r => (this.releaseIceCandidates = r)
+      );
+    }
+    return this._pc.setRemoteDescription(desc).then(() => {
+      info(this + ": Successfully set remote description");
+      if (desc.type != "rollback") {
+        this.releaseIceCandidates();
+      }
+    });
+  },
+
+  /**
+   * Tries to set the remote description and expect failure. Automatically
+   * causes the test case to fail if the call succeeds.
+   *
+   * @param {object} desc
+   *        RTCSessionDescriptionInit for the remote description request
+   * @returns {Promise}
+   *        a promise that resolve to the returned error
+   */
+  setRemoteDescriptionAndFail(desc) {
+    return this._pc
+      .setRemoteDescription(desc)
+      .then(
+        generateErrorCallback("setRemoteDescription should have failed."),
+        err => {
+          info(this + ": As expected, failed to set the remote description");
+          return err;
+        }
+      );
+  },
+
+  /**
+   * Registers a callback for the signaling state change and
+   * appends the new state to an array for logging it later.
+   */
+  logSignalingState() {
+    this.signalingStateLog = [this._pc.signalingState];
+    this._pc.addEventListener("signalingstatechange", e => {
+      var newstate = this._pc.signalingState;
+      var oldstate = this.signalingStateLog[this.signalingStateLog.length - 1];
+      if (Object.keys(signalingStateTransitions).includes(oldstate)) {
+        ok(
+          signalingStateTransitions[oldstate].includes(newstate),
+          this +
+            ": legal signaling state transition from " +
+            oldstate +
+            " to " +
+            newstate
+        );
+      } else {
+        ok(
+          false,
+          this +
+            ": old signaling state " +
+            oldstate +
+            " missing in signaling transition array"
+        );
+      }
+      this.signalingStateLog.push(newstate);
+    });
+  },
+
+  isTrackOnPC(track) {
+    return !!this.getStreamForRecvTrack(track);
+  },
+
+  allExpectedTracksAreObserved(expected, observed) {
+    return Object.keys(expected).every(trackId => observed[trackId]);
+  },
+
+  setupStreamEventHandlers(stream) {
+    const myTrackIds = new Set(stream.getTracks().map(t => t.id));
+
+    stream.addEventListener("addtrack", ({ track }) => {
+      ok(
+        !myTrackIds.has(track.id),
+        "Duplicate addtrack callback: " +
+          `stream id=${stream.id} track id=${track.id}`
+      );
+      myTrackIds.add(track.id);
+      // addtrack events happen before track events, so the track callback hasn't
+      // heard about this yet.
+      let streams = this.remoteStreamsByTrackId.get(track.id);
+      ok(
+        !streams || !streams.has(stream.id),
+        `In addtrack for stream id=${stream.id}` +
+          `there should not have been a track event for track id=${track.id} ` +
+          " containing this stream yet."
+      );
+      ok(
+        stream.getTracks().includes(track),
+        "In addtrack, stream id=" +
+          `${stream.id} should already contain track id=${track.id}`
+      );
+    });
+
+    stream.addEventListener("removetrack", ({ track }) => {
+      ok(
+        myTrackIds.has(track.id),
+        "Duplicate removetrack callback: " +
+          `stream id=${stream.id} track id=${track.id}`
+      );
+      myTrackIds.delete(track.id);
+      // Also remove the association from remoteStreamsByTrackId
+      const streams = this.remoteStreamsByTrackId.get(track.id);
+      ok(
+        streams,
+        `In removetrack for stream id=${stream.id}, track id=` +
+          `${track.id} should have had a track callback for the stream.`
+      );
+      streams.delete(stream.id);
+      ok(
+        !stream.getTracks().includes(track),
+        "In removetrack, stream id=" +
+          `${stream.id} should not contain track id=${track.id}`
+      );
+    });
+  },
+
+  setupTrackEventHandler() {
+    this._pc.addEventListener("track", ({ track, streams }) => {
+      info(`${this}: 'ontrack' event fired for ${track.id}`);
+      ok(this.isTrackOnPC(track), `Found track ${track.id}`);
+
+      let gratuitousEvent = true;
+      let streamsContainingTrack = this.remoteStreamsByTrackId.get(track.id);
+      if (!streamsContainingTrack) {
+        gratuitousEvent = false; // Told us about a new track
+        this.remoteStreamsByTrackId.set(track.id, new Set());
+        streamsContainingTrack = this.remoteStreamsByTrackId.get(track.id);
+      }
+
+      for (const stream of streams) {
+        ok(
+          stream.getTracks().includes(track),
+          `In track event, track id=${track.id}` +
+            ` should already be in stream id=${stream.id}`
+        );
+
+        if (!streamsContainingTrack.has(stream.id)) {
+          gratuitousEvent = false; // Told us about a new stream
+          streamsContainingTrack.add(stream.id);
+          this.setupStreamEventHandlers(stream);
+        }
+      }
+
+      ok(!gratuitousEvent, "track event told us something new");
+
+      // So far, we've verified consistency between the current state of the
+      // streams, addtrack/removetrack events on the streams, and track events
+      // on the peerconnection. We have also verified that we have not gotten
+      // any gratuitous events. We have not done anything to verify that the
+      // current state of affairs matches what we were expecting it to.
+
+      this.ensureMediaElement(track, "remote");
+    });
+  },
+
+  /**
+   * Either adds a given ICE candidate right away or stores it to be added
+   * later, depending on the state of the PeerConnection.
+   *
+   * @param {object} candidate
+   *        The RTCIceCandidate to be added or stored
+   */
+  storeOrAddIceCandidate(candidate) {
+    this._remote_ice_candidates.push(candidate);
+    if (this.signalingState === "closed") {
+      info("Received ICE candidate for closed PeerConnection - discarding");
+      return;
+    }
+    this.holdIceCandidates
+      .then(() => {
+        info(this + ": adding ICE candidate " + JSON.stringify(candidate));
+        return this._pc.addIceCandidate(candidate);
+      })
+      .then(() => ok(true, this + " successfully added an ICE candidate"))
+      .catch(e =>
+        // The onicecandidate callback runs independent of the test steps
+        // and therefore errors thrown from in there don't get caught by the
+        // race of the Promises around our test steps.
+        // Note: as long as we are queuing ICE candidates until the success
+        //       of sRD() this should never ever happen.
+        ok(false, this + " adding ICE candidate failed with: " + e.message)
+      );
+  },
+
+  /**
+   * Registers a callback for the ICE connection state change and
+   * appends the new state to an array for logging it later.
+   */
+  logIceConnectionState() {
+    this.iceConnectionLog = [this._pc.iceConnectionState];
+    this.ice_connection_callbacks.logIceStatus = () => {
+      var newstate = this._pc.iceConnectionState;
+      var oldstate = this.iceConnectionLog[this.iceConnectionLog.length - 1];
+      if (Object.keys(iceStateTransitions).includes(oldstate)) {
+        if (this.iceCheckingRestartExpected) {
+          is(
+            newstate,
+            "checking",
+            "iceconnectionstate event '" +
+              newstate +
+              "' matches expected state 'checking'"
+          );
+          this.iceCheckingRestartExpected = false;
+        } else if (this.iceCheckingIceRollbackExpected) {
+          is(
+            newstate,
+            "connected",
+            "iceconnectionstate event '" +
+              newstate +
+              "' matches expected state 'connected'"
+          );
+          this.iceCheckingIceRollbackExpected = false;
+        } else {
+          ok(
+            iceStateTransitions[oldstate].includes(newstate),
+            this +
+              ": legal ICE state transition from " +
+              oldstate +
+              " to " +
+              newstate
+          );
+        }
+      } else {
+        ok(
+          false,
+          this +
+            ": old ICE state " +
+            oldstate +
+            " missing in ICE transition array"
+        );
+      }
+      this.iceConnectionLog.push(newstate);
+    };
+  },
+
+  /**
+   * Resets the ICE connected Promise and allows ICE connection state monitoring
+   * to go backwards to 'checking'.
+   */
+  expectIceChecking() {
+    this.iceCheckingRestartExpected = true;
+    this.iceConnected = new Promise((resolve, reject) => {
+      this.iceConnectedResolve = resolve;
+      this.iceConnectedReject = reject;
+    });
+  },
+
+  /**
+   * Waits for ICE to either connect or fail.
+   *
+   * @returns {Promise}
+   *          resolves when connected, rejects on failure
+   */
+  waitForIceConnected() {
+    return this.iceConnected;
+  },
+
+  /**
+   * Setup a onicecandidate handler
+   *
+   * @param {object} test
+   *        A PeerConnectionTest object to which the ice candidates gets
+   *        forwarded.
+   */
+  setupIceCandidateHandler(test, candidateHandler) {
+    candidateHandler = candidateHandler || test.iceCandidateHandler.bind(test);
+
+    var resolveEndOfTrickle;
+    this.endOfTrickleIce = new Promise(r => (resolveEndOfTrickle = r));
+    this.holdIceCandidates = new Promise(r => (this.releaseIceCandidates = r));
+
+    this._pc.onicecandidate = anEvent => {
+      if (!anEvent.candidate) {
+        this._pc.onicecandidate = () =>
+          ok(
+            false,
+            this.label + " received ICE candidate after end of trickle"
+          );
+        info(this.label + ": received end of trickle ICE event");
+        ok(
+          this._pc.iceGatheringState === "complete",
+          "ICE gathering state has reached complete"
+        );
+        resolveEndOfTrickle(this.label);
+        return;
+      }
+
+      info(
+        this.label + ": iceCandidate = " + JSON.stringify(anEvent.candidate)
+      );
+      ok(anEvent.candidate.sdpMid.length, "SDP mid not empty");
+      ok(
+        anEvent.candidate.usernameFragment.length,
+        "usernameFragment not empty"
+      );
+
+      ok(
+        typeof anEvent.candidate.sdpMLineIndex === "number",
+        "SDP MLine Index needs to exist"
+      );
+      this._local_ice_candidates.push(anEvent.candidate);
+      candidateHandler(this.label, anEvent.candidate);
+    };
+  },
+
+  checkLocalMediaTracks() {
+    info(
+      `${this}: Checking local tracks ${JSON.stringify(
+        this.expectedLocalTrackInfo
+      )}`
+    );
+    const sendersWithTrack = this._pc.getSenders().filter(({ track }) => track);
+    is(
+      sendersWithTrack.length,
+      this.expectedLocalTrackInfo.length,
+      "The number of senders with a track should be equal to the number of " +
+        "expected local tracks."
+    );
+
+    // expectedLocalTrackInfo is in the same order that the tracks were added, and
+    // so should the output of getSenders.
+    this.expectedLocalTrackInfo.forEach((info, i) => {
+      const sender = sendersWithTrack[i];
+      is(sender, info.sender, `Sender ${i} should match`);
+      is(sender.track, info.track, `Track ${i} should match`);
+    });
+  },
+
+  /**
+   * Checks that we are getting the media tracks we expect.
+   */
+  checkMediaTracks() {
+    this.checkLocalMediaTracks();
+  },
+
+  checkLocalMsids() {
+    const sdp = this.localDescription.sdp;
+    const msections = sdputils.getMSections(sdp);
+    const expectedStreamIdCounts = new Map();
+    for (const { track, sender, streamId } of this.expectedLocalTrackInfo) {
+      const transceiver = this._pc
+        .getTransceivers()
+        .find(t => t.sender == sender);
+      ok(transceiver, "There should be a transceiver for each sender");
+      if (transceiver.mid) {
+        const midFinder = new RegExp(`^a=mid:${transceiver.mid}$`, "m");
+        const msection = msections.find(m => m.match(midFinder));
+        ok(
+          msection,
+          `There should be a media section for mid = ${transceiver.mid}`
+        );
+        ok(
+          msection.startsWith(`m=${track.kind}`),
+          `Media section should be of type ${track.kind}`
+        );
+        const msidFinder = new RegExp(`^a=msid:${streamId} \\S+$`, "m");
+        ok(
+          msection.match(msidFinder),
+          `Should find a=msid:${streamId} in media section` +
+            " (with any track id for now)"
+        );
+        const count = expectedStreamIdCounts.get(streamId) || 0;
+        expectedStreamIdCounts.set(streamId, count + 1);
+      }
+    }
+
+    // Check for any unexpected msids.
+    const allMsids = sdp.match(new RegExp("^a=msid:\\S+", "mg"));
+    if (!allMsids) {
+      return;
+    }
+    const allStreamIds = allMsids.map(msidAttr =>
+      msidAttr.replace("a=msid:", "")
+    );
+    allStreamIds.forEach(id => {
+      const count = expectedStreamIdCounts.get(id);
+      ok(count, `Unexpected stream id ${id} found in local description.`);
+      if (count) {
+        expectedStreamIdCounts.set(id, count - 1);
+      }
+    });
+  },
+
+  /**
+   * Check that media flow is present for the given media element by checking
+   * that it reaches ready state HAVE_ENOUGH_DATA and progresses time further
+   * than the start of the check.
+   *
+   * This ensures, that the stream being played is producing
+   * data and, in case it contains a video track, that at least one video frame
+   * has been displayed.
+   *
+   * @param {HTMLMediaElement} track
+   *        The media element to check
+   * @returns {Promise}
+   *        A promise that resolves when media data is flowing.
+   */
+  waitForMediaElementFlow(element) {
+    info("Checking data flow for element: " + element.id);
+    is(
+      element.ended,
+      !element.srcObject.active,
+      "Element ended should be the inverse of the MediaStream's active state"
+    );
+    if (element.ended) {
+      is(
+        element.readyState,
+        element.HAVE_CURRENT_DATA,
+        "Element " + element.id + " is ended and should have had data"
+      );
+      return Promise.resolve();
+    }
+
+    const haveEnoughData = (
+      element.readyState == element.HAVE_ENOUGH_DATA
+        ? Promise.resolve()
+        : haveEvent(
+            element,
+            "canplay",
+            wait(60000, new Error("Timeout for element " + element.id))
+          )
+    ).then(_ => info("Element " + element.id + " has enough data."));
+
+    const startTime = element.currentTime;
+    const timeProgressed = timeout(
+      listenUntil(element, "timeupdate", _ => element.currentTime > startTime),
+      60000,
+      "Element " + element.id + " should progress currentTime"
+    ).then();
+
+    return Promise.all([haveEnoughData, timeProgressed]);
+  },
+
+  /**
+   * Wait for RTP packet flow for the given MediaStreamTrack.
+   *
+   * @param {object} track
+   *        A MediaStreamTrack to wait for data flow on.
+   * @returns {Promise}
+   *        Returns a promise which yields a StatsReport object with RTP stats.
+   */
+  async _waitForRtpFlow(target, rtpType) {
+    const { track } = target;
+    info(`_waitForRtpFlow(${track.id}, ${rtpType})`);
+    const packets = `packets${rtpType == "outbound-rtp" ? "Sent" : "Received"}`;
+
+    const retryInterval = 500; // Time between stats checks
+    const timeout = 30000; // Timeout in ms
+    const retries = timeout / retryInterval;
+
+    for (let i = 0; i < retries; i++) {
+      info(`Checking ${rtpType} for ${track.kind} track ${track.id} try ${i}`);
+      for (const rtp of (await target.getStats()).values()) {
+        if (rtp.type != rtpType) {
+          continue;
+        }
+        if (rtp.kind != track.kind) {
+          continue;
+        }
+
+        const numPackets = rtp[packets];
+        info(`Track ${track.id} has ${numPackets} ${packets}.`);
+        if (!numPackets) {
+          continue;
+        }
+
+        ok(true, `RTP flowing for ${track.kind} track ${track.id}`);
+        return;
+      }
+      await wait(retryInterval);
+    }
+    throw new Error(
+      `Checking stats for track ${track.id} timed out after ${timeout} ms`
+    );
+  },
+
+  /**
+   * Wait for inbound RTP packet flow for the given MediaStreamTrack.
+   *
+   * @param {object} receiver
+   *        An RTCRtpReceiver to wait for data flow on.
+   * @returns {Promise}
+   *        Returns a promise that resolves once data is flowing.
+   */
+  async waitForInboundRtpFlow(receiver) {
+    return this._waitForRtpFlow(receiver, "inbound-rtp");
+  },
+
+  /**
+   * Wait for outbound RTP packet flow for the given MediaStreamTrack.
+   *
+   * @param {object} sender
+   *        An RTCRtpSender to wait for data flow on.
+   * @returns {Promise}
+   *        Returns a promise that resolves once data is flowing.
+   */
+  async waitForOutboundRtpFlow(sender) {
+    return this._waitForRtpFlow(sender, "outbound-rtp");
+  },
+
+  getExpectedActiveReceivers() {
+    return this._pc
+      .getTransceivers()
+      .filter(
+        t =>
+          !t.stopped &&
+          t.currentDirection &&
+          t.currentDirection != "inactive" &&
+          t.currentDirection != "sendonly"
+      )
+      .filter(({ receiver }) => receiver.track)
+      .map(({ mid, currentDirection, receiver }) => {
+        info(
+          `Found transceiver that should be receiving RTP: mid=${mid}` +
+            ` currentDirection=${currentDirection}` +
+            ` kind=${receiver.track.kind} track-id=${receiver.track.id}`
+        );
+        return receiver;
+      });
+  },
+
+  getExpectedSenders() {
+    return this._pc.getSenders().filter(({ track }) => track);
+  },
+
+  /**
+   * Wait for presence of video flow on all media elements and rtp flow on
+   * all sending and receiving track involved in this test.
+   *
+   * @returns {Promise}
+   *        A promise that resolves when media flows for all elements and tracks
+   */
+  waitForMediaFlow() {
+    const receivers = this.getExpectedActiveReceivers();
+    return Promise.all([
+      ...this.localMediaElements.map(el => this.waitForMediaElementFlow(el)),
+      ...this.remoteMediaElements
+        .filter(({ srcObject }) =>
+          receivers.some(({ track }) =>
+            srcObject.getTracks().some(t => t == track)
+          )
+        )
+        .map(el => this.waitForMediaElementFlow(el)),
+      ...receivers.map(receiver => this.waitForInboundRtpFlow(receiver)),
+      ...this.getExpectedSenders().map(sender =>
+        this.waitForOutboundRtpFlow(sender)
+      ),
+    ]);
+  },
+
+  /**
+   * Check that correct audio (typically a flat tone) is flowing to this
+   * PeerConnection for each transceiver that should be receiving. Uses
+   * WebAudio AnalyserNodes to compare input and output audio data in the
+   * frequency domain.
+   *
+   * @param {object} from
+   *        A PeerConnectionWrapper whose audio RTPSender we use as source for
+   *        the audio flow check.
+   * @returns {Promise}
+   *        A promise that resolves when we're receiving the tone/s from |from|.
+   */
+  async checkReceivingToneFrom(
+    audiocontext,
+    from,
+    cancel = wait(60000, new Error("Tone not detected"))
+  ) {
+    let localTransceivers = this._pc
+      .getTransceivers()
+      .filter(t => t.mid)
+      .filter(t => t.receiver.track.kind == "audio")
+      .sort((t1, t2) => t1.mid < t2.mid);
+    let remoteTransceivers = from._pc
+      .getTransceivers()
+      .filter(t => t.mid)
+      .filter(t => t.receiver.track.kind == "audio")
+      .sort((t1, t2) => t1.mid < t2.mid);
+
+    is(
+      localTransceivers.length,
+      remoteTransceivers.length,
+      "Same number of associated audio transceivers on remote and local."
+    );
+
+    for (let i = 0; i < localTransceivers.length; i++) {
+      is(
+        localTransceivers[i].mid,
+        remoteTransceivers[i].mid,
+        "Transceivers at index " + i + " have the same mid."
+      );
+
+      if (!remoteTransceivers[i].sender.track) {
+        continue;
+      }
+
+      if (
+        remoteTransceivers[i].currentDirection == "recvonly" ||
+        remoteTransceivers[i].currentDirection == "inactive"
+      ) {
+        continue;
+      }
+
+      let sendTrack = remoteTransceivers[i].sender.track;
+      let inputElem = from.getMediaElementForTrack(sendTrack, "local");
+      ok(
+        inputElem,
+        "Remote wrapper should have a media element for track id " +
+          sendTrack.id
+      );
+      let inputAudioStream = from.getStreamForSendTrack(sendTrack);
+      ok(
+        inputAudioStream,
+        "Remote wrapper should have a stream for track id " + sendTrack.id
+      );
+      let inputAnalyser = new AudioStreamAnalyser(
+        audiocontext,
+        inputAudioStream
+      );
+
+      let recvTrack = localTransceivers[i].receiver.track;
+      let outputAudioStream = this.getStreamForRecvTrack(recvTrack);
+      ok(
+        outputAudioStream,
+        "Local wrapper should have a stream for track id " + recvTrack.id
+      );
+      let outputAnalyser = new AudioStreamAnalyser(
+        audiocontext,
+        outputAudioStream
+      );
+
+      let error = null;
+      cancel.then(e => (error = e));
+
+      let indexOfMax = data =>
+        data.reduce((max, val, i) => (val >= data[max] ? i : max), 0);
+
+      await outputAnalyser.waitForAnalysisSuccess(() => {
+        if (error) {
+          throw error;
+        }
+
+        let inputData = inputAnalyser.getByteFrequencyData();
+        let outputData = outputAnalyser.getByteFrequencyData();
+
+        let inputMax = indexOfMax(inputData);
+        let outputMax = indexOfMax(outputData);
+        info(
+          `Comparing maxima; input[${inputMax}] = ${inputData[inputMax]},` +
+            ` output[${outputMax}] = ${outputData[outputMax]}`
+        );
+        if (!inputData[inputMax] || !outputData[outputMax]) {
+          return false;
+        }
+
+        // When the input and output maxima are within reasonable distance (2% of
+        // total length, which means ~10 for length 512) from each other, we can
+        // be sure that the input tone has made it through the peer connection.
+        info(`input data length: ${inputData.length}`);
+        return Math.abs(inputMax - outputMax) < inputData.length * 0.02;
+      });
+    }
+  },
+
+  /**
+   * Check that stats are present by checking for known stats.
+   */
+  async getStats(selector) {
+    const stats = await this._pc.getStats(selector);
+    const dict = {};
+    for (const [k, v] of stats.entries()) {
+      dict[k] = v;
+    }
+    info(`${this}: Got stats: ${JSON.stringify(dict)}`);
+    return stats;
+  },
+
+  /**
+   * Checks that we are getting the media streams we expect.
+   *
+   * @param {object} stats
+   *        The stats to check from this PeerConnectionWrapper
+   */
+  checkStats(stats) {
+    const isRemote = ({ type }) =>
+      ["remote-outbound-rtp", "remote-inbound-rtp"].includes(type);
+    var counters = {};
+    for (let [key, res] of stats) {
+      info("Checking stats for " + key + " : " + res);
+      // validate stats
+      ok(res.id == key, "Coherent stats id");
+      const now = performance.timeOrigin + performance.now();
+      const minimum = performance.timeOrigin;
+      const type = isRemote(res) ? "rtcp" : "rtp";
+      ok(
+        res.timestamp >= minimum,
+        `Valid ${type} timestamp ${res.timestamp} >= ${minimum} (
+            ${res.timestamp - minimum} ms)`
+      );
+      ok(
+        res.timestamp <= now,
+        `Valid ${type} timestamp ${res.timestamp} <= ${now} (
+            ${res.timestamp - now} ms)`
+      );
+      if (isRemote(res)) {
+        continue;
+      }
+      counters[res.type] = (counters[res.type] || 0) + 1;
+
+      switch (res.type) {
+        case "inbound-rtp":
+        case "outbound-rtp":
+          {
+            // Inbound tracks won't have an ssrc if RTP is not flowing.
+            // (eg; negotiated inactive)
+            ok(
+              res.ssrc || res.type == "inbound-rtp",
+              "Outbound RTP stats has an ssrc."
+            );
+
+            if (res.ssrc) {
+              // ssrc is a 32 bit number returned as an unsigned long
+              ok(!/[^0-9]/.test(`${res.ssrc}`), "SSRC is numeric");
+              ok(parseInt(res.ssrc) < Math.pow(2, 32), "SSRC is within limits");
+            }
+
+            if (res.type == "outbound-rtp") {
+              ok(res.packetsSent !== undefined, "Rtp packetsSent");
+              // We assume minimum payload to be 1 byte (guess from RFC 3550)
+              ok(res.bytesSent >= res.packetsSent, "Rtp bytesSent");
+            } else {
+              ok(res.packetsReceived !== undefined, "Rtp packetsReceived");
+              ok(res.bytesReceived >= res.packetsReceived, "Rtp bytesReceived");
+            }
+            if (res.remoteId) {
+              var rem = stats.get(res.remoteId);
+              ok(isRemote(rem), "Remote is rtcp");
+              ok(rem.localId == res.id, "Remote backlink match");
+              if (res.type == "outbound-rtp") {
+                ok(rem.type == "remote-inbound-rtp", "Rtcp is inbound");
+                if (rem.packetsLost) {
+                  ok(
+                    rem.packetsLost >= 0,
+                    "Rtcp packetsLost " + rem.packetsLost + " >= 0"
+                  );
+                  ok(
+                    rem.packetsLost < 1000,
+                    "Rtcp packetsLost " + rem.packetsLost + " < 1000"
+                  );
+                }
+                if (!this.disableRtpCountChecking) {
+                  // no guarantee which one is newer!
+                  // Note: this must change when we add a timestamp field to remote RTCP reports
+                  // and make rem.timestamp be the reception time
+                  if (res.timestamp < rem.timestamp) {
+                    info(
+                      "REVERSED timestamps: rec:" +
+                        rem.packetsReceived +
+                        " time:" +
+                        rem.timestamp +
+                        " sent:" +
+                        res.packetsSent +
+                        " time:" +
+                        res.timestamp
+                    );
+                  }
+                }
+                if (rem.jitter) {
+                  ok(rem.jitter >= 0, "Rtcp jitter " + rem.jitter + " >= 0");
+                  ok(rem.jitter < 5, "Rtcp jitter " + rem.jitter + " < 5 sec");
+                }
+                if (rem.roundTripTime) {
+                  ok(
+                    rem.roundTripTime >= 0,
+                    "Rtcp rtt " + rem.roundTripTime + " >= 0"
+                  );
+                  ok(
+                    rem.roundTripTime < 60,
+                    "Rtcp rtt " + rem.roundTripTime + " < 1 min"
+                  );
+                }
+              } else {
+                ok(rem.type == "remote-outbound-rtp", "Rtcp is outbound");
+                ok(rem.packetsSent !== undefined, "Rtcp packetsSent");
+                ok(rem.bytesSent !== undefined, "Rtcp bytesSent");
+              }
+              ok(rem.ssrc == res.ssrc, "Remote ssrc match");
+            } else {
+              info("No rtcp info received yet");
+            }
+          }
+          break;
+      }
+    }
+
+    var nin = this._pc.getTransceivers().filter(t => {
+      return (
+        !t.stopped &&
+        t.currentDirection != "inactive" &&
+        t.currentDirection != "sendonly"
+      );
+    }).length;
+    const nout = Object.keys(this.expectedLocalTrackInfo).length;
+    var ndata = this.dataChannels.length;
+
+    // TODO(Bug 957145): Restore stronger inbound-rtp test once Bug 948249 is fixed
+    //is((counters["inbound-rtp"] || 0), nin, "Have " + nin + " inbound-rtp stat(s)");
+    ok(
+      (counters["inbound-rtp"] || 0) >= nin,
+      "Have at least " + nin + " inbound-rtp stat(s) *"
+    );
+
+    is(
+      counters["outbound-rtp"] || 0,
+      nout,
+      "Have " + nout + " outbound-rtp stat(s)"
+    );
+
+    var numLocalCandidates = counters["local-candidate"] || 0;
+    var numRemoteCandidates = counters["remote-candidate"] || 0;
+    // If there are no tracks, there will be no stats either.
+    if (nin + nout + ndata > 0) {
+      ok(numLocalCandidates, "Have local-candidate stat(s)");
+      ok(numRemoteCandidates, "Have remote-candidate stat(s)");
+    } else {
+      is(numLocalCandidates, 0, "Have no local-candidate stats");
+      is(numRemoteCandidates, 0, "Have no remote-candidate stats");
+    }
+  },
+
+  /**
+   * Compares the Ice server configured for this PeerConnectionWrapper
+   * with the ICE candidates received in the RTCP stats.
+   *
+   * @param {object} stats
+   *        The stats to be verified for relayed vs. direct connection.
+   */
+  checkStatsIceConnectionType(stats, expectedLocalCandidateType) {
+    let lId;
+    let rId;
+    for (let stat of stats.values()) {
+      if (stat.type == "candidate-pair" && stat.selected) {
+        lId = stat.localCandidateId;
+        rId = stat.remoteCandidateId;
+        break;
+      }
+    }
+    isnot(
+      lId,
+      undefined,
+      "Got local candidate ID " + lId + " for selected pair"
+    );
+    isnot(
+      rId,
+      undefined,
+      "Got remote candidate ID " + rId + " for selected pair"
+    );
+    let lCand = stats.get(lId);
+    let rCand = stats.get(rId);
+    if (!lCand || !rCand) {
+      ok(
+        false,
+        "failed to find candidatepair IDs or stats for local: " +
+          lId +
+          " remote: " +
+          rId
+      );
+      return;
+    }
+
+    info(
+      "checkStatsIceConnectionType verifying: local=" +
+        JSON.stringify(lCand) +
+        " remote=" +
+        JSON.stringify(rCand)
+    );
+    expectedLocalCandidateType = expectedLocalCandidateType || "host";
+    var candidateType = lCand.candidateType;
+    if (lCand.relayProtocol === "tcp" && candidateType === "relay") {
+      candidateType = "relay-tcp";
+    }
+
+    if (lCand.relayProtocol === "tls" && candidateType === "relay") {
+      candidateType = "relay-tls";
+    }
+
+    if (expectedLocalCandidateType === "srflx" && candidateType === "prflx") {
+      // Be forgiving of prflx when expecting srflx, since that can happen due
+      // to timing.
+      candidateType = "srflx";
+    }
+
+    is(
+      candidateType,
+      expectedLocalCandidateType,
+      "Local candidate type is what we expected for selected pair"
+    );
+  },
+
+  /**
+   * Compares amount of established ICE connection according to ICE candidate
+   * pairs in the stats reporting with the expected amount of connection based
+   * on the constraints.
+   *
+   * @param {object} stats
+   *        The stats to check for ICE candidate pairs
+   * @param {object} testOptions
+   *        The test options object from the PeerConnectionTest
+   */
+  checkStatsIceConnections(stats, testOptions) {
+    var numIceConnections = 0;
+    stats.forEach(stat => {
+      if (stat.type === "candidate-pair" && stat.selected) {
+        numIceConnections += 1;
+      }
+    });
+    info("ICE connections according to stats: " + numIceConnections);
+    isnot(
+      numIceConnections,
+      0,
+      "Number of ICE connections according to stats is not zero"
+    );
+    if (testOptions.bundle) {
+      if (testOptions.rtcpmux) {
+        is(numIceConnections, 1, "stats reports exactly 1 ICE connection");
+      } else {
+        is(
+          numIceConnections,
+          2,
+          "stats report exactly 2 ICE connections for media and RTCP"
+        );
+      }
+    } else {
+      var numAudioTransceivers = this._pc
+        .getTransceivers()
+        .filter(transceiver => {
+          return (
+            !transceiver.stopped && transceiver.receiver.track.kind == "audio"
+          );
+        }).length;
+
+      var numVideoTransceivers = this._pc
+        .getTransceivers()
+        .filter(transceiver => {
+          return (
+            !transceiver.stopped && transceiver.receiver.track.kind == "video"
+          );
+        }).length;
+
+      var numExpectedTransports = numAudioTransceivers + numVideoTransceivers;
+      if (!testOptions.rtcpmux) {
+        numExpectedTransports *= 2;
+      }
+
+      if (this.dataChannels.length) {
+        ++numExpectedTransports;
+      }
+
+      info(
+        "expected audio + video + data transports: " + numExpectedTransports
+      );
+      is(
+        numIceConnections,
+        numExpectedTransports,
+        "stats ICE connections matches expected A/V transports"
+      );
+    }
+  },
+
+  expectNegotiationNeeded() {
+    if (!this.observedNegotiationNeeded) {
+      this.observedNegotiationNeeded = new Promise(resolve => {
+        this.onnegotiationneeded = resolve;
+      });
+    }
+  },
+
+  /**
+   * Property-matching function for finding a certain stat in passed-in stats
+   *
+   * @param {object} stats
+   *        The stats to check from this PeerConnectionWrapper
+   * @param {object} props
+   *        The properties to look for
+   * @returns {boolean} Whether an entry containing all match-props was found.
+   */
+  hasStat(stats, props) {
+    for (let res of stats.values()) {
+      var match = true;
+      for (let prop in props) {
+        if (res[prop] !== props[prop]) {
+          match = false;
+          break;
+        }
+      }
+      if (match) {
+        return true;
+      }
+    }
+    return false;
+  },
+
+  /**
+   * Closes the connection
+   */
+  close() {
+    this._pc.close();
+    this.localMediaElements.forEach(e => e.pause());
+    info(this + ": Closed connection.");
+  },
+
+  /**
+   * Returns the string representation of the class
+   *
+   * @returns {String} The string representation
+   */
+  toString() {
+    return "PeerConnectionWrapper (" + this.label + ")";
+  },
+};
+
+// haxx to prevent SimpleTest from failing at window.onload
+function addLoadEvent() {}
+
+function loadScript(...scripts) {
+  return Promise.all(
+    scripts.map(script => {
+      var el = document.createElement("script");
+      if (typeof scriptRelativePath === "string" && script.charAt(0) !== "/") {
+        script = scriptRelativePath + script;
+      }
+      el.src = script;
+      document.head.appendChild(el);
+      return new Promise(r => {
+        el.onload = r;
+        el.onerror = r;
+      });
+    })
+  );
+}
+
+// Ensure SimpleTest.js is loaded before other scripts.
+/* import-globals-from /testing/mochitest/tests/SimpleTest/SimpleTest.js */
+/* import-globals-from head.js */
+/* import-globals-from templates.js */
+/* import-globals-from turnConfig.js */
+/* import-globals-from dataChannel.js */
+/* import-globals-from network.js */
+/* import-globals-from sdpUtils.js */
+
+var scriptsReady = loadScript("/tests/SimpleTest/SimpleTest.js").then(() => {
+  return loadScript(
+    "head.js",
+    "templates.js",
+    "turnConfig.js",
+    "dataChannel.js",
+    "network.js",
+    "sdpUtils.js"
+  );
+});
+
+function createHTML(options) {
+  return scriptsReady.then(() => realCreateHTML(options));
+}
+
+var iceServerWebsocket;
+var iceServersArray = [];
+
+var addTurnsSelfsignedCerts = () => {
+  var gUrl = SimpleTest.getTestFileURL("addTurnsSelfsignedCert.js");
+  var gScript = SpecialPowers.loadChromeScript(gUrl);
+  var certs = [];
+  // If the ICE server is running TURNS, and includes a "cert" attribute in
+  // its JSON, we set up an override that will forgive things like
+  // self-signed for it.
+  iceServersArray.forEach(iceServer => {
+    if (iceServer.hasOwnProperty("cert")) {
+      iceServer.urls.forEach(url => {
+        if (url.startsWith("turns:")) {
+          // Assumes no port or params!
+          certs.push({ cert: iceServer.cert, hostname: url.substr(6) });
+        }
+      });
+    }
+  });
+
+  return new Promise((resolve, reject) => {
+    gScript.addMessageListener("certs-added", () => {
+      resolve();
+    });
+
+    gScript.sendAsyncMessage("add-turns-certs", certs);
+  });
+};
+
+var setupIceServerConfig = useIceServer => {
+  // We disable ICE support for HTTP proxy when using a TURN server, because
+  // mochitest uses a fake HTTP proxy to serve content, which will eat our STUN
+  // packets for TURN TCP.
+  var enableHttpProxy = enable =>
+    SpecialPowers.pushPrefEnv({
+      set: [["media.peerconnection.disable_http_proxy", !enable]],
+    });
+
+  var spawnIceServer = () =>
+    new Promise((resolve, reject) => {
+      iceServerWebsocket = new WebSocket("ws://localhost:8191/");
+      iceServerWebsocket.onopen = event => {
+        info("websocket/process bridge open, starting ICE Server...");
+        iceServerWebsocket.send("iceserver");
+      };
+
+      iceServerWebsocket.onmessage = event => {
+        // The first message will contain the iceServers configuration, subsequent
+        // messages are just logging.
+        info("ICE Server: " + event.data);
+        resolve(event.data);
+      };
+
+      iceServerWebsocket.onerror = () => {
+        reject("ICE Server error: Is the ICE server websocket up?");
+      };
+
+      iceServerWebsocket.onclose = () => {
+        info("ICE Server websocket closed");
+        reject("ICE Server gone before getting configuration");
+      };
+    });
+
+  if (!useIceServer) {
+    info("Skipping ICE Server for this test");
+    return enableHttpProxy(true);
+  }
+
+  return enableHttpProxy(false)
+    .then(spawnIceServer)
+    .then(iceServersStr => {
+      iceServersArray = JSON.parse(iceServersStr);
+    })
+    .then(addTurnsSelfsignedCerts);
+};
+
+async function runNetworkTest(testFunction, fixtureOptions = {}) {
+  let { AppConstants } = SpecialPowers.ChromeUtils.importESModule(
+    "resource://gre/modules/AppConstants.sys.mjs"
+  );
+
+  await scriptsReady;
+  await runTestWhenReady(async options => {
+    await startNetworkAndTest();
+    await setupIceServerConfig(fixtureOptions.useIceServer);
+    await testFunction(options);
+    await networkTestFinished();
+  });
+}
diff --git a/dom/media/webrtc/tests/mochitests/peerconnection_audio_forced_sample_rate.js b/dom/media/webrtc/tests/mochitests/peerconnection_audio_forced_sample_rate.js
new file mode 100644
index 0000000000..d0c647be0d
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/peerconnection_audio_forced_sample_rate.js
@@ -0,0 +1,32 @@
+// This function takes a sample-rate, and tests that audio flows correctly when
+// the sampling-rate at which the MTG runs is not one of the sampling-rates that
+// the MediaPipeline can work with.
+// It is in a separate file because we have an MTG per document, and we want to
+// test multiple sample-rates, so we include it in multiple HTML mochitest
+// files.
+async function test_peerconnection_audio_forced_sample_rate(forcedSampleRate) {
+  await scriptsReady;
+  await pushPrefs(["media.cubeb.force_sample_rate", forcedSampleRate]);
+  await runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    const ac = new AudioContext();
+    test.setMediaConstraints([{ audio: true }], []);
+    test.chain.replace("PC_LOCAL_GUM", [
+      function PC_LOCAL_WEBAUDIO_SOURCE(test) {
+        const oscillator = ac.createOscillator();
+        oscillator.type = "sine";
+        oscillator.frequency.value = 700;
+        oscillator.start();
+        const dest = ac.createMediaStreamDestination();
+        oscillator.connect(dest);
+        test.pcLocal.attachLocalStream(dest.stream);
+      },
+    ]);
+    test.chain.append([
+      function CHECK_REMOTE_AUDIO_FLOW(test) {
+        return test.pcRemote.checkReceivingToneFrom(ac, test.pcLocal);
+      },
+    ]);
+    return test.run();
+  });
+}
diff --git a/dom/media/webrtc/tests/mochitests/sdpUtils.js b/dom/media/webrtc/tests/mochitests/sdpUtils.js
new file mode 100644
index 0000000000..51cae10dba
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/sdpUtils.js
@@ -0,0 +1,398 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+var sdputils = {
+  // Finds the codec id / payload type given a codec format
+  // (e.g., "VP8", "VP9/90000"). `offset` tells us which one to use in case of
+  // multiple matches.
+  findCodecId(sdp, format, offset = 0) {
+    let regex = new RegExp("rtpmap:([0-9]+) " + format, "gi");
+    let match;
+    for (let i = 0; i <= offset; ++i) {
+      match = regex.exec(sdp);
+      if (!match) {
+        throw new Error(
+          "Couldn't find offset " +
+            i +
+            " of codec " +
+            format +
+            " while looking for offset " +
+            offset +
+            " in sdp:\n" +
+            sdp
+        );
+      }
+    }
+    // match[0] is the full matched string
+    // match[1] is the first parenthesis group
+    return match[1];
+  },
+
+  // Returns a list of all payload types, excluding rtx, in an sdp.
+  getPayloadTypes(sdp) {
+    const regex = /^a=rtpmap:([0-9]+) (?:(?!rtx).)*$/gim;
+    const pts = [];
+    for (const [line, pt] of sdp.matchAll(regex)) {
+      pts.push(pt);
+    }
+    return pts;
+  },
+
+  // Finds all the extmap ids in the given sdp.  Note that this does NOT
+  // consider m-sections, so a more generic version would need to
+  // look at each m-section separately.
+  findExtmapIds(sdp) {
+    var sdpExtmapIds = [];
+    extmapRegEx = /^a=extmap:([0-9+])/gm;
+    // must call exec on the regex to get each match in the string
+    while ((searchResults = extmapRegEx.exec(sdp)) !== null) {
+      // returned array has the matched text as the first item,
+      // and then one item for each capturing parenthesis that
+      // matched containing the text that was captured.
+      sdpExtmapIds.push(searchResults[1]);
+    }
+    return sdpExtmapIds;
+  },
+
+  findExtmapIdsUrnsDirections(sdp) {
+    var sdpExtmap = [];
+    extmapRegEx = /^a=extmap:([0-9+])([A-Za-z/]*) ([A-Za-z0-9_:\-\/\.]+)/gm;
+    // must call exec on the regex to get each match in the string
+    while ((searchResults = extmapRegEx.exec(sdp)) !== null) {
+      // returned array has the matched text as the first item,
+      // and then one item for each capturing parenthesis that
+      // matched containing the text that was captured.
+      var idUrn = [];
+      idUrn.push(searchResults[1]);
+      idUrn.push(searchResults[3]);
+      idUrn.push(searchResults[2].slice(1));
+      sdpExtmap.push(idUrn);
+    }
+    return sdpExtmap;
+  },
+
+  verify_unique_extmap_ids(sdp) {
+    const sdpExtmapIds = sdputils.findExtmapIdsUrnsDirections(sdp);
+
+    return sdpExtmapIds.reduce(function (result, item, index) {
+      const [id, urn, dir] = item;
+      ok(
+        !(id in result) || (result[id][0] === urn && result[id][1] === dir),
+        "ID " + id + " is unique ID for " + urn + " and direction " + dir
+      );
+      result[id] = [urn, dir];
+      return result;
+    }, {});
+  },
+
+  getMSections(sdp) {
+    return sdp
+      .split(new RegExp("^m=", "gm"))
+      .slice(1)
+      .map(s => "m=" + s);
+  },
+
+  getAudioMSections(sdp) {
+    return this.getMSections(sdp).filter(section =>
+      section.startsWith("m=audio")
+    );
+  },
+
+  getVideoMSections(sdp) {
+    return this.getMSections(sdp).filter(section =>
+      section.startsWith("m=video")
+    );
+  },
+
+  checkSdpAfterEndOfTrickle(description, testOptions, label) {
+    info("EOC-SDP: " + JSON.stringify(description));
+
+    const checkForTransportAttributes = msection => {
+      info("Checking msection: " + msection);
+      ok(
+        msection.includes("a=end-of-candidates"),
+        label + ": SDP contains end-of-candidates"
+      );
+
+      if (!msection.startsWith("m=application")) {
+        if (testOptions.rtcpmux) {
+          ok(
+            msection.includes("a=rtcp-mux"),
+            label + ": SDP contains rtcp-mux"
+          );
+        } else {
+          ok(msection.includes("a=rtcp:"), label + ": SDP contains rtcp port");
+        }
+      }
+    };
+
+    const hasOwnTransport = msection => {
+      const port0Check = new RegExp(/^m=\S+ 0 /).exec(msection);
+      if (port0Check) {
+        return false;
+      }
+      const midMatch = new RegExp(/\r\na=mid:(\S+)/).exec(msection);
+      if (!midMatch) {
+        return true;
+      }
+      const mid = midMatch[1];
+      const bundleGroupMatch = new RegExp(
+        "\\r\\na=group:BUNDLE \\S.* " + mid + "\\s+"
+      ).exec(description.sdp);
+      return bundleGroupMatch == null;
+    };
+
+    const msectionsWithOwnTransports = this.getMSections(
+      description.sdp
+    ).filter(hasOwnTransport);
+
+    ok(
+      msectionsWithOwnTransports.length,
+      "SDP should contain at least one msection with a transport"
+    );
+    msectionsWithOwnTransports.forEach(checkForTransportAttributes);
+
+    if (testOptions.ssrc) {
+      ok(description.sdp.includes("a=ssrc"), label + ": SDP contains a=ssrc");
+    } else {
+      ok(
+        !description.sdp.includes("a=ssrc"),
+        label + ": SDP does not contain a=ssrc"
+      );
+    }
+  },
+
+  // Note, we don't bother removing the fmtp lines, which makes a good test
+  // for some SDP parsing issues.
+  removeCodec(sdp, codec) {
+    var updated_sdp = sdp.replace(
+      new RegExp("a=rtpmap:" + codec + ".*\\/90000\\r\\n", ""),
+      ""
+    );
+    updated_sdp = updated_sdp.replace(
+      new RegExp("(RTP\\/SAVPF.*)( " + codec + ")(.*\\r\\n)", ""),
+      "$1$3"
+    );
+    updated_sdp = updated_sdp.replace(
+      new RegExp("a=rtcp-fb:" + codec + " nack\\r\\n", ""),
+      ""
+    );
+    updated_sdp = updated_sdp.replace(
+      new RegExp("a=rtcp-fb:" + codec + " nack pli\\r\\n", ""),
+      ""
+    );
+    updated_sdp = updated_sdp.replace(
+      new RegExp("a=rtcp-fb:" + codec + " ccm fir\\r\\n", ""),
+      ""
+    );
+    return updated_sdp;
+  },
+
+  removeAllButPayloadType(sdp, pt) {
+    return sdp.replace(
+      new RegExp("m=(\\w+ \\w+) UDP/TLS/RTP/SAVPF .*" + pt + ".*\\r\\n", "gi"),
+      "m=$1 UDP/TLS/RTP/SAVPF " + pt + "\r\n"
+    );
+  },
+
+  removeRtpMapForPayloadType(sdp, pt) {
+    return sdp.replace(new RegExp("a=rtpmap:" + pt + ".*\\r\\n", "gi"), "");
+  },
+
+  removeRtcpMux(sdp) {
+    return sdp.replace(/a=rtcp-mux\r\n/g, "");
+  },
+
+  removeSSRCs(sdp) {
+    return sdp.replace(/a=ssrc.*\r\n/g, "");
+  },
+
+  removeBundle(sdp) {
+    return sdp.replace(/a=group:BUNDLE .*\r\n/g, "");
+  },
+
+  reduceAudioMLineToPcmuPcma(sdp) {
+    return sdp.replace(
+      /m=audio .*\r\n/g,
+      "m=audio 9 UDP/TLS/RTP/SAVPF 0 8\r\n"
+    );
+  },
+
+  setAllMsectionsInactive(sdp) {
+    return sdp
+      .replace(/\r\na=sendrecv/g, "\r\na=inactive")
+      .replace(/\r\na=sendonly/g, "\r\na=inactive")
+      .replace(/\r\na=recvonly/g, "\r\na=inactive");
+  },
+
+  removeAllRtpMaps(sdp) {
+    return sdp.replace(/a=rtpmap:.*\r\n/g, "");
+  },
+
+  reduceAudioMLineToDynamicPtAndOpus(sdp) {
+    return sdp.replace(
+      /m=audio .*\r\n/g,
+      "m=audio 9 UDP/TLS/RTP/SAVPF 101 109\r\n"
+    );
+  },
+
+  addTiasBps(sdp, bps) {
+    return sdp.replace(/c=IN (.*)\r\n/g, "c=IN $1\r\nb=TIAS:" + bps + "\r\n");
+  },
+
+  removeSimulcastProperties(sdp) {
+    return sdp
+      .replace(/a=simulcast:.*\r\n/g, "")
+      .replace(/a=rid:.*\r\n/g, "")
+      .replace(
+        /a=extmap:[^\s]* urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id.*\r\n/g,
+        ""
+      )
+      .replace(
+        /a=extmap:[^\s]* urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id.*\r\n/g,
+        ""
+      );
+  },
+
+  transferSimulcastProperties(offer_sdp, answer_sdp) {
+    if (!offer_sdp.includes("a=simulcast:")) {
+      return answer_sdp;
+    }
+    ok(
+      offer_sdp.includes("a=simulcast:send "),
+      "Offer contains simulcast attribute"
+    );
+    var o_simul = offer_sdp.match(/simulcast:send (.*)([\n$])*/i);
+    var new_answer_sdp = answer_sdp + "a=simulcast:recv " + o_simul[1] + "\r\n";
+    ok(offer_sdp.includes("a=rid:"), "Offer contains RID attribute");
+    var o_rids = offer_sdp.match(/a=rid:(.*)/gi);
+    o_rids.forEach(o_rid => {
+      new_answer_sdp = new_answer_sdp + o_rid.replace(/send/, "recv") + "\r\n";
+    });
+    var extmap_id = offer_sdp.match(
+      "a=extmap:([0-9+])/sendonly urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id"
+    );
+    ok(extmap_id != null, "Offer contains RID RTP header extension");
+    new_answer_sdp =
+      new_answer_sdp +
+      "a=extmap:" +
+      extmap_id[1] +
+      "/recvonly urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\r\n";
+    var extmap_id = offer_sdp.match(
+      "a=extmap:([0-9+])/sendonly urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id"
+    );
+    if (extmap_id != null) {
+      new_answer_sdp =
+        new_answer_sdp +
+        "a=extmap:" +
+        extmap_id[1] +
+        "/recvonly urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\r\n";
+    }
+
+    return new_answer_sdp;
+  },
+
+  verifySdp(
+    desc,
+    expectedType,
+    offerConstraintsList,
+    offerOptions,
+    testOptions
+  ) {
+    info("Examining this SessionDescription: " + JSON.stringify(desc));
+    info("offerConstraintsList: " + JSON.stringify(offerConstraintsList));
+    info("offerOptions: " + JSON.stringify(offerOptions));
+    ok(desc, "SessionDescription is not null");
+    is(desc.type, expectedType, "SessionDescription type is " + expectedType);
+    ok(desc.sdp.length > 10, "SessionDescription body length is plausible");
+    ok(desc.sdp.includes("a=ice-ufrag"), "ICE username is present in SDP");
+    ok(desc.sdp.includes("a=ice-pwd"), "ICE password is present in SDP");
+    ok(desc.sdp.includes("a=fingerprint"), "ICE fingerprint is present in SDP");
+    //TODO: update this for loopback support bug 1027350
+    ok(
+      !desc.sdp.includes(LOOPBACK_ADDR),
+      "loopback interface is absent from SDP"
+    );
+    var requiresTrickleIce = !desc.sdp.includes("a=candidate");
+    if (requiresTrickleIce) {
+      info("No ICE candidate in SDP -> requiring trickle ICE");
+    } else {
+      info("at least one ICE candidate is present in SDP");
+    }
+
+    //TODO: how can we check for absence/presence of m=application?
+
+    var audioTracks =
+      sdputils.countTracksInConstraint("audio", offerConstraintsList) ||
+      (offerOptions && offerOptions.offerToReceiveAudio ? 1 : 0);
+
+    info("expected audio tracks: " + audioTracks);
+    if (audioTracks == 0) {
+      ok(!desc.sdp.includes("m=audio"), "audio m-line is absent from SDP");
+    } else {
+      ok(desc.sdp.includes("m=audio"), "audio m-line is present in SDP");
+      is(
+        testOptions.opus,
+        desc.sdp.includes("a=rtpmap:109 opus/48000/2"),
+        "OPUS codec is present in SDP"
+      );
+      //TODO: ideally the rtcp-mux should be for the m=audio, and not just
+      //      anywhere in the SDP (JS SDP parser bug 1045429)
+      is(
+        testOptions.rtcpmux,
+        desc.sdp.includes("a=rtcp-mux"),
+        "RTCP Mux is offered in SDP"
+      );
+    }
+
+    var videoTracks =
+      sdputils.countTracksInConstraint("video", offerConstraintsList) ||
+      (offerOptions && offerOptions.offerToReceiveVideo ? 1 : 0);
+
+    info("expected video tracks: " + videoTracks);
+    if (videoTracks == 0) {
+      ok(!desc.sdp.includes("m=video"), "video m-line is absent from SDP");
+    } else {
+      ok(desc.sdp.includes("m=video"), "video m-line is present in SDP");
+      if (testOptions.h264) {
+        ok(
+          desc.sdp.includes("a=rtpmap:126 H264/90000") ||
+            desc.sdp.includes("a=rtpmap:97 H264/90000"),
+          "H.264 codec is present in SDP"
+        );
+      } else {
+        ok(
+          desc.sdp.includes("a=rtpmap:120 VP8/90000") ||
+            desc.sdp.includes("a=rtpmap:121 VP9/90000"),
+          "VP8 or VP9 codec is present in SDP"
+        );
+      }
+      is(
+        testOptions.rtcpmux,
+        desc.sdp.includes("a=rtcp-mux"),
+        "RTCP Mux is offered in SDP"
+      );
+      is(
+        testOptions.ssrc,
+        desc.sdp.includes("a=ssrc"),
+        "a=ssrc signaled in SDP"
+      );
+    }
+
+    return requiresTrickleIce;
+  },
+
+  /**
+   * Counts the amount of audio tracks in a given media constraint.
+   *
+   * @param constraints
+   *        The contraint to be examined.
+   */
+  countTracksInConstraint(type, constraints) {
+    if (!Array.isArray(constraints)) {
+      return 0;
+    }
+    return constraints.reduce((sum, c) => sum + (c[type] ? 1 : 0), 0);
+  },
+};
diff --git a/dom/media/webrtc/tests/mochitests/simulcast.js b/dom/media/webrtc/tests/mochitests/simulcast.js
new file mode 100644
index 0000000000..0af36478c4
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/simulcast.js
@@ -0,0 +1,232 @@
+"use strict";
+/* Helper functions to munge SDP and split the sending track into
+ * separate tracks on the receiving end. This can be done in a number
+ * of ways, the one used here uses the fact that the MID and RID header
+ * extensions which are used for packet routing share the same wire
+ * format. The receiver interprets the rids from the sender as mids
+ * which allows receiving the different spatial resolutions on separate
+ * m-lines and tracks.
+ */
+
+// Borrowed from wpt, with some dependencies removed.
+
+const ridExtensions = [
+  "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id",
+  "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id",
+];
+
+function ridToMid(description, rids) {
+  const sections = SDPUtils.splitSections(description.sdp);
+  const dtls = SDPUtils.getDtlsParameters(sections[1], sections[0]);
+  const ice = SDPUtils.getIceParameters(sections[1], sections[0]);
+  const rtpParameters = SDPUtils.parseRtpParameters(sections[1]);
+  const setupValue = description.sdp.match(/a=setup:(.*)/)[1];
+  const directionValue =
+    description.sdp.match(/a=sendrecv|a=sendonly|a=recvonly|a=inactive/) ||
+    "a=sendrecv";
+  const mline = SDPUtils.parseMLine(sections[1]);
+
+  // Skip mid extension; we are replacing it with the rid extmap
+  rtpParameters.headerExtensions = rtpParameters.headerExtensions.filter(
+    ext => ext.uri != "urn:ietf:params:rtp-hdrext:sdes:mid"
+  );
+
+  for (const ext of rtpParameters.headerExtensions) {
+    if (ext.uri == "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id") {
+      ext.uri = "urn:ietf:params:rtp-hdrext:sdes:mid";
+    }
+  }
+
+  // Filter rtx as we have no way to (re)interpret rrid.
+  // Not doing this makes probing use RTX, it's not understood and ramp-up is slower.
+  rtpParameters.codecs = rtpParameters.codecs.filter(
+    c => c.name.toUpperCase() !== "RTX"
+  );
+
+  if (!rids) {
+    rids = Array.from(description.sdp.matchAll(/a=rid:(.*) send/g)).map(
+      r => r[1]
+    );
+  }
+
+  let sdp =
+    SDPUtils.writeSessionBoilerplate() +
+    SDPUtils.writeDtlsParameters(dtls, setupValue) +
+    SDPUtils.writeIceParameters(ice) +
+    "a=group:BUNDLE " +
+    rids.join(" ") +
+    "\r\n";
+  const baseRtpDescription = SDPUtils.writeRtpDescription(
+    mline.kind,
+    rtpParameters
+  );
+  for (const rid of rids) {
+    sdp +=
+      baseRtpDescription +
+      "a=mid:" +
+      rid +
+      "\r\n" +
+      "a=msid:rid-" +
+      rid +
+      " rid-" +
+      rid +
+      "\r\n";
+    sdp += directionValue + "\r\n";
+  }
+  return sdp;
+}
+
+function midToRid(description, localDescription, rids) {
+  const sections = SDPUtils.splitSections(description.sdp);
+  const dtls = SDPUtils.getDtlsParameters(sections[1], sections[0]);
+  const ice = SDPUtils.getIceParameters(sections[1], sections[0]);
+  const rtpParameters = SDPUtils.parseRtpParameters(sections[1]);
+  const setupValue = description.sdp.match(/a=setup:(.*)/)[1];
+  const directionValue =
+    description.sdp.match(/a=sendrecv|a=sendonly|a=recvonly|a=inactive/) ||
+    "a=sendrecv";
+  const mline = SDPUtils.parseMLine(sections[1]);
+
+  // Skip rid extensions; we are replacing them with the mid extmap
+  rtpParameters.headerExtensions = rtpParameters.headerExtensions.filter(
+    ext => !ridExtensions.includes(ext.uri)
+  );
+
+  for (const ext of rtpParameters.headerExtensions) {
+    if (ext.uri == "urn:ietf:params:rtp-hdrext:sdes:mid") {
+      ext.uri = "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id";
+    }
+  }
+
+  const localMid = localDescription
+    ? SDPUtils.getMid(SDPUtils.splitSections(localDescription.sdp)[1])
+    : "0";
+
+  if (!rids) {
+    rids = [];
+    for (let i = 1; i < sections.length; i++) {
+      rids.push(SDPUtils.getMid(sections[i]));
+    }
+  }
+
+  let sdp =
+    SDPUtils.writeSessionBoilerplate() +
+    SDPUtils.writeDtlsParameters(dtls, setupValue) +
+    SDPUtils.writeIceParameters(ice) +
+    "a=group:BUNDLE " +
+    localMid +
+    "\r\n";
+  sdp += SDPUtils.writeRtpDescription(mline.kind, rtpParameters);
+  // Although we are converting mids to rids, we still need a mid.
+  // The first one will be consistent with trickle ICE candidates.
+  sdp += "a=mid:" + localMid + "\r\n";
+  sdp += directionValue + "\r\n";
+
+  for (const rid of rids) {
+    const stringrid = String(rid); // allow integers
+    const choices = stringrid.split(",");
+    choices.forEach(choice => {
+      sdp += "a=rid:" + choice + " recv\r\n";
+    });
+  }
+  if (rids.length) {
+    sdp += "a=simulcast:recv " + rids.join(";") + "\r\n";
+  }
+
+  return sdp;
+}
+
+async function doOfferToSendSimulcast(offerer, answerer) {
+  await offerer.setLocalDescription();
+
+  // Is this a renegotiation? If so, we cannot remove (or reorder!) any mids,
+  // even if some rids have been removed or reordered.
+  let mids = [];
+  if (answerer.localDescription) {
+    // Renegotiation. Mids must be the same as before, because renegotiation
+    // can never remove or reorder mids, nor can it expand the simulcast
+    // envelope.
+    mids = [...answerer.localDescription.sdp.matchAll(/a=mid:(.*)/g)].map(
+      e => e[1]
+    );
+  } else {
+    // First negotiation; the mids will be exactly the same as the rids
+    const simulcastAttr = offerer.localDescription.sdp.match(
+      /a=simulcast:send (.*)/
+    );
+    if (simulcastAttr) {
+      mids = simulcastAttr[1].split(";");
+    }
+  }
+
+  const nonSimulcastOffer = ridToMid(offerer.localDescription, mids);
+  await answerer.setRemoteDescription({
+    type: "offer",
+    sdp: nonSimulcastOffer,
+  });
+}
+
+async function doAnswerToRecvSimulcast(offerer, answerer, rids) {
+  await answerer.setLocalDescription();
+  const simulcastAnswer = midToRid(
+    answerer.localDescription,
+    offerer.localDescription,
+    rids
+  );
+  await offerer.setRemoteDescription({ type: "answer", sdp: simulcastAnswer });
+}
+
+async function doOfferToRecvSimulcast(offerer, answerer, rids) {
+  await offerer.setLocalDescription();
+  const simulcastOffer = midToRid(
+    offerer.localDescription,
+    answerer.localDescription,
+    rids
+  );
+  await answerer.setRemoteDescription({ type: "offer", sdp: simulcastOffer });
+}
+
+async function doAnswerToSendSimulcast(offerer, answerer) {
+  await answerer.setLocalDescription();
+
+  // See which mids the offerer had; it will barf if we remove or reorder them
+  const mids = [...offerer.localDescription.sdp.matchAll(/a=mid:(.*)/g)].map(
+    e => e[1]
+  );
+
+  const nonSimulcastAnswer = ridToMid(answerer.localDescription, mids);
+  await offerer.setRemoteDescription({
+    type: "answer",
+    sdp: nonSimulcastAnswer,
+  });
+}
+
+async function doOfferToSendSimulcastAndAnswer(offerer, answerer, rids) {
+  await doOfferToSendSimulcast(offerer, answerer);
+  await doAnswerToRecvSimulcast(offerer, answerer, rids);
+}
+
+async function doOfferToRecvSimulcastAndAnswer(offerer, answerer, rids) {
+  await doOfferToRecvSimulcast(offerer, answerer, rids);
+  await doAnswerToSendSimulcast(offerer, answerer);
+}
+
+// This would be useful for cases other than simulcast, but we do not use it
+// anywhere else right now, nor do we have a place for wpt-friendly helpers at
+// the moment.
+function createPlaybackElement(track) {
+  const elem = document.createElement(track.kind);
+  elem.autoplay = true;
+  elem.srcObject = new MediaStream([track]);
+  elem.id = track.id;
+  return elem;
+}
+
+async function getPlaybackWithLoadedMetadata(track) {
+  const elem = createPlaybackElement(track);
+  return new Promise(resolve => {
+    elem.addEventListener("loadedmetadata", () => {
+      resolve(elem);
+    });
+  });
+}
diff --git a/dom/media/webrtc/tests/mochitests/stats.js b/dom/media/webrtc/tests/mochitests/stats.js
new file mode 100644
index 0000000000..475f8eeca9
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/stats.js
@@ -0,0 +1,1596 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+"use strict";
+
+const statsExpectedByType = {
+  "inbound-rtp": {
+    expected: [
+      "trackIdentifier",
+      "id",
+      "timestamp",
+      "type",
+      "ssrc",
+      "mediaType",
+      "kind",
+      "codecId",
+      "packetsReceived",
+      "packetsLost",
+      "packetsDiscarded",
+      "bytesReceived",
+      "jitter",
+      "lastPacketReceivedTimestamp",
+      "headerBytesReceived",
+      // Always missing from libwebrtc stats
+      // "estimatedPlayoutTimestamp",
+      "jitterBufferDelay",
+      "jitterBufferEmittedCount",
+    ],
+    optional: ["remoteId", "nackCount", "qpSum"],
+    localVideoOnly: [
+      "firCount",
+      "pliCount",
+      "framesDecoded",
+      "framesDropped",
+      "discardedPackets",
+      "framesPerSecond",
+      "frameWidth",
+      "frameHeight",
+      "framesReceived",
+      "totalDecodeTime",
+      "totalInterFrameDelay",
+      "totalProcessingDelay",
+      "totalSquaredInterFrameDelay",
+    ],
+    localAudioOnly: [
+      "totalSamplesReceived",
+      // libwebrtc doesn't seem to do FEC for video
+      "fecPacketsReceived",
+      "fecPacketsDiscarded",
+      "concealedSamples",
+      "silentConcealedSamples",
+      "concealmentEvents",
+      "insertedSamplesForDeceleration",
+      "removedSamplesForAcceleration",
+      "audioLevel",
+      "totalAudioEnergy",
+      "totalSamplesDuration",
+    ],
+    unimplemented: [
+      "mediaTrackId",
+      "transportId",
+      "associateStatsId",
+      "sliCount",
+      "packetsRepaired",
+      "fractionLost",
+      "burstPacketsLost",
+      "burstLossCount",
+      "burstDiscardCount",
+      "gapDiscardRate",
+      "gapLossRate",
+    ],
+    deprecated: ["mozRtt", "isRemote"],
+  },
+  "outbound-rtp": {
+    expected: [
+      "id",
+      "timestamp",
+      "type",
+      "ssrc",
+      "mediaType",
+      "kind",
+      "codecId",
+      "packetsSent",
+      "bytesSent",
+      "remoteId",
+      "headerBytesSent",
+      "retransmittedPacketsSent",
+      "retransmittedBytesSent",
+    ],
+    optional: ["nackCount", "qpSum"],
+    localAudioOnly: [],
+    localVideoOnly: [
+      "framesEncoded",
+      "firCount",
+      "pliCount",
+      "frameWidth",
+      "frameHeight",
+      "framesSent",
+      "hugeFramesSent",
+      "totalEncodeTime",
+      "totalEncodedBytesTarget",
+    ],
+    unimplemented: ["mediaTrackId", "transportId", "sliCount", "targetBitrate"],
+    deprecated: ["isRemote"],
+  },
+  "remote-inbound-rtp": {
+    expected: [
+      "id",
+      "timestamp",
+      "type",
+      "ssrc",
+      "mediaType",
+      "kind",
+      "codecId",
+      "packetsLost",
+      "jitter",
+      "localId",
+      "totalRoundTripTime",
+      "fractionLost",
+      "roundTripTimeMeasurements",
+    ],
+    optional: ["roundTripTime", "nackCount", "packetsReceived"],
+    unimplemented: [
+      "mediaTrackId",
+      "transportId",
+      "packetsDiscarded",
+      "associateStatsId",
+      "sliCount",
+      "packetsRepaired",
+      "burstPacketsLost",
+      "burstLossCount",
+      "burstDiscardCount",
+      "gapDiscardRate",
+      "gapLossRate",
+    ],
+    deprecated: ["mozRtt", "isRemote"],
+  },
+  "remote-outbound-rtp": {
+    expected: [
+      "id",
+      "timestamp",
+      "type",
+      "ssrc",
+      "mediaType",
+      "kind",
+      "codecId",
+      "packetsSent",
+      "bytesSent",
+      "localId",
+      "remoteTimestamp",
+    ],
+    optional: ["nackCount"],
+    unimplemented: ["mediaTrackId", "transportId", "sliCount", "targetBitrate"],
+    deprecated: ["isRemote"],
+  },
+  "media-source": {
+    expected: ["id", "timestamp", "type", "trackIdentifier", "kind"],
+    unimplemented: [
+      "audioLevel",
+      "totalAudioEnergy",
+      "totalSamplesDuration",
+      "echoReturnLoss",
+      "echoReturnLossEnhancement",
+      "droppedSamplesDuration",
+      "droppedSamplesEvents",
+      "totalCaptureDelay",
+      "totalSamplesCaptured",
+      "width",
+      "height",
+      "frames",
+      "framesPerSecond",
+    ],
+    optional: [],
+    deprecated: [],
+  },
+  csrc: { skip: true },
+  codec: {
+    expected: [
+      "timestamp",
+      "type",
+      "id",
+      "payloadType",
+      "transportId",
+      "mimeType",
+      "clockRate",
+      "sdpFmtpLine",
+    ],
+    optional: ["codecType", "channels"],
+    unimplemented: [],
+    deprecated: [],
+  },
+  "peer-connection": { skip: true },
+  "data-channel": { skip: true },
+  track: { skip: true },
+  transport: { skip: true },
+  "candidate-pair": {
+    expected: [
+      "id",
+      "timestamp",
+      "type",
+      "transportId",
+      "localCandidateId",
+      "remoteCandidateId",
+      "state",
+      "priority",
+      "nominated",
+      "writable",
+      "readable",
+      "bytesSent",
+      "bytesReceived",
+      "lastPacketSentTimestamp",
+      "lastPacketReceivedTimestamp",
+    ],
+    optional: ["selected"],
+    unimplemented: [
+      "totalRoundTripTime",
+      "currentRoundTripTime",
+      "availableOutgoingBitrate",
+      "availableIncomingBitrate",
+      "requestsReceived",
+      "requestsSent",
+      "responsesReceived",
+      "responsesSent",
+      "retransmissionsReceived",
+      "retransmissionsSent",
+      "consentRequestsSent",
+    ],
+    deprecated: [],
+  },
+  "local-candidate": {
+    expected: [
+      "id",
+      "timestamp",
+      "type",
+      "address",
+      "protocol",
+      "port",
+      "candidateType",
+      "priority",
+    ],
+    optional: ["relayProtocol", "proxied"],
+    unimplemented: ["networkType", "url", "transportId"],
+    deprecated: [
+      "candidateId",
+      "portNumber",
+      "ipAddress",
+      "componentId",
+      "mozLocalTransport",
+      "transport",
+    ],
+  },
+  "remote-candidate": {
+    expected: [
+      "id",
+      "timestamp",
+      "type",
+      "address",
+      "protocol",
+      "port",
+      "candidateType",
+      "priority",
+    ],
+    optional: ["relayProtocol", "proxied"],
+    unimplemented: ["networkType", "url", "transportId"],
+    deprecated: [
+      "candidateId",
+      "portNumber",
+      "ipAddress",
+      "componentId",
+      "mozLocalTransport",
+      "transport",
+    ],
+  },
+  certificate: { skip: true },
+};
+
+["in", "out"].forEach(pre => {
+  let s = statsExpectedByType[pre + "bound-rtp"];
+  s.optional = [...s.optional, ...s.localVideoOnly, ...s.localAudioOnly];
+});
+
+//
+//  Checks that the fields in a report conform to the expectations in
+// statExpectedByType
+//
+function checkExpectedFields(report) {
+  report.forEach(stat => {
+    let expectations = statsExpectedByType[stat.type];
+    ok(expectations, "Stats type " + stat.type + " was expected");
+    // If the type is not expected or if it is flagged for skipping continue to
+    // the next
+    if (!expectations || expectations.skip) {
+      return;
+    }
+    // Check that all required fields exist
+    expectations.expected.forEach(field => {
+      ok(
+        field in stat,
+        "Expected stat field " + stat.type + "." + field + " exists"
+      );
+    });
+    // Check that each field is either expected or optional
+    let allowed = [...expectations.expected, ...expectations.optional];
+    Object.keys(stat).forEach(field => {
+      ok(
+        allowed.includes(field),
+        "Stat field " +
+          stat.type +
+          "." +
+          field +
+          ` is allowed. ${JSON.stringify(stat)}`
+      );
+    });
+
+    //
+    // Ensure that unimplemented fields are not implemented
+    //   note: if a field is implemented it should be moved to expected or
+    //   optional.
+    //
+    expectations.unimplemented.forEach(field => {
+      ok(
+        !Object.keys(stat).includes(field),
+        "Unimplemented field " + stat.type + "." + field + " does not exist."
+      );
+    });
+
+    //
+    // Ensure that all deprecated fields are not present
+    //
+    expectations.deprecated.forEach(field => {
+      ok(
+        !Object.keys(stat).includes(field),
+        "Deprecated field " + stat.type + "." + field + " does not exist."
+      );
+    });
+  });
+}
+
+function pedanticChecks(report) {
+  // Check that report is only-maplike
+  [...report.keys()].forEach(key =>
+    is(
+      report[key],
+      undefined,
+      `Report is not dictionary like, it lacks a property for key ${key}`
+    )
+  );
+  report.forEach((statObj, mapKey) => {
+    info(`"${mapKey} = ${JSON.stringify(statObj, null, 2)}`);
+  });
+  // eslint-disable-next-line complexity
+  report.forEach((statObj, mapKey) => {
+    let tested = {};
+    // Record what fields get tested.
+    // To access a field foo without marking it as tested use stat.inner.foo
+    let stat = new Proxy(statObj, {
+      get(stat, key) {
+        if (key == "inner") {
+          return stat;
+        }
+        tested[key] = true;
+        return stat[key];
+      },
+    });
+
+    let expectations = statsExpectedByType[stat.type];
+
+    if (expectations.skip) {
+      return;
+    }
+
+    // All stats share the following attributes inherited from RTCStats
+    is(stat.id, mapKey, stat.type + ".id is the same as the report key.");
+
+    // timestamp
+    ok(stat.timestamp >= 0, stat.type + ".timestamp is not less than 0");
+    // If the timebase for the timestamp is not properly set the timestamp
+    // will appear relative to the year 1970; Bug 1495446
+    const date = new Date(stat.timestamp);
+    ok(
+      date.getFullYear() > 1970,
+      `${stat.type}.timestamp is relative to current time, date=${date}`
+    );
+    //
+    // RTCStreamStats attributes with common behavior
+    //
+    // inbound-rtp, outbound-rtp, remote-inbound-rtp, remote-outbound-rtp
+    // inherit from RTCStreamStats
+    if (
+      [
+        "inbound-rtp",
+        "outbound-rtp",
+        "remote-inbound-rtp",
+        "remote-outbound-rtp",
+      ].includes(stat.type)
+    ) {
+      const isRemote = stat.type.startsWith("remote-");
+      //
+      // Common RTCStreamStats fields
+      //
+
+      // SSRC
+      ok(stat.ssrc, stat.type + ".ssrc has a value");
+
+      // kind
+      ok(
+        ["audio", "video"].includes(stat.kind),
+        stat.type + ".kind is 'audio' or 'video'"
+      );
+
+      // mediaType, renamed to kind but remains for backward compability.
+      ok(
+        ["audio", "video"].includes(stat.mediaType),
+        stat.type + ".mediaType is 'audio' or 'video'"
+      );
+
+      ok(stat.kind == stat.mediaType, "kind equals legacy mediaType");
+
+      // codecId
+      ok(stat.codecId, `${stat.type}.codecId has a value`);
+      ok(report.has(stat.codecId), `codecId ${stat.codecId} exists in report`);
+      is(
+        report.get(stat.codecId).type,
+        "codec",
+        `codecId ${stat.codecId} in report is codec type`
+      );
+      is(
+        report.get(stat.codecId).mimeType.slice(0, 5),
+        stat.kind,
+        `codecId ${stat.codecId} in report is for a mimeType of the same ` +
+          `media type as the referencing rtp stream stat`
+      );
+
+      if (isRemote) {
+        // local id
+        if (stat.localId) {
+          ok(
+            report.has(stat.localId),
+            `localId ${stat.localId} exists in report.`
+          );
+          is(
+            report.get(stat.localId).ssrc,
+            stat.ssrc,
+            "remote ssrc and local ssrc match."
+          );
+          is(
+            report.get(stat.localId).remoteId,
+            stat.id,
+            "local object has remote object as it's own remote object."
+          );
+        }
+      } else {
+        // remote id
+        if (stat.remoteId) {
+          ok(
+            report.has(stat.remoteId),
+            `remoteId ${stat.remoteId} exists in report.`
+          );
+          is(
+            report.get(stat.remoteId).ssrc,
+            stat.ssrc,
+            "remote ssrc and local ssrc match."
+          );
+          is(
+            report.get(stat.remoteId).localId,
+            stat.id,
+            "remote object has local object as it's own local object."
+          );
+        }
+      }
+
+      // nackCount
+      if (stat.nackCount) {
+        ok(
+          stat.nackCount >= 0,
+          `${stat.type}.nackCount is sane (${stat.kind}).`
+        );
+      }
+
+      if (!isRemote && stat.inner.kind == "video") {
+        // firCount
+        ok(
+          stat.firCount >= 0 && stat.firCount < 100,
+          `${stat.type}.firCount is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.firCount}`
+        );
+
+        // pliCount
+        ok(
+          stat.pliCount >= 0 && stat.pliCount < 200,
+          `${stat.type}.pliCount is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.pliCount}`
+        );
+
+        // qpSum
+        if (stat.qpSum !== undefined) {
+          ok(
+            stat.qpSum > 0,
+            `${stat.type}.qpSum is at least 0 ` +
+              `${stat.kind} test. value=${stat.qpSum}`
+          );
+        }
+      } else {
+        is(
+          stat.qpSum,
+          undefined,
+          `${stat.type}.qpSum does not exist when stat.kind != video`
+        );
+      }
+    }
+
+    if (stat.type == "inbound-rtp") {
+      //
+      // Required fields
+      //
+
+      // trackIdentifier
+      is(typeof stat.trackIdentifier, "string");
+      isnot(stat.trackIdentifier, "");
+
+      // packetsReceived
+      ok(
+        stat.packetsReceived >= 0 && stat.packetsReceived < 10 ** 5,
+        `${stat.type}.packetsReceived is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.packetsReceived}`
+      );
+
+      // packetsDiscarded
+      ok(
+        stat.packetsDiscarded >= 0 && stat.packetsDiscarded < 100,
+        `${stat.type}.packetsDiscarded is sane number for a short test. ` +
+          `value=${stat.packetsDiscarded}`
+      );
+      // bytesReceived
+      ok(
+        stat.bytesReceived >= 0 && stat.bytesReceived < 10 ** 9, // Not a magic number, just a guess
+        `${stat.type}.bytesReceived is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.bytesReceived}`
+      );
+
+      // packetsLost
+      ok(
+        stat.packetsLost < 100,
+        `${stat.type}.packetsLost is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.packetsLost}`
+      );
+
+      // This should be much lower for audio, TODO: Bug 1330575
+      let expectedJitter = stat.kind == "video" ? 0.5 : 1;
+      // jitter
+      ok(
+        stat.jitter < expectedJitter,
+        `${stat.type}.jitter is sane number for a ${stat.kind} ` +
+          `local only test. value=${stat.jitter}`
+      );
+
+      // lastPacketReceivedTimestamp
+      ok(
+        stat.lastPacketReceivedTimestamp !== undefined,
+        `${stat.type}.lastPacketReceivedTimestamp has a value`
+      );
+
+      // headerBytesReceived
+      ok(
+        stat.headerBytesReceived >= 0 && stat.headerBytesReceived < 50000,
+        `${stat.type}.headerBytesReceived is sane for a short test. ` +
+          `value=${stat.headerBytesReceived}`
+      );
+
+      // Always missing from libwebrtc stats
+      // estimatedPlayoutTimestamp
+      // ok(
+      //   stat.estimatedPlayoutTimestamp !== undefined,
+      //   `${stat.type}.estimatedPlayoutTimestamp has a value`
+      // );
+
+      // jitterBufferEmittedCount
+      let expectedJitterBufferEmmitedCount = stat.kind == "video" ? 7 : 1000;
+      ok(
+        stat.jitterBufferEmittedCount > expectedJitterBufferEmmitedCount,
+        `${stat.type}.jitterBufferEmittedCount is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.jitterBufferEmittedCount}`
+      );
+
+      // jitterBufferDelay
+      let avgJitterBufferDelay =
+        stat.jitterBufferDelay / stat.jitterBufferEmittedCount;
+      ok(
+        avgJitterBufferDelay > 0.01 && avgJitterBufferDelay < 10,
+        `${stat.type}.jitterBufferDelay is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.jitterBufferDelay}/${stat.jitterBufferEmittedCount}=${avgJitterBufferDelay}`
+      );
+
+      //
+      // Optional fields
+      //
+
+      //
+      // Local audio only stats
+      //
+      if (stat.inner.kind != "audio") {
+        expectations.localAudioOnly.forEach(field => {
+          ok(
+            stat[field] === undefined,
+            `${stat.type} does not have field ${field}` +
+              ` when kind is not 'audio'`
+          );
+        });
+      } else {
+        expectations.localAudioOnly.forEach(field => {
+          ok(
+            stat.inner[field] !== undefined,
+            stat.type + " has field " + field + " when kind is video"
+          );
+        });
+        // totalSamplesReceived
+        ok(
+          stat.totalSamplesReceived > 1000,
+          `${stat.type}.totalSamplesReceived is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.totalSamplesReceived}`
+        );
+
+        // fecPacketsReceived
+        ok(
+          stat.fecPacketsReceived >= 0 && stat.fecPacketsReceived < 10 ** 5,
+          `${stat.type}.fecPacketsReceived is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.fecPacketsReceived}`
+        );
+
+        // fecPacketsDiscarded
+        ok(
+          stat.fecPacketsDiscarded >= 0 && stat.fecPacketsDiscarded < 100,
+          `${stat.type}.fecPacketsDiscarded is sane number for a short test. ` +
+            `value=${stat.fecPacketsDiscarded}`
+        );
+        // concealedSamples
+        ok(
+          stat.concealedSamples >= 0 &&
+            stat.concealedSamples <= stat.totalSamplesReceived,
+          `${stat.type}.concealedSamples is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.concealedSamples}`
+        );
+
+        // silentConcealedSamples
+        ok(
+          stat.silentConcealedSamples >= 0 &&
+            stat.silentConcealedSamples <= stat.concealedSamples,
+          `${stat.type}.silentConcealedSamples is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.silentConcealedSamples}`
+        );
+
+        // concealmentEvents
+        ok(
+          stat.concealmentEvents >= 0 &&
+            stat.concealmentEvents <= stat.packetsReceived,
+          `${stat.type}.concealmentEvents is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.concealmentEvents}`
+        );
+
+        // insertedSamplesForDeceleration
+        ok(
+          stat.insertedSamplesForDeceleration >= 0 &&
+            stat.insertedSamplesForDeceleration <= stat.totalSamplesReceived,
+          `${stat.type}.insertedSamplesForDeceleration is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.insertedSamplesForDeceleration}`
+        );
+
+        // removedSamplesForAcceleration
+        ok(
+          stat.removedSamplesForAcceleration >= 0 &&
+            stat.removedSamplesForAcceleration <= stat.totalSamplesReceived,
+          `${stat.type}.removedSamplesForAcceleration is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.removedSamplesForAcceleration}`
+        );
+
+        // audioLevel
+        ok(
+          stat.audioLevel >= 0 && stat.audioLevel <= 128,
+          `${stat.type}.bytesReceived is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.audioLevel}`
+        );
+
+        // totalAudioEnergy
+        ok(
+          stat.totalAudioEnergy >= 0 && stat.totalAudioEnergy <= 128,
+          `${stat.type}.totalAudioEnergy is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.totalAudioEnergy}`
+        );
+
+        // totalSamplesDuration
+        ok(
+          stat.totalSamplesDuration >= 0 && stat.totalSamplesDuration <= 300,
+          `${stat.type}.totalSamplesDuration is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.totalSamplesDuration}`
+        );
+      }
+
+      //
+      // Local video only stats
+      //
+      if (stat.inner.kind != "video") {
+        expectations.localVideoOnly.forEach(field => {
+          ok(
+            stat[field] === undefined,
+            `${stat.type} does not have field ${field}` +
+              ` when kind is not 'video'`
+          );
+        });
+      } else {
+        expectations.localVideoOnly.forEach(field => {
+          ok(
+            stat.inner[field] !== undefined,
+            stat.type + " has field " + field + " when kind is video"
+          );
+        });
+        // discardedPackets
+        ok(
+          stat.discardedPackets < 100,
+          `${stat.type}.discardedPackets is a sane number for a short test. ` +
+            `value=${stat.discardedPackets}`
+        );
+        // framesPerSecond
+        ok(
+          stat.framesPerSecond > 0 && stat.framesPerSecond < 70,
+          `${stat.type}.framesPerSecond is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.framesPerSecond}`
+        );
+
+        // framesDecoded
+        ok(
+          stat.framesDecoded > 0 && stat.framesDecoded < 1000000,
+          `${stat.type}.framesDecoded is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.framesDecoded}`
+        );
+
+        // framesDropped
+        ok(
+          stat.framesDropped >= 0 && stat.framesDropped < 100,
+          `${stat.type}.framesDropped is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.framesDropped}`
+        );
+
+        // frameWidth
+        ok(
+          stat.frameWidth > 0 && stat.frameWidth < 100000,
+          `${stat.type}.frameWidth is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.framesSent}`
+        );
+
+        // frameHeight
+        ok(
+          stat.frameHeight > 0 && stat.frameHeight < 100000,
+          `${stat.type}.frameHeight is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.frameHeight}`
+        );
+
+        // totalDecodeTime
+        ok(
+          stat.totalDecodeTime >= 0 && stat.totalDecodeTime < 300,
+          `${stat.type}.totalDecodeTime is sane for a short test. ` +
+            `value=${stat.totalDecodeTime}`
+        );
+
+        // totalProcessingDelay
+        ok(
+          stat.totalProcessingDelay < 100,
+          `${stat.type}.totalProcessingDelay is sane number for a short test ` +
+            `local only test. value=${stat.totalProcessingDelay}`
+        );
+
+        // totalInterFrameDelay
+        ok(
+          stat.totalInterFrameDelay >= 0 && stat.totalInterFrameDelay < 100,
+          `${stat.type}.totalInterFrameDelay is sane for a short test. ` +
+            `value=${stat.totalInterFrameDelay}`
+        );
+
+        // totalSquaredInterFrameDelay
+        ok(
+          stat.totalSquaredInterFrameDelay >= 0 &&
+            stat.totalSquaredInterFrameDelay < 100,
+          `${stat.type}.totalSquaredInterFrameDelay is sane for a short test. ` +
+            `value=${stat.totalSquaredInterFrameDelay}`
+        );
+
+        // framesReceived
+        ok(
+          stat.framesReceived >= 0 && stat.framesReceived < 100000,
+          `${stat.type}.framesReceived is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.framesReceived}`
+        );
+      }
+    } else if (stat.type == "remote-inbound-rtp") {
+      // roundTripTime
+      ok(
+        stat.roundTripTime >= 0,
+        `${stat.type}.roundTripTime is sane with` +
+          `value of: ${stat.roundTripTime} (${stat.kind})`
+      );
+      //
+      // Required fields
+      //
+
+      // packetsLost
+      ok(
+        stat.packetsLost < 100,
+        `${stat.type}.packetsLost is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.packetsLost}`
+      );
+
+      // jitter
+      ok(
+        stat.jitter >= 0,
+        `${stat.type}.jitter is sane number (${stat.kind}). ` +
+          `value=${stat.jitter}`
+      );
+
+      //
+      // Optional fields
+      //
+
+      // packetsReceived
+      if (stat.packetsReceived) {
+        ok(
+          stat.packetsReceived >= 0 && stat.packetsReceived < 10 ** 5,
+          `${stat.type}.packetsReceived is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.packetsReceived}`
+        );
+      }
+
+      // totalRoundTripTime
+      ok(
+        stat.totalRoundTripTime < 50000,
+        `${stat.type}.totalRoundTripTime is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.totalRoundTripTime}`
+      );
+
+      // fractionLost
+      ok(
+        stat.fractionLost < 0.2,
+        `${stat.type}.fractionLost is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.fractionLost}`
+      );
+
+      // roundTripTimeMeasurements
+      ok(
+        stat.roundTripTimeMeasurements >= 1 &&
+          stat.roundTripTimeMeasurements < 500,
+        `${stat.type}.roundTripTimeMeasurements is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.roundTripTimeMeasurements}`
+      );
+    } else if (stat.type == "outbound-rtp") {
+      //
+      // Required fields
+      //
+
+      // packetsSent
+      ok(
+        stat.packetsSent > 0 && stat.packetsSent < 10000,
+        `${stat.type}.packetsSent is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.packetsSent}`
+      );
+
+      // bytesSent
+      const audio1Min = 16000 * 60; // 128kbps
+      const video1Min = 250000 * 60; // 2Mbps
+      ok(
+        stat.bytesSent > 0 &&
+          stat.bytesSent < (stat.kind == "video" ? video1Min : audio1Min),
+        `${stat.type}.bytesSent is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.bytesSent}`
+      );
+
+      // headerBytesSent
+      ok(
+        stat.headerBytesSent > 0 &&
+          stat.headerBytesSent < (stat.kind == "video" ? video1Min : audio1Min),
+        `${stat.type}.headerBytesSent is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.headerBytesSent}`
+      );
+
+      // retransmittedPacketsSent
+      ok(
+        stat.retransmittedPacketsSent >= 0 &&
+          stat.retransmittedPacketsSent <
+            (stat.kind == "video" ? video1Min : audio1Min),
+        `${stat.type}.retransmittedPacketsSent is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.retransmittedPacketsSent}`
+      );
+
+      // retransmittedBytesSent
+      ok(
+        stat.retransmittedBytesSent >= 0 &&
+          stat.retransmittedBytesSent <
+            (stat.kind == "video" ? video1Min : audio1Min),
+        `${stat.type}.retransmittedBytesSent is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.retransmittedBytesSent}`
+      );
+
+      //
+      // Optional fields
+      //
+
+      // qpSum
+      // This is supported for all of our vpx codecs (on the encode side, see
+      // bug 1519590)
+      const mimeType = report.get(stat.codecId).mimeType;
+      if (mimeType.includes("VP")) {
+        ok(
+          stat.qpSum >= 0,
+          `${stat.type}.qpSum is a sane number (${stat.kind}) ` +
+            `for ${report.get(stat.codecId).mimeType}. value=${stat.qpSum}`
+        );
+      } else if (mimeType.includes("H264")) {
+        // OpenH264 encoder records QP so we check for either condition.
+        if (!stat.qpSum && !("qpSum" in stat)) {
+          ok(
+            !stat.qpSum && !("qpSum" in stat),
+            `${stat.type}.qpSum absent for ${report.get(stat.codecId).mimeType}`
+          );
+        } else {
+          ok(
+            stat.qpSum >= 0,
+            `${stat.type}.qpSum is a sane number (${stat.kind}) ` +
+              `for ${report.get(stat.codecId).mimeType}. value=${stat.qpSum}`
+          );
+        }
+      } else {
+        ok(
+          !stat.qpSum && !("qpSum" in stat),
+          `${stat.type}.qpSum absent for ${report.get(stat.codecId).mimeType}`
+        );
+      }
+
+      //
+      // Local video only stats
+      //
+      if (stat.inner.kind != "video") {
+        expectations.localVideoOnly.forEach(field => {
+          ok(
+            stat[field] === undefined,
+            `${stat.type} does not have field ` +
+              `${field} when kind is not 'video'`
+          );
+        });
+      } else {
+        expectations.localVideoOnly.forEach(field => {
+          ok(
+            stat.inner[field] !== undefined,
+            `${stat.type} has field ` +
+              `${field} when kind is video and isRemote is false`
+          );
+        });
+
+        // framesEncoded
+        ok(
+          stat.framesEncoded >= 0 && stat.framesEncoded < 100000,
+          `${stat.type}.framesEncoded is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.framesEncoded}`
+        );
+
+        // frameWidth
+        ok(
+          stat.frameWidth >= 0 && stat.frameWidth < 100000,
+          `${stat.type}.frameWidth is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.frameWidth}`
+        );
+
+        // frameHeight
+        ok(
+          stat.frameHeight >= 0 && stat.frameHeight < 100000,
+          `${stat.type}.frameHeight is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.frameHeight}`
+        );
+
+        // framesSent
+        ok(
+          stat.framesSent >= 0 && stat.framesSent < 100000,
+          `${stat.type}.framesSent is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.framesSent}`
+        );
+
+        // hugeFramesSent
+        ok(
+          stat.hugeFramesSent >= 0 && stat.hugeFramesSent < 100000,
+          `${stat.type}.hugeFramesSent is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.hugeFramesSent}`
+        );
+
+        // totalEncodeTime
+        ok(
+          stat.totalEncodeTime >= 0,
+          `${stat.type}.totalEncodeTime is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.totalEncodeTime}`
+        );
+
+        // totalEncodedBytesTarget
+        ok(
+          stat.totalEncodedBytesTarget > 1000,
+          `${stat.type}.totalEncodedBytesTarget is a sane number for a short ` +
+            `${stat.kind} test. value=${stat.totalEncodedBytesTarget}`
+        );
+      }
+    } else if (stat.type == "remote-outbound-rtp") {
+      //
+      // Required fields
+      //
+
+      // packetsSent
+      ok(
+        stat.packetsSent > 0 && stat.packetsSent < 10000,
+        `${stat.type}.packetsSent is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.packetsSent}`
+      );
+
+      // bytesSent
+      const audio1Min = 16000 * 60; // 128kbps
+      const video1Min = 250000 * 60; // 2Mbps
+      ok(
+        stat.bytesSent > 0 &&
+          stat.bytesSent < (stat.kind == "video" ? video1Min : audio1Min),
+        `${stat.type}.bytesSent is a sane number for a short ` +
+          `${stat.kind} test. value=${stat.bytesSent}`
+      );
+
+      ok(
+        stat.remoteTimestamp !== undefined,
+        `${stat.type}.remoteTimestamp ` + `is not undefined (${stat.kind})`
+      );
+      const ageSeconds = (stat.timestamp - stat.remoteTimestamp) / 1000;
+      // remoteTimestamp is exact (so it can be mapped to a packet), whereas
+      // timestamp has reduced precision. It is possible that
+      // remoteTimestamp occurs a millisecond into the future from
+      // timestamp. We also subtract half a millisecond when reducing
+      // precision on libwebrtc timestamps, to counteract the potential
+      // rounding up that libwebrtc may do since it tends to round its
+      // internal timestamps to whole milliseconds. In the worst case
+      // remoteTimestamp may therefore occur 2 milliseconds ahead of
+      // timestamp.
+      ok(
+        ageSeconds >= -0.002 && ageSeconds < 30,
+        `${stat.type}.remoteTimestamp is on the same timeline as ` +
+          `${stat.type}.timestamp, and no older than 30 seconds. ` +
+          `difference=${ageSeconds}s`
+      );
+    } else if (stat.type == "media-source") {
+      // trackIdentifier
+      is(typeof stat.trackIdentifier, "string");
+      isnot(stat.trackIdentifier, "");
+
+      // kind
+      is(typeof stat.kind, "string");
+      ok(stat.kind == "audio" || stat.kind == "video");
+    } else if (stat.type == "codec") {
+      //
+      // Required fields
+      //
+
+      // mimeType & payloadType
+      switch (stat.mimeType) {
+        case "audio/opus":
+          is(stat.payloadType, 109, "codec.payloadType for opus");
+          break;
+        case "video/VP8":
+          is(stat.payloadType, 120, "codec.payloadType for VP8");
+          break;
+        case "video/VP9":
+          is(stat.payloadType, 121, "codec.payloadType for VP9");
+          break;
+        case "video/H264":
+          ok(
+            stat.payloadType == 97 || stat.payloadType == 126,
+            `codec.payloadType for H264 was ${stat.payloadType}, exp. 97 or 126`
+          );
+          break;
+        default:
+          ok(
+            false,
+            `Unexpected codec.mimeType ${stat.mimeType} for payloadType ` +
+              `${stat.payloadType}`
+          );
+          break;
+      }
+
+      // transportId
+      // (no transport stats yet)
+      ok(stat.transportId, "codec.transportId is set");
+
+      // clockRate
+      if (stat.mimeType.startsWith("audio")) {
+        is(stat.clockRate, 48000, "codec.clockRate for audio/opus");
+      } else if (stat.mimeType.startsWith("video")) {
+        is(stat.clockRate, 90000, "codec.clockRate for video");
+      }
+
+      // sdpFmtpLine
+      // (not technically mandated by spec, but expected here)
+      ok(stat.sdpFmtpLine, "codec.sdpFmtpLine is set");
+      const opusParams = [
+        "maxplaybackrate",
+        "maxaveragebitrate",
+        "usedtx",
+        "stereo",
+        "useinbandfec",
+        "cbr",
+        "ptime",
+        "minptime",
+        "maxptime",
+      ];
+      const vpxParams = ["max-fs", "max-fr"];
+      const h264Params = [
+        "packetization-mode",
+        "level-asymmetry-allowed",
+        "profile-level-id",
+        "max-fs",
+        "max-cpb",
+        "max-dpb",
+        "max-br",
+        "max-mbps",
+      ];
+      for (const param of stat.sdpFmtpLine.split(";")) {
+        const [key, value] = param.split("=");
+        if (stat.payloadType == 109) {
+          ok(
+            opusParams.includes(key),
+            `codec.sdpFmtpLine param ${key}=${value} for opus`
+          );
+        } else if (stat.payloadType == 120 || stat.payloadType == 121) {
+          ok(
+            vpxParams.includes(key),
+            `codec.sdpFmtpLine param ${key}=${value} for VPx`
+          );
+        } else if (stat.payloadType == 97 || stat.payloadType == 126) {
+          ok(
+            h264Params.includes(key),
+            `codec.sdpFmtpLine param ${key}=${value} for H264`
+          );
+          if (key == "packetization-mode") {
+            if (stat.payloadType == 97) {
+              is(value, "0", "codec.sdpFmtpLine: H264 (97) packetization-mode");
+            } else if (stat.payloadType == 126) {
+              is(
+                value,
+                "1",
+                "codec.sdpFmtpLine: H264 (126) packetization-mode"
+              );
+            }
+          }
+          if (key == "profile-level-id") {
+            is(value, "42e01f", "codec.sdpFmtpLine: H264 profile-level-id");
+          }
+        }
+      }
+
+      //
+      // Optional fields
+      //
+
+      // codecType
+      ok(
+        !Object.keys(stat).includes("codecType") ||
+          stat.codecType == "encode" ||
+          stat.codecType == "decode",
+        "codec.codecType (${codec.codecType}) is an expected value or absent"
+      );
+      let numRecvStreams = 0;
+      let numSendStreams = 0;
+      const counts = {
+        "inbound-rtp": 0,
+        "outbound-rtp": 0,
+        "remote-inbound-rtp": 0,
+        "remote-outbound-rtp": 0,
+      };
+      const [kind] = stat.mimeType.split("/");
+      report.forEach(other => {
+        if (other.type == "inbound-rtp" && other.kind == kind) {
+          numRecvStreams += 1;
+        } else if (other.type == "outbound-rtp" && other.kind == kind) {
+          numSendStreams += 1;
+        }
+        if (other.codecId == stat.id) {
+          counts[other.type] += 1;
+        }
+      });
+      const expectedCounts = {
+        encode: {
+          "inbound-rtp": 0,
+          "outbound-rtp": numSendStreams,
+          "remote-inbound-rtp": numSendStreams,
+          "remote-outbound-rtp": 0,
+        },
+        decode: {
+          "inbound-rtp": numRecvStreams,
+          "outbound-rtp": 0,
+          "remote-inbound-rtp": 0,
+          "remote-outbound-rtp": numRecvStreams,
+        },
+        absent: {
+          "inbound-rtp": numRecvStreams,
+          "outbound-rtp": numSendStreams,
+          "remote-inbound-rtp": numSendStreams,
+          "remote-outbound-rtp": numRecvStreams,
+        },
+      };
+      // Note that the logic above assumes at most one sender and at most one
+      // receiver was used to generate this stats report. If more senders or
+      // receivers are present, they'd be referring to not only this codec stat,
+      // skewing `numSendStreams` and `numRecvStreams` above.
+      // This could be fixed when we support `senderId` and `receiverId` in
+      // RTCOutboundRtpStreamStats and RTCInboundRtpStreamStats respectively.
+      for (const [key, value] of Object.entries(counts)) {
+        is(
+          value,
+          expectedCounts[stat.codecType || "absent"][key],
+          `codec.codecType ${stat.codecType || "absent"} ref from ${key} stat`
+        );
+      }
+
+      // channels
+      if (stat.mimeType.startsWith("audio")) {
+        ok(stat.channels, "codec.channels should exist for audio");
+        if (stat.channels) {
+          if (stat.sdpFmtpLine.includes("stereo=1")) {
+            is(stat.channels, 2, "codec.channels for stereo audio");
+          } else {
+            is(stat.channels, 1, "codec.channels for mono audio");
+          }
+        }
+      } else {
+        ok(!stat.channels, "codec.channels should not exist for video");
+      }
+    } else if (stat.type == "candidate-pair") {
+      info("candidate-pair is: " + JSON.stringify(stat));
+      //
+      // Required fields
+      //
+
+      // transportId
+      ok(
+        stat.transportId,
+        `${stat.type}.transportId has a value. value=` +
+          `${stat.transportId} (${stat.kind})`
+      );
+
+      // localCandidateId
+      ok(
+        stat.localCandidateId,
+        `${stat.type}.localCandidateId has a value. value=` +
+          `${stat.localCandidateId} (${stat.kind})`
+      );
+
+      // remoteCandidateId
+      ok(
+        stat.remoteCandidateId,
+        `${stat.type}.remoteCandidateId has a value. value=` +
+          `${stat.remoteCandidateId} (${stat.kind})`
+      );
+
+      // priority
+      ok(
+        stat.priority,
+        `${stat.type}.priority has a value. value=` +
+          `${stat.priority} (${stat.kind})`
+      );
+
+      // readable
+      ok(
+        stat.readable,
+        `${stat.type}.readable is true. value=${stat.readable} ` +
+          `(${stat.kind})`
+      );
+
+      // writable
+      ok(
+        stat.writable,
+        `${stat.type}.writable is true. value=${stat.writable} ` +
+          `(${stat.kind})`
+      );
+
+      // state
+      if (
+        stat.state == "succeeded" &&
+        stat.selected !== undefined &&
+        stat.selected
+      ) {
+        info("candidate-pair state is succeeded and selected is true");
+        // nominated
+        ok(
+          stat.nominated,
+          `${stat.type}.nominated is true. value=${stat.nominated} ` +
+            `(${stat.kind})`
+        );
+
+        // bytesSent
+        ok(
+          stat.bytesSent > 1000,
+          `${stat.type}.bytesSent is a sane number (>1,000) for a short ` +
+            `${stat.kind} test. value=${stat.bytesSent}`
+        );
+
+        // bytesReceived
+        ok(
+          stat.bytesReceived > 500,
+          `${stat.type}.bytesReceived is a sane number (>500) for a short ` +
+            `${stat.kind} test. value=${stat.bytesReceived}`
+        );
+
+        // lastPacketSentTimestamp
+        ok(
+          stat.lastPacketSentTimestamp,
+          `${stat.type}.lastPacketSentTimestamp has a value. value=` +
+            `${stat.lastPacketSentTimestamp} (${stat.kind})`
+        );
+
+        // lastPacketReceivedTimestamp
+        ok(
+          stat.lastPacketReceivedTimestamp,
+          `${stat.type}.lastPacketReceivedTimestamp has a value. value=` +
+            `${stat.lastPacketReceivedTimestamp} (${stat.kind})`
+        );
+      } else {
+        info("candidate-pair is _not_ both state == succeeded and selected");
+        // nominated
+        ok(
+          stat.nominated !== undefined,
+          `${stat.type}.nominated exists. value=${stat.nominated} ` +
+            `(${stat.kind})`
+        );
+        ok(
+          stat.bytesSent !== undefined,
+          `${stat.type}.bytesSent exists. value=${stat.bytesSent} ` +
+            `(${stat.kind})`
+        );
+        ok(
+          stat.bytesReceived !== undefined,
+          `${stat.type}.bytesReceived exists. value=${stat.bytesReceived} ` +
+            `(${stat.kind})`
+        );
+        ok(
+          stat.lastPacketSentTimestamp !== undefined,
+          `${stat.type}.lastPacketSentTimestamp exists. value=` +
+            `${stat.lastPacketSentTimestamp} (${stat.kind})`
+        );
+        ok(
+          stat.lastPacketReceivedTimestamp !== undefined,
+          `${stat.type}.lastPacketReceivedTimestamp exists. value=` +
+            `${stat.lastPacketReceivedTimestamp} (${stat.kind})`
+        );
+      }
+
+      //
+      // Optional fields
+      //
+      // selected
+      ok(
+        stat.selected === undefined ||
+          (stat.state == "succeeded" && stat.selected) ||
+          !stat.selected,
+        `${stat.type}.selected is undefined, true when state is succeeded, ` +
+          `or false. value=${stat.selected} (${stat.kind})`
+      );
+    } else if (
+      stat.type == "local-candidate" ||
+      stat.type == "remote-candidate"
+    ) {
+      info(`candidate is ${JSON.stringify(stat)}`);
+
+      // address
+      ok(
+        stat.address,
+        `${stat.type} has address. value=${stat.address} ` + `(${stat.kind})`
+      );
+
+      // protocol
+      ok(
+        stat.protocol,
+        `${stat.type} has protocol. value=${stat.protocol} ` + `(${stat.kind})`
+      );
+
+      // port
+      ok(
+        stat.port >= 0,
+        `${stat.type} has port >= 0. value=${stat.port} ` + `(${stat.kind})`
+      );
+      ok(
+        stat.port <= 65535,
+        `${stat.type} has port <= 65535. value=${stat.port} ` + `(${stat.kind})`
+      );
+
+      // candidateType
+      ok(
+        stat.candidateType,
+        `${stat.type} has candidateType. value=${stat.candidateType} ` +
+          `(${stat.kind})`
+      );
+
+      // priority
+      ok(
+        stat.priority > 0 && stat.priority < 2 ** 32 - 1,
+        `${stat.type} has priority between 1 and 2^32 - 1 inc. ` +
+          `value=${stat.priority} (${stat.kind})`
+      );
+
+      // relayProtocol
+      if (stat.type == "local-candidate" && stat.candidateType == "relay") {
+        ok(
+          stat.relayProtocol,
+          `relay ${stat.type} has relayProtocol. value=${stat.relayProtocol} ` +
+            `(${stat.kind})`
+        );
+      } else {
+        is(
+          stat.relayProtocol,
+          undefined,
+          `relayProtocol is undefined for candidates that are not relay and ` +
+            `local. value=${stat.relayProtocol} (${stat.kind})`
+        );
+      }
+
+      // proxied
+      if (stat.proxied) {
+        ok(
+          stat.proxied == "proxied" || stat.proxied == "non-proxied",
+          `${stat.type} has proxied. value=${stat.proxied} (${stat.kind})`
+        );
+      }
+    }
+
+    //
+    // Ensure everything was tested
+    //
+    [...expectations.expected, ...expectations.optional].forEach(field => {
+      ok(
+        Object.keys(tested).includes(field),
+        `${stat.type}.${field} was tested.`
+      );
+    });
+  });
+}
+
+function dumpStats(stats) {
+  const dict = {};
+  for (const [k, v] of stats.entries()) {
+    dict[k] = v;
+  }
+  info(`Got stats: ${JSON.stringify(dict)}`);
+}
+
+async function waitForSyncedRtcp(pc) {
+  // Ensures that RTCP is present
+  let ensureSyncedRtcp = async () => {
+    let report = await pc.getStats();
+    for (const v of report.values()) {
+      if (v.type.endsWith("bound-rtp") && !(v.remoteId || v.localId)) {
+        info(`${v.id} is missing remoteId or localId: ${JSON.stringify(v)}`);
+        return null;
+      }
+      if (v.type == "remote-inbound-rtp" && v.roundTripTime === undefined) {
+        info(`${v.id} is missing roundTripTime: ${JSON.stringify(v)}`);
+        return null;
+      }
+    }
+    return report;
+  };
+  // Returns true if there is proof in aStats of rtcp flow for all remote stats
+  // objects, compared to baseStats.
+  const hasAllRtcpUpdated = (baseStats, stats) => {
+    let hasRtcpStats = false;
+    for (const v of stats.values()) {
+      if (v.type == "remote-outbound-rtp") {
+        hasRtcpStats = true;
+        if (!v.remoteTimestamp) {
+          // `remoteTimestamp` is 0 or not present.
+          return false;
+        }
+        if (v.remoteTimestamp <= baseStats.get(v.id)?.remoteTimestamp) {
+          // `remoteTimestamp` has not advanced further than the base stats,
+          // i.e., no new sender report has been received.
+          return false;
+        }
+      } else if (v.type == "remote-inbound-rtp") {
+        hasRtcpStats = true;
+        // The ideal thing here would be to check `reportsReceived`, but it's
+        // not yet implemented.
+        if (!v.packetsReceived) {
+          // `packetsReceived` is 0 or not present.
+          return false;
+        }
+        if (v.packetsReceived <= baseStats.get(v.id)?.packetsReceived) {
+          // `packetsReceived` has not advanced further than the base stats,
+          // i.e., no new receiver report has been received.
+          return false;
+        }
+      }
+    }
+    return hasRtcpStats;
+  };
+  let attempts = 0;
+  const baseStats = await pc.getStats();
+  // Time-units are MS
+  const waitPeriod = 100;
+  const maxTime = 20000;
+  for (let totalTime = maxTime; totalTime > 0; totalTime -= waitPeriod) {
+    try {
+      let syncedStats = await ensureSyncedRtcp();
+      if (syncedStats && hasAllRtcpUpdated(baseStats, syncedStats)) {
+        dumpStats(syncedStats);
+        return syncedStats;
+      }
+    } catch (e) {
+      info(e);
+      info(e.stack);
+      throw e;
+    }
+    attempts += 1;
+    info(`waitForSyncedRtcp: no sync on attempt ${attempts}, retrying.`);
+    await wait(waitPeriod);
+  }
+  throw Error(
+    "Waiting for synced RTCP timed out after at least " + maxTime + "ms"
+  );
+}
+
+function checkSenderStats(senderStats, streamCount) {
+  const outboundRtpReports = [];
+  const remoteInboundRtpReports = [];
+  for (const v of senderStats.values()) {
+    if (v.type == "outbound-rtp") {
+      outboundRtpReports.push(v);
+    } else if (v.type == "remote-inbound-rtp") {
+      remoteInboundRtpReports.push(v);
+    }
+  }
+  is(
+    outboundRtpReports.length,
+    streamCount,
+    `Sender with ${streamCount} simulcast streams has ${streamCount} outbound-rtp reports`
+  );
+  is(
+    remoteInboundRtpReports.length,
+    streamCount,
+    `Sender with ${streamCount} simulcast streams has ${streamCount} remote-inbound-rtp reports`
+  );
+  for (const outboundRtpReport of outboundRtpReports) {
+    is(
+      outboundRtpReports.filter(r => r.ssrc == outboundRtpReport.ssrc).length,
+      1,
+      "Simulcast send track SSRCs are distinct"
+    );
+    const remoteReports = remoteInboundRtpReports.filter(
+      r => r.id == outboundRtpReport.remoteId
+    );
+    is(
+      remoteReports.length,
+      1,
+      "Simulcast send tracks have exactly one remote counterpart"
+    );
+    const remoteInboundRtpReport = remoteReports[0];
+    is(
+      outboundRtpReport.ssrc,
+      remoteInboundRtpReport.ssrc,
+      "SSRC matches for outbound-rtp and remote-inbound-rtp"
+    );
+  }
+}
+
+function PC_LOCAL_TEST_LOCAL_STATS(test) {
+  return waitForSyncedRtcp(test.pcLocal._pc).then(stats => {
+    checkExpectedFields(stats);
+    pedanticChecks(stats);
+    return Promise.all([
+      test.pcLocal._pc.getSenders().map(async s => {
+        checkSenderStats(
+          await s.getStats(),
+          Math.max(1, s.getParameters()?.encodings?.length ?? 0)
+        );
+      }),
+    ]);
+  });
+}
+
+function PC_REMOTE_TEST_REMOTE_STATS(test) {
+  return waitForSyncedRtcp(test.pcRemote._pc).then(stats => {
+    checkExpectedFields(stats);
+    pedanticChecks(stats);
+    return Promise.all([
+      test.pcRemote._pc.getSenders().map(async s => {
+        checkSenderStats(
+          await s.getStats(),
+          s.track ? Math.max(1, s.getParameters()?.encodings?.length ?? 0) : 0
+        );
+      }),
+    ]);
+  });
+}
diff --git a/dom/media/webrtc/tests/mochitests/templates.js b/dom/media/webrtc/tests/mochitests/templates.js
new file mode 100644
index 0000000000..6b7750fd2c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/templates.js
@@ -0,0 +1,615 @@
+/**
+ * Default list of commands to execute for a PeerConnection test.
+ */
+
+const STABLE = "stable";
+const HAVE_LOCAL_OFFER = "have-local-offer";
+const HAVE_REMOTE_OFFER = "have-remote-offer";
+const CLOSED = "closed";
+
+const ICE_NEW = "new";
+const GATH_NEW = "new";
+const GATH_GATH = "gathering";
+const GATH_COMPLETE = "complete";
+
+function deltaSeconds(date1, date2) {
+  return (date2.getTime() - date1.getTime()) / 1000;
+}
+
+function dumpSdp(test) {
+  if (typeof test._local_offer !== "undefined") {
+    dump("ERROR: SDP offer: " + test._local_offer.sdp.replace(/[\r]/g, ""));
+  }
+  if (typeof test._remote_answer !== "undefined") {
+    dump("ERROR: SDP answer: " + test._remote_answer.sdp.replace(/[\r]/g, ""));
+  }
+
+  if (
+    test.pcLocal &&
+    typeof test.pcLocal._local_ice_candidates !== "undefined"
+  ) {
+    dump(
+      "pcLocal._local_ice_candidates: " +
+        JSON.stringify(test.pcLocal._local_ice_candidates) +
+        "\n"
+    );
+    dump(
+      "pcLocal._remote_ice_candidates: " +
+        JSON.stringify(test.pcLocal._remote_ice_candidates) +
+        "\n"
+    );
+    dump(
+      "pcLocal._ice_candidates_to_add: " +
+        JSON.stringify(test.pcLocal._ice_candidates_to_add) +
+        "\n"
+    );
+  }
+  if (
+    test.pcRemote &&
+    typeof test.pcRemote._local_ice_candidates !== "undefined"
+  ) {
+    dump(
+      "pcRemote._local_ice_candidates: " +
+        JSON.stringify(test.pcRemote._local_ice_candidates) +
+        "\n"
+    );
+    dump(
+      "pcRemote._remote_ice_candidates: " +
+        JSON.stringify(test.pcRemote._remote_ice_candidates) +
+        "\n"
+    );
+    dump(
+      "pcRemote._ice_candidates_to_add: " +
+        JSON.stringify(test.pcRemote._ice_candidates_to_add) +
+        "\n"
+    );
+  }
+
+  if (test.pcLocal && typeof test.pcLocal.iceConnectionLog !== "undefined") {
+    dump(
+      "pcLocal ICE connection state log: " +
+        test.pcLocal.iceConnectionLog +
+        "\n"
+    );
+  }
+  if (test.pcRemote && typeof test.pcRemote.iceConnectionLog !== "undefined") {
+    dump(
+      "pcRemote ICE connection state log: " +
+        test.pcRemote.iceConnectionLog +
+        "\n"
+    );
+  }
+
+  if (
+    test.pcLocal &&
+    test.pcRemote &&
+    typeof test.pcLocal.setRemoteDescDate !== "undefined" &&
+    typeof test.pcRemote.setLocalDescDate !== "undefined"
+  ) {
+    var delta = deltaSeconds(
+      test.pcLocal.setRemoteDescDate,
+      test.pcRemote.setLocalDescDate
+    );
+    dump(
+      "Delay between pcLocal.setRemote <-> pcRemote.setLocal: " + delta + "\n"
+    );
+  }
+  if (
+    test.pcLocal &&
+    test.pcRemote &&
+    typeof test.pcLocal.setRemoteDescDate !== "undefined" &&
+    typeof test.pcLocal.setRemoteDescStableEventDate !== "undefined"
+  ) {
+    var delta = deltaSeconds(
+      test.pcLocal.setRemoteDescDate,
+      test.pcLocal.setRemoteDescStableEventDate
+    );
+    dump(
+      "Delay between pcLocal.setRemote <-> pcLocal.signalingStateStable: " +
+        delta +
+        "\n"
+    );
+  }
+  if (
+    test.pcLocal &&
+    test.pcRemote &&
+    typeof test.pcRemote.setLocalDescDate !== "undefined" &&
+    typeof test.pcRemote.setLocalDescStableEventDate !== "undefined"
+  ) {
+    var delta = deltaSeconds(
+      test.pcRemote.setLocalDescDate,
+      test.pcRemote.setLocalDescStableEventDate
+    );
+    dump(
+      "Delay between pcRemote.setLocal <-> pcRemote.signalingStateStable: " +
+        delta +
+        "\n"
+    );
+  }
+}
+
+// We need to verify that at least one candidate has been (or will be) gathered.
+function waitForAnIceCandidate(pc) {
+  return new Promise(resolve => {
+    if (!pc.localRequiresTrickleIce || pc._local_ice_candidates.length) {
+      resolve();
+    } else {
+      // In some circumstances, especially when both PCs are on the same
+      // browser, even though we are connected, the connection can be
+      // established without receiving a single candidate from one or other
+      // peer.  So we wait for at least one...
+      pc._pc.addEventListener("icecandidate", resolve);
+    }
+  }).then(() => {
+    ok(
+      pc._local_ice_candidates.length,
+      pc + " received local trickle ICE candidates"
+    );
+    isnot(
+      pc._pc.iceGatheringState,
+      GATH_NEW,
+      pc + " ICE gathering state is not 'new'"
+    );
+  });
+}
+
+async function checkTrackStats(pc, track, outbound) {
+  const audio = track.kind == "audio";
+  const msg =
+    `${pc} stats ${outbound ? "outbound " : "inbound "}` +
+    `${audio ? "audio" : "video"} rtp track id ${track.id}`;
+  const stats = await pc.getStats(track);
+  ok(
+    pc.hasStat(stats, {
+      type: outbound ? "outbound-rtp" : "inbound-rtp",
+      kind: audio ? "audio" : "video",
+    }),
+    `${msg} - found expected stats`
+  );
+  ok(
+    !pc.hasStat(stats, {
+      type: outbound ? "inbound-rtp" : "outbound-rtp",
+    }),
+    `${msg} - did not find extra stats with wrong direction`
+  );
+  ok(
+    !pc.hasStat(stats, {
+      kind: audio ? "video" : "audio",
+    }),
+    `${msg} - did not find extra stats with wrong media type`
+  );
+}
+
+function checkAllTrackStats(pc) {
+  return Promise.all([
+    ...pc
+      .getExpectedActiveReceivers()
+      .map(({ track }) => checkTrackStats(pc, track, false)),
+    ...pc
+      .getExpectedSenders()
+      .map(({ track }) => checkTrackStats(pc, track, true)),
+  ]);
+}
+
+// Commands run once at the beginning of each test, even when performing a
+// renegotiation test.
+var commandsPeerConnectionInitial = [
+  function PC_LOCAL_SETUP_ICE_LOGGER(test) {
+    test.pcLocal.logIceConnectionState();
+  },
+
+  function PC_REMOTE_SETUP_ICE_LOGGER(test) {
+    test.pcRemote.logIceConnectionState();
+  },
+
+  function PC_LOCAL_SETUP_SIGNALING_LOGGER(test) {
+    test.pcLocal.logSignalingState();
+  },
+
+  function PC_REMOTE_SETUP_SIGNALING_LOGGER(test) {
+    test.pcRemote.logSignalingState();
+  },
+
+  function PC_LOCAL_SETUP_TRACK_HANDLER(test) {
+    test.pcLocal.setupTrackEventHandler();
+  },
+
+  function PC_REMOTE_SETUP_TRACK_HANDLER(test) {
+    test.pcRemote.setupTrackEventHandler();
+  },
+
+  function PC_LOCAL_CHECK_INITIAL_SIGNALINGSTATE(test) {
+    is(
+      test.pcLocal.signalingState,
+      STABLE,
+      "Initial local signalingState is 'stable'"
+    );
+  },
+
+  function PC_REMOTE_CHECK_INITIAL_SIGNALINGSTATE(test) {
+    is(
+      test.pcRemote.signalingState,
+      STABLE,
+      "Initial remote signalingState is 'stable'"
+    );
+  },
+
+  function PC_LOCAL_CHECK_INITIAL_ICE_STATE(test) {
+    is(
+      test.pcLocal.iceConnectionState,
+      ICE_NEW,
+      "Initial local ICE connection state is 'new'"
+    );
+  },
+
+  function PC_REMOTE_CHECK_INITIAL_ICE_STATE(test) {
+    is(
+      test.pcRemote.iceConnectionState,
+      ICE_NEW,
+      "Initial remote ICE connection state is 'new'"
+    );
+  },
+
+  function PC_LOCAL_CHECK_INITIAL_CAN_TRICKLE_SYNC(test) {
+    is(
+      test.pcLocal._pc.canTrickleIceCandidates,
+      null,
+      "Local trickle status should start out unknown"
+    );
+  },
+
+  function PC_REMOTE_CHECK_INITIAL_CAN_TRICKLE_SYNC(test) {
+    is(
+      test.pcRemote._pc.canTrickleIceCandidates,
+      null,
+      "Remote trickle status should start out unknown"
+    );
+  },
+];
+
+var commandsGetUserMedia = [
+  function PC_LOCAL_GUM(test) {
+    return test.pcLocal.getAllUserMediaAndAddStreams(test.pcLocal.constraints);
+  },
+
+  function PC_REMOTE_GUM(test) {
+    return test.pcRemote.getAllUserMediaAndAddStreams(
+      test.pcRemote.constraints
+    );
+  },
+];
+
+var commandsPeerConnectionOfferAnswer = [
+  function PC_LOCAL_SETUP_ICE_HANDLER(test) {
+    test.pcLocal.setupIceCandidateHandler(test);
+  },
+
+  function PC_REMOTE_SETUP_ICE_HANDLER(test) {
+    test.pcRemote.setupIceCandidateHandler(test);
+  },
+
+  function PC_LOCAL_CREATE_OFFER(test) {
+    return test.createOffer(test.pcLocal).then(offer => {
+      is(
+        test.pcLocal.signalingState,
+        STABLE,
+        "Local create offer does not change signaling state"
+      );
+    });
+  },
+
+  function PC_LOCAL_SET_LOCAL_DESCRIPTION(test) {
+    return test
+      .setLocalDescription(test.pcLocal, test.originalOffer, HAVE_LOCAL_OFFER)
+      .then(() => {
+        is(
+          test.pcLocal.signalingState,
+          HAVE_LOCAL_OFFER,
+          "signalingState after local setLocalDescription is 'have-local-offer'"
+        );
+      });
+  },
+
+  function PC_REMOTE_GET_OFFER(test) {
+    test._local_offer = test.originalOffer;
+    test._offer_constraints = test.pcLocal.constraints;
+    test._offer_options = test.pcLocal.offerOptions;
+    return Promise.resolve();
+  },
+
+  function PC_REMOTE_SET_REMOTE_DESCRIPTION(test) {
+    return test
+      .setRemoteDescription(test.pcRemote, test._local_offer, HAVE_REMOTE_OFFER)
+      .then(() => {
+        is(
+          test.pcRemote.signalingState,
+          HAVE_REMOTE_OFFER,
+          "signalingState after remote setRemoteDescription is 'have-remote-offer'"
+        );
+      });
+  },
+
+  function PC_REMOTE_CHECK_CAN_TRICKLE_SYNC(test) {
+    is(
+      test.pcRemote._pc.canTrickleIceCandidates,
+      true,
+      "Remote thinks that local can trickle"
+    );
+  },
+
+  function PC_LOCAL_SANE_LOCAL_SDP(test) {
+    test.pcLocal.localRequiresTrickleIce = sdputils.verifySdp(
+      test._local_offer,
+      "offer",
+      test._offer_constraints,
+      test._offer_options,
+      test.testOptions
+    );
+  },
+
+  function PC_REMOTE_SANE_REMOTE_SDP(test) {
+    test.pcRemote.remoteRequiresTrickleIce = sdputils.verifySdp(
+      test._local_offer,
+      "offer",
+      test._offer_constraints,
+      test._offer_options,
+      test.testOptions
+    );
+  },
+
+  function PC_REMOTE_CREATE_ANSWER(test) {
+    return test.createAnswer(test.pcRemote).then(answer => {
+      is(
+        test.pcRemote.signalingState,
+        HAVE_REMOTE_OFFER,
+        "Remote createAnswer does not change signaling state"
+      );
+    });
+  },
+
+  function PC_REMOTE_SET_LOCAL_DESCRIPTION(test) {
+    return test
+      .setLocalDescription(test.pcRemote, test.originalAnswer, STABLE)
+      .then(() => {
+        is(
+          test.pcRemote.signalingState,
+          STABLE,
+          "signalingState after remote setLocalDescription is 'stable'"
+        );
+      });
+  },
+
+  function PC_LOCAL_GET_ANSWER(test) {
+    test._remote_answer = test.originalAnswer;
+    test._answer_constraints = test.pcRemote.constraints;
+    return Promise.resolve();
+  },
+
+  function PC_LOCAL_SET_REMOTE_DESCRIPTION(test) {
+    return test
+      .setRemoteDescription(test.pcLocal, test._remote_answer, STABLE)
+      .then(() => {
+        is(
+          test.pcLocal.signalingState,
+          STABLE,
+          "signalingState after local setRemoteDescription is 'stable'"
+        );
+      });
+  },
+
+  function PC_REMOTE_SANE_LOCAL_SDP(test) {
+    test.pcRemote.localRequiresTrickleIce = sdputils.verifySdp(
+      test._remote_answer,
+      "answer",
+      test._offer_constraints,
+      test._offer_options,
+      test.testOptions
+    );
+  },
+  function PC_LOCAL_SANE_REMOTE_SDP(test) {
+    test.pcLocal.remoteRequiresTrickleIce = sdputils.verifySdp(
+      test._remote_answer,
+      "answer",
+      test._offer_constraints,
+      test._offer_options,
+      test.testOptions
+    );
+  },
+
+  function PC_LOCAL_CHECK_CAN_TRICKLE_SYNC(test) {
+    is(
+      test.pcLocal._pc.canTrickleIceCandidates,
+      true,
+      "Local thinks that remote can trickle"
+    );
+  },
+
+  function PC_LOCAL_WAIT_FOR_ICE_CONNECTED(test) {
+    return test.pcLocal.waitForIceConnected().then(() => {
+      info(
+        test.pcLocal +
+          ": ICE connection state log: " +
+          test.pcLocal.iceConnectionLog
+      );
+    });
+  },
+
+  function PC_REMOTE_WAIT_FOR_ICE_CONNECTED(test) {
+    return test.pcRemote.waitForIceConnected().then(() => {
+      info(
+        test.pcRemote +
+          ": ICE connection state log: " +
+          test.pcRemote.iceConnectionLog
+      );
+    });
+  },
+
+  function PC_LOCAL_VERIFY_ICE_GATHERING(test) {
+    return waitForAnIceCandidate(test.pcLocal);
+  },
+
+  function PC_REMOTE_VERIFY_ICE_GATHERING(test) {
+    return waitForAnIceCandidate(test.pcRemote);
+  },
+
+  function PC_LOCAL_WAIT_FOR_MEDIA_FLOW(test) {
+    return test.pcLocal.waitForMediaFlow();
+  },
+
+  function PC_REMOTE_WAIT_FOR_MEDIA_FLOW(test) {
+    return test.pcRemote.waitForMediaFlow();
+  },
+
+  function PC_LOCAL_CHECK_STATS(test) {
+    return test.pcLocal.getStats().then(stats => {
+      test.pcLocal.checkStats(stats);
+    });
+  },
+
+  function PC_REMOTE_CHECK_STATS(test) {
+    return test.pcRemote.getStats().then(stats => {
+      test.pcRemote.checkStats(stats);
+    });
+  },
+
+  function PC_LOCAL_CHECK_ICE_CONNECTION_TYPE(test) {
+    return test.pcLocal.getStats().then(stats => {
+      test.pcLocal.checkStatsIceConnectionType(
+        stats,
+        test.testOptions.expectedLocalCandidateType
+      );
+    });
+  },
+
+  function PC_REMOTE_CHECK_ICE_CONNECTION_TYPE(test) {
+    return test.pcRemote.getStats().then(stats => {
+      test.pcRemote.checkStatsIceConnectionType(
+        stats,
+        test.testOptions.expectedRemoteCandidateType
+      );
+    });
+  },
+
+  function PC_LOCAL_CHECK_ICE_CONNECTIONS(test) {
+    return test.pcLocal.getStats().then(stats => {
+      test.pcLocal.checkStatsIceConnections(stats, test.testOptions);
+    });
+  },
+
+  function PC_REMOTE_CHECK_ICE_CONNECTIONS(test) {
+    return test.pcRemote.getStats().then(stats => {
+      test.pcRemote.checkStatsIceConnections(stats, test.testOptions);
+    });
+  },
+
+  function PC_LOCAL_CHECK_MSID(test) {
+    return test.pcLocal.checkLocalMsids();
+  },
+  function PC_REMOTE_CHECK_MSID(test) {
+    return test.pcRemote.checkLocalMsids();
+  },
+
+  function PC_LOCAL_CHECK_TRACK_STATS(test) {
+    return checkAllTrackStats(test.pcLocal);
+  },
+  function PC_REMOTE_CHECK_TRACK_STATS(test) {
+    return checkAllTrackStats(test.pcRemote);
+  },
+  function PC_LOCAL_VERIFY_SDP_AFTER_END_OF_TRICKLE(test) {
+    if (test.pcLocal.endOfTrickleSdp) {
+      /* In case the endOfTrickleSdp promise is resolved already it will win the
+       * race because it gets evaluated first. But if endOfTrickleSdp is still
+       * pending the rejection will win the race. */
+      return Promise.race([
+        test.pcLocal.endOfTrickleSdp,
+        Promise.reject("No SDP"),
+      ]).then(
+        sdp =>
+          sdputils.checkSdpAfterEndOfTrickle(
+            sdp,
+            test.testOptions,
+            test.pcLocal.label
+          ),
+        () =>
+          info(
+            "pcLocal: Gathering is not complete yet, skipping post-gathering SDP check"
+          )
+      );
+    }
+  },
+  function PC_REMOTE_VERIFY_SDP_AFTER_END_OF_TRICKLE(test) {
+    if (test.pcRemote.endOfTrickleSdp) {
+      /* In case the endOfTrickleSdp promise is resolved already it will win the
+       * race because it gets evaluated first. But if endOfTrickleSdp is still
+       * pending the rejection will win the race. */
+      return Promise.race([
+        test.pcRemote.endOfTrickleSdp,
+        Promise.reject("No SDP"),
+      ]).then(
+        sdp =>
+          sdputils.checkSdpAfterEndOfTrickle(
+            sdp,
+            test.testOptions,
+            test.pcRemote.label
+          ),
+        () =>
+          info(
+            "pcRemote: Gathering is not complete yet, skipping post-gathering SDP check"
+          )
+      );
+    }
+  },
+];
+
+function PC_LOCAL_REMOVE_ALL_BUT_H264_FROM_OFFER(test) {
+  isnot(
+    test.originalOffer.sdp.search("H264/90000"),
+    -1,
+    "H.264 should be present in the SDP offer"
+  );
+  test.originalOffer.sdp = sdputils.removeCodec(
+    sdputils.removeCodec(
+      sdputils.removeCodec(test.originalOffer.sdp, 120),
+      121,
+      97
+    )
+  );
+  info("Updated H264 only offer: " + JSON.stringify(test.originalOffer));
+}
+
+function PC_LOCAL_REMOVE_BUNDLE_FROM_OFFER(test) {
+  test.originalOffer.sdp = sdputils.removeBundle(test.originalOffer.sdp);
+  info("Updated no bundle offer: " + JSON.stringify(test.originalOffer));
+}
+
+function PC_LOCAL_REMOVE_RTCPMUX_FROM_OFFER(test) {
+  test.originalOffer.sdp = sdputils.removeRtcpMux(test.originalOffer.sdp);
+  info("Updated no RTCP-Mux offer: " + JSON.stringify(test.originalOffer));
+}
+
+function PC_LOCAL_REMOVE_SSRC_FROM_OFFER(test) {
+  test.originalOffer.sdp = sdputils.removeSSRCs(test.originalOffer.sdp);
+  info("Updated no SSRCs offer: " + JSON.stringify(test.originalOffer));
+}
+
+function PC_REMOTE_REMOVE_SSRC_FROM_ANSWER(test) {
+  test.originalAnswer.sdp = sdputils.removeSSRCs(test.originalAnswer.sdp);
+  info("Updated no SSRCs answer: " + JSON.stringify(test.originalAnswer));
+}
+
+var addRenegotiation = (chain, commands, checks) => {
+  chain.append(commands);
+  chain.append(commandsPeerConnectionOfferAnswer);
+  if (checks) {
+    chain.append(checks);
+  }
+};
+
+var addRenegotiationAnswerer = (chain, commands, checks) => {
+  chain.append(function SWAP_PC_LOCAL_PC_REMOTE(test) {
+    var temp = test.pcLocal;
+    test.pcLocal = test.pcRemote;
+    test.pcRemote = temp;
+  });
+  addRenegotiation(chain, commands, checks);
+};
diff --git a/dom/media/webrtc/tests/mochitests/test_1488832.html b/dom/media/webrtc/tests/mochitests/test_1488832.html
new file mode 100644
index 0000000000..8798994b24
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_1488832.html
@@ -0,0 +1,37 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+<script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<iframe id="testframe"></iframe>
+<script>
+"use strict";
+
+createHTML({
+  title: "gUM shutdown race",
+  bug: "1488832"
+});
+
+runTest(async () => {
+  testframe.srcdoc = `
+  <html>
+  <head>
+  <script>
+  function start() {
+    for (let i = 0; i < 16; i++) {
+      window.navigator.mediaDevices.getUserMedia({video: true})
+      setTimeout('location.reload()', 100)
+    }
+  }
+  document.addEventListener('DOMContentLoaded', start)
+  </` + `script>
+  </head>
+  </html>`;
+
+  await wait(10000);
+  testframe.srcdoc = "";
+});
+</script>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_1717318.html b/dom/media/webrtc/tests/mochitests/test_1717318.html
new file mode 100644
index 0000000000..425bd29e7e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_1717318.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>PC construct with no global object (bug 1717318)</title>
+  <script type="text/javascript" src="/MochiKit/MochiKit.js"></script>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<script class="testbody" type="text/javascript">
+"use strict";
+
+// nsIArray is not special here, it could be pretty much any interface.
+// We do this outside the try block just in case someday the interface is
+// removed.
+const dummyInterface = SpecialPowers.Components.interfaces.nsIArray;
+ok(dummyInterface, "nsIArray should exist");
+try {
+  // Just don't crash.
+  SpecialPowers.Components.classes["@mozilla.org/peerconnection;1"]
+      .createInstance(dummyInterface);
+} catch (e) {}
+
+</script>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_a_noOp.html b/dom/media/webrtc/tests/mochitests/test_a_noOp.html
new file mode 100644
index 0000000000..971f5d7666
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_a_noOp.html
@@ -0,0 +1,32 @@
+<!DOCTYPE HTML>
+<html>
+<!--
+https://bugzilla.mozilla.org/show_bug.cgi?id=1264772
+-->
+<head>
+  <title>Test for Bug 1264772</title>
+  <script type="text/javascript" src="/MochiKit/MochiKit.js"></script>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<a target="_blank" href="https://bugzilla.mozilla.org/show_bug.cgi?id=1264772">Mozilla Bug 1264772</a>
+<p id="display"></p>
+<div id="content" style="display: none">
+</div>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+/** Test for Bug 1264772 **/
+// The WebRTC tests seem to have more problems with intermittents (at
+// least on Android) if they run first in a test run. This is a dummy test
+// to ensure that the browser is ready prior to running any actual WebRTC
+// tests.
+//
+// Note: mochitests are run in alphabetical order, so it is not sufficient
+//       for this test to appear first in the manifest.
+ok(true, "test passed");
+
+</script>
+</pre>
+</body>
diff --git a/dom/media/webrtc/tests/mochitests/test_dataChannel_basicAudio.html b/dom/media/webrtc/tests/mochitests/test_dataChannel_basicAudio.html
new file mode 100644
index 0000000000..06ca9562ad
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_dataChannel_basicAudio.html
@@ -0,0 +1,25 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "796895",
+    title: "Basic data channel audio connection"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    addInitialDataChannel(test.chain);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_dataChannel_basicAudioVideo.html b/dom/media/webrtc/tests/mochitests/test_dataChannel_basicAudioVideo.html
new file mode 100644
index 0000000000..ea534ca2e7
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_dataChannel_basicAudioVideo.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "796891",
+    title: "Basic data channel audio/video connection"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    addInitialDataChannel(test.chain);
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             [{audio: true}, {video: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_dataChannel_basicAudioVideoCombined.html b/dom/media/webrtc/tests/mochitests/test_dataChannel_basicAudioVideoCombined.html
new file mode 100644
index 0000000000..d5409986ec
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_dataChannel_basicAudioVideoCombined.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "796891",
+    title: "Basic data channel audio/video connection"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    addInitialDataChannel(test.chain);
+    test.setMediaConstraints([{audio: true, video: true}],
+                             [{audio: true, video: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_dataChannel_basicAudioVideoNoBundle.html b/dom/media/webrtc/tests/mochitests/test_dataChannel_basicAudioVideoNoBundle.html
new file mode 100644
index 0000000000..7dc22d86ad
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_dataChannel_basicAudioVideoNoBundle.html
@@ -0,0 +1,27 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1016476",
+    title: "Basic data channel audio/video connection without bundle"
+  });
+
+var test;
+runNetworkTest(function (options) {
+  options = options || { };
+  options.bundle = false;
+  test = new PeerConnectionTest(options);
+  addInitialDataChannel(test.chain);
+  test.setMediaConstraints([{audio: true}, {video: true}],
+                           [{audio: true}, {video: true}]);
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_dataChannel_basicDataOnly.html b/dom/media/webrtc/tests/mochitests/test_dataChannel_basicDataOnly.html
new file mode 100644
index 0000000000..98e72f7a21
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_dataChannel_basicDataOnly.html
@@ -0,0 +1,24 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "796894",
+    title: "Basic datachannel only connection"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    addInitialDataChannel(test.chain);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_dataChannel_basicVideo.html b/dom/media/webrtc/tests/mochitests/test_dataChannel_basicVideo.html
new file mode 100644
index 0000000000..90f2d7caff
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_dataChannel_basicVideo.html
@@ -0,0 +1,25 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "796889",
+    title: "Basic data channel video connection"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    addInitialDataChannel(test.chain);
+    test.setMediaConstraints([{video: true}], [{video: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_dataChannel_bug1013809.html b/dom/media/webrtc/tests/mochitests/test_dataChannel_bug1013809.html
new file mode 100644
index 0000000000..e36caebab4
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_dataChannel_bug1013809.html
@@ -0,0 +1,27 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "796895",
+    title: "Basic data channel audio connection"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    addInitialDataChannel(test.chain);
+    var sld = test.chain.remove("PC_REMOTE_SET_LOCAL_DESCRIPTION");
+    test.chain.insertAfter("PC_LOCAL_SET_REMOTE_DESCRIPTION", sld);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_dataChannel_dataOnlyBufferedAmountLow.html b/dom/media/webrtc/tests/mochitests/test_dataChannel_dataOnlyBufferedAmountLow.html
new file mode 100644
index 0000000000..26767e0865
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_dataChannel_dataOnlyBufferedAmountLow.html
@@ -0,0 +1,25 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1051685",
+    title: "Verify bufferedAmountLowThreshold works"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    addInitialDataChannel(test.chain);
+    test.chain.insertAfter('PC_REMOTE_CHECK_ICE_CONNECTIONS', commandsCheckLargeXfer);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_dataChannel_dtlsVersions.html b/dom/media/webrtc/tests/mochitests/test_dataChannel_dtlsVersions.html
new file mode 100644
index 0000000000..6f0cbc5d3d
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_dataChannel_dtlsVersions.html
@@ -0,0 +1,38 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1284103",
+    title: "Test basic data channel audio connection for supported DTLS versions"
+  });
+
+  async function testDtlsVersion(options, version) {
+    await SpecialPowers.pushPrefEnv({
+      set: [
+        ["media.peerconnection.dtls.version.min", version],
+        ["media.peerconnection.dtls.version.max", version]
+      ]
+    });
+
+    const test = new PeerConnectionTest(options);
+    addInitialDataChannel(test.chain);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+
+    await test.run();
+  }
+
+  runNetworkTest(async (options) => {
+    // 770 = DTLS 1.0, 771 = DTLS 1.2, 772 = DTLS 1.3
+    for (var version = 770; version <= 772; version++) {
+      await testDtlsVersion(options, version);
+    }
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_dataChannel_hostnameObfuscation.html b/dom/media/webrtc/tests/mochitests/test_dataChannel_hostnameObfuscation.html
new file mode 100644
index 0000000000..d0790fb9c9
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_dataChannel_hostnameObfuscation.html
@@ -0,0 +1,59 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1592620",
+    title: "Blocklist to disable hostname obfuscation"
+  });
+
+  async function testBlocklist(options, blocklistEntry, shouldBeObfuscated) {
+    let test = new PeerConnectionTest(options);
+    addInitialDataChannel(test.chain);
+
+    if (blocklistEntry !== null) {
+      await SpecialPowers.pushPrefEnv({
+        set: [
+          ["media.peerconnection.ice.obfuscate_host_addresses.blocklist",
+           blocklistEntry]
+        ]
+      });
+    }
+
+    test.chain.insertAfter('PC_LOCAL_WAIT_FOR_ICE_CONNECTED', [
+      async function CHECK_LOCAL_CANDIDATES() {
+        const stats = await test.pcLocal.getStats();
+        stats.forEach(s => {
+          if (s.type === 'local-candidate') {
+            if (shouldBeObfuscated) {
+              ok(s.address.includes(".local"), "address should be obfuscated");
+            } else {
+              ok(!s.address.includes(".local"), "address should not be obfuscated");
+            }
+          }
+        });
+      }]);
+
+    await test.run();
+  }
+
+  runNetworkTest(async (options) => {
+    await SpecialPowers.pushPrefEnv({
+      set: [["media.peerconnection.ice.obfuscate_host_addresses", true]]
+    });
+    await testBlocklist(options, null, true);
+    await testBlocklist(options, "", true);
+    await testBlocklist(options, "example.com", true);
+    await testBlocklist(options, "mochi.test", false);
+    await testBlocklist(options, "example.com,mochi.test", false);
+    await testBlocklist(options, "*.test", false);
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_dataChannel_noOffer.html b/dom/media/webrtc/tests/mochitests/test_dataChannel_noOffer.html
new file mode 100644
index 0000000000..a6e9fa5214
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_dataChannel_noOffer.html
@@ -0,0 +1,33 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "856319",
+    title: "Don't offer m=application unless createDataChannel is called first"
+  });
+
+  runNetworkTest(async function () {
+    const pc = new RTCPeerConnection();
+
+    // necessary to circumvent bug 864109
+    const options = { offerToReceiveAudio: true };
+
+    const errorCallback = generateErrorCallback();
+    try {
+      const offer = await pc.createOffer(options);
+      ok(!offer.sdp.includes("m=application"),
+        "m=application is not contained in the SDP");
+    } catch(e) {
+      errorCallback(e);
+    }
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_dataChannel_stats.html b/dom/media/webrtc/tests/mochitests/test_dataChannel_stats.html
new file mode 100644
index 0000000000..4498e2d23a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_dataChannel_stats.html
@@ -0,0 +1,50 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1218356",
+    title: "DataChannel stats"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    test.chain.remove('PC_LOCAL_CHECK_STATS');
+    test.chain.remove('PC_REMOTE_CHECK_STATS');
+    addInitialDataChannel(test.chain);
+    test.chain.removeAfter("PC_REMOTE_CHECK_ICE_CONNECTIONS");
+    test.chain.insertAfter("PC_REMOTE_CHECK_ICE_CONNECTIONS",
+      async function TEST_DATA_CHANNEL_STATS(test) {
+        const channel = test.pcLocal.dataChannels[0];
+        test.pcRemote.dataChannels[0].onbufferedamountlow = () => {};
+        test.pcRemote.dataChannels[0].send(`Sending Message`);
+        channel.onbufferedamountlow = () => {};
+        const event = await new Promise( r => channel.onmessage = r);
+        info(`Received message: "${event.data}"`);
+        const report = await test.pcLocal.getStats();
+        info(`Received Stats ${JSON.stringify([...report.values()], null, 2)}\n`);
+        const stats = [...report.values()].find(block => block.type == "data-channel");
+        info(`DataChannel stats ${JSON.stringify(stats, null, 2)}`);
+        is(stats.label, channel.label, 'DataChannel stats has correct label');
+        is(stats.protocol, channel.protocol,
+           'DataChannel stats has correct protocol');
+        is(stats.dataChannelIdentifier, channel.id,
+           'DataChannel stats has correct dataChannelIdentifier');
+        is(stats.state, channel.readyState, 'DataChannel has correct state');
+        is(stats.bytesReceived, 15, 'DataChannel has correct bytesReceived');
+        is(stats.bytesSent, 0, 'DataChannel has correct bytesSent');
+        is(stats.messagesReceived, 1,
+           'DataChannel has correct messagesReceived');
+        is(stats.messagesSent, 0, 'DataChannel has correct messagesSent');
+    });
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_defaultAudioConstraints.html b/dom/media/webrtc/tests/mochitests/test_defaultAudioConstraints.html
new file mode 100644
index 0000000000..8e0db48fff
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_defaultAudioConstraints.html
@@ -0,0 +1,80 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+"use strict";
+
+createHTML({
+  title: "Test that the audio constraints that observe at the audio constraints we expect.",
+  bug: "1509842"
+});
+
+runTest(async () => {
+  // We need a real device to get a MediaEngine supporting constraints
+  let audioDevice = SpecialPowers.getCharPref("media.audio_loopback_dev", "");
+  if (!audioDevice) {
+    todo(false, "No device set by framework. Try --use-test-media-devices");
+    return;
+  }
+
+  // Get a gUM track with the default settings, check that they are what we
+  // expect.
+  let stream = await navigator.mediaDevices.getUserMedia({ audio: true });
+  let track = stream.getAudioTracks()[0];
+  let defaultSettings = track.getSettings();
+
+  is(defaultSettings.echoCancellation, true,
+      "Echo cancellation should be ON by default.");
+  is(defaultSettings.noiseSuppression, true,
+      "Noise suppression should be ON by default.");
+  is(defaultSettings.autoGainControl, true,
+      "Automatic gain control should be ON by default.");
+
+  track.stop();
+
+  // This is UA-dependant, and belongs in a Mochitest, not in a WPT.
+  // When a gUM track has been requested with `echoCancellation` OFF, check that
+  // `noiseSuppression` and `autoGainControl` are off as well.
+  stream =
+    await navigator.mediaDevices.getUserMedia({audio:{echoCancellation: false}});
+  track = stream.getAudioTracks()[0];
+  defaultSettings = track.getSettings();
+
+  is(defaultSettings.echoCancellation, false,
+      "Echo cancellation should be OFF when requested.");
+  is(defaultSettings.noiseSuppression, false,
+      `Noise suppression should be OFF when echoCancellation is the only
+      constraint and is OFF.`);
+  is(defaultSettings.autoGainControl, false,
+      `Automatic gain control should be OFF when echoCancellation is the only
+      constraint and is OFF.`);
+
+  track.stop();
+
+  // When a gUM track has been requested with `echoCancellation` OFF, check that
+  // `noiseSuppression` and `autoGainControl` are not OFF as well if another
+  // constraint has been specified.
+  stream =
+    await navigator.mediaDevices.getUserMedia({audio:{echoCancellation: false,
+                                                      autoGainControl: true}});
+  track = stream.getAudioTracks()[0];
+  defaultSettings = track.getSettings();
+
+  is(defaultSettings.echoCancellation, false,
+      "Echo cancellation should be OFF when requested.");
+  is(defaultSettings.noiseSuppression, false,
+      `Noise suppression should be OFF when echoCancellation is OFF and another
+      constraint has been specified.`);
+  is(defaultSettings.autoGainControl, true,
+      "Auto gain control should be ON when requested.");
+
+  track.stop();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_enumerateDevices.html b/dom/media/webrtc/tests/mochitests/test_enumerateDevices.html
new file mode 100644
index 0000000000..48bec0006a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_enumerateDevices.html
@@ -0,0 +1,141 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({ title: "Run enumerateDevices code", bug: "1046245" });
+/**
+  Tests covering enumerateDevices API and deviceId constraint. Exercise code.
+*/
+
+async function mustSucceedWithStream(msg, f) {
+  try {
+    const stream = await f();
+    for (const track of stream.getTracks()) {
+      track.stop();
+    }
+    ok(true, msg + " must succeed");
+  } catch (e) {
+    is(e.name, null, msg + " must succeed: " + e.message);
+  }
+}
+
+async function mustFailWith(msg, reason, constraint, f) {
+  try {
+    await f();
+    ok(false, msg + " must fail");
+  } catch(e) {
+    is(e.name, reason, msg + " must fail: " + e.message);
+    if (constraint) {
+      is(e.constraint, constraint, msg + " must fail w/correct constraint.");
+    }
+  }
+}
+
+const gUM = c => navigator.mediaDevices.getUserMedia(c);
+
+const kinds = ["videoinput", "audioinput", "audiooutput"];
+
+function validateDevice({kind, label, deviceId, groupId}) {
+  ok(kinds.includes(kind), "Known device kind");
+  is(deviceId.length, 44, "deviceId length id as expected for Firefox");
+  ok(label.length !== undefined, "Device label: " + label);
+  isnot(groupId, "", "groupId must be present.");
+}
+
+runTest(async () => {
+  await pushPrefs(["media.navigator.streams.fake", true]);
+
+  // Validate enumerated devices after gUM.
+  for (const track of (await gUM({video: true, audio: true})).getTracks()) {
+    track.stop();
+  }
+
+  let devices = await navigator.mediaDevices.enumerateDevices();
+  ok(devices.length, "At least one device found");
+  const jsoned = JSON.parse(JSON.stringify(devices));
+  is(jsoned[0].kind, devices[0].kind, "kind survived serializer");
+  is(jsoned[0].deviceId, devices[0].deviceId, "deviceId survived serializer");
+  for (const device of devices) {
+    validateDevice(device);
+    if (device.kind == "audiooutput") continue;
+    // Test deviceId constraint
+    let deviceId = device.deviceId;
+    let constraints = (device.kind == "videoinput") ? { video: { deviceId } }
+                                                    : { audio: { deviceId } };
+    for (const track of (await gUM(constraints)).getTracks()) {
+      is(typeof(track.label), "string", "Track label is a string");
+      is(track.label, device.label, "Track label is the device label");
+      track.stop();
+    }
+  }
+
+  const unknownId = "unknown9qHr8B0JIbcHlbl9xR+jMbZZ8WyoPfpCXPfc=";
+
+  // Check deviceId failure paths for video.
+
+  await mustSucceedWithStream("unknown plain deviceId on video",
+                              () => gUM({ video: { deviceId: unknownId } }));
+  await mustSucceedWithStream("unknown plain deviceId on audio",
+                              () => gUM({ audio: { deviceId: unknownId } }));
+  await mustFailWith("unknown exact deviceId on video",
+                     "OverconstrainedError", "deviceId",
+                     () => gUM({ video: { deviceId: { exact: unknownId } } }));
+  await mustFailWith("unknown exact deviceId on audio",
+                     "OverconstrainedError", "deviceId",
+                     () => gUM({ audio: { deviceId: { exact: unknownId } } }));
+
+  // Check that deviceIds are stable for same origin and differ across origins.
+
+  const path = "/tests/dom/media/webrtc/tests/mochitests/test_enumerateDevices_iframe.html";
+  const origins = ["https://example.com", "https://test1.example.com"];
+  info(window.location);
+
+  const haveDevicesMap = new Promise(resolve => {
+    const map = new Map();
+    window.addEventListener("message", ({origin, data}) => {
+      ok(origins.includes(origin), "Got message from expected origin");
+      map.set(origin, JSON.parse(data));
+      if (map.size < origins.length) return;
+      resolve(map);
+    });
+  });
+
+  await Promise.all(origins.map(origin => {
+    const iframe = document.createElement("iframe");
+    iframe.src = origin + path;
+    iframe.allow = "camera;microphone;speaker-selection";
+    info(iframe.src);
+    document.documentElement.appendChild(iframe);
+    return new Promise(resolve => iframe.onload = resolve);
+  }));
+  let devicesMap = await haveDevicesMap;
+  let [sameOriginDevices, differentOriginDevices] = origins.map(o => devicesMap.get(o));
+
+  is(sameOriginDevices.length, devices.length, "same origin same devices");
+  is(differentOriginDevices.length, devices.length, "cross origin same devices");
+  [...sameOriginDevices, ...differentOriginDevices].forEach(d => validateDevice(d));
+
+  for (const device of sameOriginDevices) {
+    ok(devices.find(d => d.deviceId == device.deviceId),
+       "Same origin deviceId for " + device.label + " must match");
+  }
+  for (const device of differentOriginDevices) {
+    ok(!devices.find(d => d.deviceId == device.deviceId),
+         "Different origin deviceId for " + device.label + " must be different");
+  }
+
+  // Check the special case of no devices found.
+  await pushPrefs(["media.navigator.streams.fake", false],
+                  ["media.audio_loopback_dev", "none"],
+                  ["media.video_loopback_dev", "none"]);
+  devices = await navigator.mediaDevices.enumerateDevices();
+  is(devices.length, 0, "No devices");
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_enumerateDevices_getUserMediaFake.html b/dom/media/webrtc/tests/mochitests/test_enumerateDevices_getUserMediaFake.html
new file mode 100644
index 0000000000..7952bcba1b
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_enumerateDevices_getUserMediaFake.html
@@ -0,0 +1,63 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <meta charset="utf-8">
+  <script src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+  <script>
+"use strict";
+
+createHTML({
+  title: "Test labeled devices or speakers aren't exposed in enumerateDevices() after fake getUserMedia()",
+  bug: "1743524"
+});
+
+runTest(async () => {
+  await pushPrefs(
+    ["media.setsinkid.enabled", true],
+    // This test uses real devices because fake devices are not grouped with
+    // audiooutput devices.
+    ["media.navigator.streams.fake", false]);
+  const devices = navigator.mediaDevices;
+  {
+    // `fake:true` means that getUserMedia() resolves without any permission
+    // check, and so this should not be sufficient to expose real device info.
+    const stream = await devices.getUserMedia({ audio: true, fake: true });
+    stream.getTracks()[0].stop();
+    const list = await devices.enumerateDevices();
+    const labeledDevices = list.filter(({label}) => label != "");
+    is(labeledDevices.length, 0, "must be zero labeled devices after fake gUM");
+    const outputDevices = list.filter(({kind}) => kind == "audiooutput");
+    is(outputDevices.length, 0, "must be zero output devices after fake gUM");
+  }
+  {
+    // Check without `fake:true` to verify assumptions about existing devices.
+    let stream;
+    try {
+      stream = await devices.getUserMedia({ audio: true });
+      stream.getTracks()[0].stop();
+    } catch (e) {
+      if (e.name == "NotFoundError" &&
+          navigator.userAgent.includes("Mac OS X")) {
+        todo(false, "Expecting no real audioinput device on Mac test machines");
+        return;
+      }
+      throw e;
+    }
+    {
+      const list = await devices.enumerateDevices();
+      const audioDevices = list.filter(({kind}) => kind.includes("audio"));
+      ok(audioDevices.length, "have audio devices after real gUM");
+      const unlabeledAudioDevices = audioDevices.filter(({label}) => !label);
+      is(unlabeledAudioDevices.length, 0,
+         "must be zero unlabeled audio devices after real gUM");
+
+      const outputDevices = list.filter(({kind}) => kind == "audiooutput");
+      isnot(outputDevices.length, 0, "have output devices after real gUM");
+    }
+  }
+});
+  </script>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_enumerateDevices_iframe.html b/dom/media/webrtc/tests/mochitests/test_enumerateDevices_iframe.html
new file mode 100644
index 0000000000..beea3a4f97
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_enumerateDevices_iframe.html
@@ -0,0 +1,28 @@
+<!DOCTYPE HTML>
+<html>
+<body>
+<pre id="test">
+<script type="application/javascript">
+/**
+  Runs inside iframe in test_enumerateDevices.html.
+*/
+
+const pushPrefs = (...p) => SpecialPowers.pushPrefEnv({set: p});
+const gUM = c => navigator.mediaDevices.getUserMedia(c);
+
+(async () => {
+  await pushPrefs(["media.navigator.streams.fake", true]);
+
+  // Validate enumerated devices after gUM.
+  for (const track of (await gUM({video: true, audio: true})).getTracks()) {
+    track.stop();
+  }
+
+  const devices = await navigator.mediaDevices.enumerateDevices();
+  parent.postMessage(JSON.stringify(devices), "https://example.com:443");
+
+})().catch(e => setTimeout(() => { throw e; }));
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_enumerateDevices_iframe_pre_gum.html b/dom/media/webrtc/tests/mochitests/test_enumerateDevices_iframe_pre_gum.html
new file mode 100644
index 0000000000..f2dc2d1f65
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_enumerateDevices_iframe_pre_gum.html
@@ -0,0 +1,22 @@
+<!DOCTYPE HTML>
+<html>
+<body>
+<pre id="test">
+<script type="application/javascript">
+/**
+  Runs inside iframe in test_enumerateDevices_legacy.html.
+*/
+
+const pushPrefs = (...p) => SpecialPowers.pushPrefEnv({set: p});
+
+(async () => {
+  await pushPrefs(["media.navigator.streams.fake", true]);
+
+  const devices = await navigator.mediaDevices.enumerateDevices();
+  parent.postMessage(JSON.stringify(devices), "https://example.com:443");
+
+})().catch(e => setTimeout(() => { throw e; }));
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_enumerateDevices_legacy.html b/dom/media/webrtc/tests/mochitests/test_enumerateDevices_legacy.html
new file mode 100644
index 0000000000..c599f2b599
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_enumerateDevices_legacy.html
@@ -0,0 +1,147 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({ title: "Run enumerateDevices code", bug: "1046245" });
+/**
+  This is a modified copy of test_enumerateDevices.html testing the
+  enumerateDevices() legacy version and deviceId constraint.
+*/
+
+async function mustSucceedWithStream(msg, f) {
+  try {
+    const stream = await f();
+    for (const track of stream.getTracks()) {
+      track.stop();
+    }
+    ok(true, msg + " must succeed");
+  } catch (e) {
+    is(e.name, null, msg + " must succeed: " + e.message);
+  }
+}
+
+async function mustFailWith(msg, reason, constraint, f) {
+  try {
+    await f();
+    ok(false, msg + " must fail");
+  } catch(e) {
+    is(e.name, reason, msg + " must fail: " + e.message);
+    if (constraint) {
+      is(e.constraint, constraint, msg + " must fail w/correct constraint.");
+    }
+  }
+}
+
+const gUM = c => navigator.mediaDevices.getUserMedia(c);
+
+const kinds = ["videoinput", "audioinput", "audiooutput"];
+
+function validateDevice({kind, label, deviceId, groupId}) {
+  ok(kinds.includes(kind), "Known device kind");
+  is(deviceId.length, 44, "deviceId length id as expected for Firefox");
+  ok(label.length !== undefined, "Device label: " + label);
+  isnot(groupId, "", "groupId must be present.");
+}
+
+runTest(async () => {
+  await pushPrefs(["media.navigator.streams.fake", true],
+                  ["media.devices.enumerate.legacy.enabled", true]);
+
+  // Validate enumerated devices before gUM (legacy).
+
+  let devices = await navigator.mediaDevices.enumerateDevices();
+  ok(devices.length, "At least one device found");
+  const jsoned = JSON.parse(JSON.stringify(devices));
+  is(jsoned[0].kind, devices[0].kind, "kind survived serializer");
+  is(jsoned[0].deviceId, devices[0].deviceId, "deviceId survived serializer");
+  for (const device of devices) {
+    validateDevice(device);
+    if (device.kind == "audiooutput") continue;
+    is(device.label, "", "Device label is empty");
+    // Test deviceId constraint
+    let deviceId = device.deviceId;
+    let constraints = (device.kind == "videoinput") ? { video: { deviceId } }
+                                                    : { audio: { deviceId } };
+    let namedDevices;
+    for (const track of (await gUM(constraints)).getTracks()) {
+      is(typeof(track.label), "string", "Track label is a string");
+      isnot(track.label.length, 0, "Track label is not empty");
+      if (!namedDevices) {
+        namedDevices = await navigator.mediaDevices.enumerateDevices();
+      }
+      const namedDevice = namedDevices.find(d => d.deviceId == device.deviceId);
+      is(track.label, namedDevice.label, "Track label is the device label");
+      track.stop();
+    }
+  }
+
+  const unknownId = "unknown9qHr8B0JIbcHlbl9xR+jMbZZ8WyoPfpCXPfc=";
+
+  // Check deviceId failure paths for video.
+
+  await mustSucceedWithStream("unknown plain deviceId on video",
+                              () => gUM({ video: { deviceId: unknownId } }));
+  await mustSucceedWithStream("unknown plain deviceId on audio",
+                              () => gUM({ audio: { deviceId: unknownId } }));
+  await mustFailWith("unknown exact deviceId on video",
+                     "OverconstrainedError", "deviceId",
+                     () => gUM({ video: { deviceId: { exact: unknownId } } }));
+  await mustFailWith("unknown exact deviceId on audio",
+                     "OverconstrainedError", "deviceId",
+                     () => gUM({ audio: { deviceId: { exact: unknownId } } }));
+
+  // Check that deviceIds are stable for same origin and differ across origins.
+
+  const path = "/tests/dom/media/webrtc/tests/mochitests/test_enumerateDevices_iframe_pre_gum.html";
+  const origins = ["https://example.com", "https://test1.example.com"];
+  info(window.location);
+
+  const haveDevicesMap = new Promise(resolve => {
+    const map = new Map();
+    window.addEventListener("message", ({origin, data}) => {
+      ok(origins.includes(origin), "Got message from expected origin");
+      map.set(origin, JSON.parse(data));
+      if (map.size < origins.length) return;
+      resolve(map);
+    });
+  });
+
+  await Promise.all(origins.map(origin => {
+    const iframe = document.createElement("iframe");
+    iframe.src = origin + path;
+    iframe.allow = "camera;microphone;speaker-selection";
+    info(iframe.src);
+    document.documentElement.appendChild(iframe);
+    return new Promise(resolve => iframe.onload = resolve);
+  }));
+  let devicesMap = await haveDevicesMap;
+  let [sameOriginDevices, differentOriginDevices] = origins.map(o => devicesMap.get(o));
+
+  is(sameOriginDevices.length, devices.length, "same origin same devices");
+  is(differentOriginDevices.length, devices.length, "cross origin same devices");
+  [...sameOriginDevices, ...differentOriginDevices].forEach(d => validateDevice(d));
+
+  for (const device of sameOriginDevices) {
+    ok(devices.find(d => d.deviceId == device.deviceId),
+       "Same origin deviceId for " + device.label + " must match");
+  }
+  for (const device of differentOriginDevices) {
+    ok(!devices.find(d => d.deviceId == device.deviceId),
+         "Different origin deviceId for " + device.label + " must be different");
+  }
+
+  // Check the special case of no devices found.
+  await pushPrefs(["media.navigator.streams.fake", false],
+                  ["media.audio_loopback_dev", "none"],
+                  ["media.video_loopback_dev", "none"]);
+  devices = await navigator.mediaDevices.enumerateDevices();
+  is(devices.length, 0, "No devices");
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_enumerateDevices_navigation.html b/dom/media/webrtc/tests/mochitests/test_enumerateDevices_navigation.html
new file mode 100644
index 0000000000..bf7650223f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_enumerateDevices_navigation.html
@@ -0,0 +1,54 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<iframe id="iframe" srcdoc="<script>
+  window.enumerateDevices = () =>
+    navigator.mediaDevices.enumerateDevices();
+  </script>"
+    width="100%" height="50%" frameborder="1">
+</iframe>
+<pre id="test">
+<script type="application/javascript">
+createHTML({ title: "Suspend enumerateDevices code ", bug: "1479840" });
+/**
+  This test covers the case that the enumerateDevices method is suspended by
+  navigating away the current window. In order to implement that the enumeration
+  is executed in an iframe which is cleared before the enumeration has been resolved
+*/
+
+runTest(async () => {
+  // Run enumerate devices and mesure the time it will take.
+  const start = new Date().getTime();
+  try {
+    await iframe.contentWindow.enumerateDevices();
+  } catch (e) {
+    info("Failed to enumerate devices, error: " + e);
+  }
+  const elapsed = new Date().getTime() - start;
+
+  // Run again and navigate away. Expected to remain pending.
+  let p = iframe.contentWindow.enumerateDevices()
+  p.then( devices => {
+    ok(false, "Enumerate devices promise resolved unexpectedly, found " + devices.length + " devices.");
+  })
+  .catch ( error => {
+    ok(false, "Enumerate devices promise rejected unexpectedly: " + error);
+  });
+  iframe.srcdoc = "";
+
+  // Wait enough time.
+  try {
+    await timeout(p, 5 * elapsed, "timeout");
+    ok(false, "Enumerate devices promise resolved unexpectedly");
+  } catch (e) {
+    is(e.message, "timeout", "We should time out without enumerateDevices rejecting");
+  }
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_fingerprinting_resistance.html b/dom/media/webrtc/tests/mochitests/test_fingerprinting_resistance.html
new file mode 100644
index 0000000000..7e9cd5a219
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_fingerprinting_resistance.html
@@ -0,0 +1,112 @@
+<!DOCTYPE html>
+<html>
+<head>
+<meta charset="utf-8">
+<script src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<script>
+/* global SimpleTest SpecialPowers */
+
+async function testEnumerateDevices(expectDevices) {
+  let devices = await navigator.mediaDevices.enumerateDevices();
+  if (!expectDevices) {
+    SimpleTest.is(devices.length, 0, "testEnumerateDevices: No devices");
+    return;
+  }
+  let cams = devices.filter((device) => device.kind == "videoinput");
+  let mics = devices.filter((device) => device.kind == "audioinput");
+  SimpleTest.ok((cams.length == 1) && (mics.length == 1),
+                "testEnumerateDevices: a microphone and a camera");
+}
+
+async function testGetUserMedia(expectDevices) {
+  const constraints = [
+    {audio: true},
+    {video: true},
+    {audio: true, video: true},
+    {video: {width: {min: 1e9}}}, // impossible
+    {audio: {channelCount: {exact: 1e3}}}, // impossible
+  ];
+  for (let constraint of constraints) {
+    let message = "getUserMedia(" + JSON.stringify(constraint) + ")";
+    try {
+      let stream = await navigator.mediaDevices.getUserMedia(constraint);
+      SimpleTest.ok(expectDevices, message + " resolved");
+      if (!expectDevices) {
+        continue;
+      }
+
+      // We only do testGetUserMedia(true) when privacy.resistFingerprinting
+      // is true, test if MediaStreamTrack.label is spoofed.
+      for (let track of stream.getTracks()) {
+        switch (track.kind) {
+          case "audio":
+            SimpleTest.is(track.label, "Internal Microphone", "AudioStreamTrack.label");
+            break;
+          case "video":
+            SimpleTest.is(track.label, "Internal Camera", "VideoStreamTrack.label");
+            break;
+          default:
+            SimpleTest.ok(false, "Unknown kind: " + track.kind);
+            break;
+        }
+        track.stop();
+      }
+    } catch (e) {
+      if (!expectDevices) {
+        SimpleTest.is(e.name, "NotAllowedError", message + " throws NotAllowedError");
+      } else {
+        SimpleTest.ok(false, message + " failed: " + e);
+      }
+    }
+  }
+}
+
+async function testDevices() {
+  await SpecialPowers.pushPrefEnv({
+    set: [
+      ["privacy.resistFingerprinting", true],
+      ["media.navigator.streams.fake", true]
+    ]
+  });
+  await testEnumerateDevices(true); // should list a microphone and a camera
+  await testGetUserMedia(true); // should get audio and video streams
+}
+
+async function testNoDevices() {
+  await SpecialPowers.pushPrefEnv({
+    set: [
+      ["privacy.resistFingerprinting", false],
+      ["media.navigator.permission.device", false],
+      ["media.navigator.streams.fake", false],
+      ["media.audio_loopback_dev", "foo"],
+      ["media.video_loopback_dev", "bar"]
+    ]
+  });
+  await testEnumerateDevices(false); // should list nothing
+  await SpecialPowers.pushPrefEnv({
+    set: [
+      ["privacy.resistFingerprinting", true]
+    ]
+  });
+  await testEnumerateDevices(true); // should list a microphone and a camera
+  await testGetUserMedia(false); // should reject with NotAllowedError
+}
+
+createHTML({
+  title: "Neutralize the threat of fingerprinting of media devices API when 'privacy.resistFingerprinting' is true",
+  bug: "1372073"
+});
+
+runTest(async () => {
+  // Make sure enumerateDevices and getUserMedia work when
+  // privacy.resistFingerprinting is true.
+  await testDevices();
+
+  // Test that absence of devices can't be detected.
+  await testNoDevices();
+});
+</script>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_forceSampleRate.html b/dom/media/webrtc/tests/mochitests/test_forceSampleRate.html
new file mode 100644
index 0000000000..c5a9820aaa
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_forceSampleRate.html
@@ -0,0 +1,23 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the pref media.cubeb.force_sample_rate</title>
+  <script type="text/javascript" src="/MochiKit/MochiKit.js"></script>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<script class="testbody" type="text/javascript">
+const WEIRD_SAMPLE_RATE = 44101;
+
+SimpleTest.waitForExplicitFinish();
+SpecialPowers.pushPrefEnv({"set": [
+  ["media.cubeb.force_sample_rate", WEIRD_SAMPLE_RATE]
+]}).then(function() {
+  var ac = new AudioContext();
+  is(ac.sampleRate, WEIRD_SAMPLE_RATE, "Forced sample-rate set successfully.");
+  SimpleTest.finish();
+});
+</script>
+</pre>
+</body>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_GC_MediaStream.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_GC_MediaStream.html
new file mode 100644
index 0000000000..5aa0e64947
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_GC_MediaStream.html
@@ -0,0 +1,59 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  "use strict";
+
+  createHTML({
+    title: "MediaStreams can be garbage collected",
+    bug: "1407542"
+  });
+
+  let SpecialStream = SpecialPowers.wrap(MediaStream);
+
+  async function testGC(stream, numCopies, copy) {
+    let startStreams = await SpecialStream.countUnderlyingStreams();
+
+    let copies = new Array(numCopies).fill(0).map(() => copy(stream));
+    ok(await SpecialStream.countUnderlyingStreams() > startStreams,
+        "MediaStreamTrack constructor creates more underlying streams");
+
+    copies = [];
+    await new Promise(r => SpecialPowers.exactGC(r));
+    is(await SpecialStream.countUnderlyingStreams(), startStreams,
+       "MediaStreamTracks should have been collected");
+  }
+
+  runTest(async () => {
+    // We do not need LoopbackTone because it is not used
+    // and creates extra streams that affect the result
+    DISABLE_LOOPBACK_TONE = true;
+
+    let gUMStream = await getUserMedia({video: true});
+    info("Testing GC of track-array constructor with cloned tracks");
+    await testGC(gUMStream, 10, s => new MediaStream(s.getTracks().map(t => t.clone())));
+
+    info("Testing GC of empty constructor plus addTrack with cloned tracks");
+    await testGC(gUMStream, 10, s => {
+      let s2 = new MediaStream();
+      s.getTracks().forEach(t => s2.addTrack(t.clone()));
+      return s2;
+    });
+
+    info("Testing GC of cloned stream");
+    await testGC(gUMStream, 10, s => s.clone());
+
+    info("Testing GC of gUM stream");
+    gUMStream = null;
+    await new Promise(r => SpecialPowers.exactGC(r));
+    is(await SpecialStream.countUnderlyingStreams(), 0,
+       "Original gUM stream should be collectable");
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_active_autoplay.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_active_autoplay.html
new file mode 100644
index 0000000000..c1a39cdd4c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_active_autoplay.html
@@ -0,0 +1,61 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<video id="testAutoplay" autoplay></video>
+<script type="application/javascript">
+"use strict";
+
+const video = document.getElementById("testAutoplay");
+var stream;
+var otherVideoTrack;
+var otherAudioTrack;
+
+createHTML({
+  title: "MediaStream can be autoplayed in media element after going inactive and then active",
+  bug: "1208316"
+});
+
+runTest(() => getUserMedia({audio: true, video: true}).then(s => {
+  stream = s;
+  otherVideoTrack = stream.getVideoTracks()[0].clone();
+  otherAudioTrack = stream.getAudioTracks()[0].clone();
+
+  video.srcObject = stream;
+  return haveEvent(video, "playing", wait(5000, new Error("Timeout")));
+})
+.then(() => {
+  ok(!video.ended, "Video element should be playing after adding a gUM stream");
+  stream.getTracks().forEach(t => t.stop());
+  return haveEvent(video, "ended", wait(5000, new Error("Timeout")));
+})
+.then(() => {
+  ok(video.ended, "Video element should be ended");
+  stream.addTrack(otherVideoTrack);
+  return haveEvent(video, "playing", wait(5000, new Error("Timeout")));
+})
+.then(() => {
+  ok(!video.ended, "Video element should be playing after adding a video track");
+  stream.getTracks().forEach(t => t.stop());
+  return haveEvent(video, "ended", wait(5000, new Error("Timeout")));
+})
+.then(() => {
+  ok(video.ended, "Video element should be ended");
+  stream.addTrack(otherAudioTrack);
+  return haveEvent(video, "playing", wait(5000, new Error("Timeout")));
+})
+.then(() => {
+  ok(!video.ended, "Video element should be playing after adding a audio track");
+  stream.getTracks().forEach(t => t.stop());
+  return haveEvent(video, "ended", wait(5000, new Error("Timeout")));
+})
+.then(() => {
+  ok(video.ended, "Video element should be ended");
+}));
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_addTrackRemoveTrack.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_addTrackRemoveTrack.html
new file mode 100644
index 0000000000..27dad2519f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_addTrackRemoveTrack.html
@@ -0,0 +1,169 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  "use strict";
+
+  createHTML({
+    title: "MediaStream's addTrack() and removeTrack() with getUserMedia streams Test",
+    bug: "1103188"
+  });
+
+  runTest(() => Promise.resolve()
+    .then(() => getUserMedia({audio: true})).then(stream =>
+      getUserMedia({video: true}).then(otherStream => {
+        info("Test addTrack()ing a video track to an audio-only gUM stream");
+        var track = stream.getTracks()[0];
+        var otherTrack = otherStream.getTracks()[0];
+
+        stream.addTrack(track);
+        checkMediaStreamContains(stream, [track], "Re-added audio");
+
+        stream.addTrack(otherTrack);
+        checkMediaStreamContains(stream, [track, otherTrack], "Added video");
+
+        var testElem = createMediaElement('video', 'testAddTrackAudioVideo');
+        var playback = new MediaStreamPlayback(testElem, stream);
+        return playback.playMedia(false);
+    }))
+    .then(() => getUserMedia({video: true})).then(stream =>
+      getUserMedia({video: true}).then(otherStream => {
+        info("Test addTrack()ing a video track to a video-only gUM stream");
+        var track = stream.getTracks()[0];
+        var otherTrack = otherStream.getTracks()[0];
+
+        stream.addTrack(track);
+        checkMediaStreamContains(stream, [track], "Re-added video");
+
+        stream.addTrack(otherTrack);
+        checkMediaStreamContains(stream, [track, otherTrack], "Added video");
+
+        var test = createMediaElement('video', 'testAddTrackDoubleVideo');
+        var playback = new MediaStreamPlayback(test, stream);
+        return playback.playMedia(false);
+    }))
+    .then(() => getUserMedia({video: true})).then(stream =>
+      getUserMedia({video: true}).then(otherStream => {
+        info("Test removeTrack() existing and added video tracks from a video-only gUM stream");
+        var track = stream.getTracks()[0];
+        var otherTrack = otherStream.getTracks()[0];
+
+        stream.removeTrack(otherTrack);
+        checkMediaStreamContains(stream, [track], "Removed non-existing video");
+
+        stream.addTrack(otherTrack);
+        checkMediaStreamContains(stream, [track, otherTrack], "Added video");
+
+        stream.removeTrack(otherTrack);
+        checkMediaStreamContains(stream, [track], "Removed added video");
+
+        stream.removeTrack(otherTrack);
+        checkMediaStreamContains(stream, [track], "Re-removed added video");
+
+        stream.removeTrack(track);
+        checkMediaStreamContains(stream, [], "Removed original video");
+
+        var elem = createMediaElement('video', 'testRemoveAllVideo');
+        var loadeddata = false;
+        elem.onloadeddata = () => { loadeddata = true; elem.onloadeddata = null; };
+        elem.srcObject = stream;
+        elem.play();
+        return wait(500).then(() => {
+          ok(!loadeddata, "Stream without tracks shall not raise 'loadeddata' on media element");
+          elem.pause();
+          elem.srcObject = null;
+        })
+        .then(() => {
+          stream.addTrack(track);
+          checkMediaStreamContains(stream, [track], "Re-added added-then-removed track");
+          var playback = new MediaStreamPlayback(elem, stream);
+          return playback.playMedia(false);
+        })
+        .then(() => otherTrack.stop());
+    }))
+    .then(() => getUserMedia({ audio: true })).then(audioStream =>
+      getUserMedia({ video: true }).then(videoStream => {
+        info("Test adding track and removing the original");
+        var audioTrack = audioStream.getTracks()[0];
+        var videoTrack = videoStream.getTracks()[0];
+        videoStream.removeTrack(videoTrack);
+        audioStream.addTrack(videoTrack);
+
+        checkMediaStreamContains(videoStream, [], "1, Removed original track");
+        checkMediaStreamContains(audioStream, [audioTrack, videoTrack],
+                                 "2, Added external track");
+
+        var elem = createMediaElement('video', 'testAddRemoveOriginalTrackVideo');
+        var playback = new MediaStreamPlayback(elem, audioStream);
+        return playback.playMedia(false);
+      }))
+    .then(() => getUserMedia({ audio: true, video: true })).then(stream => {
+      info("Test removing stopped tracks");
+      stream.getTracks().forEach(t => {
+        t.stop();
+        stream.removeTrack(t);
+      });
+      checkMediaStreamContains(stream, [], "Removed stopped tracks");
+    })
+    .then(() => {
+      var ac = new AudioContext();
+
+      var osc1k = createOscillatorStream(ac, 1000);
+      var audioTrack1k = osc1k.getTracks()[0];
+
+      var osc5k = createOscillatorStream(ac, 5000);
+      var audioTrack5k = osc5k.getTracks()[0];
+
+      var osc10k = createOscillatorStream(ac, 10000);
+      var audioTrack10k = osc10k.getTracks()[0];
+
+      var stream = osc1k;
+      return Promise.resolve().then(() => {
+        info("Analysing audio output with original 1k track");
+        var analyser = new AudioStreamAnalyser(ac, stream);
+        return analyser.waitForAnalysisSuccess(array =>
+                 array[analyser.binIndexForFrequency(50)] < 50 &&
+                 array[analyser.binIndexForFrequency(1000)] > 200 &&
+                 array[analyser.binIndexForFrequency(5000)] < 50 &&
+                 array[analyser.binIndexForFrequency(10000)] < 50);
+      }).then(() => {
+        info("Analysing audio output with removed original 1k track and added 5k track");
+        stream.removeTrack(audioTrack1k);
+        stream.addTrack(audioTrack5k);
+        var analyser = new AudioStreamAnalyser(ac, stream);
+        return analyser.waitForAnalysisSuccess(array =>
+                 array[analyser.binIndexForFrequency(1000)] < 50 &&
+                 array[analyser.binIndexForFrequency(5000)] > 200 &&
+                 array[analyser.binIndexForFrequency(10000)] < 50);
+      }).then(() => {
+        info("Analysing audio output with removed 5k track and added 10k track");
+        stream.removeTrack(audioTrack5k);
+        stream.addTrack(audioTrack10k);
+        var analyser = new AudioStreamAnalyser(ac, stream);
+        return analyser.waitForAnalysisSuccess(array =>
+                 array[analyser.binIndexForFrequency(1000)] < 50 &&
+                 array[analyser.binIndexForFrequency(5000)] < 50 &&
+                 array[analyser.binIndexForFrequency(10000)] > 200);
+      }).then(() => {
+        info("Analysing audio output with re-added 1k, 5k and added 10k tracks");
+        stream.addTrack(audioTrack1k);
+        stream.addTrack(audioTrack5k);
+        var analyser = new AudioStreamAnalyser(ac, stream);
+        return analyser.waitForAnalysisSuccess(array =>
+                 array[analyser.binIndexForFrequency(50)] < 50 &&
+                 array[analyser.binIndexForFrequency(1000)] > 200 &&
+                 array[analyser.binIndexForFrequency(2500)] < 50 &&
+                 array[analyser.binIndexForFrequency(5000)] > 200 &&
+                 array[analyser.binIndexForFrequency(7500)] < 50 &&
+                 array[analyser.binIndexForFrequency(10000)] > 200 &&
+                 array[analyser.binIndexForFrequency(11000)] < 50);
+      });
+    }));
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_addtrack_removetrack_events.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_addtrack_removetrack_events.html
new file mode 100644
index 0000000000..833653ebb2
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_addtrack_removetrack_events.html
@@ -0,0 +1,110 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+"use strict";
+
+createHTML({
+  title: "MediaStream's 'addtrack' and 'removetrack' events shouldn't fire on manual operations",
+  bug: "1208328"
+});
+
+var spinEventLoop = () => new Promise(r => setTimeout(r, 0));
+
+var stream;
+var clone;
+var newStream;
+var tracks = [];
+
+var addTrack = track => {
+  info("Adding track " + track.id);
+  stream.addTrack(track);
+};
+var removeTrack = track => {
+  info("Removing track " + track.id);
+  stream.removeTrack(track);
+};
+var stopTrack = track => {
+  if (track.readyState == "live") {
+    info("Stopping track " + track.id);
+  }
+  track.stop();
+};
+
+runTest(() => getUserMedia({audio: true, video: true})
+  .then(s => {
+    stream = s;
+    clone = s.clone();
+    stream.addEventListener("addtrack", function onAddtrack(event) {
+      ok(false, "addtrack fired unexpectedly for track " + event.track.id);
+    });
+    stream.addEventListener("removetrack", function onRemovetrack(event) {
+      ok(false, "removetrack fired unexpectedly for track " + event.track.id);
+    });
+
+    return getUserMedia({audio: true, video: true});
+  })
+  .then(s => {
+    newStream = s;
+    info("Stopping an original track");
+    stopTrack(stream.getTracks()[0]);
+
+    return spinEventLoop();
+  })
+  .then(() => {
+    info("Removing original tracks");
+    stream.getTracks().forEach(t => (stream.removeTrack(t), tracks.push(t)));
+
+    return spinEventLoop();
+  })
+  .then(() => {
+    info("Adding other gUM tracks");
+    newStream.getTracks().forEach(t => addTrack(t))
+
+    return spinEventLoop();
+  })
+  .then(() => {
+    info("Adding cloned tracks");
+    let clone = stream.clone();
+    clone.getTracks().forEach(t => addTrack(t));
+
+    return spinEventLoop();
+  })
+  .then(() => {
+    info("Removing a clone");
+    removeTrack(clone.getTracks()[0]);
+
+    return spinEventLoop();
+  })
+  .then(() => {
+    info("Stopping clones");
+    clone.getTracks().forEach(t => stopTrack(t));
+
+    return spinEventLoop();
+  })
+  .then(() => {
+    info("Stopping originals");
+    stream.getTracks().forEach(t => stopTrack(t));
+    tracks.forEach(t => stopTrack(t));
+
+    return spinEventLoop();
+  })
+  .then(() => {
+    info("Removing remaining tracks");
+    stream.getTracks().forEach(t => removeTrack(t));
+
+    return spinEventLoop();
+  })
+  .then(() => {
+      // Test MediaStreamTrackEvent required args here.
+      mustThrowWith("MediaStreamTrackEvent without required args",
+                    "TypeError", () => new MediaStreamTrackEvent("addtrack", {}));
+  }));
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_audioCapture.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_audioCapture.html
new file mode 100644
index 0000000000..2cc649a321
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_audioCapture.html
@@ -0,0 +1,104 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioCapture </title>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script>
+
+(async () => {
+  // Get an opus file containing a sine wave at maximum amplitude, of duration
+  // `lengthSeconds`, and of frequency `frequency`.
+  async function getSineWaveFile(frequency, lengthSeconds) {
+    const off = new OfflineAudioContext(1, lengthSeconds * 48000, 48000);
+    const osc = off.createOscillator();
+    const rec = new MediaRecorder(off.destination,
+                                  {mimeType: "audio/ogg; codecs=opus"});
+    osc.frequency.value = frequency;
+    osc.connect(off.destination);
+    osc.start();
+    rec.start();
+    off.startRendering();
+    const {data} = await new Promise(r => rec.ondataavailable = r);
+    return data;
+  }
+
+  await createHTML({
+    bug: "1156472",
+    title: "Test AudioCapture with regular HTMLMediaElement, AudioContext, " +
+           "and HTMLMediaElement playing a MediaStream",
+    visible: true
+  });
+
+  await runTestWhenReady(async () => {
+    /**
+     * Get two HTMLMediaElements:
+     * - One playing a sine tone from a blob (of an opus file created on the fly)
+     * - One being the output for an AudioContext's OscillatorNode, connected to
+     *   a MediaSourceDestinationNode.
+     *
+     * Also, use the AudioContext playing through its AudioDestinationNode another
+     * tone, using another OscillatorNode.
+     *
+     * Capture the output of the document, feed that back into the AudioContext,
+     * with an AnalyserNode, and check the frequency content to make sure we
+     * have recorded the three sources.
+     *
+     * The three sine tones have frequencies far apart from each other, so that we
+     * can check that the spectrum of the capture stream contains three
+     * components with a high magnitude.
+     */
+    const wavtone = createMediaElement("audio", "WaveTone");
+    const acTone = createMediaElement("audio", "audioContextTone");
+    const ac = new AudioContext();
+
+    const oscThroughMediaElement = ac.createOscillator();
+    oscThroughMediaElement.frequency.value = 1000;
+    const oscThroughAudioDestinationNode = ac.createOscillator();
+    oscThroughAudioDestinationNode.frequency.value = 5000;
+    const msDest = ac.createMediaStreamDestination();
+
+    oscThroughMediaElement.connect(msDest);
+    oscThroughAudioDestinationNode.connect(ac.destination);
+
+    acTone.srcObject = msDest.stream;
+
+    const blob = await getSineWaveFile(10000, 10);
+    wavtone.src = URL.createObjectURL(blob);
+    oscThroughMediaElement.start();
+    oscThroughAudioDestinationNode.start();
+    wavtone.loop = true;
+    wavtone.play();
+    acTone.play();
+
+    const constraints = {audio: {mediaSource: "audioCapture"}};
+
+    const stream = await getUserMedia(constraints);
+    try {
+      const analyser = new AudioStreamAnalyser(ac, stream);
+      analyser.enableDebugCanvas();
+      await analyser.waitForAnalysisSuccess(array => {
+        // We want to find three frequency components here, around 1000, 5000
+        // and 10000Hz. Frequency are logarithmic. Also make sure we have low
+        // energy in between, not just a flat white noise.
+        return (array[analyser.binIndexForFrequency(50)]    < 50 &&
+                array[analyser.binIndexForFrequency(1000)]  > 200 &&
+                array[analyser.binIndexForFrequency(2500)]  < 50 &&
+                array[analyser.binIndexForFrequency(5000)]  > 200 &&
+                array[analyser.binIndexForFrequency(7500)]  < 50 &&
+                array[analyser.binIndexForFrequency(10000)] > 200);
+      });
+    } finally {
+      for (let t of stream.getTracks()) {
+        t.stop();
+      }
+      ac.close();
+    }
+  });
+})();
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_audioConstraints.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_audioConstraints.html
new file mode 100644
index 0000000000..162e83063a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_audioConstraints.html
@@ -0,0 +1,93 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+"use strict";
+
+createHTML({
+  title: "Test that microphone getSettings report correct settings after applyConstraints",
+  bug: "1447982",
+});
+
+function testTrackAgainstAudioConstraints(track, audioConstraints) {
+  let constraints = track.getConstraints();
+  is(constraints.autoGainControl, audioConstraints.autoGainControl,
+     "Should report correct autoGainControl constraint");
+  is(constraints.echoCancellation, audioConstraints.echoCancellation,
+     "Should report correct echoCancellation constraint");
+  is(constraints.noiseSuppression, audioConstraints.noiseSuppression,
+     "Should report correct noiseSuppression constraint");
+
+  let settings = track.getSettings();
+  is(settings.autoGainControl, audioConstraints.autoGainControl,
+     "Should report correct autoGainControl setting");
+  is(settings.echoCancellation, audioConstraints.echoCancellation,
+     "Should report correct echoCancellation setting");
+  is(settings.noiseSuppression, audioConstraints.noiseSuppression,
+     "Should report correct noiseSuppression setting");
+}
+
+async function testAudioConstraints(track, audioConstraints) {
+  // We applyConstraints() first and do a fresh gUM later, to avoid
+  // testing multiple concurrent captures at different settings.
+
+  info(`Testing applying constraints ${JSON.stringify(audioConstraints)} ` +
+       `to track with settings ${JSON.stringify(track.getSettings())}`);
+  await track.applyConstraints(audioConstraints);
+  testTrackAgainstAudioConstraints(track, audioConstraints);
+
+  info("Testing fresh gUM request with audio constraints " +
+       JSON.stringify(audioConstraints));
+  let stream = await getUserMedia({audio: audioConstraints});
+  testTrackAgainstAudioConstraints(stream.getTracks()[0], audioConstraints);
+  stream.getTracks().forEach(t => t.stop());
+}
+
+runTest(async () => {
+  let audioDevice = SpecialPowers.getCharPref("media.audio_loopback_dev", "");
+  if (!audioDevice) {
+    ok(false, "No device set by framework. Try --use-test-media-devices");
+    return;
+  }
+
+  let supportedConstraints = navigator.mediaDevices.getSupportedConstraints();
+  is(supportedConstraints.autoGainControl, true,
+     "autoGainControl constraint should be supported");
+  is(supportedConstraints.echoCancellation, true,
+     "echoCancellation constraint should be supported");
+  is(supportedConstraints.noiseSuppression, true,
+     "noiseSuppression constraint should be supported");
+
+  let egn = (e, g, n) => ({
+    echoCancellation: e,
+    autoGainControl: g,
+    noiseSuppression: n
+  });
+
+  let stream = await getUserMedia({
+    audio: egn(true, true, true),
+  });
+  let track = stream.getTracks()[0];
+  let audioConstraintsToTest = [
+    egn(false, true,  true),
+    egn(true,  false, true),
+    egn(true,  true,  false),
+    egn(false, false, true),
+    egn(false, true,  false),
+    egn(true,  false, false),
+    egn(false, false, false),
+    egn(true,  true,  true),
+  ];
+  for (let audioConstraints of audioConstraintsToTest) {
+    await testAudioConstraints(track, audioConstraints);
+  }
+  track.stop();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_audioConstraints_concurrentIframes.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_audioConstraints_concurrentIframes.html
new file mode 100644
index 0000000000..d07dbc41f1
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_audioConstraints_concurrentIframes.html
@@ -0,0 +1,157 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  title: "getUserMedia in multiple iframes with different constraints",
+  bug: "1404977"
+});
+/**
+  * Verify that we can successfully call getUserMedia for the same device in
+  * multiple iframes concurrently. This is checked by creating a number of
+  * iframes and performing a separate getUserMedia call in each. We verify the
+  * stream returned by that call has the same constraints as requested both
+  * immediately after the call and after all gUM calls have been made. The test
+  * then verifies the streams can be played.
+  */
+runTest(async function() {
+  // Compare constraints and return a string with the differences in
+  // echoCancellation, autoGainControl, and noiseSuppression. The string
+  // will be empty if there are no differences.
+  function getConstraintDifferenceString(constraints, otherConstraints) {
+    let diffString = "";
+    if (constraints.echoCancellation != otherConstraints.echoCancellation) {
+      diffString += "echoCancellation different: " +
+                    `${constraints.echoCancellation} != ${otherConstraints.echoCancellation}, `;
+    }
+    if (constraints.autoGainControl != otherConstraints.autoGainControl) {
+      diffString += "autoGainControl different: " +
+                    `${constraints.autoGainControl} != ${otherConstraints.autoGainControl}, `;
+    }
+    if (constraints.noiseSuppression != otherConstraints.noiseSuppression) {
+      diffString += "noiseSuppression different: " +
+                    `${constraints.noiseSuppression} != ${otherConstraints.noiseSuppression}, `;
+    }
+    // Replace trailing comma and space if any
+    return diffString.replace(/, $/, "");
+  }
+
+  // We need a real device to get a MediaEngine supporting constraints
+  let audioDevice = SpecialPowers.getCharPref("media.audio_loopback_dev", "");
+  if (!audioDevice) {
+    todo(false, "No device set by framework. Try --use-test-media-devices");
+    return;
+  }
+
+  let egn = (e, g, n) => ({
+    echoCancellation: e,
+    autoGainControl: g,
+    noiseSuppression: n
+  });
+
+  let allConstraintCombinations = [
+    egn(false, false, false),
+    egn(true,  false, false),
+    egn(false, true,  false),
+    egn(false, false, true),
+    egn(true,  true,  false),
+    egn(true,  false, true),
+    egn(false, true,  true),
+    egn(true,  true,  true),
+  ];
+
+  // TODO: We would like to be able to perform an arbitrary number of gUM calls
+  // at once, but issues with pulse and audio IPC mean on some systems we're
+  // limited to as few as 2 concurrent calls. To avoid issues we chunk test runs
+  // to only two calls at a time. The while, splice and GC lines can be removed,
+  // the extra scope removed and allConstraintCombinations can be renamed to
+  // constraintCombinations once this issue is resolved. See bug 1480489.
+  while (allConstraintCombinations.length) {
+    {
+      let constraintCombinations = allConstraintCombinations.splice(0, 2);
+      // Array to store objects that associate information used in our test such as
+      // constraints, iframes, gum streams, and various promises.
+      let testCases = [];
+
+      for (let constraints of constraintCombinations) {
+        let testCase = {requestedConstraints: constraints};
+        // Provide an id for logging, labeling related elements.
+        testCase.id = `testCase.` +
+                      `e=${constraints.echoCancellation}.` +
+                      `g=${constraints.noiseSuppression}.` +
+                      `n=${constraints.noiseSuppression}`;
+        testCases.push(testCase);
+        testCase.iframe = document.createElement("iframe");
+        testCase.iframeLoadedPromise = new Promise((resolve, reject) => {
+          testCase.iframe.onload = () => { resolve(); };
+        });
+        document.body.appendChild(testCase.iframe);
+      }
+      is(testCases.length,
+        constraintCombinations.length,
+        "Should have created a testcase for each constraint");
+
+      // Wait for all iframes to be loaded
+      await Promise.all(testCases.map(tc => tc.iframeLoadedPromise));
+
+      // Start a tone at our top level page so the gUM calls will record something
+      // should we wish to verify their recording in future.
+      let tone = new LoopbackTone(new AudioContext, TEST_AUDIO_FREQ);
+      tone.start();
+
+      // One by one see if we can grab a gUM stream per iframe
+      for (let testCase of testCases) {
+        // Use normal gUM rather than our test helper as the test harness was
+        // not made to be used inside iframes.
+        testCase.gumStream =
+          await testCase.iframe.contentWindow.navigator.mediaDevices.getUserMedia({audio: testCase.requestedConstraints})
+          .catch(e => Promise.reject(`getUserMedia calls should not fail! Failed at ${testCase.id} with: ${e}!`));
+        let differenceString = getConstraintDifferenceString(
+          testCase.requestedConstraints,
+          testCase.gumStream.getAudioTracks()[0].getSettings());
+        ok(!differenceString,
+          `gUM stream for ${testCase.id} should have the same constraints as were ` +
+          `requested from gUM. Differences: ${differenceString}`);
+      }
+
+      // Once all streams are collected, make sure the constraints haven't been
+      // mutated by another gUM call.
+      for (let testCase of testCases) {
+        let differenceString = getConstraintDifferenceString(
+          testCase.requestedConstraints,
+          testCase.gumStream.getAudioTracks()[0].getSettings());
+        ok(!differenceString,
+          `gUM stream for ${testCase.id} should not have had constraints altered after ` +
+          `all gUM calls are done. Differences: ${differenceString}`);
+      }
+
+      // We do not currently have tests to verify the behaviour of the different
+      // constraints. Once we do we should do further verification here. See
+      // bug 1406372, bug 1406376, and bug 1406377.
+
+      for (let testCase of testCases) {
+        let testAudio = createMediaElement("audio", `testAudio.${testCase.id}`);
+        let playback = new MediaStreamPlayback(testAudio, testCase.gumStream);
+        await playback.playMediaWithoutStoppingTracks(false);
+      }
+
+      // Stop the tracks for each stream, we left them running above via
+      // playMediaWithoutStoppingTracks to make sure they can play concurrently.
+      for (let testCase of testCases) {
+        testCase.gumStream.getTracks().map(t => t.stop());
+        document.body.removeChild(testCase.iframe);
+      }
+
+      tone.stop();
+    }
+    await new Promise(r => SpecialPowers.exactGC(r));
+  }
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_audioConstraints_concurrentStreams.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_audioConstraints_concurrentStreams.html
new file mode 100644
index 0000000000..f5b5e784ea
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_audioConstraints_concurrentStreams.html
@@ -0,0 +1,123 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  title: "getUserMedia multiple times, concurrently, and with different constraints",
+  bug: "1404977"
+});
+/**
+  * Verify that we can successfully call getUserMedia multiple times for the
+  * same device, concurrently. This is checked by calling getUserMedia a number
+  * of times with different constraints. We verify that the stream returned by
+  * that call has the same constraints as requested both immediately after the
+  * call and after all gUM calls have been made. The test then verifies the
+  * streams can be played.
+  */
+runTest(async function() {
+  // Compare constraints and return a string with the differences in
+  // echoCancellation, autoGainControl, and noiseSuppression. The string
+  // will be empty if there are no differences.
+  function getConstraintDifferenceString(constraints, otherConstraints) {
+    let diffString = "";
+    if (constraints.echoCancellation != otherConstraints.echoCancellation) {
+      diffString += "echoCancellation different: " +
+                    `${constraints.echoCancellation} != ${otherConstraints.echoCancellation}, `;
+    }
+    if (constraints.autoGainControl != otherConstraints.autoGainControl) {
+      diffString += "autoGainControl different: " +
+                    `${constraints.autoGainControl} != ${otherConstraints.autoGainControl}, `;
+    }
+    if (constraints.noiseSuppression != otherConstraints.noiseSuppression) {
+      diffString += "noiseSuppression different: " +
+                    `${constraints.noiseSuppression} != ${otherConstraints.noiseSuppression}, `;
+    }
+    // Replace trailing comma and space if any
+    return diffString.replace(/, $/, "");
+  }
+
+  // We need a real device to get a MediaEngine supporting constraints
+  let audioDevice = SpecialPowers.getCharPref("media.audio_loopback_dev", "");
+  if (!audioDevice) {
+    todo(false, "No device set by framework. Try --use-test-media-devices");
+    return;
+  }
+
+  let egn = (e, g, n) => ({
+    echoCancellation: e,
+    autoGainControl: g,
+    noiseSuppression: n
+  });
+
+  let constraintCombinations = [
+    egn(false, false, false),
+    egn(true,  false, false),
+    egn(false, true,  false),
+    egn(false, false, true),
+    egn(true,  true,  false),
+    egn(true,  false, true),
+    egn(false, true,  true),
+    egn(true,  true,  true),
+  ];
+
+  // Array to store objects that associate information used in our test such as
+  // constraints, gum streams, and various promises.
+  let testCases = [];
+
+  for (let constraints of constraintCombinations) {
+    let testCase = {requestedConstraints: constraints};
+    // Provide an id for logging, labeling related elements.
+    testCase.id = `testCase.` +
+                  `e=${constraints.echoCancellation}.` +
+                  `g=${constraints.noiseSuppression}.` +
+                  `n=${constraints.noiseSuppression}`;
+    testCases.push(testCase);
+    testCase.gumStream =
+      await getUserMedia({audio: testCase.requestedConstraints})
+      .catch(e => Promise.reject(`getUserMedia calls should not fail! Failed at ${testCase.id} with: ${e}!`));
+    let differenceString = getConstraintDifferenceString(
+      testCase.requestedConstraints,
+      testCase.gumStream.getAudioTracks()[0].getSettings());
+      ok(!differenceString,
+        `gUM stream for ${testCase.id} should have the same constraints as were ` +
+        `requested from gUM. Differences: ${differenceString}`);
+  }
+  is(testCases.length,
+    constraintCombinations.length,
+    "Should have a stream for each constraint");
+
+  // Once all streams are collected, make sure the constraints haven't been
+  // mutated by another gUM call.
+  for (let testCase of testCases) {
+    let differenceString = getConstraintDifferenceString(
+      testCase.requestedConstraints,
+      testCase.gumStream.getAudioTracks()[0].getSettings());
+    ok(!differenceString,
+      `gUM stream for ${testCase.id} should not have had constraints altered after ` +
+      `all gUM calls are done. Differences: ${differenceString}`);
+  }
+
+  // We do not currently have tests to verify the behaviour of the different
+  // constraints. Once we do we should do further verificaiton here. See
+  // bug 1406372, bug 1406376, and bug 1406377.
+
+  for (let testCase of testCases) {
+    let testAudio = createMediaElement("audio", `testAudio.${testCase.id}`);
+    let playback = new MediaStreamPlayback(testAudio, testCase.gumStream);
+    await playback.playMediaWithoutStoppingTracks(false);
+  }
+
+  // Stop the tracks for each stream, we left them running above via
+  // playMediaWithoutStoppingTracks to make sure they can play concurrently.
+  for (let testCase of testCases) {
+    testCase.gumStream.getTracks().map(t => t.stop());
+  }
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicAudio.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicAudio.html
new file mode 100644
index 0000000000..b4775b4244
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicAudio.html
@@ -0,0 +1,27 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({ title: "getUserMedia Basic Audio Test", bug: "781534" });
+  /**
+   * Run a test to verify that we can complete a start and stop media playback
+   * cycle for an audio MediaStream on an audio HTMLMediaElement.
+   */
+  runTest(function () {
+    var testAudio = createMediaElement('audio', 'testAudio');
+    var constraints = {audio: true};
+
+    return getUserMedia(constraints).then(stream => {
+      var playback = new MediaStreamPlayback(testAudio, stream);
+      return playback.playMedia(false);
+    });
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicAudio_loopback.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicAudio_loopback.html
new file mode 100644
index 0000000000..10bf669c00
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicAudio_loopback.html
@@ -0,0 +1,99 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+
+<script>
+  createHTML({
+    title: "getUserMedia Basic Audio Test Loopback",
+    bug: "1406350",
+    visible: true
+  });
+
+  /**
+   * Run a test to verify the use of LoopbackTone as audio input.
+   */
+  runTest(async () => {
+    if (!SpecialPowers.getCharPref("media.audio_loopback_dev", "")) {
+      todo(false, "No loopback device set by framework. Try --use-test-media-devices");
+      return;
+    }
+
+    // At this point DefaultLoopbackTone has been instantiated
+    // automatically on frequency TEST_AUDIO_FREQ (440 Hz). Verify
+    // that a tone is detected on that frequency.
+    info("Capturing at default frequency");
+    const stream = await getUserMedia({audio: true});
+
+    try {
+      const audioContext = new AudioContext();
+      const analyser = new AudioStreamAnalyser(audioContext, stream);
+      analyser.enableDebugCanvas();
+      await analyser.waitForAnalysisSuccess(array => {
+        // High energy on 1000 Hz low energy around that
+        const freg_50Hz   = array[analyser.binIndexForFrequency(50)];
+        const freq        = array[analyser.binIndexForFrequency(TEST_AUDIO_FREQ)];
+        const freq_2000Hz = array[analyser.binIndexForFrequency(2000)];
+
+        info("Analysing audio frequency - low:target:high = "
+                + freg_50Hz + ':' + freq + ':' + freq_2000Hz);
+        return freg_50Hz < 50 && freq > 200 && freq_2000Hz < 50;
+      });
+
+      // Use the LoopbackTone API to change the frequency of the default tone.
+      // Verify that a tone is detected on the new frequency (800 Hz).
+      info("Change loopback tone frequency");
+      DefaultLoopbackTone.changeFrequency(800);
+      await analyser.waitForAnalysisSuccess(array => {
+        const freg_50Hz   = array[analyser.binIndexForFrequency(50)];
+        const freq        = array[analyser.binIndexForFrequency(800)];
+        const freq_2000Hz = array[analyser.binIndexForFrequency(2000)];
+
+        info("Analysing audio frequency - low:target:high = "
+                + freg_50Hz + ':' + freq + ':' + freq_2000Hz);
+        return freg_50Hz < 50 && freq > 200 && freq_2000Hz < 50;
+      });
+
+      // Create a second tone at a different frequency.
+      // Verify that both tones are detected.
+      info("Multiple loopback tones");
+      DefaultLoopbackTone.changeFrequency(TEST_AUDIO_FREQ);
+      const second_tone = new LoopbackTone(audioContext, 2000);
+      second_tone.start();
+      await analyser.waitForAnalysisSuccess(array => {
+        const freg_50Hz   = array[analyser.binIndexForFrequency(50)];
+        const freq        = array[analyser.binIndexForFrequency(TEST_AUDIO_FREQ)];
+        const freq_2000Hz = array[analyser.binIndexForFrequency(2000)];
+        const freq_4000Hz = array[analyser.binIndexForFrequency(4000)];
+
+        info("Analysing audio frequency - low:target1:target2:high = "
+                + freg_50Hz + ':' + freq + ':' + freq_2000Hz + ':' + freq_4000Hz);
+        return freg_50Hz < 50 && freq > 200 && freq_2000Hz > 200 && freq_4000Hz < 50;
+      });
+
+      // Stop all tones and verify that there is no audio on the given frequencies.
+      info("Stop all loopback tones");
+      DefaultLoopbackTone.stop();
+      second_tone.stop()
+      await analyser.waitForAnalysisSuccess(array => {
+        const freg_50Hz   = array[analyser.binIndexForFrequency(50)];
+        const freq        = array[analyser.binIndexForFrequency(TEST_AUDIO_FREQ)];
+        const freq_2000Hz = array[analyser.binIndexForFrequency(2000)];
+
+        info("Analysing audio frequency - low:target:high = "
+                + freg_50Hz + ':' + freq + ':' + freq_2000Hz);
+        return freg_50Hz < 50 && freq < 50 && freq_2000Hz < 50;
+      });
+    } finally {
+      for (let t of stream.getTracks()) {
+        t.stop();
+      }
+    }
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicScreenshare.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicScreenshare.html
new file mode 100644
index 0000000000..cc73de77da
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicScreenshare.html
@@ -0,0 +1,260 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "getUserMedia Basic Screenshare Test",
+    bug: "1211656",
+    visible: true,
+  });
+
+  const {AppConstants} =
+    SpecialPowers.ChromeUtils.import("resource://gre/modules/AppConstants.jsm");
+
+  // Since the MacOS backend captures in the wrong rgb color profile we need
+  // large thresholds there, and they vary greatly by color. We define
+  // thresholds per platform and color here to still allow the test to run.
+  // Since the colors used (red, green, blue, white) consist only of "pure"
+  // components (0 or 255 for each component), the high thresholds on Mac will
+  // still be able to catch an error where the image is corrupt, or if frames
+  // don't flow.
+  const thresholds = {
+    // macos: captures in the display rgb color profile, which we treat as
+    // sRGB, which is most likely wrong. These thresholds are needed currently
+    // in CI. See bug 1827606.
+    "macosx": { "red": 120, "green": 135, "blue": 35, "white": 10 },
+    // windows: rounding errors in 1) conversion to I420 (the capture),
+    // 2) downscaling, 3) conversion to RGB (for rendering).
+    "win": { "red": 5, "green": 5, "blue": 10, "white": 5 },
+    // linux: rounding errors in 1) conversion to I420 (the capture),
+    // 2) downscaling, 3) conversion to RGB (for rendering).
+    "linux": { "red": 5, "green": 5, "blue": 10, "white": 5 },
+    // android: we don't have a screen capture backend.
+    "android": { "red": 0, "green": 0, "blue": 0, "white": 0 },
+    // other: here just because it's supported by AppConstants.platform.
+    "other": { "red": 0, "green": 0, "blue": 0, "white": 0 },
+  };
+
+  const verifyScreenshare =
+      async (video, helper, upleft, upright, downleft, downright) => {
+    if (video.readyState < video.HAVE_CURRENT_DATA) {
+      info("Waiting for data");
+      await new Promise(r => video.onloadeddata = r);
+    }
+
+    // We assume video size will not change. Offsets help to account for a
+    // square fullscreen-canvas, while the screen is rectangular.
+    const offsetX = Math.max(0, video.videoWidth - video.videoHeight) / 2;
+    const offsetY = Math.max(0, video.videoHeight - video.videoWidth) / 2;
+
+    const verifyAround = async (internalX, internalY, color) => {
+      // Pick a couple of samples around a coordinate to check for a color.
+      // We check multiple rows and columns, to avoid most artifact issues.
+      let areaSamples = [
+        {dx: 0, dy: 0},
+        {dx: 1, dy: 3},
+        {dx: 8, dy: 5},
+      ];
+      const threshold = thresholds[AppConstants.platform][color.name];
+      for (let {dx, dy} of areaSamples) {
+        const x = offsetX + dx + internalX;
+        const y = offsetY + dy + internalY;
+        info(`Checking pixel (${[x,y]}) of total resolution `
+          + `${video.videoWidth}x${video.videoHeight} against ${color.name}.`);
+        let lastPixel = [-1, -1, -1, -1];
+        await helper.waitForPixel(video, px => {
+          lastPixel = Array.from(px);
+          return helper.isPixel(px, color, threshold);
+        }, {
+          offsetX: x,
+          offsetY: y,
+          cancel: wait(30000).then(_ =>
+            new Error(`Checking ${[x,y]} against ${color.name} timed out. ` +
+                      `Got [${lastPixel}]. Threshold ${threshold}.`)),
+        });
+        ok(true, `Pixel (${[x,y]}) passed. Got [${lastPixel}].`);
+      }
+    };
+
+    const screenSizeSq = Math.min(video.videoWidth, video.videoHeight);
+
+    info("Waiting for upper left quadrant to become " + upleft.name);
+    await verifyAround(screenSizeSq / 4, screenSizeSq / 4, upleft);
+
+    info("Waiting for upper right quadrant to become " + upright.name);
+    await verifyAround(screenSizeSq * 3 / 4, screenSizeSq / 4, upright);
+
+    info("Waiting for lower left quadrant to become " + downleft.name);
+    await verifyAround(screenSizeSq / 4, screenSizeSq * 3 / 4, downleft);
+
+    info("Waiting for lower right quadrant to become " + downright.name);
+    await verifyAround(screenSizeSq * 3 / 4, screenSizeSq * 3 / 4, downright);
+  };
+
+  /**
+   * Run a test to verify that we can complete a start and stop media playback
+   * cycle for a screenshare MediaStream on a video HTMLMediaElement.
+   */
+  runTest(async function () {
+    await pushPrefs(
+      ["full-screen-api.enabled", true],
+      ["full-screen-api.allow-trusted-requests-only", false],
+      ["full-screen-api.transition-duration.enter", "0 0"],
+      ["full-screen-api.transition-duration.leave", "0 0"],
+    );
+
+    // Improve real estate for screenshots
+    const test = document.getElementById("test");
+    test.setAttribute("style", "height:0;margin:0;");
+    const display = document.getElementById("display");
+    display.setAttribute("style", "margin:0;");
+    const testVideo = createMediaElement('video', 'testVideo');
+    testVideo.removeAttribute("width");
+    testVideo.removeAttribute("height");
+    testVideo.setAttribute("style", "max-height:240px;");
+
+    const canvas = document.createElement("canvas");
+    canvas.width = canvas.height = 20;
+    document.getElementById("content").appendChild(canvas);
+    const draw = ([upleft, upright, downleft, downright]) => {
+      helper.drawColor(canvas, helper[upleft], {offsetX: 0, offsetY: 0});
+      helper.drawColor(canvas, helper[upright], {offsetX: 10, offsetY: 0});
+      helper.drawColor(canvas, helper[downleft], {offsetX: 0, offsetY: 10});
+      helper.drawColor(canvas, helper[downright], {offsetX: 10, offsetY: 10});
+    };
+    const helper = new CaptureStreamTestHelper2D(1, 1);
+
+    const doVerify = async (stream, [upleft, upright, downleft, downright]) => {
+      // Reset from potential earlier verification runs.
+      testVideo.srcObject = null;
+      const playback = new MediaStreamPlayback(testVideo, stream);
+      playback.startMedia();
+      await playback.verifyPlaying();
+      const settings = stream.getTracks()[0].getSettings();
+      is(settings.width, testVideo.videoWidth,
+         "Width setting should match video width");
+      is(settings.height, testVideo.videoHeight,
+         "Height setting should match video height");
+      await SpecialPowers.wrap(canvas).requestFullscreen();
+      try {
+        await verifyScreenshare(testVideo, helper, helper[upleft], helper[upright],
+                                helper[downleft], helper[downright]);
+      } finally {
+        await playback.stopTracksForStreamInMediaPlayback();
+        await SpecialPowers.wrap(document).exitFullscreen();
+        // We wait a bit extra here to make sure we have completely left
+        // fullscreen when the --screenshot-on-fail screenshot is captured.
+        await wait(300);
+      }
+    };
+
+    info("Testing screenshare without constraints");
+    SpecialPowers.wrap(document).notifyUserGestureActivation();
+    let stream = await getUserMedia({video: {mediaSource: "screen"}});
+    let settings = stream.getTracks()[0].getSettings();
+    ok(settings.width <= 8192,
+       `Width setting ${settings.width} should be set after gUM (or 0 per bug 1453247)`);
+    ok(settings.height <= 8192,
+       `Height setting ${settings.height} should be set after gUM (or 0 per bug 1453247)`);
+    let colors = ["red", "blue", "green", "white"];
+    draw(colors);
+    await doVerify(stream, colors);
+    const screenWidth = testVideo.videoWidth;
+    const screenHeight = testVideo.videoHeight;
+
+    info("Testing screenshare with size and framerate constraints");
+    SpecialPowers.wrap(document).notifyUserGestureActivation();
+    for (const track of stream.getTracks()) {
+      track.stop();
+    }
+    stream = await getUserMedia({
+      video: {
+        mediaSource: 'screen',
+        width: {
+          min: '10',
+          max: '100'
+        },
+        height: {
+          min: '10',
+          max: '100'
+        },
+        frameRate: {
+          min: '10',
+          max: '15'
+        },
+      },
+    });
+    settings = stream.getTracks()[0].getSettings();
+    ok(settings.width == 0 || (settings.width >= 10 && settings.width <= 100),
+       `Width setting ${settings.width} should be correct after gUM (or 0 per bug 1453247)`);
+    ok(settings.height == 0 || (settings.height >= 10 && settings.height <= 100),
+       `Height setting ${settings.height} should be correct after gUM (or 0 per bug 1453247)`);
+    colors = ["green", "red", "white", "blue"];
+    draw(colors);
+    const streamClone = stream.clone();
+    await doVerify(streamClone, colors);
+    settings = stream.getTracks()[0].getSettings();
+    ok(settings.width >= 10 && settings.width <= 100,
+       `Width setting ${settings.width} should be within constraints`);
+    ok(settings.height >= 10 && settings.height <= 100,
+       `Height setting ${settings.height} should be within constraints`);
+    is(settings.width, testVideo.videoWidth,
+       "Width setting should match video width");
+    is(settings.height, testVideo.videoHeight,
+       "Height setting should match video height");
+    let expectedHeight = (screenHeight * settings.width) / screenWidth;
+    ok(Math.abs(expectedHeight - settings.height) <= 1,
+       "Aspect ratio after constrained gUM should be close enough");
+
+    info("Testing modifying screenshare with applyConstraints");
+    testVideo.srcObject = stream;
+    testVideo.play();
+    await new Promise(r => testVideo.onloadeddata = r);
+    const resize = haveEvent(
+      testVideo, "resize", wait(5000, new Error("Timeout waiting for resize")));
+    await stream.getVideoTracks()[0].applyConstraints({
+      mediaSource: 'screen',
+      width: 200,
+      height: 200,
+      frameRate: {
+        min: '5',
+        max: '10'
+      }
+    });
+    // getSettings() should report correct size as soon as applyConstraints()
+    // resolves - bug 1453259. Until fixed, check that we at least report
+    // something sane.
+    const newSettings = stream.getTracks()[0].getSettings();
+    ok(newSettings.width > settings.width && newSettings.width < screenWidth,
+       `Width setting ${newSettings.width} should have increased after applyConstraints`);
+    ok(newSettings.height > settings.height && newSettings.height < screenHeight,
+       `Height setting ${newSettings.height} should have increased after applyConstraints`);
+    await resize;
+    settings = stream.getTracks()[0].getSettings();
+    ok(settings.width > 100 && settings.width < screenWidth,
+       `Width setting ${settings.width} should have increased after first frame after applyConstraints`);
+    ok(settings.height > 100 && settings.height < screenHeight,
+       `Height setting ${settings.height} should have increased after first frame after applyConstraints`);
+    is(settings.width, testVideo.videoWidth,
+       "Width setting should match video width");
+    is(settings.height, testVideo.videoHeight,
+       "Height setting should match video height");
+    expectedHeight = (screenHeight * settings.width) / screenWidth;
+    ok(Math.abs(expectedHeight - settings.height) <= 1,
+       "Aspect ratio after applying constraints should be close enough");
+    colors = ["white", "green", "blue", "red"];
+    draw(colors);
+    await doVerify(stream, colors);
+    for (const track of [...stream.getTracks(), ...streamClone.getTracks()]) {
+      track.stop();
+    }
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicTabshare.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicTabshare.html
new file mode 100644
index 0000000000..635cf387d4
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicTabshare.html
@@ -0,0 +1,67 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "getUserMedia Basic Tabshare Test",
+    bug: "1193075"
+  });
+  /**
+   * Run a test to verify that we can complete a start and stop media playback
+   * cycle for a tabshare MediaStream on a video HTMLMediaElement.
+   *
+   * Additionally, exercise applyConstraints code for tabshare viewport offset.
+   */
+  runTest(function () {
+    var testVideo = createMediaElement('video', 'testVideo');
+
+    return Promise.resolve()
+      .then(() => pushPrefs(["media.getusermedia.browser.enabled", true]))
+      .then(() => {
+        SpecialPowers.wrap(document).notifyUserGestureActivation();
+        return getUserMedia({
+          video: { mediaSource: "browser",
+                   scrollWithPage: true },
+        });
+      })
+      .then(stream => {
+        var playback = new MediaStreamPlayback(testVideo, stream);
+        return playback.playMedia(false);
+      })
+      .then(() => getUserMedia({
+        video: {
+          mediaSource: "browser",
+          viewportOffsetX: 0,
+          viewportOffsetY: 0,
+          viewportWidth: 100,
+          viewportHeight: 100
+        },
+      }))
+      .then(stream => {
+        var playback = new MediaStreamPlayback(testVideo, stream);
+        playback.startMedia(false);
+        return playback.verifyPlaying()
+          .then(() => Promise.all([
+            () => testVideo.srcObject.getVideoTracks()[0].applyConstraints({
+              mediaSource: "browser",
+              viewportOffsetX: 10,
+              viewportOffsetY: 50,
+              viewportWidth: 90,
+              viewportHeight: 50
+            }),
+            () => listenUntil(testVideo, "resize", () => true)
+          ]))
+          .then(() => playback.verifyPlaying()) // still playing
+          .then(() => playback.stopTracksForStreamInMediaPlayback())
+          .then(() => playback.detachFromMediaElement());
+      });
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicVideo.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicVideo.html
new file mode 100644
index 0000000000..786d9f2e4b
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicVideo.html
@@ -0,0 +1,30 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "getUserMedia Basic Video Test",
+    bug: "781534"
+  });
+  /**
+   * Run a test to verify that we can complete a start and stop media playback
+   * cycle for an video MediaStream on a video HTMLMediaElement.
+   */
+  runTest(function () {
+    var testVideo = createMediaElement('video', 'testVideo');
+    var constraints = {video: true};
+
+    return getUserMedia(constraints).then(stream => {
+      var playback = new MediaStreamPlayback(testVideo, stream);
+      return playback.playMedia(false);
+    });
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicVideoAudio.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicVideoAudio.html
new file mode 100644
index 0000000000..5218bf7301
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicVideoAudio.html
@@ -0,0 +1,30 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "getUserMedia Basic Video & Audio Test",
+    bug: "781534"
+  });
+  /**
+   * Run a test to verify that we can complete a start and stop media playback
+   * cycle for a video and audio MediaStream on a video HTMLMediaElement.
+   */
+  runTest(function () {
+    var testVideoAudio = createMediaElement('video', 'testVideoAudio');
+    var constraints = {video: true, audio: true};
+
+    return getUserMedia(constraints).then(stream => {
+      var playback = new MediaStreamPlayback(testVideoAudio, stream);
+      return playback.playMedia(false);
+    });
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicVideo_playAfterLoadedmetadata.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicVideo_playAfterLoadedmetadata.html
new file mode 100644
index 0000000000..fbab1b4357
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicVideo_playAfterLoadedmetadata.html
@@ -0,0 +1,42 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "getUserMedia Basic Video shall receive 'loadedmetadata' without play()ing",
+    bug: "1149494"
+  });
+  /**
+   * Run a test to verify that we will always get 'loadedmetadata' from a video
+   * HTMLMediaElement playing a gUM MediaStream.
+   */
+  runTest(() => {
+    var testVideo = createMediaElement('video', 'testVideo');
+    var constraints = {video: true};
+
+    return getUserMedia(constraints).then(stream => {
+      var playback = new MediaStreamPlayback(testVideo, stream);
+      var video = playback.mediaElement;
+
+      video.srcObject = stream;
+      return new Promise(resolve => {
+        ok(playback.mediaElement.paused,
+           "Media element should be paused before play()ing");
+        video.addEventListener('loadedmetadata', function() {
+          ok(video.videoWidth > 0, "Expected nonzero video width");
+          ok(video.videoHeight > 0, "Expected nonzero video width");
+          resolve();
+        });
+      })
+      .then(() => stream.getTracks().forEach(t => t.stop()));
+    });
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicWindowshare.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicWindowshare.html
new file mode 100644
index 0000000000..7b27944bdc
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_basicWindowshare.html
@@ -0,0 +1,39 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "getUserMedia Basic Windowshare Test",
+    bug: "1038926"
+  });
+  /**
+   * Run a test to verify that we can complete a start and stop media playback
+   * cycle for an screenshare MediaStream on a video HTMLMediaElement.
+   */
+  runTest(async function () {
+    const testVideo = createMediaElement('video', 'testVideo');
+    const constraints = {
+      video: { mediaSource: "window" },
+    };
+
+    try {
+      await getUserMedia(constraints);
+      ok(false, "Should require user gesture");
+    } catch (e) {
+      is(e.name, "InvalidStateError");
+    }
+
+    SpecialPowers.wrap(document).notifyUserGestureActivation();
+    const stream = await getUserMedia(constraints);
+    const playback = new MediaStreamPlayback(testVideo, stream);
+    return playback.playMedia(false);
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_bug1223696.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_bug1223696.html
new file mode 100644
index 0000000000..6af7b69d70
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_bug1223696.html
@@ -0,0 +1,54 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  "use strict";
+
+  createHTML({
+    title: "Testing that removeTrack+addTrack of video tracks still render the correct track in a media element",
+    bug: "1223696",
+    visible: true
+  });
+
+  runTest(async function() {
+    const stream = await getUserMedia({audio:true, video: true});
+    info("Test addTrack()ing a video track to an audio-only gUM stream");
+
+    const video = createMediaElement("video", "test_video_track");
+    video.srcObject = stream;
+    video.play();
+
+    await haveEvent(video, "loadeddata", wait(5000, new Error("Timeout")));
+    info("loadeddata");
+
+    const removedTrack = stream.getVideoTracks()[0];
+    stream.removeTrack(removedTrack);
+
+    const h = new CaptureStreamTestHelper2D();
+    const emitter = new VideoFrameEmitter(h.grey, h.grey);
+    emitter.start();
+
+    stream.addTrack(emitter.stream().getVideoTracks()[0]);
+
+    checkMediaStreamContains(stream, [stream.getAudioTracks()[0],
+                                      emitter.stream().getVideoTracks()[0]]);
+
+    await h.pixelMustBecome(video, h.grey, {
+      threshold: 5,
+      infoString: "The canvas track should be rendered by the media element",
+    });
+
+    emitter.stop();
+    for (const t of [removedTrack, ...stream.getAudioTracks()]) {
+      t.stop();
+    }
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_callbacks.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_callbacks.html
new file mode 100644
index 0000000000..14c6cc7e7f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_callbacks.html
@@ -0,0 +1,35 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "navigator.mozGetUserMedia Callback Test",
+    bug: "1119593"
+  });
+  /**
+   * Check that the old fashioned callback-based function works.
+   */
+  runTest(function () {
+    var testAudio = createMediaElement('audio', 'testAudio');
+    var constraints = {audio: true};
+
+    SimpleTest.waitForExplicitFinish();
+    return new Promise(resolve =>
+      navigator.mozGetUserMedia(constraints, stream => {
+        checkMediaStreamTracks(constraints, stream);
+
+        var playback = new MediaStreamPlayback(testAudio, stream);
+        return playback.playMedia(false)
+          .then(() => resolve(), generateErrorCallback());
+      }, generateErrorCallback())
+    );
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_constraints.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_constraints.html
new file mode 100644
index 0000000000..d6439ce9d6
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_constraints.html
@@ -0,0 +1,166 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script src="mediaStreamPlayback.js"></script>
+  <script src="constraints.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({ title: "Test getUserMedia constraints", bug: "882145" });
+/**
+  Tests covering gUM constraints API for audio, video and fake video. Exercise
+  successful parsing code and ensure that unknown required constraints and
+  overconstraining cases produce appropriate errors.
+*/
+var tests = [
+  // Each test here tests a different constraint or codepath.
+  { message: "unknown required constraint on video ignored",
+    constraints: { video: { somethingUnknown: { exact: 0 } } },
+    error: null },
+  { message: "unknown required constraint on audio ignored",
+    constraints: { audio: { somethingUnknown: { exact: 0 } } },
+    error: null },
+  { message: "audio overconstrained by facingMode ignored",
+    constraints: { audio: { facingMode: { exact: 'left' } } },
+    error: null },
+  { message: "full screensharing requires permission",
+    constraints: { video: { mediaSource: 'screen' } },
+    error: "NotAllowedError" },
+  { message: "application screensharing no longer exists",
+    constraints: { video: { mediaSource: 'application' } },
+    error: "OverconstrainedError" },
+  { message: "window screensharing requires permission",
+    constraints: { video: { mediaSource: 'window' } },
+    error: "NotAllowedError" },
+  { message: "browser screensharing requires permission",
+    constraints: { video: { mediaSource: 'browser' } },
+    error: "NotAllowedError" },
+  { message: "unknown mediaSource in video fails",
+    constraints: { video: { mediaSource: 'uncle' } },
+    error: "OverconstrainedError",
+    constraint: "mediaSource" },
+  { message: "unknown mediaSource in audio fails",
+    constraints: { audio: { mediaSource: 'uncle' } },
+    error: "OverconstrainedError",
+    constraint: "mediaSource" },
+  { message: "emtpy constraint fails",
+    constraints: { },
+    error: "TypeError" },
+  { message: "Triggering mock failure in default video device fails",
+    constraints: { video: { deviceId: 'bad device' } },
+    error: "NotReadableError" },
+  { message: "Triggering mock failure in default audio device fails",
+    constraints: { audio: { deviceId: 'bad device' } },
+    error: "NotReadableError" },
+  { message: "Success-path: optional video facingMode + audio ignoring facingMode",
+    constraints: { audio: { mediaSource: 'microphone',
+                            facingMode: 'left',
+                            foo: 0,
+                            advanced: [{ facingMode: 'environment' },
+                                       { facingMode: 'user' },
+                                       { bar: 0 }] },
+                   video: { mediaSource: 'camera',
+                            foo: 0,
+                            advanced: [{ facingMode: 'environment' },
+                                       { facingMode: ['user'] },
+                                       { facingMode: ['left', 'right', 'user'] },
+                                       { bar: 0 }] } },
+    error: null },
+  { message: "legacy facingMode ignored",
+    constraints: { video: { mandatory: { facingMode: 'left' } } },
+    error: null },
+];
+
+var mustSupport = [
+  'width', 'height', 'frameRate', 'facingMode', 'deviceId', 'groupId',
+  'echoCancellation', 'noiseSuppression', 'autoGainControl', 'channelCount',
+
+  // Yet to add:
+  //  'aspectRatio', 'volume', 'sampleRate', 'sampleSize', 'latency'
+
+  // http://fluffy.github.io/w3c-screen-share/#screen-based-video-constraints
+  // OBE by http://w3c.github.io/mediacapture-screen-share
+  'mediaSource',
+
+  // Experimental https://bugzilla.mozilla.org/show_bug.cgi?id=1131568#c3
+  'browserWindow', 'scrollWithPage',
+  'viewportOffsetX', 'viewportOffsetY', 'viewportWidth', 'viewportHeight',
+];
+
+var mustFailWith = (msg, reason, constraint, f) =>
+  f().then(() => ok(false, msg + " must fail"), e => {
+    is(e.name, reason, msg + " must fail: " + e.message);
+    if (constraint !== undefined) {
+      is(e.constraint, constraint, msg + " must fail w/correct constraint.");
+    }
+  });
+
+/**
+ * Starts the test run by running through each constraint
+ * test by verifying that the right resolution and rejection is fired.
+ */
+
+runTest(() => pushPrefs(
+    // This test expects fake devices, particularly for the 'triggering mock
+    // failure *' steps. So explicitly disable loopback and setup fakes
+    ['media.audio_loopback_dev', ''],
+    ['media.video_loopback_dev', ''],
+    ['media.navigator.streams.fake', true]
+  )
+  .then(() => {
+    // Check supported constraints first.
+    var dict = navigator.mediaDevices.getSupportedConstraints();
+    var supported = Object.keys(dict);
+
+    mustSupport.forEach(key => ok(supported.includes(key) && dict[key],
+                                  "Supports " + key));
+
+    var unexpected = supported.filter(key => !mustSupport.includes(key));
+    is(unexpected.length, 0,
+       "Unanticipated support (please update test): " + unexpected);
+  })
+  .then(() => pushPrefs(["media.getusermedia.browser.enabled", false],
+                        ["media.getusermedia.screensharing.enabled", false]))
+  .then(() => tests.reduce((p, test) => p.then(
+    () => {
+      SpecialPowers.wrap(document).notifyUserGestureActivation();
+      return getUserMedia(test.constraints);
+    })
+    .then(stream => {
+      is(null, test.error, test.message);
+      stream.getTracks().forEach(t => t.stop());
+    }, e => {
+      is(e.name, test.error, test.message + ": " + e.message);
+      if (test.constraint) {
+        is(e.constraint, test.constraint,
+           test.message + " w/correct constraint.");
+      }
+    }), Promise.resolve()))
+  .then(() => getUserMedia({video: true, audio: true}))
+  .then(stream => stream.getVideoTracks()[0].applyConstraints({ width: 320 })
+    .then(() => stream.getAudioTracks()[0].applyConstraints({ }))
+    .then(() => {
+      stream.getTracks().forEach(track => track.stop());
+      ok(true, "applyConstraints code exercised");
+    }))
+  // TODO: Test outcome once fake devices support constraints (Bug 1088621)
+  .then(() => mustFailWith("applyConstraints fails on non-Gum tracks",
+                           "OverconstrainedError", "",
+                           () => (new AudioContext())
+                               .createMediaStreamDestination().stream
+                               .getAudioTracks()[0].applyConstraints()))
+  .then(() => mustFailWith(
+      "getUserMedia with unsatisfied required constraint",
+      "OverconstrainedError", "deviceId",
+      () => getUserMedia({ audio: true,
+                           video: { deviceId: { exact: "unheardof" } } })))
+  .then(() => mustFailWith(
+      "getUserMedia with unsatisfied required constraint array",
+      "OverconstrainedError", "deviceId",
+      () => getUserMedia({ audio: true,
+                           video: { deviceId: { exact: ["a", "b"] } } }))));
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_cubebDisabled.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_cubebDisabled.html
new file mode 100644
index 0000000000..54142aeb77
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_cubebDisabled.html
@@ -0,0 +1,42 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "getUserMedia with Cubeb Disabled Test",
+    bug: "1443525"
+  });
+  /**
+   * Run a test to verify we fail gracefully if we cannot fetch a cubeb context
+   * during a gUM call.
+   */
+  runTest(async function () {
+    info("Get user media with cubeb disabled starting");
+    // Push prefs to ensure no cubeb context and no fake streams.
+    await pushPrefs(["media.cubeb.force_null_context", true],
+                    ["media.navigator.permission.device", false],
+                    ["media.navigator.streams.fake", false]);
+
+    // Request audio only, to avoid cams
+    let constraints = {audio: true, video: false};
+    let stream;
+    try {
+      stream = await getUserMedia(constraints);
+    } catch (e) {
+      // We've got no audio backend, so we expect gUM to fail.
+      ok(e.name == "NotFoundError", "Expected NotFoundError due to no audio tracks!");
+      return;
+    }
+    // If we're not on android we should not have gotten a stream without a cubeb context!
+    ok(false, "getUserMedia not expected to succeed when cubeb is disabled, but it did!");
+  });
+
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_cubebDisabledFakeStreams.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_cubebDisabledFakeStreams.html
new file mode 100644
index 0000000000..f8150cc4c1
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_cubebDisabledFakeStreams.html
@@ -0,0 +1,43 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "getUserMedia fake stream with Cubeb Disabled Test",
+    bug: "1443525"
+  });
+  /**
+   * Run a test to verify we can still return a fake stream even if we cannot
+   * get a cubeb context. See also Bug 1434477
+   */
+  runTest(async function () {
+    info("Get user media with cubeb disabled and fake tracks starting");
+    // Push prefs to ensure no cubeb context and fake streams
+    await pushPrefs(["media.cubeb.force_null_context", true],
+                    ["media.navigator.streams.fake", true],
+                    ['media.audio_loopback_dev', '']);
+    let testAudio = createMediaElement('audio', 'testAudio');
+    // Request audio only, to avoid cams
+    let constraints = {audio: true, video: false};
+    let stream;
+    try {
+      stream = await getUserMedia(constraints);
+    } catch (e) {
+      // We've got no audio backend, so we expect gUM to fail
+      ok(false, `Did not expect to fail, but got ${e}`);
+      return;
+    }
+    ok(stream, "getUserMedia should get a stream!");
+    let playback = new MediaStreamPlayback(testAudio, stream);
+    return playback.playMedia(false);
+  });
+
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_getTrackById.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_getTrackById.html
new file mode 100644
index 0000000000..161bf631e3
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_getTrackById.html
@@ -0,0 +1,50 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "Basic getTrackById test of gUM stream",
+    bug: "1208390",
+  });
+
+  runTest(() => {
+    var constraints = {audio: true, video: true};
+    return getUserMedia(constraints).then(stream => {
+      is(stream.getTrackById(""), null,
+         "getTrackById of non-matching string should return null");
+
+      let audioTrack = stream.getAudioTracks()[0];
+      is(stream.getTrackById(audioTrack.id), audioTrack,
+         "getTrackById with matching id should return the track");
+
+      let videoTrack = stream.getVideoTracks()[0];
+      is(stream.getTrackById(videoTrack.id), videoTrack,
+         "getTrackById with matching id should return the track");
+
+      stream.removeTrack(audioTrack);
+      is(stream.getTrackById(audioTrack.id), null,
+         "getTrackById with id of removed track should return null");
+
+      let newStream = new MediaStream();
+      is(newStream.getTrackById(videoTrack.id), null,
+         "getTrackById with id of track in other stream should return null");
+
+      newStream.addTrack(audioTrack);
+      is(newStream.getTrackById(audioTrack.id), audioTrack,
+         "getTrackByid with matching id should return the track");
+
+      newStream.addTrack(videoTrack);
+      is(newStream.getTrackById(videoTrack.id), videoTrack,
+         "getTrackByid with matching id should return the track");
+      [audioTrack, videoTrack].forEach(t => t.stop());
+    });
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_gumWithinGum.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_gumWithinGum.html
new file mode 100644
index 0000000000..86a7aa5606
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_gumWithinGum.html
@@ -0,0 +1,38 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({title: "getUserMedia within getUserMedia", bug: "822109" });
+  /**
+   * Run a test that we can complete a playback cycle for a video,
+   * then upon completion, do a playback cycle with audio, such that
+   * the audio gum call happens within the video gum call.
+   */
+  runTest(function () {
+    return getUserMedia({video: true})
+      .then(videoStream => {
+        var testVideo = createMediaElement('video', 'testVideo');
+        var videoPlayback = new MediaStreamPlayback(testVideo,
+                                                    videoStream);
+
+        return videoPlayback.playMediaWithoutStoppingTracks(false)
+          .then(() => getUserMedia({audio: true}))
+          .then(audioStream => {
+            var testAudio = createMediaElement('audio', 'testAudio');
+            var audioPlayback = new MediaStreamPlayback(testAudio,
+                                                        audioStream);
+
+            return audioPlayback.playMedia(false);
+          })
+          .then(() => videoStream.getTracks().forEach(t => t.stop()));
+      });
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_loadedmetadata.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_loadedmetadata.html
new file mode 100644
index 0000000000..d6efac4650
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_loadedmetadata.html
@@ -0,0 +1,39 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "getUserMedia in media element should have video dimensions on loadedmetadata",
+    bug: "1240478"
+  });
+  /**
+   * Tests that assigning a stream to a media element results in the
+   * "loadedmetadata" event without having to play() the media element.
+   *
+   * Also makes sure that the video size has been set on "loadedmetadata".
+   */
+  runTest(function () {
+    var v = document.createElement("video");
+    document.body.appendChild(v);
+    v.preload = "metadata";
+
+    var constraints = {video: true, audio: true};
+    return getUserMedia(constraints).then(stream => new Promise(resolve => {
+      v.srcObject = stream;
+      v.onloadedmetadata = () => {
+        isnot(v.videoWidth, 0, "videoWidth shall be set on 'loadedmetadata'");
+        isnot(v.videoHeight, 0, "videoHeight shall be set on 'loadedmetadata'");
+        resolve();
+      };
+    })
+    .then(() => stream.getTracks().forEach(t => t.stop())));
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaElementCapture_audio.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaElementCapture_audio.html
new file mode 100644
index 0000000000..3b9e00896c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaElementCapture_audio.html
@@ -0,0 +1,116 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script>
+
+createHTML({
+  bug: "1259788",
+  title: "Test CaptureStream audio content on HTMLMediaElement playing a gUM MediaStream",
+  visible: true
+});
+
+var audioContext;
+var gUMAudioElement;
+var analyser;
+runTest(() => getUserMedia({audio: { echoCancellation: false }})
+  .then(stream => {
+    gUMAudioElement = createMediaElement("audio", "gUMAudio");
+    gUMAudioElement.srcObject = stream;
+
+    audioContext = new AudioContext();
+    info("Capturing");
+
+    analyser = new AudioStreamAnalyser(audioContext,
+                                       gUMAudioElement.mozCaptureStream());
+    analyser.enableDebugCanvas();
+    return analyser.waitForAnalysisSuccess(array =>
+      array[analyser.binIndexForFrequency(50)]              < 50 &&
+      array[analyser.binIndexForFrequency(TEST_AUDIO_FREQ)] > 200 &&
+      array[analyser.binIndexForFrequency(2500)]            < 50);
+  })
+  .then(() => {
+    info("Audio flowing. Pausing.");
+    gUMAudioElement.pause();
+
+    return analyser.waitForAnalysisSuccess(array =>
+      array[analyser.binIndexForFrequency(50)]              < 50 &&
+      array[analyser.binIndexForFrequency(TEST_AUDIO_FREQ)] < 50 &&
+      array[analyser.binIndexForFrequency(2500)]            < 50);
+  })
+  .then(() => {
+    info("Audio stopped flowing. Playing.");
+    gUMAudioElement.play();
+
+    return analyser.waitForAnalysisSuccess(array =>
+      array[analyser.binIndexForFrequency(50)]              < 50 &&
+      array[analyser.binIndexForFrequency(TEST_AUDIO_FREQ)] > 200 &&
+      array[analyser.binIndexForFrequency(2500)]            < 50);
+  })
+  .then(() => {
+    info("Audio flowing. Removing source.");
+    var stream = gUMAudioElement.srcObject;
+    gUMAudioElement.srcObject = null;
+
+    return analyser.waitForAnalysisSuccess(array =>
+      array[analyser.binIndexForFrequency(50)]              < 50 &&
+      array[analyser.binIndexForFrequency(TEST_AUDIO_FREQ)] < 50 &&
+      array[analyser.binIndexForFrequency(2500)]            < 50)
+        .then(() => stream);
+  })
+  .then(stream => {
+    info("Audio stopped flowing. Setting source.");
+    gUMAudioElement.srcObject = stream;
+
+    return analyser.waitForAnalysisSuccess(array =>
+      array[analyser.binIndexForFrequency(50)]              < 50 &&
+      array[analyser.binIndexForFrequency(TEST_AUDIO_FREQ)] > 200 &&
+      array[analyser.binIndexForFrequency(2500)]            < 50);
+  })
+  .then(() => {
+    info("Audio flowing from new source. Adding a track.");
+    let oscillator = audioContext.createOscillator();
+    oscillator.type = 'sine';
+    oscillator.frequency.value = 2000;
+    oscillator.start();
+
+    let oscOut = audioContext.createMediaStreamDestination();
+    oscillator.connect(oscOut);
+
+    gUMAudioElement.srcObject.addTrack(oscOut.stream.getTracks()[0]);
+
+    return analyser.waitForAnalysisSuccess(array =>
+      array[analyser.binIndexForFrequency(50)]              < 50 &&
+      array[analyser.binIndexForFrequency(TEST_AUDIO_FREQ)] > 200 &&
+      array[analyser.binIndexForFrequency(1500)]            < 50 &&
+      array[analyser.binIndexForFrequency(2000)]            > 200 &&
+      array[analyser.binIndexForFrequency(2500)]            < 50);
+  })
+  .then(() => {
+    info("Audio flowing from new track. Removing a track.");
+
+    const gUMTrack = gUMAudioElement.srcObject.getTracks()[0];
+    gUMAudioElement.srcObject.removeTrack(gUMTrack);
+
+    is(gUMAudioElement.srcObject.getTracks().length, 1,
+       "A track should have been removed");
+
+    return analyser.waitForAnalysisSuccess(array =>
+      array[analyser.binIndexForFrequency(50)]              < 50 &&
+      array[analyser.binIndexForFrequency(TEST_AUDIO_FREQ)] < 50 &&
+      array[analyser.binIndexForFrequency(1500)]            < 50 &&
+      array[analyser.binIndexForFrequency(2000)]            > 200 &&
+      array[analyser.binIndexForFrequency(2500)]            < 50)
+        .then(() => [gUMTrack, ...gUMAudioElement.srcObject.getTracks()]
+            .forEach(t => t.stop()));
+  })
+  .then(() => ok(true, "Test passed."))
+  .catch(e => ok(false, "Test failed: " + e + (e.stack ? "\n" + e.stack : ""))));
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaElementCapture_tracks.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaElementCapture_tracks.html
new file mode 100644
index 0000000000..a747e75de9
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaElementCapture_tracks.html
@@ -0,0 +1,179 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script>
+
+createHTML({
+  bug: "1259788",
+  title: "Test CaptureStream track output on HTMLMediaElement playing a gUM MediaStream",
+  visible: true
+});
+
+let audioElement;
+let audioCaptureStream;
+let videoElement;
+let videoCaptureStream;
+let untilEndedElement;
+let streamUntilEnded;
+const tracks = [];
+runTest(async () => {
+  try {
+    let stream = await getUserMedia({audio: true, video: true});
+    // We need to test with multiple tracks. We add an extra of each kind.
+    for (const track of stream.getTracks()) {
+      stream.addTrack(track.clone());
+    }
+
+    audioElement = createMediaElement("audio", "gUMAudio");
+    audioElement.srcObject = stream;
+
+    await haveEvent(audioElement, "loadedmetadata", wait(50000, new Error("Timeout")));
+
+    info("Capturing audio element (loadedmetadata -> captureStream)");
+    audioCaptureStream = audioElement.mozCaptureStream();
+
+    is(audioCaptureStream.getAudioTracks().length, 2,
+      "audio element should capture two audio tracks");
+    is(audioCaptureStream.getVideoTracks().length, 0,
+      "audio element should not capture any video tracks");
+
+    await haveNoEvent(audioCaptureStream, "addtrack");
+
+    videoElement = createMediaElement("video", "gUMVideo");
+
+    info("Capturing video element (captureStream -> loadedmetadata)");
+    videoCaptureStream = videoElement.mozCaptureStream();
+    videoElement.srcObject = audioElement.srcObject.clone();
+
+    is(videoCaptureStream.getTracks().length, 0,
+      "video element should have no tracks before metadata known");
+
+    await haveEventsButNoMore(
+      videoCaptureStream, "addtrack", 3, wait(50000, new Error("No event")));
+
+    is(videoCaptureStream.getAudioTracks().length, 2,
+      "video element should capture two audio tracks");
+    is(videoCaptureStream.getVideoTracks().length, 1,
+      "video element should capture one video track at most");
+
+    info("Testing dynamically adding audio track to audio element");
+    audioElement.srcObject.addTrack(
+      audioElement.srcObject.getAudioTracks()[0].clone());
+    await haveEventsButNoMore(
+      audioCaptureStream, "addtrack", 1, wait(50000, new Error("No event")));
+
+    is(audioCaptureStream.getAudioTracks().length, 3,
+      "Audio element should have three audio tracks captured.");
+
+    info("Testing dynamically adding video track to audio element");
+    audioElement.srcObject.addTrack(
+      audioElement.srcObject.getVideoTracks()[0].clone());
+    await haveNoEvent(audioCaptureStream, "addtrack");
+
+    is(audioCaptureStream.getVideoTracks().length, 0,
+      "Audio element should have no video tracks captured.");
+
+    info("Testing dynamically adding audio track to video element");
+    videoElement.srcObject.addTrack(
+      videoElement.srcObject.getAudioTracks()[0].clone());
+    await haveEventsButNoMore(
+      videoCaptureStream, "addtrack", 1, wait(50000, new Error("Timeout")));
+
+    is(videoCaptureStream.getAudioTracks().length, 3,
+      "Captured video stream should have three audio tracks captured.");
+
+    info("Testing dynamically adding video track to video element");
+    videoElement.srcObject.addTrack(
+      videoElement.srcObject.getVideoTracks()[0].clone());
+    await haveNoEvent(videoCaptureStream, "addtrack");
+
+    is(videoCaptureStream.getVideoTracks().length, 1,
+      "Captured video stream should have at most one video tracks captured.");
+
+    info("Testing track removal.");
+    tracks.push(...videoElement.srcObject.getTracks());
+    for (const track of videoElement.srcObject.getVideoTracks().reverse()) {
+      videoElement.srcObject.removeTrack(track);
+    }
+    is(videoCaptureStream.getVideoTracks().length, 1,
+      "Captured video should have still have one video track.");
+
+    await haveEvent(videoCaptureStream.getVideoTracks()[0], "ended",
+      wait(50000, new Error("Timeout")));
+    await haveEvent(videoCaptureStream, "removetrack",
+      wait(50000, new Error("Timeout")));
+
+    is(videoCaptureStream.getVideoTracks().length, 0,
+      "Captured video stream should have no video tracks after removal.");
+
+
+    info("Testing source reset.");
+    stream = await getUserMedia({audio: true, video: true});
+    videoElement.srcObject = stream;
+    for (const track of videoCaptureStream.getTracks()) {
+      await Promise.race(videoCaptureStream.getTracks().map(
+        t => haveEvent(t, "ended", wait(50000, new Error("Timeout"))))
+      );
+      await haveEvent(videoCaptureStream, "removetrack", wait(50000, new Error("Timeout")));
+    }
+    await haveEventsButNoMore(
+      videoCaptureStream, "addtrack", 2, wait(50000, new Error("Timeout")));
+    is(videoCaptureStream.getAudioTracks().length, 1,
+      "Captured video stream should have one audio track");
+
+    is(videoCaptureStream.getVideoTracks().length, 1,
+      "Captured video stream should have one video track");
+
+    info("Testing CaptureStreamUntilEnded");
+    untilEndedElement =
+      createMediaElement("video", "gUMVideoUntilEnded");
+    untilEndedElement.srcObject = audioElement.srcObject;
+
+    await haveEvent(untilEndedElement, "loadedmetadata",
+      wait(50000, new Error("Timeout")));
+
+    streamUntilEnded = untilEndedElement.mozCaptureStreamUntilEnded();
+
+    is(streamUntilEnded.getAudioTracks().length, 3,
+      "video element should capture all 3 audio tracks until ended");
+    is(streamUntilEnded.getVideoTracks().length, 1,
+      "video element should capture only 1 video track until ended");
+
+    for (const track of untilEndedElement.srcObject.getTracks()) {
+      track.stop();
+    }
+
+    await haveEvent(untilEndedElement, "ended", wait(50000, new Error("Timeout")));
+    for (const track of streamUntilEnded.getTracks()) {
+      await Promise.race(streamUntilEnded.getTracks().map(
+        t => haveEvent(t, "ended", wait(50000, new Error("Timeout"))))
+      );
+      await haveEvent(streamUntilEnded, "removetrack", wait(50000, new Error("Timeout")));
+    }
+
+    info("Element and tracks ended. Ensuring that new tracks aren't created.");
+    untilEndedElement.srcObject = videoElement.srcObject;
+    await haveEventsButNoMore(
+      untilEndedElement, "loadedmetadata", 1, wait(50000, new Error("Timeout")));
+
+    is(streamUntilEnded.getTracks().length, 0, "Should have no tracks");
+  } catch(e) {
+    ok(false, "Test failed: " + e + (e && e.stack ? "\n" + e.stack : ""));
+  } finally {
+    if (videoElement) {
+      tracks.push(...videoElement.srcObject.getTracks());
+    }
+    for(const track of tracks) {
+      track.stop();
+    }
+  }
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaElementCapture_video.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaElementCapture_video.html
new file mode 100644
index 0000000000..d177e93bfb
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaElementCapture_video.html
@@ -0,0 +1,91 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script>
+
+createHTML({
+  bug: "1259788",
+  title: "Test CaptureStream video content on HTMLMediaElement playing a gUM MediaStream",
+  visible: true
+});
+
+var gUMVideoElement;
+var captureStreamElement;
+
+const pausedTimeout = 1000;
+let h;
+
+runTest(async () => {
+  try {
+    await pushPrefs(
+      // This test expects fake video devices, as it expects captured frames to
+      // shift over time, which is not currently provided by loopback devices
+      ['media.video_loopback_dev', ''],
+      ['media.navigator.streams.fake', true]);
+
+    let stream = await getUserMedia({video: true});
+    h = new VideoStreamHelper();
+    gUMVideoElement =
+      createMediaElement("video", "gUMVideo");
+    gUMVideoElement.srcObject = stream;
+    gUMVideoElement.play();
+
+    info("Capturing");
+    captureStreamElement =
+      createMediaElement("video", "captureStream");
+    captureStreamElement.srcObject = gUMVideoElement.mozCaptureStream();
+    captureStreamElement.play();
+
+    await h.checkVideoPlaying(captureStreamElement);
+
+    // Adding a dummy audio track to the stream will keep a consuming media
+    // element from ending.
+    // We could also solve it by repeatedly play()ing or autoplay, but then we
+    // wouldn't be sure the media element stopped rendering video because it
+    // went to the ended state or because there were no frames for the track.
+    let osc = createOscillatorStream(new AudioContext(), 1000);
+    captureStreamElement.srcObject.addTrack(osc.getTracks()[0]);
+
+    info("Video flowing. Pausing.");
+    gUMVideoElement.pause();
+    await h.checkVideoPaused(captureStreamElement, { time: pausedTimeout });
+
+    info("Video stopped flowing. Playing.");
+    gUMVideoElement.play();
+    await h.checkVideoPlaying(captureStreamElement);
+
+    info("Video flowing. Removing source.");
+    stream = gUMVideoElement.srcObject;
+    gUMVideoElement.srcObject = null;
+    await h.checkVideoPaused(captureStreamElement, { time: pausedTimeout });
+
+    info("Video stopped flowing. Setting source.");
+    gUMVideoElement.srcObject = stream;
+    await h.checkVideoPlaying(captureStreamElement);
+
+    info("Video flowing. Changing source by track manipulation. Remove first.");
+    let track = gUMVideoElement.srcObject.getTracks()[0];
+    gUMVideoElement.srcObject.removeTrack(track);
+    await h.checkVideoPaused(captureStreamElement, { time: pausedTimeout });
+
+    info("Video paused. Changing source by track manipulation. Add first.");
+    gUMVideoElement.srcObject.addTrack(track);
+    gUMVideoElement.play();
+    await h.checkVideoPlaying(captureStreamElement);
+
+    gUMVideoElement.srcObject.getTracks().forEach(t => t.stop());
+    ok(true, "Test passed.");
+  } catch (e) {
+    ok(false, "Test failed: " + e + (e.stack ? "\n" + e.stack : ""));
+  }
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaStreamClone.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaStreamClone.html
new file mode 100644
index 0000000000..029ce77dd0
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaStreamClone.html
@@ -0,0 +1,258 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+"use strict";
+
+createHTML({
+  title: "MediaStream.clone()",
+  bug: "1208371"
+});
+
+runTest(async () => {
+  await pushPrefs(
+    ["media.getusermedia.camera.stop_on_disable.enabled", true],
+    ["media.getusermedia.camera.stop_on_disable.delay_ms", 0],
+    ["media.getusermedia.microphone.stop_on_disable.enabled", true],
+    ["media.getusermedia.microphone.stop_on_disable.delay_ms", 0]);
+
+  let gUMStream = await getUserMedia({audio: true, video: true});
+  {
+    info("Test clone()ing an audio/video gUM stream");
+    let clone = gUMStream.clone();
+
+    checkMediaStreamCloneAgainstOriginal(clone, gUMStream);
+    checkMediaStreamTrackCloneAgainstOriginal(clone.getAudioTracks()[0],
+                                              gUMStream.getAudioTracks()[0]);
+    checkMediaStreamTrackCloneAgainstOriginal(clone.getVideoTracks()[0],
+                                              gUMStream.getVideoTracks()[0]);
+
+    isnot(clone.id.length, 0, "Stream clone should have an id string");
+    isnot(clone.getAudioTracks()[0].id.length, 0,
+          "Audio track clone should have an id string");
+    isnot(clone.getVideoTracks()[0].id.length, 0,
+          "Audio track clone should have an id string");
+
+    info("Playing from track clones");
+    let test = createMediaElement('video', 'testClonePlayback');
+    let playback = new MediaStreamPlayback(test, clone);
+    await playback.playMedia(false);
+  }
+
+  {
+    info("Test addTrack()ing a video track to a stream without affecting its clone");
+    let stream = new MediaStream(gUMStream.getVideoTracks());
+    let otherStream = await getUserMedia({video: true});
+    let track = stream.getTracks()[0];
+    let otherTrack = otherStream.getTracks()[0];
+
+    let streamClone = stream.clone();
+    let trackClone = streamClone.getTracks()[0];
+    checkMediaStreamContains(streamClone, [trackClone], "Initial clone");
+
+    stream.addTrack(otherTrack);
+    checkMediaStreamContains(stream, [track, otherTrack],
+                             "Added video to original");
+    checkMediaStreamContains(streamClone, [trackClone],
+                             "Clone not affected");
+
+    stream.removeTrack(track);
+    streamClone.addTrack(track);
+    checkMediaStreamContains(streamClone, [trackClone, track],
+                             "Added video to clone");
+    checkMediaStreamContains(stream, [otherTrack],
+                             "Original not affected");
+
+    // Not part of streamClone. Does not get stopped by the playback test.
+    otherTrack.stop();
+
+    let test = createMediaElement('video', 'testClonePlayback');
+    let playback = new MediaStreamPlayback(test, streamClone);
+    await playback.playMedia(false);
+  }
+
+  {
+    info("Test cloning a stream into inception");
+    let stream = gUMStream.clone()
+    let clone = stream;
+    let clones = Array(10).fill().map(() => clone = clone.clone());
+    let inceptionClone = clones.pop();
+    checkMediaStreamCloneAgainstOriginal(inceptionClone, stream);
+    stream.getTracks().forEach(t => (stream.removeTrack(t),
+                                     inceptionClone.addTrack(t)));
+    is(inceptionClone.getAudioTracks().length, 2,
+       "The inception clone should contain the original audio track and a track clone");
+    is(inceptionClone.getVideoTracks().length, 2,
+       "The inception clone should contain the original video track and a track clone");
+
+    let test = createMediaElement('video', 'testClonePlayback');
+    let playback = new MediaStreamPlayback(test, inceptionClone);
+    await playback.playMedia(false);
+    clones.forEach(c => c.getTracks().forEach(t => t.stop()));
+    stream.getTracks().forEach(t => t.stop());
+  }
+
+  {
+    info("Test adding tracks from many stream clones to the original stream");
+    let stream = gUMStream.clone();
+
+    const LOOPS = 3;
+    for (let i = 0; i < LOOPS; i++) {
+      stream.clone().getTracks().forEach(t => stream.addTrack(t));
+    }
+    is(stream.getAudioTracks().length, Math.pow(2, LOOPS),
+       "The original track should contain the original audio track and all the audio clones");
+    is(stream.getVideoTracks().length, Math.pow(2, LOOPS),
+       "The original track should contain the original video track and all the video clones");
+    stream.getTracks().forEach(t1 => is(stream.getTracks()
+                                              .filter(t2 => t1.id == t2.id)
+                                              .length,
+                                        1, "Each track should be unique"));
+
+    let test = createMediaElement('video', 'testClonePlayback');
+    let playback = new MediaStreamPlayback(test, stream);
+    await playback.playMedia(false);
+  }
+
+  {
+    info("Testing audio content routing with MediaStream.clone()");
+    let ac = new AudioContext();
+
+    let osc1kOriginal = createOscillatorStream(ac, 1000);
+    let audioTrack1kOriginal = osc1kOriginal.getTracks()[0];
+    let audioTrack1kClone = osc1kOriginal.clone().getTracks()[0];
+
+    let osc5kOriginal = createOscillatorStream(ac, 5000);
+    let audioTrack5kOriginal = osc5kOriginal.getTracks()[0];
+    let audioTrack5kClone = osc5kOriginal.clone().getTracks()[0];
+
+    info("Analysing audio output of original stream (1k + 5k)");
+    let stream = new MediaStream();
+    stream.addTrack(audioTrack1kOriginal);
+    stream.addTrack(audioTrack5kOriginal);
+
+    let analyser = new AudioStreamAnalyser(ac, stream);
+    await analyser.waitForAnalysisSuccess(array =>
+            array[analyser.binIndexForFrequency(50)] < 50 &&
+            array[analyser.binIndexForFrequency(1000)] > 200 &&
+            array[analyser.binIndexForFrequency(3000)] < 50 &&
+            array[analyser.binIndexForFrequency(5000)] > 200 &&
+            array[analyser.binIndexForFrequency(10000)] < 50);
+
+    info("Waiting for original tracks to stop");
+    stream.getTracks().forEach(t => t.stop());
+    await analyser.waitForAnalysisSuccess(array =>
+            array[analyser.binIndexForFrequency(50)] < 50 &&
+            // WebAudioDestination streams do not handle stop()
+            // XXX Should they? Plan to resolve that in bug 1208384.
+            // array[analyser.binIndexForFrequency(1000)] < 50 &&
+            array[analyser.binIndexForFrequency(3000)] < 50 &&
+            // array[analyser.binIndexForFrequency(5000)] < 50 &&
+            array[analyser.binIndexForFrequency(10000)] < 50);
+    analyser.disconnect();
+
+    info("Analysing audio output of stream clone (1k + 5k)");
+    stream = new MediaStream();
+    stream.addTrack(audioTrack1kClone);
+    stream.addTrack(audioTrack5kClone);
+
+    analyser = new AudioStreamAnalyser(ac, stream);
+    await analyser.waitForAnalysisSuccess(array =>
+            array[analyser.binIndexForFrequency(50)] < 50 &&
+            array[analyser.binIndexForFrequency(1000)] > 200 &&
+            array[analyser.binIndexForFrequency(3000)] < 50 &&
+            array[analyser.binIndexForFrequency(5000)] > 200 &&
+            array[analyser.binIndexForFrequency(10000)] < 50);
+    analyser.disconnect();
+
+    info("Analysing audio output of clone of clone (1k + 5k)");
+    stream = new MediaStream([audioTrack1kClone, audioTrack5kClone]).clone();
+
+    analyser = new AudioStreamAnalyser(ac, stream);
+    await analyser.waitForAnalysisSuccess(array =>
+            array[analyser.binIndexForFrequency(50)] < 50 &&
+            array[analyser.binIndexForFrequency(1000)] > 200 &&
+            array[analyser.binIndexForFrequency(3000)] < 50 &&
+            array[analyser.binIndexForFrequency(5000)] > 200 &&
+            array[analyser.binIndexForFrequency(10000)] < 50);
+    analyser.disconnect();
+
+    info("Analysing audio output of clone() + addTrack()ed tracks (1k + 5k)");
+    stream = new MediaStream(new MediaStream([ audioTrack1kClone
+                                             , audioTrack5kClone
+                                             ]).clone().getTracks());
+
+    analyser = new AudioStreamAnalyser(ac, stream);
+    await analyser.waitForAnalysisSuccess(array =>
+            array[analyser.binIndexForFrequency(50)] < 50 &&
+            array[analyser.binIndexForFrequency(1000)] > 200 &&
+            array[analyser.binIndexForFrequency(3000)] < 50 &&
+            array[analyser.binIndexForFrequency(5000)] > 200 &&
+            array[analyser.binIndexForFrequency(10000)] < 50);
+    analyser.disconnect();
+
+    info("Analysing audio output of clone()d tracks in original stream (1k) " +
+         "and clone()d tracks in stream clone (5k)");
+    stream = new MediaStream([audioTrack1kClone, audioTrack5kClone]);
+    let streamClone = stream.clone();
+
+    stream.getTracks().forEach(t => stream.removeTrack(t));
+    stream.addTrack(streamClone.getTracks()[0]);
+    streamClone.removeTrack(streamClone.getTracks()[0]);
+
+    analyser = new AudioStreamAnalyser(ac, stream);
+    await analyser.waitForAnalysisSuccess(array =>
+            array[analyser.binIndexForFrequency(50)] < 50 &&
+            array[analyser.binIndexForFrequency(1000)] > 200 &&
+            array[analyser.binIndexForFrequency(3000)] < 50 &&
+            array[analyser.binIndexForFrequency(5000)] < 50);
+    analyser.disconnect();
+
+    let cloneAnalyser = new AudioStreamAnalyser(ac, streamClone);
+    await cloneAnalyser.waitForAnalysisSuccess(array =>
+            array[cloneAnalyser.binIndexForFrequency(1000)] < 50 &&
+            array[cloneAnalyser.binIndexForFrequency(3000)] < 50 &&
+            array[cloneAnalyser.binIndexForFrequency(5000)] > 200 &&
+            array[cloneAnalyser.binIndexForFrequency(10000)] < 50);
+    cloneAnalyser.disconnect();
+
+    info("Analysing audio output enabled and disabled tracks that don't affect each other");
+    stream = new MediaStream([audioTrack1kClone, audioTrack5kClone]);
+    let clone = stream.clone();
+
+    stream.getTracks()[0].enabled = true;
+    stream.getTracks()[1].enabled = false;
+
+    clone.getTracks()[0].enabled = false;
+    clone.getTracks()[1].enabled = true;
+
+    analyser = new AudioStreamAnalyser(ac, stream);
+    await analyser.waitForAnalysisSuccess(array =>
+            array[analyser.binIndexForFrequency(50)] < 50 &&
+            array[analyser.binIndexForFrequency(1000)] > 200 &&
+            array[analyser.binIndexForFrequency(3000)] < 50 &&
+            array[analyser.binIndexForFrequency(5000)] < 50);
+    analyser.disconnect();
+
+    cloneAnalyser = new AudioStreamAnalyser(ac, clone);
+    await cloneAnalyser.waitForAnalysisSuccess(array =>
+            array[cloneAnalyser.binIndexForFrequency(1000)] < 50 &&
+            array[cloneAnalyser.binIndexForFrequency(3000)] < 50 &&
+            array[cloneAnalyser.binIndexForFrequency(5000)] > 200 &&
+            array[cloneAnalyser.binIndexForFrequency(10000)] < 50);
+    cloneAnalyser.disconnect();
+
+    // Restore original tracks
+    stream.getTracks().forEach(t => t.enabled = true);
+  }
+
+  gUMStream.getTracks().forEach(t => t.stop());
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaStreamConstructors.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaStreamConstructors.html
new file mode 100644
index 0000000000..4ea6e3f444
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaStreamConstructors.html
@@ -0,0 +1,171 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  "use strict";
+
+  createHTML({
+    title: "MediaStream constructors with getUserMedia streams Test",
+    bug: "1070216"
+  });
+
+  var audioContext = new AudioContext();
+  var videoElement;
+
+  runTest(() => Promise.resolve()
+    .then(() => videoElement = createMediaElement('video', 'constructorsTest'))
+    .then(() => getUserMedia({video: true})).then(gUMStream => {
+      info("Test default constructor with video");
+      ok(gUMStream.active, "gUMStream with one track should be active");
+      var track = gUMStream.getTracks()[0];
+
+      var stream = new MediaStream();
+      ok(!stream.active, "New MediaStream should be inactive");
+      checkMediaStreamContains(stream, [], "Default constructed stream");
+
+      stream.addTrack(track);
+      ok(stream.active, "MediaStream should be active after adding a track");
+      checkMediaStreamContains(stream, [track], "Added video track");
+
+      var playback = new MediaStreamPlayback(videoElement, stream);
+      return playback.playMedia(false).then(() => {
+        ok(!gUMStream.active, "gUMStream should be inactive after stopping");
+        ok(!stream.active, "stream with stopped tracks should be inactive");
+      });
+    })
+    .then(() => getUserMedia({video: true})).then(gUMStream => {
+      info("Test copy constructor with gUM stream");
+      ok(gUMStream.active, "gUMStream with one track should be active");
+      var track = gUMStream.getTracks()[0];
+
+      var stream = new MediaStream(gUMStream);
+      ok(stream.active, "List constructed MediaStream should be active");
+      checkMediaStreamContains(stream, [track], "Copy constructed video track");
+
+      var playback = new MediaStreamPlayback(videoElement, stream);
+      return playback.playMedia(false).then(() => {
+        ok(!gUMStream.active, "gUMStream should be inactive after stopping");
+        ok(!stream.active, "stream with stopped tracks should be inactive");
+      });
+    })
+    .then(() => getUserMedia({video: true})).then(gUMStream => {
+      info("Test list constructor with empty list");
+      ok(gUMStream.active, "gUMStream with one track should be active");
+      var track = gUMStream.getTracks()[0];
+
+      var stream = new MediaStream([]);
+      ok(!stream.active, "Empty-list constructed MediaStream should be inactive");
+      checkMediaStreamContains(stream, [], "Empty-list constructed stream");
+
+      stream.addTrack(track);
+      ok(stream.active, "MediaStream should be active after adding a track");
+      checkMediaStreamContains(stream, [track], "Added video track");
+
+      var playback = new MediaStreamPlayback(videoElement, stream);
+      return playback.playMedia(false).then(() => {
+        ok(!gUMStream.active, "gUMStream should be inactive after stopping");
+        ok(!stream.active, "stream with stopped tracks should be inactive");
+      });
+    })
+    .then(() => getUserMedia({audio: true, video: true})).then(gUMStream => {
+      info("Test list constructor with a gUM audio/video stream");
+      ok(gUMStream.active, "gUMStream with two tracks should be active");
+      var audioTrack = gUMStream.getAudioTracks()[0];
+      var videoTrack = gUMStream.getVideoTracks()[0];
+
+      var stream = new MediaStream([audioTrack, videoTrack]);
+      ok(stream.active, "List constructed MediaStream should be active");
+      checkMediaStreamContains(stream, [audioTrack, videoTrack],
+                               "List constructed audio and video tracks");
+
+      var playback = new MediaStreamPlayback(videoElement, stream);
+      return playback.playMedia(false).then(() => {
+        ok(!gUMStream.active, "gUMStream should be inactive after stopping");
+        ok(!stream.active, "stream with stopped tracks should be inactive");
+      });
+    })
+    .then(() => getUserMedia({video: true})).then(gUMStream => {
+      info("Test list constructor with gUM-video and WebAudio tracks");
+      ok(gUMStream.active, "gUMStream with one track should be active");
+      var audioStream = createOscillatorStream(audioContext, 2000);
+      ok(audioStream.active, "WebAudio stream should be active");
+
+      var audioTrack = audioStream.getTracks()[0];
+      var videoTrack = gUMStream.getTracks()[0];
+
+      var stream = new MediaStream([audioTrack, videoTrack]);
+      ok(stream.active, "List constructed MediaStream should be active");
+      checkMediaStreamContains(stream, [audioTrack, videoTrack],
+                               "List constructed WebAudio and gUM-video tracks");
+
+      var playback = new MediaStreamPlayback(videoElement, stream);
+      return playback.playMedia(false).then(() => {
+        gUMStream.getTracks().forEach(t => t.stop());
+        ok(!gUMStream.active, "gUMStream should be inactive after stopping");
+        ok(!stream.active, "stream with stopped tracks should be inactive");
+      });
+    })
+    .then(() => {
+      var osc1k = createOscillatorStream(audioContext, 1000);
+      var audioTrack1k = osc1k.getTracks()[0];
+
+      var osc5k = createOscillatorStream(audioContext, 5000);
+      var audioTrack5k = osc5k.getTracks()[0];
+
+      var osc10k = createOscillatorStream(audioContext, 10000);
+      var audioTrack10k = osc10k.getTracks()[0];
+
+      return Promise.resolve().then(() => {
+        info("Analysing audio output with empty default constructed stream");
+        var stream = new MediaStream();
+        var analyser = new AudioStreamAnalyser(audioContext, stream);
+        return analyser.waitForAnalysisSuccess(array =>
+          array[analyser.binIndexForFrequency(1000)]  < 50 &&
+          array[analyser.binIndexForFrequency(5000)]  < 50 &&
+          array[analyser.binIndexForFrequency(10000)] < 50)
+          .then(() => analyser.disconnect());
+      }).then(() => {
+        info("Analysing audio output with copy constructed 5k stream");
+        var stream = new MediaStream(osc5k);
+        is(stream.active, osc5k.active,
+           "Copy constructed MediaStream should preserve active state");
+        var analyser = new AudioStreamAnalyser(audioContext, stream);
+        return analyser.waitForAnalysisSuccess(array =>
+          array[analyser.binIndexForFrequency(1000)]  < 50 &&
+          array[analyser.binIndexForFrequency(5000)]  > 200 &&
+          array[analyser.binIndexForFrequency(10000)] < 50)
+          .then(() => analyser.disconnect());
+      }).then(() => {
+        info("Analysing audio output with empty-list constructed stream");
+        var stream = new MediaStream([]);
+        var analyser = new AudioStreamAnalyser(audioContext, stream);
+        return analyser.waitForAnalysisSuccess(array =>
+          array[analyser.binIndexForFrequency(1000)]  < 50 &&
+          array[analyser.binIndexForFrequency(5000)]  < 50 &&
+          array[analyser.binIndexForFrequency(10000)] < 50)
+          .then(() => analyser.disconnect());
+      }).then(() => {
+        info("Analysing audio output with list constructed 1k, 5k and 10k tracks");
+        var stream = new MediaStream([audioTrack1k, audioTrack5k, audioTrack10k]);
+        ok(stream.active,
+           "List constructed MediaStream from WebAudio should be active");
+        var analyser = new AudioStreamAnalyser(audioContext, stream);
+        return analyser.waitForAnalysisSuccess(array =>
+          array[analyser.binIndexForFrequency(50)]    < 50 &&
+          array[analyser.binIndexForFrequency(1000)]  > 200 &&
+          array[analyser.binIndexForFrequency(2500)]  < 50 &&
+          array[analyser.binIndexForFrequency(5000)]  > 200 &&
+          array[analyser.binIndexForFrequency(7500)]  < 50 &&
+          array[analyser.binIndexForFrequency(10000)] > 200 &&
+          array[analyser.binIndexForFrequency(11000)] < 50)
+          .then(() => analyser.disconnect());
+      });
+    }));
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaStreamTrackClone.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaStreamTrackClone.html
new file mode 100644
index 0000000000..e5e0764427
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_mediaStreamTrackClone.html
@@ -0,0 +1,170 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  "use strict";
+
+  createHTML({
+    title: "MediaStreamTrack.clone()",
+    bug: "1208371"
+  });
+
+  var testSingleTrackClonePlayback = constraints =>
+      getUserMedia(constraints).then(stream => {
+    info("Test clone()ing an " + constraints + " gUM track");
+    var track = stream.getTracks()[0];
+    var clone = track.clone();
+
+    checkMediaStreamTrackCloneAgainstOriginal(clone, track);
+
+    info("Stopping original track");
+    track.stop();
+
+    info("Creating new stream for clone");
+    var cloneStream = new MediaStream([clone]);
+    checkMediaStreamContains(cloneStream, [clone]);
+
+    info("Testing playback of track clone");
+    var test = createMediaElement('video', 'testClonePlayback');
+    var playback = new MediaStreamPlayback(test, cloneStream);
+    return playback.playMedia(false);
+  });
+
+  runTest(() => Promise.resolve()
+    .then(() => testSingleTrackClonePlayback({audio: true}))
+    .then(() => testSingleTrackClonePlayback({video: true}))
+    .then(() => getUserMedia({video: true})).then(stream => {
+      info("Test cloning a track into inception");
+      var track = stream.getTracks()[0];
+      var clone = track;
+      var clones = Array(10).fill().map(() => clone = clone.clone());
+      var inceptionClone = clones.pop();
+      checkMediaStreamTrackCloneAgainstOriginal(inceptionClone, track);
+
+      var cloneStream = new MediaStream();
+      cloneStream.addTrack(inceptionClone);
+
+      // cloneStream is now essentially the same as stream.clone();
+      checkMediaStreamCloneAgainstOriginal(cloneStream, stream);
+
+      var test = createMediaElement('video', 'testClonePlayback');
+      var playback = new MediaStreamPlayback(test, cloneStream);
+      return playback.playMedia(false).then(() => {
+          info("Testing that clones of ended tracks are ended");
+          cloneStream.clone().getTracks().forEach(t =>
+            is(t.readyState, "ended", "Track " + t.id + " should be ended"));
+        })
+        .then(() => {
+          clones.forEach(t => t.stop());
+          track.stop();
+        });
+    })
+    .then(() => getUserMedia({audio: true, video: true})).then(stream => {
+      info("Test adding many track clones to the original stream");
+
+      const LOOPS = 3;
+      for (var i = 0; i < LOOPS; i++) {
+        stream.getTracks().forEach(t => stream.addTrack(t.clone()));
+      }
+      is(stream.getVideoTracks().length, Math.pow(2, LOOPS),
+         "The original track should contain the original video track and all the video clones");
+      stream.getTracks().forEach(t1 => is(stream.getTracks()
+                                                .filter(t2 => t1.id == t2.id)
+                                                .length,
+                                          1, "Each track should be unique"));
+
+      var test = createMediaElement('video', 'testClonePlayback');
+      var playback = new MediaStreamPlayback(test, stream);
+      return playback.playMedia(false);
+    })
+    .then(() => {
+      info("Testing audio content routing with MediaStreamTrack.clone()");
+      var ac = new AudioContext();
+
+      var osc1kOriginal = createOscillatorStream(ac, 1000);
+      var audioTrack1kOriginal = osc1kOriginal.getTracks()[0];
+      var audioTrack1kClone = audioTrack1kOriginal.clone();
+
+      var osc5kOriginal = createOscillatorStream(ac, 5000);
+      var audioTrack5kOriginal = osc5kOriginal.getTracks()[0];
+      var audioTrack5kClone = audioTrack5kOriginal.clone();
+
+      return Promise.resolve().then(() => {
+        info("Analysing audio output enabled and disabled tracks that don't affect each other");
+        audioTrack1kOriginal.enabled = true;
+        audioTrack5kOriginal.enabled = false;
+
+        audioTrack1kClone.enabled = false;
+        audioTrack5kClone.enabled = true;
+
+        var analyser =
+          new AudioStreamAnalyser(ac, new MediaStream([audioTrack1kOriginal,
+                                                      audioTrack5kOriginal]));
+        return analyser.waitForAnalysisSuccess(array =>
+                 array[analyser.binIndexForFrequency(50)] < 50 &&
+                 array[analyser.binIndexForFrequency(1000)] > 200 &&
+                 array[analyser.binIndexForFrequency(3000)] < 50 &&
+                 array[analyser.binIndexForFrequency(5000)] < 50)
+          .then(() => analyser.disconnect())
+          .then(() => {
+            var cloneAnalyser =
+              new AudioStreamAnalyser(ac, new MediaStream([audioTrack1kClone,
+                                                           audioTrack5kClone]));
+            return cloneAnalyser.waitForAnalysisSuccess(array =>
+                     array[cloneAnalyser.binIndexForFrequency(1000)] < 50 &&
+                     array[cloneAnalyser.binIndexForFrequency(3000)] < 50 &&
+                     array[cloneAnalyser.binIndexForFrequency(5000)] > 200 &&
+                     array[cloneAnalyser.binIndexForFrequency(10000)] < 50)
+              .then(() => cloneAnalyser.disconnect());
+          })
+          // Restore original tracks
+          .then(() => [audioTrack1kOriginal,
+                       audioTrack5kOriginal,
+                       audioTrack1kClone,
+                       audioTrack5kClone].forEach(t => t.enabled = true));
+      }).then(() => {
+        info("Analysing audio output of 1k original and 5k clone.");
+        var stream = new MediaStream();
+        stream.addTrack(audioTrack1kOriginal);
+        stream.addTrack(audioTrack5kClone);
+
+        var analyser = new AudioStreamAnalyser(ac, stream);
+        return analyser.waitForAnalysisSuccess(array =>
+                 array[analyser.binIndexForFrequency(50)] < 50 &&
+                 array[analyser.binIndexForFrequency(1000)] > 200 &&
+                 array[analyser.binIndexForFrequency(3000)] < 50 &&
+                 array[analyser.binIndexForFrequency(5000)] > 200 &&
+                 array[analyser.binIndexForFrequency(10000)] < 50)
+          .then(() => {
+            info("Waiting for tracks to stop");
+            stream.getTracks().forEach(t => t.stop());
+            return analyser.waitForAnalysisSuccess(array =>
+                     array[analyser.binIndexForFrequency(50)] < 50 &&
+                     array[analyser.binIndexForFrequency(1000)] < 50 &&
+                     array[analyser.binIndexForFrequency(3000)] < 50 &&
+                     array[analyser.binIndexForFrequency(5000)] < 50 &&
+                     array[analyser.binIndexForFrequency(10000)] < 50);
+          }).then(() => analyser.disconnect());
+      }).then(() => {
+        info("Analysing audio output of clones of clones (1kx2 + 5kx4)");
+        var stream = new MediaStream([audioTrack1kClone.clone(),
+                                      audioTrack5kOriginal.clone().clone().clone().clone()]);
+
+        var analyser = new AudioStreamAnalyser(ac, stream);
+        return analyser.waitForAnalysisSuccess(array =>
+                 array[analyser.binIndexForFrequency(50)] < 50 &&
+                 array[analyser.binIndexForFrequency(1000)] > 200 &&
+                 array[analyser.binIndexForFrequency(3000)] < 50 &&
+                 array[analyser.binIndexForFrequency(5000)] > 200 &&
+                 array[analyser.binIndexForFrequency(10000)] < 50)
+          .then(() => analyser.disconnect());
+      });
+    }));
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_nonDefaultRate.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_nonDefaultRate.html
new file mode 100644
index 0000000000..8a6ac8c62b
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_nonDefaultRate.html
@@ -0,0 +1,37 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "getUserMedia feed to a graph with non default rate",
+    bug: "1387454",
+  });
+
+  /**
+   * Run a test to verify that when we use the streem from a gUM to an AudioContext
+   * with non default rate the connection fails. (gUM is always on default rate).
+   */
+  runTest(async () => {
+    // Since we do not examine the stream we do not need loopback.
+    DISABLE_LOOPBACK_TONE = true;
+    const stream = await getUserMedia({audio: true});
+    const nonDefaultRate = 32000;
+    const ac = new AudioContext({sampleRate: nonDefaultRate});
+    mustThrowWith(
+      "Connect stream with graph of different sample rate",
+      "NotSupportedError", () => {
+        ac.createMediaStreamSource(stream);
+      }
+    );
+    for (let t of stream.getTracks()) {
+      t.stop();
+    }
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_peerIdentity.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_peerIdentity.html
new file mode 100644
index 0000000000..c4dfb9acb8
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_peerIdentity.html
@@ -0,0 +1,51 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+  <script type="application/javascript" src="blacksilence.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  title: "Test getUserMedia peerIdentity Constraint",
+  bug: "942367"
+});
+async function theTest() {
+  async function testPeerIdentityConstraint(withConstraint) {
+    const config = { audio: true, video: true };
+    if (withConstraint) {
+      config.peerIdentity = 'user@example.com';
+    }
+    info('getting media with constraints: ' + JSON.stringify(config));
+    const stream = await getUserMedia(config);
+    for (const track of stream.getTracks()) {
+      const recorder = new MediaRecorder(new MediaStream([track]));
+      try {
+        recorder.start();
+        ok(!withConstraint,
+          `gUM ${track.kind} track without peerIdentity must not throw`);
+        recorder.stop();
+      } catch(e) {
+        ok(withConstraint,
+          `gUM ${track.kind} track with peerIdentity must throw`);
+      }
+    }
+    await Promise.all([
+      audioIsSilence(withConstraint, stream),
+      videoIsBlack(withConstraint, stream),
+    ]);
+    stream.getTracks().forEach(t => t.stop());
+  };
+
+  // both without and with the constraint
+  await testPeerIdentityConstraint(false);
+  await testPeerIdentityConstraint(true);
+}
+
+runTest(theTest);
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_permission.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_permission.html
new file mode 100644
index 0000000000..cd02c7326c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_permission.html
@@ -0,0 +1,104 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({ title: "Test getUserMedia in iframes", bug: "1371741" });
+/**
+  Tests covering enumerateDevices API and deviceId constraint. Exercise code.
+*/
+
+// Call gUM in iframe.
+async function iframeGum(dict, iframe = document.createElement("iframe")) {
+  Object.assign(iframe, dict);
+  if (dict.src) {
+    info(`<iframe src="${dict.src}" sandbox="${dict.sandbox}">`);
+  } else {
+    info(`<iframe srcdoc sandbox="${dict.sandbox}">`);
+  }
+  document.documentElement.appendChild(iframe);
+
+  const once = (t, msg) => new Promise(r => t.addEventListener(msg, r, { once: true }));
+  const haveMessage = once(window, "message");
+  await new Promise(resolve => iframe.onload = resolve);
+  return (await haveMessage).data;
+};
+
+runTest(async () => {
+  const path = "/tests/dom/media/webrtc/tests/mochitests/test_getUserMedia_permission_iframe.html";
+
+  async function sourceFn() {
+    try {
+      const gUM = c => navigator.mediaDevices.getUserMedia(c);
+      let message;
+      let stream;
+      try {
+        stream = await gUM({ video: true });
+        message = 'success';
+      } catch(e) {
+        message = e.name;
+      }
+      parent.postMessage(message, 'https://example.com:443');
+
+      if (message == "success") {
+        stream.getTracks().forEach(track => track.stop());
+      }
+    } catch (e) {
+      setTimeout(() => { throw e; });
+    }
+  }
+
+  const source = `<html\><script\>(${sourceFn.toString()})()</script\></html\>`;
+
+  // Test gUM in sandboxed vs. regular iframe.
+
+  for (const origin of [window.location.origin, "https://test1.example.com"]) {
+    const src = origin + path;
+    is(await iframeGum({ src, sandbox: "allow-scripts" }),
+       "NotAllowedError", "gUM fails in sandboxed iframe " + origin);
+  }
+  is(await iframeGum({
+       src: path,
+       sandbox: "allow-scripts allow-same-origin",
+     }),
+     "success", "gUM works in regular same-origin iframe");
+  is(await iframeGum({
+       src: `https://test1.example.com${path}`,
+       sandbox: "allow-scripts allow-same-origin",
+     }),
+     "NotAllowedError", "gUM fails in regular cross-origin iframe");
+
+  // Test gUM in sandboxed vs regular srcdoc iframe
+
+  const iframeSrcdoc = document.createElement("iframe");
+  iframeSrcdoc.srcdoc = source;
+  is(await iframeGum({ sandbox: "allow-scripts" }, iframeSrcdoc),
+     "NotAllowedError", "gUM fails in sandboxed srcdoc iframe");
+  is(await iframeGum({ sandbox: "allow-scripts allow-same-origin" }, iframeSrcdoc),
+     "success", "gUM works in regular srcdoc iframe");
+
+  // Test gUM in sandboxed vs regular blob iframe
+
+  const blob = new Blob([source], {type : "text/html"});
+	let src = URL.createObjectURL(blob);
+  is(await iframeGum({ src, sandbox: "allow-scripts" }),
+     "NotAllowedError", "gUM fails in sandboxed blob iframe");
+  is(await iframeGum({ src, sandbox: "allow-scripts allow-same-origin"}),
+     "success", "gUM works in regular blob iframe");
+  URL.revokeObjectURL(src);
+
+  // data iframes always have null-principals
+
+  src = `data:text/html;base64,${btoa(source)}`;
+  is(await iframeGum({ src, sandbox: "allow-scripts" }),
+     "NotAllowedError", "gUM fails in sandboxed data iframe");
+  is(await iframeGum({ src, sandbox: "allow-scripts allow-same-origin"}),
+     "NotAllowedError", "gUM fails in regular data iframe");
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_permission_iframe.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_permission_iframe.html
new file mode 100644
index 0000000000..732c2cf98c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_permission_iframe.html
@@ -0,0 +1,30 @@
+<!DOCTYPE HTML>
+<html>
+<body>
+<pre id="test">
+<script type="application/javascript">
+/**
+  Runs inside iframe in test_getUserMedia_permission.html.
+*/
+
+const gUM = c => navigator.mediaDevices.getUserMedia(c);
+
+(async () => {
+  let message;
+  let stream;
+  try {
+    stream = await gUM({ video: true });
+    message = "success";
+  } catch(e) {
+    message = e.name;
+  }
+  parent.postMessage(message, "https://example.com:443");
+
+  if (message == "success") {
+    stream.getTracks().forEach(track => track.stop());
+  }
+})().catch(e => setTimeout(() => { throw e; }));
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_playAudioTwice.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_playAudioTwice.html
new file mode 100644
index 0000000000..30d168bf38
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_playAudioTwice.html
@@ -0,0 +1,25 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({title: "getUserMedia Play Audio Twice", bug: "822109" });
+  /**
+   * Run a test that we can complete an audio playback cycle twice in a row.
+   */
+  runTest(function () {
+    return getUserMedia({audio: true}).then(audioStream => {
+      var testAudio = createMediaElement('audio', 'testAudio');
+      var playback = new MediaStreamPlayback(testAudio, audioStream);
+
+      return playback.playMediaWithoutStoppingTracks(false)
+        .then(() => playback.playMedia(true));
+    });
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_playVideoAudioTwice.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_playVideoAudioTwice.html
new file mode 100644
index 0000000000..7b5e6effd1
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_playVideoAudioTwice.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({title: "getUserMedia Play Video and Audio Twice", bug: "822109" });
+  /**
+   * Run a test that we can complete a video playback cycle twice in a row.
+   */
+  runTest(function () {
+    return getUserMedia({video: true, audio: true}).then(stream => {
+      var testVideo = createMediaElement('video', 'testVideo');
+      var playback = new MediaStreamPlayback(testVideo, stream);
+
+      return playback.playMediaWithoutStoppingTracks(false)
+        .then(() => playback.playMedia(true));
+    });
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_playVideoTwice.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_playVideoTwice.html
new file mode 100644
index 0000000000..2890f45eab
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_playVideoTwice.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({ title: "getUserMedia Play Video Twice", bug: "822109" });
+  /**
+   * Run a test that we can complete a video playback cycle twice in a row.
+   */
+  runTest(function () {
+    return getUserMedia({video: true}).then(stream => {
+      var testVideo = createMediaElement('video', 'testVideo');
+      var streamPlayback = new MediaStreamPlayback(testVideo, stream);
+
+      return streamPlayback.playMediaWithoutStoppingTracks(false)
+        .then(() => streamPlayback.playMedia(true));
+    });
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_scarySources.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_scarySources.html
new file mode 100644
index 0000000000..782110823e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_scarySources.html
@@ -0,0 +1,51 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+
+createHTML({title: "Detect screensharing sources that are firefox", bug: "1311048"});
+
+const Services = SpecialPowers.Services;
+
+let observe = topic => new Promise(r => Services.obs.addObserver(function o(...args) {
+  Services.obs.removeObserver(o, topic);
+  r(args.map(x => SpecialPowers.wrap(x)));
+}, topic));
+
+let getDevices = async constraints => {
+  SpecialPowers.wrap(document).notifyUserGestureActivation();
+  let [{ windowID, innerWindowID, callID, devices }] = await Promise.race([
+    getUserMedia(constraints),
+    observe("getUserMedia:request")
+  ]);
+  let window = Services.wm.getOuterWindowWithId(windowID);
+  return devices.map(SpecialPowers.wrapCallback(d => d.QueryInterface(Ci.nsIMediaDevice)));
+};
+
+runTest(async () => {
+  await pushPrefs(["media.navigator.permission.disabled", true],
+                  ["media.navigator.permission.force", true]);
+  let devices = await getDevices({video: { mediaSource: "window" }});
+  ok(devices.length, "Found one or more windows.");
+  devices = Array.prototype.filter.call(devices, d => d.scary);
+  ok(devices.length, "Found one or more scary windows (our own counts).");
+  devices = devices.filter(d => d.rawName.includes("MochiTest"));
+  ok(devices.length,
+     "Our own window is among the scary: "
+     + devices.map(d => `"${d.rawName}"`));
+
+  devices = await getDevices({video: { mediaSource: "screen" }});
+  let numScreens = devices.length;
+  ok(numScreens, "Found one or more screens.");
+  devices = Array.prototype.filter.call(devices, d => d.scary);
+  is(devices.length, numScreens, "All screens are scary.");
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_spinEventLoop.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_spinEventLoop.html
new file mode 100644
index 0000000000..ae691785f5
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_spinEventLoop.html
@@ -0,0 +1,28 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({ title: "getUserMedia Basic Audio Test", bug: "1208656" });
+  /**
+   * Run a test to verify that we can spin the event loop from within a mozGUM callback.
+   */
+  runTest(() => {
+    var testAudio = createMediaElement('audio', 'testAudio');
+    return new Promise((resolve, reject) => {
+      navigator.mozGetUserMedia({ audio: true }, stream => {
+        SpecialPowers.spinEventLoop(window);
+        ok(true, "Didn't crash");
+        stream.getTracks().forEach(t => t.stop());
+        resolve();
+      }, () => {});
+    });
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_trackCloneCleanup.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_trackCloneCleanup.html
new file mode 100644
index 0000000000..60077ec73b
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_trackCloneCleanup.html
@@ -0,0 +1,32 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  "use strict";
+
+  createHTML({
+    title: "Stopping a MediaStreamTrack and its clones should deallocate the device",
+    bug: "1294605"
+  });
+
+  runTest(async () => {
+    await pushPrefs(["media.navigator.permission.fake", true]);
+    const stream = await getUserMedia({audio: true, video: true, fake: true});
+    const clone = stream.clone();
+    stream.getTracks().forEach(t => t.stop());
+    stream.clone().getTracks().forEach(t => stream.addTrack(t));
+    is(stream.getTracks().filter(t => t.readyState == "live").length, 0,
+       "Cloning ended tracks should make them ended");
+    [...stream.getTracks(), ...clone.getTracks()].forEach(t => t.stop());
+
+    // Bug 1295352: better to be explicit about noGum here wrt future refactoring.
+    return noGum();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_getUserMedia_trackEnded.html b/dom/media/webrtc/tests/mochitests/test_getUserMedia_trackEnded.html
new file mode 100644
index 0000000000..b275f4555f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_getUserMedia_trackEnded.html
@@ -0,0 +1,68 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<iframe id="iframe" onload="iframeLoaded()" srcdoc="
+  <script type='application/javascript'>
+  document.gUM = (constraints, success, failure) =>
+    navigator.mediaDevices.getUserMedia(constraints).then(success, failure);
+  </script>">
+</iframe>
+<script type="application/javascript">
+  "use strict";
+  let iframeLoadedPromise = {};
+  iframeLoadedPromise.promise = new Promise(r => {
+    iframeLoadedPromise.resolve = r;
+  });;
+  function iframeLoaded() {
+    iframeLoadedPromise.resolve();
+  }
+
+  createHTML({
+    title: "getUserMedia MediaStreamTrack 'ended' event on navigating",
+    bug: "1208373",
+  });
+
+  runTest(async () => {
+    await iframeLoadedPromise.promise;
+    let iframe = document.getElementById("iframe");
+    let stream;
+    // We're passing callbacks into a method in the iframe here, because
+    // a Promise created in the iframe is unusable after the iframe has
+    // navigated away (see bug 1269400 for details).
+    return new Promise((resolve, reject) =>
+        iframe.contentDocument.gUM({audio: true, video: true}, resolve, reject))
+      .then(s => {
+        // We're cloning a stream containing identical tracks (an original
+        // and its clone) to test that ended works both for originals
+        // clones when they're both owned by the same MediaStream.
+        // (Bug 1274221)
+        stream = new MediaStream([].concat(s.getTracks(), s.getTracks())
+                                   .map(t => t.clone())).clone();
+        var allTracksEnded = Promise.all(stream.getTracks().map(t => {
+          info("Set up ended handler for track " + t.id);
+          return haveEvent(t, "ended", wait(50000))
+            .then(event => {
+              info("ended handler invoked for track " + t.id);
+              is(event.target, t, "Target should be correct");
+            }, e => e ? Promise.reject(e)
+                      : ok(false, "ended event never raised for track " + t.id));
+        }));
+        stream.getTracks().forEach(t =>
+          is(t.readyState, "live",
+             "Non-ended track should have readyState 'live'"));
+        iframe.srcdoc = "";
+        info("iframe has been reset. Waiting for tracks to end.");
+        return allTracksEnded;
+      })
+      .then(() => stream.getTracks().forEach(t =>
+        is(t.readyState, "ended",
+           "Ended track should have readyState 'ended'")));
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_groupId.html b/dom/media/webrtc/tests/mochitests/test_groupId.html
new file mode 100644
index 0000000000..f2aefe5e80
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_groupId.html
@@ -0,0 +1,53 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({ title: "Test group id of MediaDeviceInfo", bug: "1213453" });
+
+async function getDefaultDevices() {
+  const devices = await navigator.mediaDevices.enumerateDevices();
+  is(devices.length, 2, "Two fake devices found.");
+
+  devices.forEach(d => isnot(d.groupId, "", "GroupId is included in every device"));
+
+  const videos = devices.filter(d => d.kind == "videoinput");
+  is(videos.length, 1, "One video device found.");
+  const audios = devices.filter(d => d.kind == "audioinput");
+  is(audios.length, 1, "One microphone device found.");
+
+  return {audio: audios[0], video: videos[0]};
+}
+
+runTest(async () => {
+  // Force fake devices in order to be able to change camera name by pref.
+  await pushPrefs(["media.navigator.streams.fake", true],
+                  ["media.audio_loopback_dev", ""],
+                  ["media.video_loopback_dev", ""]);
+
+  const afterGum = await navigator.mediaDevices.getUserMedia({
+    video: true, audio: true
+  });
+  afterGum.getTracks().forEach(track => track.stop());
+
+  let {audio, video} = await getDefaultDevices();
+
+  /* The low level method to correlate groupIds is by device names.
+   * Use a similar comparison here to verify that it works.
+   * Multiple devices of the same device name are not expected in
+   * automation. */
+  isnot(audio.label, video.label, "Audio label differs from video");
+  isnot(audio.groupId, video.groupId, "Not the same groupIds");
+  // Change video name to match.
+  await pushPrefs(["media.getusermedia.fake-camera-name", audio.label]);
+  ({audio, video} = await getDefaultDevices());
+  is(audio.label, video.label, "Audio label matches video");
+  is(audio.groupId, video.groupId, "GroupIds should be the same");
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_multi_mics.html b/dom/media/webrtc/tests/mochitests/test_multi_mics.html
new file mode 100644
index 0000000000..95bcbfd3e4
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_multi_mics.html
@@ -0,0 +1,66 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+"use strict";
+
+createHTML({
+  title: "Test the ability of opening multiple microphones via gUM",
+  bug: "1238038",
+});
+
+runTest(async () => {
+  // Ensure we use the real microphones by disabling loopback devices and fake devices.
+  await pushPrefs(["media.audio_loopback_dev", ""], ["media.navigator.streams.fake", false]);
+
+  try {
+    let devices = await navigator.mediaDevices.enumerateDevices();
+    // Create constraints
+    let constraints = [];
+    devices.forEach((device) => {
+      if (device.kind === "audioinput") {
+        constraints.push({
+          audio: { deviceId: { exact: device.deviceId } },
+        });
+      }
+    });
+    if (constraints.length >= 2) {
+      // Create constraints
+      let constraints = [];
+      devices.forEach((device) => {
+        if (device.kind === "audioinput") {
+          constraints.push({
+            audio: { deviceId: { exact: device.deviceId } },
+          });
+        }
+      });
+      // Open microphones by the constraints
+      let mediaStreams = [];
+      for (let c of constraints) {
+        let stream = await navigator.mediaDevices.getUserMedia(c);
+        dump("MediaStream: " + stream.id + " for device: " + c.audio.deviceId.exact + " is created\n");
+        mediaStreams.push(stream);
+      }
+      // Close microphones
+      for (let stream of mediaStreams) {
+        for (let track of stream.getTracks()) {
+          track.stop();
+        }
+        dump("Stop all tracks in MediaStream: " + stream.id + "\n");
+      }
+      mediaStreams = [];
+    } else {
+      dump("Skip test since we need at least two microphones\n");
+    }
+  } catch (e) {
+    ok(false, e.name + ": " + e.message);
+  }
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_ondevicechange.html b/dom/media/webrtc/tests/mochitests/test_ondevicechange.html
new file mode 100644
index 0000000000..4358d9d748
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_ondevicechange.html
@@ -0,0 +1,180 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <meta charset="utf-8">
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<script type="application/javascript">
+"use strict";
+
+createHTML({
+  title: "ondevicechange tests",
+  bug: "1152383"
+});
+
+async function resolveOnEvent(target, name) {
+  return new Promise(r => target.addEventListener(name, r, {once: true}));
+}
+let eventCount = 0;
+async function triggerVideoDevicechange() {
+  ++eventCount;
+  // "media.getusermedia.fake-camera-name" specifies the name of the single
+  // fake video camera.
+  // Changing the pref imitates replacing one device with another.
+  return pushPrefs(["media.getusermedia.fake-camera-name",
+                    `devicechange ${eventCount}`])
+}
+function addIframe() {
+  const iframe = document.createElement("iframe");
+  // Workaround for bug 1743933
+  iframe.loadPromise = resolveOnEvent(iframe, "load");
+  document.documentElement.appendChild(iframe);
+  return iframe;
+}
+
+runTest(async () => {
+  // A toplevel Window and an iframe Windows are compared for devicechange
+  // events.
+  const iframe1 = addIframe();
+  const iframe2 = addIframe();
+  await Promise.all([
+    iframe1.loadPromise,
+    iframe2.loadPromise,
+    pushPrefs(
+      // Use the fake video backend to trigger devicechange events.
+      ["media.navigator.streams.fake", true],
+      // Loopback would override fake.
+      ["media.video_loopback_dev", ""],
+      // Make fake devices count as real, permission-wise, or devicechange
+      // events won't be exposed
+      ["media.navigator.permission.fake", true],
+      // For gUM.
+      ["media.navigator.permission.disabled", true]
+    ),
+  ]);
+  const topDevices = navigator.mediaDevices;
+  const frame1Devices = iframe1.contentWindow.navigator.mediaDevices;
+  const frame2Devices = iframe2.contentWindow.navigator.mediaDevices;
+  // Initialization of MediaDevices::mLastPhysicalDevices is triggered when
+  // ondevicechange is set but tests "media.getusermedia.fake-camera-name"
+  // asynchronously.  Wait for getUserMedia() completion to ensure that the
+  // pref has been read before doDevicechanges() changes it.
+  frame1Devices.ondevicechange = () => {};
+  const topEventPromise = resolveOnEvent(topDevices, "devicechange");
+  const frame2EventPromise = resolveOnEvent(frame2Devices, "devicechange");
+  (await frame1Devices.getUserMedia({video: true})).getTracks()[0].stop();
+
+  await Promise.all([
+    resolveOnEvent(frame1Devices, "devicechange"),
+    triggerVideoDevicechange(),
+  ]);
+  ok(true,
+     "devicechange event is fired when gUM has been in use");
+  // The number of devices has not changed.  Race a settled Promise to check
+  // that no devicechange event has been received in frame2.
+  const racer = {};
+  is(await Promise.race([frame2EventPromise, racer]), racer,
+     "devicechange event is NOT fired in iframe2 for replaced device when " +
+     "gUM has NOT been in use");
+  // getUserMedia() is invoked on frame2Devices after a first device list
+  // change but before returning to the previous state, in order to test that
+  // the device set is compared with the set after previous device list
+  // changes regardless of whether a "devicechange" event was previously
+  // dispatched.
+  (await frame2Devices.getUserMedia({video: true})).getTracks()[0].stop();
+  // Revert device list change.
+  await Promise.all([
+    resolveOnEvent(frame1Devices, "devicechange"),
+    resolveOnEvent(frame2Devices, "devicechange"),
+    SpecialPowers.popPrefEnv(),
+  ]);
+  ok(true,
+     "devicechange event is fired on return to previous list " +
+     "after gUM has been is use");
+
+  const frame1EventPromise1 = resolveOnEvent(frame1Devices, "devicechange");
+  while (true) {
+    const racePromise = Promise.race([
+      frame1EventPromise1,
+      // 100ms is half the coalescing time in MediaManager::DeviceListChanged().
+      wait(100, {type: "wait done"}),
+    ]);
+    await triggerVideoDevicechange();
+    if ((await racePromise).type == "devicechange") {
+      ok(true,
+         "devicechange event is fired even when hardware changes continue");
+      break;
+    }
+  }
+
+  is(await Promise.race([topEventPromise, racer]), racer,
+     "devicechange event is NOT fired for device replacements when " +
+     "gUM has NOT been in use");
+
+  if (navigator.userAgent.includes("Android")) {
+    todo(false, "test assumes Firefox-for-Desktop specific API and behavior");
+    return;
+  }
+  // Open a new tab, which is expected to receive focus and hide the first tab.
+  const tab = window.open();
+  SimpleTest.registerCleanupFunction(() => tab.close());
+  await Promise.all([
+    resolveOnEvent(document, 'visibilitychange'),
+    resolveOnEvent(tab, 'focus'),
+  ]);
+  ok(tab.document.hasFocus(), "tab.document.hasFocus()");
+  await Promise.all([
+    resolveOnEvent(tab, 'blur'),
+    SpecialPowers.spawnChrome([], function focusUrlBar() {
+      this.browsingContext.topChromeWindow.gURLBar.focus();
+    }),
+  ]);
+  ok(!tab.document.hasFocus(), "!tab.document.hasFocus()");
+  is(document.visibilityState, 'hidden', 'visibilityState')
+  const frame1EventPromise2 = resolveOnEvent(frame1Devices, "devicechange");
+  const tabDevices = tab.navigator.mediaDevices;
+  tabDevices.ondevicechange = () => {};
+  const tabStream = await tabDevices.getUserMedia({video: true});
+  // Trigger and await two devicechanges on tabDevices to wait long enough to
+  // provide that a devicechange on another MediaDevices would be received.
+  for (let i = 0; i < 2; ++i) {
+    await Promise.all([
+      resolveOnEvent(tabDevices, "devicechange"),
+      triggerVideoDevicechange(),
+    ]);
+  };
+  is(await Promise.race([frame1EventPromise2, racer]), racer,
+     "devicechange event is NOT fired while tab is in background");
+  tab.close();
+  await resolveOnEvent(document, 'visibilitychange');
+  is(document.visibilityState, 'visible', 'visibilityState')
+  await frame1EventPromise2;
+  ok(true, "devicechange event IS fired when tab returns to foreground");
+
+  const audioLoopbackDev =
+        SpecialPowers.getCharPref("media.audio_loopback_dev", "");
+  if (!navigator.userAgent.includes("Linux")) {
+    todo_isnot(audioLoopbackDev, "", "audio_loopback_dev");
+    return;
+  }
+  isnot(audioLoopbackDev, "", "audio_loopback_dev");
+  await Promise.all([
+    resolveOnEvent(topDevices, "devicechange"),
+    pushPrefs(["media.audio_loopback_dev", "none"]),
+  ]);
+  ok(true,
+     "devicechange event IS fired when last audio device is removed and " +
+     "gUM has NOT been in use");
+  await Promise.all([
+    resolveOnEvent(topDevices, "devicechange"),
+    pushPrefs(["media.audio_loopback_dev", audioLoopbackDev]),
+  ]);
+  ok(true,
+     "devicechange event IS fired when first audio device is added and " +
+     "gUM has NOT been in use");
+});
+
+</script>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_addAudioTrackToExistingVideoStream.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_addAudioTrackToExistingVideoStream.html
new file mode 100644
index 0000000000..b09d7ffeb5
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_addAudioTrackToExistingVideoStream.html
@@ -0,0 +1,55 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1246310",
+    title: "Renegotiation: add audio track to existing video-only stream",
+  });
+
+  runNetworkTest(function (options) {
+    SimpleTest.requestCompleteLog();
+    const test = new PeerConnectionTest(options);
+    test.chain.replace("PC_LOCAL_GUM",
+      [
+        function PC_LOCAL_GUM_ATTACH_VIDEO_ONLY(test) {
+          var localConstraints = {audio: true, video: true};
+          test.setMediaConstraints([{video: true}], []);
+          return getUserMedia(localConstraints)
+            .then(s => test.originalGumStream = s)
+            .then(() => is(test.originalGumStream.getAudioTracks().length, 1,
+                           "Should have 1 audio track"))
+            .then(() => is(test.originalGumStream.getVideoTracks().length, 1,
+                           "Should have 1 video track"))
+            .then(() => test.pcLocal.attachLocalTrack(
+                          test.originalGumStream.getVideoTracks()[0],
+                          test.originalGumStream));
+        },
+      ]
+    );
+    addRenegotiation(test.chain,
+      [
+        function PC_LOCAL_ATTACH_SECOND_TRACK_AUDIO(test) {
+          test.setMediaConstraints([{audio: true, video: true}], []);
+          return test.pcLocal.attachLocalTrack(
+                   test.originalGumStream.getAudioTracks()[0],
+                   test.originalGumStream);
+        },
+      ],
+      [
+        function PC_CHECK_REMOTE_AUDIO_FLOW(test) {
+          return test.pcRemote.checkReceivingToneFrom(new AudioContext(), test.pcLocal);
+        }
+      ]
+    );
+
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_addDataChannel.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_addDataChannel.html
new file mode 100644
index 0000000000..c7536214e5
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_addDataChannel.html
@@ -0,0 +1,33 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: add DataChannel"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    addRenegotiation(test.chain,
+                     commandsCreateDataChannel,
+                     commandsCheckDataChannel);
+
+    // Insert before the second PC_LOCAL_WAIT_FOR_MEDIA_FLOW
+    test.chain.insertBefore('PC_LOCAL_WAIT_FOR_MEDIA_FLOW',
+                            commandsWaitForDataChannel,
+                            false,
+                            1);
+
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_addDataChannelNoBundle.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_addDataChannelNoBundle.html
new file mode 100644
index 0000000000..6ad754336c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_addDataChannelNoBundle.html
@@ -0,0 +1,44 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: add DataChannel"
+  });
+
+  runNetworkTest(function (options) {
+    options = options || { };
+    options.bundle = false;
+    const test = new PeerConnectionTest(options);
+    addRenegotiation(test.chain,
+                     commandsCreateDataChannel.concat(
+                       [
+                         function PC_LOCAL_EXPECT_ICE_CHECKING(test) {
+                           test.pcLocal.expectIceChecking();
+                         },
+                         function PC_REMOTE_EXPECT_ICE_CHECKING(test) {
+                           test.pcRemote.expectIceChecking();
+                         },
+                       ]
+                      ),
+                     commandsCheckDataChannel);
+
+    // Insert before the second PC_LOCAL_WAIT_FOR_MEDIA_FLOW
+    test.chain.insertBefore('PC_LOCAL_WAIT_FOR_MEDIA_FLOW',
+                            commandsWaitForDataChannel,
+                            false,
+                            1);
+
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_addSecondAudioStream.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_addSecondAudioStream.html
new file mode 100644
index 0000000000..61a0250887
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_addSecondAudioStream.html
@@ -0,0 +1,45 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: add second audio stream"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    addRenegotiation(test.chain,
+      [
+        function PC_LOCAL_ADD_SECOND_STREAM(test) {
+          test.setMediaConstraints([{audio: true}, {audio: true}],
+                                   [{audio: true}]);
+          return test.pcLocal.getAllUserMediaAndAddStreams([{audio: true}]);
+        },
+      ],
+      [
+        function PC_REMOTE_CHECK_ADDED_TRACK(test) {
+          // We test both tracks to avoid an ordering problem
+          is(test.pcRemote._pc.getReceivers().length, 2,
+             "pcRemote should have two receivers");
+          return Promise.all(test.pcRemote._pc.getReceivers().map(r => {
+            const analyser = new AudioStreamAnalyser(
+                new AudioContext(), new MediaStream([r.track]));
+            const freq = analyser.binIndexForFrequency(TEST_AUDIO_FREQ);
+            return analyser.waitForAnalysisSuccess(arr => arr[freq] > 200);
+          }));
+        },
+      ]
+    );
+
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_addSecondAudioStreamNoBundle.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_addSecondAudioStreamNoBundle.html
new file mode 100644
index 0000000000..32d0564717
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_addSecondAudioStreamNoBundle.html
@@ -0,0 +1,53 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: add second audio stream, no bundle"
+  });
+
+  runNetworkTest(function (options = {}) {
+    options.bundle = false;
+    const test = new PeerConnectionTest(options);
+    addRenegotiation(test.chain,
+      [
+        function PC_LOCAL_ADD_SECOND_STREAM(test) {
+          test.setMediaConstraints([{audio: true}, {audio: true}],
+                                   [{audio: true}]);
+          // Since this is a NoBundle variant, adding a track will cause us to
+          // go back to checking.
+          test.pcLocal.expectIceChecking();
+          return test.pcLocal.getAllUserMediaAndAddStreams([{audio: true}]);
+        },
+        function PC_REMOTE_EXPECT_ICE_CHECKING(test) {
+          test.pcRemote.expectIceChecking();
+        },
+      ],
+      [
+        function PC_REMOTE_CHECK_ADDED_TRACK(test) {
+          // We test both tracks to avoid an ordering problem
+          is(test.pcRemote._pc.getReceivers().length, 2,
+             "pcRemote should have two receivers");
+          return Promise.all(test.pcRemote._pc.getReceivers().map(r => {
+            const analyser = new AudioStreamAnalyser(
+                new AudioContext(), new MediaStream([r.track]));
+            const freq = analyser.binIndexForFrequency(TEST_AUDIO_FREQ);
+            return analyser.waitForAnalysisSuccess(arr => arr[freq] > 200);
+          }));
+        },
+      ]
+    );
+
+    // TODO(bug 1093835): figure out how to verify if media flows through the new stream
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_addSecondVideoStream.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_addSecondVideoStream.html
new file mode 100644
index 0000000000..1565958d01
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_addSecondVideoStream.html
@@ -0,0 +1,53 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: add second video stream"
+  });
+
+  runNetworkTest(async function (options) {
+    // Use fake video here since the native fake device on linux doesn't
+    // change color as needed by checkVideoPlaying() below.
+    await pushPrefs(
+      ['media.video_loopback_dev', ''],
+      ['media.navigator.streams.fake', true]);
+    // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+    if (navigator.userAgent.includes("Android")) {
+      await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                      ["media.webrtc.hw.h264.enabled", false]);
+    }
+
+    const test = new PeerConnectionTest(options);
+    addRenegotiation(test.chain,
+      [
+        function PC_LOCAL_ADD_SECOND_STREAM(test) {
+          test.setMediaConstraints([{video: true}, {video: true}],
+                                   [{video: true}]);
+          return test.pcLocal.getAllUserMediaAndAddStreams([{video: true}]);
+        },
+      ],
+      [
+        function PC_REMOTE_CHECK_VIDEO_FLOW(test) {
+          const h = new VideoStreamHelper();
+          is(test.pcRemote.remoteMediaElements.length, 2,
+             "Should have two remote media elements after renegotiation");
+          return Promise.all(test.pcRemote.remoteMediaElements.map(video =>
+            h.checkVideoPlaying(video)));
+        },
+      ]
+    );
+
+    test.setMediaConstraints([{video: true, fake: true}], [{video: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_addSecondVideoStreamNoBundle.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_addSecondVideoStreamNoBundle.html
new file mode 100644
index 0000000000..2857100998
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_addSecondVideoStreamNoBundle.html
@@ -0,0 +1,60 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: add second video stream, no bundle"
+  });
+
+  runNetworkTest(async function (options = {}) {
+    // Use fake video here since the native fake device on linux doesn't
+    // change color as needed by checkVideoPlaying() below.
+    await pushPrefs(
+      ['media.video_loopback_dev', ''],
+      ['media.navigator.streams.fake', true]);
+    // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+    if (navigator.userAgent.includes("Android")) {
+      await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                      ["media.webrtc.hw.h264.enabled", false]);
+    }
+
+    options.bundle = false;
+    const test = new PeerConnectionTest(options);
+    addRenegotiation(test.chain,
+      [
+        function PC_LOCAL_ADD_SECOND_STREAM(test) {
+          test.setMediaConstraints([{video: true}, {video: true}],
+                                   [{video: true}]);
+          // Since this is a NoBundle variant, adding a track will cause us to
+          // go back to checking.
+          test.pcLocal.expectIceChecking();
+          return test.pcLocal.getAllUserMediaAndAddStreams([{video: true}]);
+        },
+        function PC_REMOTE_EXPECT_ICE_CHECKING(test) {
+          test.pcRemote.expectIceChecking();
+        },
+      ],
+      [
+        function PC_REMOTE_CHECK_VIDEO_FLOW(test) {
+          const h = new VideoStreamHelper();
+          is(test.pcRemote.remoteMediaElements.length, 2,
+             "Should have two remote media elements after renegotiation");
+          return Promise.all(test.pcRemote.remoteMediaElements.map(video =>
+            h.checkVideoPlaying(video)));
+        },
+      ]
+    );
+
+    test.setMediaConstraints([{video: true}], [{video: true}]);
+    await test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_addtrack_removetrack_events.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_addtrack_removetrack_events.html
new file mode 100644
index 0000000000..ff9ca9a772
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_addtrack_removetrack_events.html
@@ -0,0 +1,75 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+"use strict";
+
+createHTML({
+  title: "MediaStream's 'addtrack' and 'removetrack' events with gUM",
+  bug: "1208328"
+});
+
+runNetworkTest(function (options) {
+  let test = new PeerConnectionTest(options);
+  let eventsPromise;
+  addRenegotiation(test.chain,
+    [
+      function PC_LOCAL_SWAP_VIDEO_TRACKS(test) {
+        return getUserMedia({video: true}).then(stream => {
+          var videoTransceiver = test.pcLocal._pc.getTransceivers()[1];
+          is(videoTransceiver.currentDirection, "sendonly",
+             "Video transceiver's current direction is sendonly");
+          is(videoTransceiver.direction, "sendrecv",
+             "Video transceiver's desired direction is sendrecv");
+
+          const localStream = test.pcLocal._pc.getLocalStreams()[0];
+          ok(localStream, "Should have local stream");
+
+          const remoteStream = test.pcRemote._pc.getRemoteStreams()[0];
+          ok(remoteStream, "Should have remote stream");
+
+          const newTrack = stream.getTracks()[0];
+
+          const videoSenderIndex =
+            test.pcLocal._pc.getSenders().findIndex(s => s.track.kind == "video");
+          isnot(videoSenderIndex, -1, "Should have video sender");
+
+          test.pcLocal.removeSender(videoSenderIndex);
+          is(videoTransceiver.direction, "recvonly",
+             "Video transceiver should be recvonly after removeTrack");
+          test.pcLocal.attachLocalTrack(stream.getTracks()[0], localStream);
+          is(videoTransceiver.direction, "recvonly",
+             "Video transceiver should be recvonly after addTrack");
+
+          eventsPromise = haveEvent(remoteStream, "addtrack",
+              wait(50000, new Error("No addtrack event for " + newTrack.id)))
+            .then(trackEvent => {
+              ok(trackEvent instanceof MediaStreamTrackEvent,
+                 "Expected event to be instance of MediaStreamTrackEvent");
+              is(trackEvent.type, "addtrack",
+                 "Expected addtrack event type");
+              is(trackEvent.track.readyState, "live",
+                 "added track should be live");
+            })
+            .then(() => haveNoEvent(remoteStream, "addtrack"));
+        });
+      },
+    ],
+    [
+      function PC_REMOTE_CHECK_EVENTS(test) {
+        return eventsPromise;
+      },
+    ]
+  );
+
+  test.setMediaConstraints([{audio: true, video: true}], []);
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_answererAddSecondAudioStream.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_answererAddSecondAudioStream.html
new file mode 100644
index 0000000000..d9b01bf722
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_answererAddSecondAudioStream.html
@@ -0,0 +1,32 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: answerer adds second audio stream"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    addRenegotiationAnswerer(test.chain,
+      [
+        function PC_LOCAL_ADD_SECOND_STREAM(test) {
+          test.setMediaConstraints([{audio: true}, {audio: true}],
+                                   [{audio: true}]);
+          return test.pcLocal.getAllUserMediaAndAddStreams([{audio: true}]);
+        },
+      ]
+    );
+
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_audioChannels.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioChannels.html
new file mode 100644
index 0000000000..f6e77f8271
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioChannels.html
@@ -0,0 +1,102 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+
+createHTML({
+  bug: "1765005",
+  title: "Verify audio channel limits for each negotiated audio codec",
+});
+
+const matchesChannels = (sdp, codec, channels) =>
+  !!sdp.match(new RegExp(`a=rtpmap:\\d* ${codec}\/\\d*\/${channels}\r\n`, "g")) ||
+  (channels <= 1 &&
+   !!sdp.match(new RegExp(`a=rtpmap:\\d* ${codec}\/\\d*\r\n`, "g")));
+
+async function testAudioChannels(track, codec, channels, accepted, expectedChannels) {
+  const pc1 = new RTCPeerConnection();
+  const pc2 = new RTCPeerConnection();
+  try {
+    pc1.addTrack(track);
+    pc1.onicecandidate = e => pc2.addIceCandidate(e.candidate);
+    pc2.onicecandidate = e => pc1.addIceCandidate(e.candidate);
+    await pc1.setLocalDescription();
+    await pc2.setRemoteDescription(pc1.localDescription);
+    let {type, sdp} = await pc2.createAnswer();
+    sdp = sdp.replace(new RegExp(`a=rtpmap:(\\d*) ${codec}\/(\\d*)\/?\\d*\r\n`, "g"),
+                                 `a=rtpmap:$1 ${codec}/$2/${channels}\r\n`);
+    const payloadType = Number(sdputils.findCodecId(sdp, codec));
+    sdp = sdputils.removeAllButPayloadType(sdp, payloadType);
+    ok(matchesChannels(sdp, codec, channels), "control");
+    await pc2.setLocalDescription({type, sdp});
+    is(matchesChannels(pc2.localDescription.sdp, codec, channels),
+       accepted,
+       `test pc2.localDescription`);
+    try {
+      await pc1.setRemoteDescription(pc2.localDescription);
+      ok(expectedChannels, "SRD should succeed iff we're expecting channels");
+      const [receiver] = pc2.getReceivers();
+      await new Promise(r => receiver.track.onunmute = r);
+      let stats = await receiver.getStats();
+      let inboundStat = [...stats.values()].find(({type}) => type == "inbound-rtp");
+      if (!inboundStat) {
+        info("work around bug 1765215"); // TODO bug 1765215
+        await new Promise(r => setTimeout(r, 200));
+        stats = await receiver.getStats();
+        inboundStat = [...stats.values()].find(({type}) => type == "inbound-rtp");
+      }
+      ok(inboundStat, "has inbound-rtp stats after track unmute (w/workaround)");
+      const codecStat = stats.get(inboundStat.codecId);
+      ok(codecStat.mimeType.includes(codec), "correct codec");
+      is(codecStat.payloadType, payloadType, "correct payloadType");
+      is(codecStat.channels, expectedChannels, "expected channels");
+    } catch (e) {
+      ok(!expectedChannels, "SRD should fail iff we're not expecting channels");
+    }
+  } finally {
+    pc1.close();
+    pc2.close();
+  }
+}
+
+runNetworkTest(async () => {
+  const [track] = (await navigator.mediaDevices.getUserMedia({audio: true}))
+      .getAudioTracks();
+  try {
+    for (let [codec, channels, accepted, expectedChannels] of [
+      ["opus", 2, true, 2],
+      ["opus", 1, true, 0],
+      ["opus", 1000, true, 0],
+      ["G722", 1, true, 1],
+      ["G722", 2, true, 0],
+      ["G722", 1000, true, 0],
+      ["PCMU", 1, true, 1],
+      ["PCMU", 2, false, 1],
+      ["PCMU", 1000, false, 1],
+      ["PCMA", 1, true, 1],
+      ["PCMA", 2, false, 1],
+      ["PCMA", 1000, false, 1]
+    ]) {
+      const testName = `${codec} with ${channels} channel(s) is ` +
+          `${accepted? "accepted" : "ignored"} and produces ` +
+          `${expectedChannels || "no"} channels`;
+      try {
+        info(`Testing that ${testName}`);
+        await testAudioChannels(track, codec, channels, accepted, expectedChannels);
+      } catch (e) {
+        ok(false, `Error testing that ${testName}: ${e}\n${e.stack}`);
+      }
+    }
+  } finally {
+    track.stop();
+  }
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_audioCodecs.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioCodecs.html
new file mode 100644
index 0000000000..8874436e3b
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioCodecs.html
@@ -0,0 +1,81 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1395853",
+    title: "Verify audio content over WebRTC for every audio codec",
+  });
+
+  // We match the format member against the sdp to figure out the payload type,
+  // So all other present codecs can be removed.
+  const codecs = [ "opus", "G722", "PCMU", "PCMA" ];
+
+  async function testAudioCodec(options = {}, codec) {
+    // sdputils checks for opus as part of its sdp sanity test
+    options.opus = codec == "opus";
+
+    let test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}], []);
+
+    test.chain.insertBefore("PC_LOCAL_SET_LOCAL_DESCRIPTION", [
+      function PC_LOCAL_FILTER_OUT_CODECS() {
+        let otherCodec = codecs.find(c => c != codec);
+        let otherId = sdputils.findCodecId(test.originalOffer.sdp, otherCodec);
+
+        let id = sdputils.findCodecId(test.originalOffer.sdp, codec);
+        test.originalOffer.sdp =
+          sdputils.removeAllButPayloadType(test.originalOffer.sdp, id);
+
+        ok(!test.originalOffer.sdp.match(new RegExp(`m=.*UDP/TLS/RTP/SAVPF.* ${otherId}[^0-9]`, "gi")),
+          `Other codec ${otherId} should be removed after filtering`);
+        ok(test.originalOffer.sdp.match(new RegExp(`m=.*UDP/TLS/RTP/SAVPF.* ${id}[^0-9]`, "gi")),
+          `Tested codec ${id} should remain after filtering`);
+
+        for (let c of codecs.filter(c => c != codec)) {
+          // Remove rtpmaps for the other codecs so sdp sanity tests pass.
+          let id = sdputils.findCodecId(test.originalOffer.sdp, c);
+          test.originalOffer.sdp =
+            sdputils.removeRtpMapForPayloadType(test.originalOffer.sdp, id);
+        }
+
+        ok(!test.originalOffer.sdp.match(new RegExp(`a=rtpmap:${otherId}.*\\r\\n`, "gi")),
+           `Rtpmap of other codec ${otherId} should be removed after filtering`);
+        ok(test.originalOffer.sdp.match(new RegExp(`a=rtpmap:${id}.*\\r\\n`, "gi")),
+           `Rtpmap of tested codec should remain after filtering`);
+      },
+    ]);
+
+    test.chain.append([
+      async function CHECK_AUDIO_FLOW() {
+        try {
+          await test.pcRemote.checkReceivingToneFrom(new AudioContext(), test.pcLocal);
+          ok(true, "input and output audio data matches");
+        } catch(e) {
+          ok(false, `No audio flow: ${e}`);
+        }
+      },
+    ]);
+
+    await test.run();
+  }
+
+  runNetworkTest(async (options) => {
+    for (let codec of codecs) {
+      info(`Testing audio for codec ${codec}`);
+      try {
+        await testAudioCodec(options, codec);
+      } catch(e) {
+        ok(false, `Error in test for codec ${codec}: ${e}\n${e.stack}`);
+      }
+      info(`Tested audio for codec ${codec}`);
+    }
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_audioContributingSources.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioContributingSources.html
new file mode 100644
index 0000000000..333b40a888
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioContributingSources.html
@@ -0,0 +1,144 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1363667",
+    title: "Test audio receiver getContributingSources"
+  });
+
+  // test_peerConnection_audioSynchronizationSources.html tests
+  // much of the functionality of getContributingSources as the implementation
+  // is shared.
+  var testGetContributingSources = async (test) => {
+    const remoteReceiver = test.pcRemote.getReceivers()[0];
+    const localReceiver = test.pcLocal.getReceivers()[0];
+
+    // Check that getContributingSources is empty as there is no MCU
+    is(remoteReceiver.getContributingSources().length, 0,
+       "remote contributing sources is empty");
+    is(localReceiver.getContributingSources().length, 0,
+       "local contributing sources is empty");
+    // Wait for the next JS event loop iteration, to clear the cache
+    await Promise.resolve().then();
+    // Insert new entries as if there were an MCU
+    const csrc0 = 124756;
+    const timestamp0 = performance.now() + performance.timeOrigin;
+    const rtpTimestamp0 = 11111;
+    const hasAudioLevel0 = true;
+    // Audio level as expected to be received in RTP
+    const audioLevel0 = 34;
+    // Audio level as expected to be returned
+    const expectedAudioLevel0 = 10 ** (-audioLevel0 / 20);
+
+    SpecialPowers.wrap(remoteReceiver).mozInsertAudioLevelForContributingSource(
+        csrc0,
+        timestamp0,
+        rtpTimestamp0,
+        hasAudioLevel0,
+        audioLevel0);
+
+    const csrc1 = 5786;
+    const timestamp1 = timestamp0 - 200;
+    const rtpTimestamp1 = 22222;
+    const hasAudioLevel1 = false;
+    const audioLevel1 = 0;
+
+    SpecialPowers.wrap(remoteReceiver).mozInsertAudioLevelForContributingSource(
+        csrc1,
+        timestamp1,
+        rtpTimestamp1,
+        hasAudioLevel1,
+        audioLevel1);
+
+    const csrc2 = 93487;
+    const timestamp2 = timestamp0 - 200;
+    const rtpTimestamp2 = 333333;
+    const hasAudioLevel2 = true;
+    const audioLevel2 = 127;
+
+    SpecialPowers.wrap(remoteReceiver).mozInsertAudioLevelForContributingSource(
+        csrc2,
+        timestamp2,
+        rtpTimestamp2,
+        hasAudioLevel2,
+        audioLevel2);
+
+    const contributingSources = remoteReceiver.getContributingSources();
+    is(contributingSources.length, 3,
+       "Expected number of contributing sources");
+
+    // Check that both inserted were returned
+    const source0 = contributingSources.find(c => c.source == csrc0);
+    ok(source0, "first csrc was found");
+
+    const source1 = contributingSources.find(c => c.source == csrc1);
+    ok(source1, "second csrsc was found");
+
+    // Add a small margin of error in the timestamps
+    const compareTimestamps = (ts1, ts2) => Math.abs(ts1 - ts2) < 100;
+
+    // Check the CSRC with audioLevel
+    const isWithinErr = Math.abs(source0.audioLevel - expectedAudioLevel0)
+        < expectedAudioLevel0 / 50;
+    ok(isWithinErr,
+       `Contributing source has correct audio level. (${source0.audioLevel})`);
+    ok(compareTimestamps(source0.timestamp, timestamp0),
+        `Contributing source has correct timestamp (got ${source0.timestamp}), expected ${timestamp0}`);
+    is(source0.rtpTimestamp, rtpTimestamp0,
+       `Contributing source has correct RTP timestamp (${source0.rtpTimestamp}`);
+    // Check the CSRC without audioLevel
+    is(source1.audioLevel, undefined,
+       `Contributing source has no audio level. (${source1.audioLevel})`);
+    ok(compareTimestamps(source1.timestamp, timestamp1),
+        `Contributing source has correct timestamp (got ${source1.timestamp}, expected ${timestamp1})`);
+    is(source1.rtpTimestamp, rtpTimestamp1,
+       `Contributing source has correct RTP timestamp (${source1.rtpTimestamp}`);
+    // Check that a received RTP audio level 127 is exactly 0
+    const source2 = contributingSources.find(c => c.source == csrc2);
+    ok(source2, "third csrc was found");
+    is(source2.audioLevel, 0,
+      `Contributing source has audio level of 0 when RTP audio level is 127`);
+    // Check caching
+    is(JSON.stringify(contributingSources),
+       JSON.stringify(remoteReceiver.getContributingSources()),
+       "getContributingSources is cached");
+    // Check that sources are sorted in descending order by time stamp
+    const timestamp3 =  performance.now() + performance.timeOrigin;
+    const rtpTimestamp3 = 44444;
+    // Larger offsets are further back in time
+    const testOffsets = [3, 7, 5, 6, 1, 4];
+    for (const offset of testOffsets) {
+      SpecialPowers.wrap(localReceiver).mozInsertAudioLevelForContributingSource(
+          offset, // Using offset for SSRC for convenience
+          timestamp3 - offset,
+          rtpTimestamp3,
+          true,
+          offset);
+    }
+    const sources = localReceiver.getContributingSources();
+    const sourceOffsets = sources.map(s => s.source);
+    is(JSON.stringify(sourceOffsets),
+       JSON.stringify([...testOffsets].sort((a, b) => a - b)),
+          `Contributing sources are sorted in descending order by timestamp:`
+          + ` ${JSON.stringify(sources)}`);
+  };
+
+  var test;
+  runNetworkTest(async function(options) {
+    test = new PeerConnectionTest(options);
+    test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW",
+      [testGetContributingSources]);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    test.pcLocal.audioElementsOnly = true;
+    await pushPrefs(["privacy.reduceTimerPrecision", false]);
+    await test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_audioRenegotiationInactiveAnswer.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioRenegotiationInactiveAnswer.html
new file mode 100644
index 0000000000..6d3a23b57a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioRenegotiationInactiveAnswer.html
@@ -0,0 +1,69 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="sdpUtils.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1213773",
+    title: "Renegotiation: answerer uses a=inactive for audio"
+  });
+
+  runNetworkTest(function (options) {
+    const helper = new AudioStreamHelper();
+
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}], []);
+    let haveFirstUnmuteEvent;
+
+    test.chain.insertBefore("PC_REMOTE_SET_LOCAL_DESCRIPTION", [
+      function PC_REMOTE_SETUP_ONUNMUTE_1() {
+        haveFirstUnmuteEvent = haveEvent(test.pcRemote._pc.getReceivers()[0].track, "unmute");
+      }
+    ]);
+
+    test.chain.append([
+      function PC_REMOTE_CHECK_AUDIO_UNMUTED() {
+        return haveFirstUnmuteEvent;
+      },
+      function PC_REMOTE_CHECK_AUDIO_FLOWING() {
+        return helper.checkAudioFlowing(test.pcRemote._pc.getRemoteStreams()[0]);
+      }
+    ]);
+
+    addRenegotiation(test.chain, []);
+
+    test.chain.insertAfter("PC_LOCAL_GET_ANSWER", [
+        function PC_LOCAL_REWRITE_REMOTE_SDP_INACTIVE(test) {
+          test._remote_answer.sdp =
+            sdputils.setAllMsectionsInactive(test._remote_answer.sdp);
+        }
+    ], false, 1);
+
+    test.chain.append([
+      function PC_REMOTE_CHECK_AUDIO_NOT_FLOWING() {
+        return helper.checkAudioNotFlowing(test.pcRemote._pc.getRemoteStreams()[0]);
+      }
+    ]);
+
+    test.chain.remove("PC_REMOTE_CHECK_STATS", 1);
+    test.chain.remove("PC_LOCAL_CHECK_STATS", 1);
+    test.chain.remove("PC_REMOTE_WAIT_FOR_MEDIA_FLOW", 1);
+
+    addRenegotiation(test.chain, []);
+
+    test.chain.append([
+      function PC_REMOTE_CHECK_AUDIO_FLOWING_2() {
+        return helper.checkAudioFlowing(test.pcRemote._pc.getRemoteStreams()[0]);
+      }
+    ]);
+
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_audioSynchronizationSources.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioSynchronizationSources.html
new file mode 100644
index 0000000000..32603b2e40
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioSynchronizationSources.html
@@ -0,0 +1,95 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1363667",
+    title: "Test audio receiver getSynchronizationSources"
+  });
+
+  var waitForSyncSources = async (test) => {
+    let receivers = [...test.pcRemote.getReceivers(),
+                     ...test.pcLocal.getReceivers()];
+    is(receivers.length, 2, "Expected number of receivers");
+    // Wait for sync sources
+    while (true) {
+        if (receivers[0].getSynchronizationSources().length &&
+            receivers[1].getSynchronizationSources().length) {
+        break;
+      }
+      await wait(250);
+    }
+  };
+
+  var testGetSynchronizationSources = async (test) => {
+    await waitForSyncSources(test);
+    let receivers = [...test.pcRemote.getReceivers(),
+                     ...test.pcLocal.getReceivers()];
+    is(receivers.length, 2,
+       `Expected number of receivers is 2. (${receivers.length})`);
+    for (let recv of receivers) {
+      let syncSources = recv.getSynchronizationSources();
+      ok(syncSources,
+         "Receiver has Synchronization sources " + JSON.stringify(syncSources));
+      is(syncSources.length, 1, "Each receiver has only a single sync source");
+      let source = recv.getSynchronizationSources()[0];
+      ok(source.audioLevel !== null,
+         `Synchronization source has audio level. (${source.audioLevel})`);
+      ok(source.audioLevel >= 0.0,
+         `Synchronization source audio level >= 0.0 (${source.audioLevel})`);
+      ok(source.audioLevel <= 1.0,
+         `Synchronization source audio level <= 1.0 (${source.audioLevel})`);
+      ok(source.timestamp,
+         `Synchronization source has timestamp (${source.timestamp})`);
+      const ageSeconds =
+        (window.performance.now() + window.performance.timeOrigin -
+         source.timestamp) / 1000;
+      ok(ageSeconds >= 0,
+         `Synchronization source timestamp is in the past`);
+      ok(ageSeconds < 2.5,
+         `Synchronization source timestamp is close to now`);
+      is(source.voiceActivityFlag, undefined,
+        "Synchronization source unsupported voiceActivity is undefined");
+    }
+  };
+
+  var testSynchronizationSourceCached = async (test) => {
+    await waitForSyncSources(test);
+    let receivers = [...test.pcRemote.getReceivers(),
+                     ...test.pcLocal.getReceivers()];
+    is(receivers.length, 2,
+       `Expected number of receivers is 2. (${receivers.length})`);
+    let sourceSets = [[],[]];
+    for (let sourceSet of sourceSets) {
+      for (let recv of receivers) {
+        let sources = recv.getSynchronizationSources();
+        is(sources.length, 1,
+           `Expected number of sources is 1. (${sources.length})`);
+        sourceSet.push(sources);
+      }
+      // Busy wait 1s before trying again
+      let endTime = performance.now() + 1000;
+      while (performance.now() < endTime) {};
+    }
+    is(JSON.stringify(sourceSets[0]), JSON.stringify(sourceSets[1]),
+       "Subsequent getSynchronizationSources calls are cached.");
+  };
+
+  var test;
+  runNetworkTest(function(options) {
+    test = new PeerConnectionTest(options);
+    test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW",
+      [testGetSynchronizationSources,
+       testSynchronizationSourceCached]);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    test.pcLocal.audioElementsOnly = true;
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_audioSynchronizationSourcesUnidirectional.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioSynchronizationSourcesUnidirectional.html
new file mode 100644
index 0000000000..6d66614e91
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_audioSynchronizationSourcesUnidirectional.html
@@ -0,0 +1,54 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1439001",
+    title: "Test audio unidirectional getSynchronizationSources"
+  });
+
+  var waitForSyncSources = async (test) => {
+    let receiver = test.pcRemote.getReceivers()[0];
+    ok(receiver, "Remote has a receiver");
+    // Wait for remote sync source
+    while (!receiver.getSynchronizationSources().length) {
+      await wait(250);
+    }
+    is(receiver.getSynchronizationSources().length, 1,
+       "Remote receiver has a synchronization source");
+    // Make sure local has no sync source
+    is(test.pcLocal.getReceivers()[0].getSynchronizationSources().length, 0,
+       "Local receiver has no synchronization source");
+  };
+  /*
+   * Test to make sure that in unidirectional calls, the receiving end has
+   * synchronization sources with audio levels, and the sending end has none.
+   */
+  var testGetSynchronizationSourcesUnidirectional = async (test) => {
+    await waitForSyncSources(test);
+    let receiver = test.pcRemote.getReceivers()[0];
+    let syncSources = receiver.getSynchronizationSources();
+    ok(syncSources.length,
+       "Receiver has Synchronization sources " + JSON.stringify(syncSources));
+    is(syncSources.length, 1, "Receiver has only a single sync source");
+    let syncSource = syncSources[0];
+    ok(syncSource.audioLevel !== undefined, "SynchronizationSource has audioLevel");
+  };
+
+  var test;
+  runNetworkTest(function(options) {
+    test = new PeerConnectionTest(options);
+    test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW",
+      [testGetSynchronizationSourcesUnidirectional]);
+    test.setMediaConstraints([{audio: true}], []);
+    test.pcLocal.audioElementsOnly = true;
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudio.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudio.html
new file mode 100644
index 0000000000..5fd10a67f9
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudio.html
@@ -0,0 +1,25 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "796892",
+    title: "Basic audio-only peer connection"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    // pc.js uses video elements by default, we want to test audio elements here
+    test.pcLocal.audioElementsOnly = true;
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioDynamicPtMissingRtpmap.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioDynamicPtMissingRtpmap.html
new file mode 100644
index 0000000000..a076bf80f1
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioDynamicPtMissingRtpmap.html
@@ -0,0 +1,36 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1246011",
+    title: "Offer with dynamic PT but missing rtpmap"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    options = options || { };
+    // we want Opus to get selected and 101 to be ignored
+    options.opus = true;
+    test = new PeerConnectionTest(options);
+    test.chain.insertBefore("PC_REMOTE_GET_OFFER", [
+      function PC_LOCAL_REDUCE_MLINE_REMOVE_RTPMAPS(test) {
+        test.originalOffer.sdp =
+          sdputils.reduceAudioMLineToDynamicPtAndOpus(test.originalOffer.sdp);
+        test.originalOffer.sdp =
+          sdputils.removeAllRtpMaps(test.originalOffer.sdp);
+        test.originalOffer.sdp = test.originalOffer.sdp + "a=rtpmap:109 opus/48000/2\r\n";
+        info("SDP with dyn PT and no Rtpmap: " + JSON.stringify(test.originalOffer));
+      }
+    ]);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelay.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelay.html
new file mode 100644
index 0000000000..180abc075a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelay.html
@@ -0,0 +1,47 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="nonTrickleIce.js"></script>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1231975",
+  title: "Basic audio-only peer connection with port dependent NAT, for verifying UDP relay"
+});
+
+// This test uses the NAT simulator, which doesn't work in https, so we turn
+// on getUserMedia in http, which requires a reload.
+if (!("mediaDevices" in navigator)) {
+  SpecialPowers.pushPrefEnv({set: [['media.devices.insecure.enabled', true]]},
+                            () => location.reload());
+} else {
+  runNetworkTest(async (options = {}) => {
+    await pushPrefs(
+        ['media.peerconnection.ice.obfuscate_host_addresses', false],
+        ['media.peerconnection.nat_simulator.filtering_type', 'PORT_DEPENDENT'],
+        ['media.peerconnection.nat_simulator.mapping_type', 'PORT_DEPENDENT'],
+        ['media.peerconnection.nat_simulator.block_tcp', true],
+        ['media.getusermedia.insecure.enabled', true]);
+    options.expectedLocalCandidateType = "srflx";
+    options.expectedRemoteCandidateType = "relay";
+    // If both have TURN, it is a toss-up which one will end up using a
+    // relay.
+    options.turn_disabled_local = true;
+    const test = new PeerConnectionTest(options);
+    // Make sure we don't end up choosing the wrong thing due to delays in
+    // trickle. Once we are willing to accept trickle after ICE success, we
+    // can maybe wait a bit to allow things to stabilize.
+    // TODO(bug 1238249)
+    makeOffererNonTrickle(test.chain);
+    makeAnswererNonTrickle(test.chain);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    await test.run();
+  }, { useIceServer: true });
+}
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelayTCP.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelayTCP.html
new file mode 100644
index 0000000000..7bb51764bd
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelayTCP.html
@@ -0,0 +1,42 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1231975",
+  title: "Basic audio-only peer connection with port dependent NAT that blocks UDP"
+});
+
+// This test uses the NAT simulator, which doesn't work in https, so we turn
+// on getUserMedia in http, which requires a reload.
+if (!("mediaDevices" in navigator)) {
+  SpecialPowers.pushPrefEnv({set: [['media.devices.insecure.enabled', true]]},
+                            () => location.reload());
+} else {
+  runNetworkTest(async (options = {}) => {
+    await pushPrefs(
+        ['media.peerconnection.ice.obfuscate_host_addresses', false],
+        ['media.peerconnection.nat_simulator.filtering_type', 'PORT_DEPENDENT'],
+        ['media.peerconnection.nat_simulator.mapping_type', 'PORT_DEPENDENT'],
+        ['media.peerconnection.nat_simulator.block_udp', true],
+        ['media.peerconnection.nat_simulator.block_tcp', false],
+        ['media.peerconnection.nat_simulator.block_tls', true],
+        ['media.peerconnection.ice.loopback', true],
+        ['media.getusermedia.insecure.enabled', true]);
+    options.expectedLocalCandidateType = "relay-tcp";
+    options.expectedRemoteCandidateType = "relay-tcp";
+    // No reason to wait for gathering to complete like the other NAT tests,
+    // since relayed-tcp is the only thing that can work.
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    await test.run();
+  }, { useIceServer: true });
+}
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelayTCPWithStun300.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelayTCPWithStun300.html
new file mode 100644
index 0000000000..43ea6aaea7
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelayTCPWithStun300.html
@@ -0,0 +1,54 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="nonTrickleIce.js"></script>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "857668",
+  title: "Basic audio-only peer connection with UDP-blocking NAT, for verifying TCP relay with STUN 300 responses"
+});
+
+// This test uses the NAT simulator, which doesn't work in https, so we turn
+// on getUserMedia in http, which requires a reload.
+if (!("mediaDevices" in navigator)) {
+  SpecialPowers.pushPrefEnv({set: [['media.devices.insecure.enabled', true]]},
+                            () => location.reload());
+} else {
+  runNetworkTest(async (options = {}) => {
+    await pushPrefs(
+        ['media.peerconnection.ice.obfuscate_host_addresses', false],
+        ['media.peerconnection.nat_simulator.filtering_type', 'ENDPOINT_INDEPENDENT'],
+        ['media.peerconnection.nat_simulator.mapping_type', 'ENDPOINT_INDEPENDENT'],
+        ['media.peerconnection.nat_simulator.block_udp', true],
+        ['media.peerconnection.nat_simulator.block_tcp', false],
+        ['media.peerconnection.ice.loopback', true],
+        ['media.getusermedia.insecure.enabled', true]);
+    options.expectedLocalCandidateType = "relay-tcp";
+    options.expectedRemoteCandidateType = "relay-tcp";
+    const turnServer = iceServersArray.find(server => "username" in server);
+    const turnRedirectPort = turnServer.turn_redirect_port;
+    const turnHostname = getTurnHostname(turnServer.urls[0]);
+    turnServer.urls = [`turn:${turnHostname}:${turnRedirectPort}?transport=tcp`];
+    // Override turn servers so we can test simulated redirects
+    options.config_local = {iceServers: [turnServer]};
+    options.config_remote = {iceServers: [turnServer]};
+    const test = new PeerConnectionTest(options);
+    // Make sure we don't end up choosing the wrong thing due to delays in
+    // trickle. Once we are willing to accept trickle after ICE success, we
+    // can maybe wait a bit to allow things to stabilize.
+    // TODO(bug 1238249)
+    makeOffererNonTrickle(test.chain);
+    makeAnswererNonTrickle(test.chain);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    await test.run();
+    await SpecialPowers.popPrefEnv();
+  }, { useIceServer: true });
+}
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelayTLS.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelayTLS.html
new file mode 100644
index 0000000000..7446401f87
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelayTLS.html
@@ -0,0 +1,41 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1231975",
+  title: "Basic audio-only peer connection with port dependent NAT that blocks STUN"
+});
+
+// This test uses the NAT simulator, which doesn't work in https, so we turn
+// on getUserMedia in http, which requires a reload.
+if (!("mediaDevices" in navigator)) {
+  SpecialPowers.pushPrefEnv({set: [['media.devices.insecure.enabled', true]]},
+                            () => location.reload());
+} else {
+  runNetworkTest(async (options = {}) => {
+    await pushPrefs(
+        ['media.peerconnection.ice.obfuscate_host_addresses', false],
+        ['media.peerconnection.nat_simulator.filtering_type', 'PORT_DEPENDENT'],
+        ['media.peerconnection.nat_simulator.mapping_type', 'PORT_DEPENDENT'],
+        ['media.peerconnection.nat_simulator.block_udp', true],
+        ['media.peerconnection.nat_simulator.block_tcp', true],
+        ['media.peerconnection.ice.loopback', true],
+        ['media.getusermedia.insecure.enabled', true]);
+    options.expectedLocalCandidateType = "relay-tls";
+    options.expectedRemoteCandidateType = "relay-tls";
+    // No reason to wait for gathering to complete like the other NAT tests,
+    // since relayed-tcp is the only thing that can work.
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    await test.run();
+  }, { useIceServer: true });
+}
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelayWithStun300.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelayWithStun300.html
new file mode 100644
index 0000000000..286e67bc2f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATRelayWithStun300.html
@@ -0,0 +1,53 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="nonTrickleIce.js"></script>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "857668",
+  title: "Basic audio-only peer connection with port dependent NAT, for verifying UDP relay with STUN 300 responses"
+});
+
+// This test uses the NAT simulator, which doesn't work in https, so we turn
+// on getUserMedia in http, which requires a reload.
+if (!("mediaDevices" in navigator)) {
+  SpecialPowers.pushPrefEnv({set: [['media.devices.insecure.enabled', true]]},
+                            () => location.reload());
+} else {
+  runNetworkTest(async (options = {}) => {
+    await pushPrefs(
+        ['media.peerconnection.ice.obfuscate_host_addresses', false],
+        ['media.peerconnection.nat_simulator.filtering_type', 'PORT_DEPENDENT'],
+        ['media.peerconnection.nat_simulator.mapping_type', 'PORT_DEPENDENT'],
+        ['media.peerconnection.nat_simulator.block_tcp', true],
+        ['media.getusermedia.insecure.enabled', true]);
+    options.expectedLocalCandidateType = "srflx";
+    options.expectedRemoteCandidateType = "relay";
+    const turnServer = iceServersArray.find(server => "username" in server);
+    const turnRedirectPort = turnServer.turn_redirect_port;
+    const turnHostname = getTurnHostname(turnServer.urls[0]);
+    turnServer.urls = [`turn:${turnHostname}:${turnRedirectPort}`];
+    // Override turn servers so we can test redirects
+    options.config_remote = {iceServers: [turnServer]};
+    // If both have TURN, it is a toss-up which one will end up using a
+    // relay, so we disable TURN for one side.
+    options.turn_disabled_local = true;
+    const test = new PeerConnectionTest(options);
+    // Make sure we don't end up choosing the wrong thing due to delays in
+    // trickle. Once we are willing to accept trickle after ICE success, we
+    // can maybe wait a bit to allow things to stabilize.
+    // TODO(bug 1238249)
+    makeOffererNonTrickle(test.chain);
+    makeAnswererNonTrickle(test.chain);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    await test.run();
+  }, { useIceServer: true });
+}
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATSrflx.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATSrflx.html
new file mode 100644
index 0000000000..78fa8bcb2c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNATSrflx.html
@@ -0,0 +1,44 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="nonTrickleIce.js"></script>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1231975",
+  title: "Basic audio-only peer connection with endpoint independent NAT, for verifying UDP srflx"
+});
+
+// This test uses the NAT simulator, which doesn't work in https, so we turn
+// on getUserMedia in http, which requires a reload.
+if (!("mediaDevices" in navigator)) {
+  SpecialPowers.pushPrefEnv({set: [['media.devices.insecure.enabled', true]]},
+                            () => location.reload());
+} else {
+  runNetworkTest(async (options = {}) => {
+    await pushPrefs(
+        ['media.peerconnection.ice.obfuscate_host_addresses', false],
+        ['media.peerconnection.nat_simulator.filtering_type', 'ENDPOINT_INDEPENDENT'],
+        ['media.peerconnection.nat_simulator.mapping_type', 'ENDPOINT_INDEPENDENT'],
+        ['media.peerconnection.nat_simulator.block_tcp', true],
+        ['media.getusermedia.insecure.enabled', true]);
+    options.expectedLocalCandidateType = "srflx";
+    options.expectedRemoteCandidateType = "srflx";
+    const test = new PeerConnectionTest(options);
+    // Make sure we don't end up choosing the wrong thing due to delays in
+    // trickle. Once we are willing to accept trickle after ICE success, we
+    // can maybe wait a bit to allow things to stabilize.
+    // TODO(bug 1238249)
+    makeOffererNonTrickle(test.chain);
+    makeAnswererNonTrickle(test.chain);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    await test.run();
+  }, { useIceServer: true });
+}
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNoisyUDPBlock.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNoisyUDPBlock.html
new file mode 100644
index 0000000000..297121cd94
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioNoisyUDPBlock.html
@@ -0,0 +1,41 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1231975",
+  title: "Basic audio-only peer connection where UDP sockets return errors on send"
+});
+
+// This test uses the NAT simulator, which doesn't work in https, so we turn
+// on getUserMedia in http, which requires a reload.
+if (!("mediaDevices" in navigator)) {
+  SpecialPowers.pushPrefEnv({set: [['media.devices.insecure.enabled', true]]},
+                            () => location.reload());
+} else {
+  runNetworkTest(async (options = {}) => {
+    await pushPrefs(
+        ['media.peerconnection.ice.obfuscate_host_addresses', false],
+        ['media.peerconnection.nat_simulator.filtering_type', 'PORT_DEPENDENT'],
+        ['media.peerconnection.nat_simulator.mapping_type', 'PORT_DEPENDENT'],
+        ['media.peerconnection.nat_simulator.block_udp', true],
+        ['media.peerconnection.nat_simulator.error_code_for_drop', 3 /*R_INTERNAL*/],
+        ['media.peerconnection.nat_simulator.block_tls', true],
+        ['media.getusermedia.insecure.enabled', true]);
+    options.expectedLocalCandidateType = "relay-tcp";
+    options.expectedRemoteCandidateType = "relay-tcp";
+    // No reason to wait for gathering to complete like the other NAT tests,
+    // since relayed-tcp is the only thing that can work.
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    await test.run();
+  }, { useIceServer: true });
+}
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioPcmaPcmuOnly.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioPcmaPcmuOnly.html
new file mode 100644
index 0000000000..f0fe721b8e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioPcmaPcmuOnly.html
@@ -0,0 +1,39 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1221837",
+    title: "Only offer PCMA and PMCU in mline (no rtpmaps)"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    options = options || { };
+    options.opus = false;
+    test = new PeerConnectionTest(options);
+    test.chain.insertBefore("PC_REMOTE_GET_OFFER", [
+      function PC_LOCAL_REDUCE_MLINE_REMOVE_RTPMAPS(test) {
+        test.originalOffer.sdp =
+          sdputils.reduceAudioMLineToPcmuPcma(test.originalOffer.sdp);
+        test.originalOffer.sdp =
+          sdputils.removeAllRtpMaps(test.originalOffer.sdp);
+        info("SDP without Rtpmaps: " + JSON.stringify(test.originalOffer));
+      }
+    ]);
+    test.chain.insertAfter("PC_REMOTE_SANE_LOCAL_SDP", [
+      function PC_REMOTE_VERIFY_PCMU(test) {
+        ok(test._remote_answer.sdp.includes("a=rtpmap:0 PCMU/8000"), "PCMU codec is present in SDP");
+      }
+    ]);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioRelayPolicy.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioRelayPolicy.html
new file mode 100644
index 0000000000..ced57ff8a3
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioRelayPolicy.html
@@ -0,0 +1,83 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1663746",
+  title: "Basic tests for relay ice policy"
+});
+
+runNetworkTest(async () => {
+  await pushPrefs(
+      // Enable mDNS, since there are some checks we want to run with that
+      ['media.peerconnection.ice.obfuscate_host_addresses', true]);
+
+  const offerer = new RTCPeerConnection({iceServers: iceServersArray, iceTransportPolicy: 'relay'});
+  const answerer = new RTCPeerConnection({iceServers: iceServersArray});
+
+  offerer.onicecandidate = e => {
+    if (e.candidate) {
+      ok(!e.candidate.candidate.includes(' host '), 'IceTransportPolicy \"relay\" should prevent the advertisement of host candidates');
+      ok(!e.candidate.candidate.includes(' srflx '), 'IceTransportPolicy \"relay\" should prevent the advertisement of srflx candidates');
+    }
+    answerer.addIceCandidate(e.candidate);
+  };
+
+  answerer.onicecandidate = e => {
+    if (e.candidate && e.candidate.candidate.includes(' host ')) {
+      ok(e.candidate.candidate.includes('.local'), 'When obfuscate_host_addresses is true, we expect host candidates to use mDNS');
+    }
+    offerer.addIceCandidate(e.candidate);
+  };
+
+  const offererConnected = new Promise(r => {
+    offerer.oniceconnectionstatechange = () => {
+      if (offerer.iceConnectionState == 'connected') {
+        r();
+      }
+    };
+  });
+
+  const answererConnected = new Promise(r => {
+    answerer.oniceconnectionstatechange = () => {
+      if (answerer.iceConnectionState == 'connected') {
+        r();
+      }
+    };
+  });
+
+  const offer = await offerer.createOffer({offerToReceiveAudio: true});
+  await Promise.all([offerer.setLocalDescription(offer), answerer.setRemoteDescription(offer)]);
+  const answer = await answerer.createAnswer();
+  await Promise.all([answerer.setLocalDescription(answer), offerer.setRemoteDescription(answer)]);
+
+  info('Waiting for ICE to connect');
+  await Promise.all([offererConnected, answererConnected]);
+
+  const offererStats = await offerer.getStats();
+  const localCandidates = [...offererStats.values()].filter(stat => stat.type == 'local-candidate');
+  const remoteCandidates = [...offererStats.values()].filter(stat => stat.type == 'remote-candidate');
+  isnot(localCandidates, []);
+  isnot(remoteCandidates, []);
+
+  const localNonRelayCandidates =
+      localCandidates.filter(cand => cand.candidateType != 'relay');
+  is(localNonRelayCandidates.length, 0, `There should only be local relay candidates, because we are using the "relay" IceTransportPolicy, but we got ${JSON.stringify(localNonRelayCandidates)}`);
+
+  const remoteHostCandidates =
+      remoteCandidates.filter(cand => cand.candidateType == 'host');
+  is(remoteHostCandidates.length, 0, `There should be no remote host candidates in the stats, because mDNS resolution should have been disabled by the "relay" IceTransportPolicy, but we got ${JSON.stringify(remoteHostCandidates)}`);
+
+  offerer.close();
+  answerer.close();
+
+  await SpecialPowers.popPrefEnv();
+}, { useIceServer: true });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioRequireEOC.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioRequireEOC.html
new file mode 100644
index 0000000000..afad4550d4
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioRequireEOC.html
@@ -0,0 +1,35 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1167443",
+    title: "Basic audio-only peer connection which waits for end-of-candidates"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    test.chain.replace("PC_LOCAL_VERIFY_SDP_AFTER_END_OF_TRICKLE", [
+      function PC_LOCAL_REQUIRE_SDP_AFTER_END_OF_TRICKLE(test) {
+        return test.pcLocal.endOfTrickleSdp.then(sdp =>
+          sdputils.checkSdpAfterEndOfTrickle(sdp, test.testOptions, test.pcLocal.label));
+      }
+    ]);
+    test.chain.replace("PC_REMOTE_VERIFY_SDP_AFTER_END_OF_TRICKLE", [
+      function PC_REMOTE_REQUIRE_SDP_AFTER_END_OF_TRICKLE(test) {
+        return test.pcRemote.endOfTrickleSdp.then(sdp =>
+          sdputils.checkSdpAfterEndOfTrickle(sdp, test.testOptions, test.pcRemote.label));
+      }
+    ]);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVerifyRtpHeaderExtensions.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVerifyRtpHeaderExtensions.html
new file mode 100644
index 0000000000..f28a990bd2
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVerifyRtpHeaderExtensions.html
@@ -0,0 +1,63 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="parser_rtp.js"></script>
+  <script type="application/javascript" src="sdpUtils.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1416932",
+    title: "Basic audio-only peer connection and verify rtp header extensions"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    // pc.js uses video elements by default, we want to test audio elements here
+    test.pcLocal.audioElementsOnly = true;
+
+    let getRtpPacket = (pc) => {
+      // we only examine received packets
+      let sending = false;
+      pc.mozEnablePacketDump(0, "rtp", sending);
+      return new Promise((res, rej) =>
+        pc.mozSetPacketCallback((...args) => {
+          res([...args]);
+          pc.mozSetPacketCallback(() => {});
+          pc.mozDisablePacketDump(0, "rtp", sending);
+        })
+      );
+    }
+
+    const pc = SpecialPowers.wrap(test.pcRemote._pc);
+    const haveFirstPacket = getRtpPacket(pc);
+
+    test.chain.insertBefore('PC_REMOTE_WAIT_FOR_MEDIA_FLOW', [
+      async function PC_REMOTE_CHECK_RTP_HEADER_EXTS_AGAINST_SDP() {
+
+        const sdpExtmapIds = sdputils.findExtmapIds(test.originalAnswer.sdp);
+
+        const [level, type, sending, data] = await haveFirstPacket;
+        const extensions = ParseRtpPacket(data).header.extensions;
+
+        // make sure we got the same number of rtp header extensions in
+        // the received packet as were negotiated in the sdp.  Then
+        // check to make sure each of the received extension ids were in
+        // the sdp.
+        is(sdpExtmapIds.length, extensions.length, "number of received ids match sdp ids");
+        // note, we are comparing a number (from the parsed rtp packet)
+        // and a string (from the answer sdp)
+        ok(extensions.every((ext) => sdpExtmapIds.includes(""+ext.id)), "extension id arrays equivalent");
+      }
+    ]);
+
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideo.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideo.html
new file mode 100644
index 0000000000..c2c2d43f09
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideo.html
@@ -0,0 +1,24 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "796890",
+    title: "Basic audio/video (separate) peer connection"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             [{audio: true}, {video: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoCombined.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoCombined.html
new file mode 100644
index 0000000000..02a561f9b8
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoCombined.html
@@ -0,0 +1,24 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "796890",
+    title: "Basic audio/video (combined) peer connection"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true, video: true}],
+                             [{audio: true, video: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoNoBundle.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoNoBundle.html
new file mode 100644
index 0000000000..cae7f6617f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoNoBundle.html
@@ -0,0 +1,25 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1016476",
+    title: "Basic audio/video peer connection with no Bundle"
+  });
+
+  runNetworkTest(options => {
+    options = options || { };
+    options.bundle = false;
+    var test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             [{audio: true}, {video: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoNoBundleNoRtcpMux.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoNoBundleNoRtcpMux.html
new file mode 100644
index 0000000000..49b0136752
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoNoBundleNoRtcpMux.html
@@ -0,0 +1,39 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1167443",
+    title: "Basic audio & video call with disabled bundle and disabled RTCP-Mux"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    options = options || { };
+    options.bundle = false;
+    options.rtcpmux = false;
+    test = new PeerConnectionTest(options);
+    test.chain.replace("PC_LOCAL_VERIFY_SDP_AFTER_END_OF_TRICKLE", [
+      function PC_LOCAL_REQUIRE_SDP_AFTER_END_OF_TRICKLE(test) {
+        return test.pcLocal.endOfTrickleSdp .then(sdp =>
+          sdputils.checkSdpAfterEndOfTrickle(sdp, test.testOptions, test.pcLocal.label));
+      }
+    ]);
+    test.chain.replace("PC_REMOTE_VERIFY_SDP_AFTER_END_OF_TRICKLE", [
+      function PC_REMOTE_REQUIRE_SDP_AFTER_END_OF_TRICKLE(test) {
+        return test.pcRemote.endOfTrickleSdp .then(sdp =>
+          sdputils.checkSdpAfterEndOfTrickle(sdp, test.testOptions, test.pcRemote.label));
+      }
+    ]);
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             [{audio: true}, {video: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoNoRtcpMux.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoNoRtcpMux.html
new file mode 100644
index 0000000000..48524604ba
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoNoRtcpMux.html
@@ -0,0 +1,38 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1167443",
+    title: "Basic audio & video call with disabled RTCP-Mux"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    options = options || { };
+    options.rtcpmux = false;
+    test = new PeerConnectionTest(options);
+    test.chain.replace("PC_LOCAL_VERIFY_SDP_AFTER_END_OF_TRICKLE", [
+      function PC_LOCAL_REQUIRE_SDP_AFTER_END_OF_TRICKLE(test) {
+        return test.pcLocal.endOfTrickleSdp .then(sdp =>
+          sdputils.checkSdpAfterEndOfTrickle(sdp, test.testOptions, test.pcLocal.label));
+      }
+    ]);
+    test.chain.replace("PC_REMOTE_VERIFY_SDP_AFTER_END_OF_TRICKLE", [
+      function PC_REMOTE_REQUIRE_SDP_AFTER_END_OF_TRICKLE(test) {
+        return test.pcRemote.endOfTrickleSdp .then(sdp =>
+          sdputils.checkSdpAfterEndOfTrickle(sdp, test.testOptions, test.pcRemote.label));
+      }
+    ]);
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             [{audio: true}, {video: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoTransceivers.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoTransceivers.html
new file mode 100644
index 0000000000..181d089d26
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoTransceivers.html
@@ -0,0 +1,31 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1290948",
+    title: "Basic audio/video with addTransceiver"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             [{audio: true}, {video: true}]);
+    test.chain.replace("PC_LOCAL_GUM",
+      [
+        function PC_LOCAL_GUM_TRANSCEIVERS(test) {
+          return test.pcLocal.getAllUserMediaAndAddTransceivers(test.pcLocal.constraints);
+        }
+      ]);
+
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoVerifyExtmap.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoVerifyExtmap.html
new file mode 100644
index 0000000000..e3da00bfa5
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoVerifyExtmap.html
@@ -0,0 +1,97 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1406529",
+    title: "Verify SDP extmap attribute for sendrecv connection"
+  });
+
+  var test;
+  runNetworkTest(async function (options) {
+    await pushPrefs(["media.navigator.video.use_transport_cc", true]);
+
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             [{audio: true}, {video: true}]);
+
+    test.chain.insertAfter('PC_LOCAL_SET_LOCAL_DESCRIPTION', [
+      async function PC_LOCAL_CHECK_SDP_OFFER_EXTMAP() {
+        sdputils.verify_unique_extmap_ids(test.originalOffer.sdp);
+
+        const audio = sdputils.findExtmapIdsUrnsDirections(
+            sdputils.getAudioMSections(test.originalOffer.sdp));
+        const expected_audio = [
+          /* Please modify this list when you add or remove RTP header
+          extensions. */
+          ["1", "urn:ietf:params:rtp-hdrext:ssrc-audio-level", ""],
+          ["2", "urn:ietf:params:rtp-hdrext:csrc-audio-level", "recvonly"],
+          ["3", "urn:ietf:params:rtp-hdrext:sdes:mid", ""],
+        ];
+        // *Ugh* ...
+        ok(JSON.stringify(audio) ===
+           JSON.stringify(expected_audio),
+           "List of offer audio URNs meets expected values");
+
+        const video = sdputils.findExtmapIdsUrnsDirections(
+            sdputils.getVideoMSections(test.originalOffer.sdp));
+        const expected_video = [
+          /* Please modify this list when you add or remove RTP header
+          extensions. */
+          ["3", "urn:ietf:params:rtp-hdrext:sdes:mid", ""],
+          ["4", "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", ""],
+          ["5", "urn:ietf:params:rtp-hdrext:toffset", ""],
+          ["6", "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay", "recvonly"],
+          ["7", "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01", ""],
+        ];
+        // *Ugh* ...
+        ok(JSON.stringify(video) ===
+           JSON.stringify(expected_video),
+           "List of offer video URNs meets expected values");
+      }
+    ]);
+
+    test.chain.removeAfter('PC_REMOTE_SET_LOCAL_DESCRIPTION');
+    test.chain.append([
+      async function PC_REMOTE_CHECK_SDP_ANSWER_EXTMAP() {
+        sdputils.verify_unique_extmap_ids(test.originalAnswer.sdp);
+
+        const audio = sdputils.findExtmapIdsUrnsDirections(
+            sdputils.getAudioMSections(test.originalAnswer.sdp));
+        const expected_audio = [
+          /* Please modify this list when you add or remove RTP header
+          extensions. */
+          ["1", "urn:ietf:params:rtp-hdrext:ssrc-audio-level",""],
+          ["3", "urn:ietf:params:rtp-hdrext:sdes:mid",""],
+        ];
+        // *Ugh* ...
+        ok(JSON.stringify(audio) ===
+           JSON.stringify(expected_audio),
+           "List of answer audio URNs meets expected values");
+
+        const video = sdputils.findExtmapIdsUrnsDirections(
+            sdputils.getVideoMSections(test.originalAnswer.sdp));
+        const expected_video = [
+          /* Please modify this list when you add or remove RTP header
+          extensions. */
+          ["3", "urn:ietf:params:rtp-hdrext:sdes:mid",""],
+          ["4", "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time",""],
+          ["5", "urn:ietf:params:rtp-hdrext:toffset",""],
+          ["7", "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01", ""],
+        ];
+        ok(JSON.stringify(video) ===
+           JSON.stringify(expected_video),
+           "List of answer video URNs meets expected values");
+      }
+    ]);
+
+    await test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoVerifyExtmapSendonly.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoVerifyExtmapSendonly.html
new file mode 100644
index 0000000000..6cbc9e4c00
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoVerifyExtmapSendonly.html
@@ -0,0 +1,97 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1406529",
+    title: "Verify SDP extmap attribute for sendonly connection"
+  });
+
+  var test;
+  runNetworkTest(async function (options) {
+    await pushPrefs(["media.navigator.video.use_transport_cc", true]);
+
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             []);
+
+    test.chain.insertAfter('PC_LOCAL_SET_LOCAL_DESCRIPTION', [
+      async function PC_LOCAL_CHECK_SDP_OFFER_EXTMAP() {
+        sdputils.verify_unique_extmap_ids(test.originalOffer.sdp);
+
+        const audio = sdputils.findExtmapIdsUrnsDirections(
+            sdputils.getAudioMSections(test.originalOffer.sdp));
+        const expected_audio = [
+          /* Please modify this list when you add or remove RTP header
+          extensions. */
+          ["1", "urn:ietf:params:rtp-hdrext:ssrc-audio-level", ""],
+          ["2", "urn:ietf:params:rtp-hdrext:csrc-audio-level", "recvonly"],
+          ["3", "urn:ietf:params:rtp-hdrext:sdes:mid", ""],
+        ];
+        // *Ugh* ...
+        ok(JSON.stringify(audio) ===
+           JSON.stringify(expected_audio),
+           "List of offer audio URNs meets expected values");
+
+        const video = sdputils.findExtmapIdsUrnsDirections(
+            sdputils.getVideoMSections(test.originalOffer.sdp));
+        const expected_video = [
+          /* Please modify this list when you add or remove RTP header
+          extensions. */
+          ["3", "urn:ietf:params:rtp-hdrext:sdes:mid", ""],
+          ["4", "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time", ""],
+          ["5", "urn:ietf:params:rtp-hdrext:toffset", ""],
+          ["6", "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay", "recvonly"],
+          ["7", "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01", ""],
+        ];
+        // *Ugh* ...
+        ok(JSON.stringify(video) ===
+           JSON.stringify(expected_video),
+           "List of offer video URNs meets expected values");
+      }
+    ]);
+
+    test.chain.removeAfter('PC_REMOTE_SET_LOCAL_DESCRIPTION');
+    test.chain.append([
+      async function PC_REMOTE_CHECK_SDP_ANSWER_EXTMAP() {
+        sdputils.verify_unique_extmap_ids(test.originalAnswer.sdp);
+
+        const audio = sdputils.findExtmapIdsUrnsDirections(
+            sdputils.getAudioMSections(test.originalAnswer.sdp));
+        const expected_audio = [
+          /* Please modify this list when you add or remove RTP header
+          extensions. */
+          ["1", "urn:ietf:params:rtp-hdrext:ssrc-audio-level",""],
+          ["3", "urn:ietf:params:rtp-hdrext:sdes:mid",""],
+        ];
+        // *Ugh* ...
+        ok(JSON.stringify(audio) ===
+           JSON.stringify(expected_audio),
+           "List of answer audio URNs meets expected values");
+
+        const video = sdputils.findExtmapIdsUrnsDirections(
+            sdputils.getVideoMSections(test.originalAnswer.sdp));
+        const expected_video = [
+          /* Please modify this list when you add or remove RTP header
+          extensions. */
+          ["3", "urn:ietf:params:rtp-hdrext:sdes:mid",""],
+          ["4", "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time",""],
+          ["5", "urn:ietf:params:rtp-hdrext:toffset",""],
+          ["7", "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01", ""],
+        ];
+        ok(JSON.stringify(video) ===
+           JSON.stringify(expected_video),
+           "List of answer video URNs meets expected values");
+      }
+    ]);
+
+    await test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoVerifyTooLongMidFails.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoVerifyTooLongMidFails.html
new file mode 100644
index 0000000000..70d27b48c6
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudioVideoVerifyTooLongMidFails.html
@@ -0,0 +1,47 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1427009",
+    title: "Test mid longer than 16 characters fails"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    options = options || { };
+    options.bundle = false;
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             [{audio: true}, {video: true}]);
+
+    test.chain.replaceAfter("PC_LOCAL_CREATE_OFFER",
+      [
+        function PC_LOCAL_MUNGE_OFFER_SDP(test) {
+          test.originalOffer.sdp =
+            test.originalOffer.sdp.replace(/a=mid:.*\r\n/g,
+                                           "a=mid:really_long_mid_over_16_chars\r\n");
+        },
+        function PC_LOCAL_EXPECT_SET_LOCAL_DESCRIPTION_FAIL(test) {
+          return test.setLocalDescription(test.pcLocal,
+                                          test.originalOffer,
+                                          HAVE_LOCAL_OFFER)
+           .then(() => ok(false, "setLocalDescription must fail"),
+                 // This needs to be RTCError once we support it, and once we
+                 // stop allowing any modification, InvalidModificationError
+                 e => is(e.name, "OperationError",
+                         "setLocalDescription must fail and did"));
+        }
+      ], 0 // first occurance
+    );
+
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudio_forced_higher_rate.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudio_forced_higher_rate.html
new file mode 100644
index 0000000000..95bfb06514
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudio_forced_higher_rate.html
@@ -0,0 +1,19 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="peerconnection_audio_forced_sample_rate.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1437366",
+  title: "Basic audio-only peer connection, with the MTG running at a rate not supported by the MediaPipeline (49000Hz)"
+});
+
+test_peerconnection_audio_forced_sample_rate(49000);
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudio_forced_lower_rate.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudio_forced_lower_rate.html
new file mode 100644
index 0000000000..aab9778971
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicAudio_forced_lower_rate.html
@@ -0,0 +1,19 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="peerconnection_audio_forced_sample_rate.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1437366",
+  title: "Basic audio-only peer connection, with the MTG running at a rate not supported by the MediaPipeline (24000Hz)"
+});
+
+test_peerconnection_audio_forced_sample_rate(24000);
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicH264Video.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicH264Video.html
new file mode 100644
index 0000000000..072c35da39
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicH264Video.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1040346",
+    title: "Basic H.264 GMP video-only peer connection"
+  });
+
+  var test;
+  runNetworkTest(async function (options) {
+    matchPlatformH264CodecPrefs();
+    options = options || { };
+    options.h264 = true;
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{video: true}], [{video: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicScreenshare.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicScreenshare.html
new file mode 100644
index 0000000000..93148ac5fd
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicScreenshare.html
@@ -0,0 +1,56 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1039666",
+    title: "Basic screenshare-only peer connection"
+  });
+
+  async function supportedVideoPayloadTypes() {
+    const pc = new RTCPeerConnection();
+    const offer = await pc.createOffer({offerToReceiveVideo: true});
+    return sdputils.getPayloadTypes(offer.sdp);
+  }
+
+  async function testScreenshare(payloadType) {
+    const options = {};
+    options.h264 = payloadType == 97 || payloadType == 126;
+    const test = new PeerConnectionTest(options);
+    const constraints = {
+      video: { mediaSource: "screen" },
+    };
+    test.setMediaConstraints([constraints], []);
+    test.chain.insertAfterEach("PC_LOCAL_CREATE_OFFER", [
+      function PC_LOCAL_ISOLATE_CODEC() {
+        info(`Forcing payload type ${payloadType}. Note that other associated ` +
+             `payload types, like RTX, are removed too.`);
+        test.originalOffer.sdp =
+          sdputils.removeAllButPayloadType(test.originalOffer.sdp, payloadType);
+      },
+    ]);
+    await test.run();
+  }
+
+  runNetworkTest(async () => {
+    await matchPlatformH264CodecPrefs();
+    const pts = await supportedVideoPayloadTypes();
+    ok(pts.includes("120"), "VP8 is supported");
+    ok(pts.includes("121"), "VP9 is supported");
+    if (pts.length > 2) {
+      is(pts.length, 4, "Expected VP8, VP9 and two variants of H264");
+      ok(pts.includes("97"), "H264 with no packetization-mode is supported");
+      ok(pts.includes("126"), "H264 with packetization-mode=1 is supported");
+    }
+    for (const pt of pts) {
+      await testScreenshare(pt);
+    }
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicVideo.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicVideo.html
new file mode 100644
index 0000000000..4a0655d696
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicVideo.html
@@ -0,0 +1,23 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "796888",
+    title: "Basic video-only peer connection"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{video: true}], [{video: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicVideoVerifyRtpHeaderExtensions.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicVideoVerifyRtpHeaderExtensions.html
new file mode 100644
index 0000000000..7874e52a10
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicVideoVerifyRtpHeaderExtensions.html
@@ -0,0 +1,82 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="parser_rtp.js"></script>
+  <script type="application/javascript" src="sdpUtils.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1416932",
+    title: "Basic video-only peer connection and verify rtp header extensions"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{video: true}], [{video: true}]);
+
+    let getRtpPacketWithExtension = (pc, extension) => {
+      // we only examine received packets
+      let sending = false;
+      pc.mozEnablePacketDump(0, "rtp", sending);
+      return new Promise((res, rej) =>
+        pc.mozSetPacketCallback((...args) => {
+          const packet = ParseRtpPacket(args[3]);
+          info(`midId = ${extension} packet = ${JSON.stringify(packet, null, 2)}`);
+          if (packet.header.extensions.find(e => e.id == extension) !== undefined) {
+            res(packet);
+            pc.mozSetPacketCallback(() => {});
+            pc.mozDisablePacketDump(0, "rtp", sending);
+          }
+        })
+      );
+    }
+
+    let havePacketWithMid;
+    let sdpExtmaps;
+
+    // MID can stop being sent when acked causing failures if packets are checked later.
+    // Starting packet sniffer before PC_LOCAL_SET_REMOTE_DESCRIPTION to be ready
+    // to inspect packets ahead of any packets arriving.
+    test.chain.insertBefore('PC_LOCAL_SET_REMOTE_DESCRIPTION', [
+      function PC_REMOTE_FIND_RTP_PACKETS_WITH_MIDID() {
+
+        sdpExtmaps = sdputils.findExtmapIdsUrnsDirections(test.originalAnswer.sdp);
+        const [midId] = sdpExtmaps.find(([, urn]) => urn == "urn:ietf:params:rtp-hdrext:sdes:mid");
+        const pc = SpecialPowers.wrap(test.pcRemote._pc);
+        havePacketWithMid = getRtpPacketWithExtension(pc, midId);
+      }
+    ]);
+
+    test.chain.insertBefore('PC_REMOTE_WAIT_FOR_MEDIA_FLOW', [
+      async function PC_REMOTE_CHECK_RTP_HEADER_EXTS_AGAINST_SDP() {
+
+        const sdpExtmapIds = sdpExtmaps.map(e => e[0]);
+        const packet = await havePacketWithMid;
+        const extIds = packet.header.extensions.map(e => `${e.id}`);
+        // make sure we got the same number of rtp header extensions in
+        // the received packet as were negotiated in the sdp.  Then
+        // check to make sure each of the received extension ids were in
+        // the sdp.
+        is(sdpExtmapIds.length, extIds.length,
+           `number of sdp ids match received ids ` +
+           `${JSON.stringify(sdpExtmapIds)} == ${JSON.stringify(extIds)}\n` +
+           `sdp = ${test.originalAnswer.sdp}\n` +
+           `packet = ${JSON.stringify(packet, null, 2)}`);
+        // note, we are comparing a number (from the parsed rtp packet)
+        // and a string (from the answer sdp)
+        ok(extIds.every(id => sdpExtmapIds.includes(id)) &&
+           sdpExtmapIds.every(id => extIds.includes(id)),
+          `extension id arrays equivalent`);
+      }
+    ]);
+
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_basicWindowshare.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicWindowshare.html
new file mode 100644
index 0000000000..1cfb0797db
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_basicWindowshare.html
@@ -0,0 +1,25 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1038926",
+    title: "Basic windowshare-only peer connection"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    const constraints = {
+      video: { mediaSource: "window" },
+    };
+    test.setMediaConstraints([constraints], []);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1013809.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1013809.html
new file mode 100644
index 0000000000..a8c7004793
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1013809.html
@@ -0,0 +1,25 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1013809",
+    title: "Audio-only peer connection with swapped setLocal and setRemote steps"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    var sld = test.chain.remove("PC_REMOTE_SET_LOCAL_DESCRIPTION");
+    test.chain.insertAfter("PC_LOCAL_SET_REMOTE_DESCRIPTION", sld);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1042791.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1042791.html
new file mode 100644
index 0000000000..a84dcf9d09
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1042791.html
@@ -0,0 +1,36 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1040346",
+    title: "Basic H.264 GMP video-only peer connection"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    options = options || { };
+    options.h264 = true;
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{video: true}], [{video: true}]);
+    test.chain.removeAfter("PC_LOCAL_CREATE_OFFER");
+
+    test.chain.append([
+      function PC_LOCAL_VERIFY_H264_OFFER(test) {
+        ok(!test.pcLocal._latest_offer.sdp.toLowerCase().includes("profile-level-id=0x42e0"),
+           "H264 offer does not contain profile-level-id=0x42e0");
+        ok(test.pcLocal._latest_offer.sdp.toLowerCase().includes("profile-level-id=42e0"),
+           "H264 offer contains profile-level-id=42e0");
+      }
+    ]);
+
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1227781.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1227781.html
new file mode 100644
index 0000000000..41e4aec457
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1227781.html
@@ -0,0 +1,37 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1227781",
+    title: "Test with invalid TURN server"
+  });
+
+  const turnConfig = {
+    iceServers: [
+      {
+        username: "mozilla",
+        credential: "mozilla",
+        url: "turn:test@10.0.0.1",
+      },
+    ],
+  };
+  runNetworkTest(function (options) {
+    let exception = false;
+    try {
+      new RTCPeerConnection(turnConfig);
+    } catch (e) {
+      info(e);
+      exception = true;
+    }
+    is(exception, true, "Exception fired");
+    ok("Success");
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1512281.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1512281.html
new file mode 100644
index 0000000000..e6451becea
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1512281.html
@@ -0,0 +1,47 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1512281",
+    title: "Test that RTCP sender and receiver stats are not swapped"
+  });
+
+const ensure_missing_rtcp = async stats => {
+	const rtcp_stats = [...stats.values()].filter(
+		s => s.type.endsWith("bound-rtp") &&
+			 s.isRemote == true).map(s => JSON.stringify(s))
+	is(rtcp_stats, [],
+	   "There are no RTCP stats when RTCP reception is turned off");
+};
+
+const PC_LOCAL_TEST_FOR_MISSING_RTCP = async test =>
+	await ensure_missing_rtcp(await test.pcLocal.getStats());
+
+const PC_REMOTE_TEST_FOR_MISSING_RTCP = async test =>
+  await ensure_missing_rtcp(await test.pcRemote.getStats());
+
+runNetworkTest(async options => {
+  await pushPrefs(["media.webrtc.net.force_disable_rtcp_reception", true]);
+
+  const test = new PeerConnectionTest(options);
+
+  test.chain.insertAfter("PC_LOCAL_WAIT_FOR_MEDIA_FLOW",
+    [PC_LOCAL_TEST_FOR_MISSING_RTCP]);
+
+  test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW",
+    [PC_REMOTE_TEST_FOR_MISSING_RTCP]);
+
+  test.setMediaConstraints([{audio: true}, {video: true}],
+                           [{audio: true}, {video: true}]);
+  await test.run();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1773067.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1773067.html
new file mode 100644
index 0000000000..9e6d79a107
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug1773067.html
@@ -0,0 +1,32 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1773067",
+    title: "getStats on a closed peer connection should fail, not hang, " +
+           " until bug 1056433 is fixed"
+  });
+
+  // TODO: Bug 1056433 removes the need for this test
+  runNetworkTest(async function () {
+    let errorName;
+    try {
+      const pc = new RTCPeerConnection();
+      pc.close();
+      await pc.getStats();
+    } catch(e) {
+      errorName = e.name;
+    }
+    is(errorName,
+       "InvalidStateError",
+       "getStats on closed peer connection fails instead of hanging");
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_bug822674.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug822674.html
new file mode 100644
index 0000000000..fceb2c2a1d
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug822674.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "822674",
+    title: "RTCPeerConnection isn't a true javascript object as it should be"
+  });
+
+  runNetworkTest(function () {
+    var pc = new RTCPeerConnection();
+
+    pc.thereIsNeverGoingToBeAPropertyWithThisNameOnThisInterface = 1;
+    is(pc.thereIsNeverGoingToBeAPropertyWithThisNameOnThisInterface, 1,
+       "Can set expandos on an RTCPeerConnection");
+
+    pc = null;
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_bug825703.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug825703.html
new file mode 100644
index 0000000000..5cd168af8a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug825703.html
@@ -0,0 +1,140 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "825703",
+    title: "RTCConfiguration valid/invalid permutations"
+  });
+
+// ^^^ Don't insert data above this line without adjusting line number below!
+var lineNumberAndFunction = {
+// <--- 16 is the line this must be.
+  line: 17, func: () => new RTCPeerConnection().onaddstream = () => {}
+};
+
+var makePC = (config, expected_error) => {
+  var exception;
+  try {
+    new RTCPeerConnection(config).close();
+  } catch (e) {
+    exception = e;
+  }
+  is((exception? exception.name : "success"), expected_error || "success",
+     "RTCPeerConnection(" + JSON.stringify(config) + ")");
+};
+
+// The order of properties in objects is not guaranteed in JavaScript, so this
+// transform produces json-comparable dictionaries. The resulting copy is only
+// meant to be used in comparisons (e.g. array-ness is not preserved).
+
+var toComparable = o =>
+    (typeof o != 'object' || !o)? o : Object.keys(o).sort().reduce((co, key) => {
+  co[key] = toComparable(o[key]);
+  return co;
+}, {});
+
+// This is a test of the iceServers parsing code + readable errors
+runNetworkTest(() => {
+  var exception = null;
+
+  try {
+    new RTCPeerConnection().close();
+  } catch (e) {
+    exception = e;
+  }
+  ok(!exception, "RTCPeerConnection() succeeds");
+  exception = null;
+
+  // Some overlap still with WPT RTCConfiguration-iceServers.html
+
+  makePC({ iceServers: [
+    { urls:"stun:127.0.0.1" },
+    { urls:"stun:localhost", foo:"" },
+    { urls: ["stun:127.0.0.1", "stun:localhost"] },
+    { urls:"stuns:localhost", foo:"" },
+    { urls:"turn:[::1]:3478", username:"p", credential:"p" },
+    { urls:"turn:[::1]:3478", username:"", credential:"" },
+    { urls:"turns:[::1]:3478", username:"", credential:"" },
+    { urls:"turn:localhost:3478?transport=udp", username:"p", credential:"p" },
+    { urls: ["turn:[::1]:3478", "turn:localhost"], username:"p", credential:"p" },
+    { urls:"turns:localhost:3478?transport=udp", username:"p", credential:"p" },
+    { url:"stun:localhost", foo:"" },
+    { url:"turn:localhost", username:"p", credential:"p" }
+  ]});
+
+  makePC({ iceServers: [{ urls:"http:0.0.0.0" }] }, "SyntaxError");
+
+  try {
+    new RTCPeerConnection({ iceServers: [{ url:"http:0.0.0.0" }] }).close();
+  } catch (e) {
+    ok(e.message.indexOf("http") > 0,
+       "RTCPeerConnection() constructor has readable exceptions");
+  }
+
+  // Test getConfiguration
+  const config = {
+    bundlePolicy: "max-bundle",
+    iceTransportPolicy: "relay",
+    peerIdentity: null,
+    certificates: [],
+    iceServers: [
+      { urls: ["stun:127.0.0.1", "stun:localhost"], credentialType:"password" },
+      { urls: ["turn:[::1]:3478"], username:"p", credential:"p", credentialType:"password" },
+    ],
+  };
+  // Make sure sdpSemantics is not exposed in getConfiguration
+  const configWithExtraProps = Object.assign({},
+                                             config,
+                                             {sdpSemantics: "plan-b"});
+  ok("sdpSemantics" in configWithExtraProps, "sdpSemantics control");
+
+  const pc = new RTCPeerConnection(configWithExtraProps);
+  is(JSON.stringify(toComparable(pc.getConfiguration())),
+     JSON.stringify(toComparable(config)), "getConfiguration");
+  pc.close();
+
+  var push = prefs => SpecialPowers.pushPrefEnv(prefs);
+
+  return Promise.resolve()
+  // This set of tests are setting the about:config User preferences for default
+  // ice servers and checking the outputs when RTCPeerConnection() is
+  // invoked. See Bug 1167922 for more information.
+  .then(() => push({ set: [['media.peerconnection.default_iceservers', ""]] })
+    .then(() => makePC())
+    .then(() => push({ set: [['media.peerconnection.default_iceservers', "k"]] }))
+    .then(() => makePC())
+    .then(() => push({ set: [['media.peerconnection.default_iceservers', "[{\"urls\": [\"stun:stun.services.mozilla.com\"]}]"]] }))
+    .then(() => makePC()))
+  // This set of tests check that warnings work. See Bug 1254839 for more.
+  .then(() => {
+    let promise = new Promise(resolve => {
+      SpecialPowers.registerConsoleListener(msg => {
+        if (msg.message.includes("onaddstream")) {
+          SpecialPowers.postConsoleSentinel();
+          resolve(msg.message);
+        }
+      });
+    });
+    lineNumberAndFunction.func();
+    return promise;
+  }).then(warning => {
+    is(warning.split('"')[1],
+       "WebRTC: onaddstream is deprecated! Use peerConnection.ontrack instead.",
+       "warning logged");
+    var remainder = warning.split('"').slice(2).join('"');
+    info(remainder);
+    ok(remainder.includes('file: "' + window.location + '"'),
+       "warning has this file");
+    ok(remainder.includes('line: ' + lineNumberAndFunction.line),
+       "warning has correct line number");
+    });
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_bug827843.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug827843.html
new file mode 100644
index 0000000000..06cfde9e5d
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug827843.html
@@ -0,0 +1,50 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "827843",
+    title: "Ensure that localDescription and remoteDescription are null after close"
+  });
+
+var steps = [
+  function CHECK_SDP_ON_CLOSED_PC(test) {
+    var description;
+    var exception = null;
+
+    test.pcLocal.close();
+
+    try { description = test.pcLocal.localDescription; } catch (e) { exception = e; }
+    ok(exception, "Attempt to access localDescription of pcLocal after close throws exception");
+    exception = null;
+
+    try { description = test.pcLocal.remoteDescription; } catch (e) { exception = e; }
+    ok(exception, "Attempt to access remoteDescription of pcLocal after close throws exception");
+    exception = null;
+
+    test.pcRemote.close();
+
+    try  { description = test.pcRemote.localDescription; } catch (e) { exception = e; }
+    ok(exception, "Attempt to access localDescription of pcRemote after close throws exception");
+    exception = null;
+
+    try  { description = test.pcRemote.remoteDescription; } catch (e) { exception = e; }
+    ok(exception, "Attempt to access remoteDescription of pcRemote after close throws exception");
+  }
+];
+
+var test;
+runNetworkTest(() => {
+  test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.chain.append(steps);
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_bug834153.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug834153.html
new file mode 100644
index 0000000000..6d8ca2a7ce
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_bug834153.html
@@ -0,0 +1,36 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "834153",
+    title: "Queue CreateAnswer in PeerConnection.js"
+  });
+
+  runNetworkTest(function () {
+    var pc1 = new RTCPeerConnection();
+    var pc2 = new RTCPeerConnection();
+
+    return pc1.createOffer({ offerToReceiveAudio: true }).then(offer => {
+      // The whole point of this test is not to wait for the
+      // setRemoteDescription call to succesfully complete, so we
+      // don't wait for it to succeed.
+      pc2.setRemoteDescription(offer);
+      return pc2.createAnswer();
+    })
+    .then(answer => is(answer.type, "answer", "CreateAnswer created an answer"))
+    .catch(reason => ok(false, reason.message))
+    .then(() => {
+      pc1.close();
+      pc2.close();
+    })
+    .catch(reason => ok(false, reason.message));
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_callbacks.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_callbacks.html
new file mode 100644
index 0000000000..4c890e4400
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_callbacks.html
@@ -0,0 +1,86 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "PeerConnection using callback functions",
+    bug: "1119593",
+    visible: true
+  });
+
+// This still aggressively uses promises, but it is testing that the callback functions
+// are properly in place.
+
+// wrapper that turns a callback-based function call into a promise
+function pcall(o, f, beforeArg) {
+  return new Promise((resolve, reject) => {
+    var args = [resolve, reject];
+    if (typeof beforeArg !== 'undefined') {
+      args.unshift(beforeArg);
+    }
+    info('Calling ' + f.name);
+    f.apply(o, args);
+  });
+}
+
+var pc1 = new RTCPeerConnection();
+var pc2 = new RTCPeerConnection();
+
+var pc2_haveRemoteOffer = new Promise(resolve => {
+  pc2.onsignalingstatechange =
+    e => (e.target.signalingState == "have-remote-offer") && resolve();
+});
+var pc1_stable = new Promise(resolve => {
+  pc1.onsignalingstatechange =
+    e => (e.target.signalingState == "stable") && resolve();
+});
+
+pc1.onicecandidate = e => {
+  pc2_haveRemoteOffer
+    .then(() => !e.candidate || pcall(pc2, pc2.addIceCandidate, e.candidate))
+    .catch(generateErrorCallback());
+};
+pc2.onicecandidate = e => {
+  pc1_stable
+    .then(() => !e.candidate || pcall(pc1, pc1.addIceCandidate, e.candidate))
+    .catch(generateErrorCallback());
+};
+
+var v1, v2;
+var delivered = new Promise(resolve => {
+  pc2.onaddstream = e => {
+    v2.srcObject = e.stream;
+    resolve(e.stream);
+  };
+});
+
+runNetworkTest(function() {
+  v1 = createMediaElement('video', 'v1');
+  v2 = createMediaElement('video', 'v2');
+  var canPlayThrough = new Promise(resolve => v2.canplaythrough = resolve);
+  is(v2.currentTime, 0, "v2.currentTime is zero at outset");
+
+  // not testing legacy gUM here
+  return navigator.mediaDevices.getUserMedia({ video: true, audio: true })
+    .then(stream => pc1.addStream(v1.srcObject = stream))
+    .then(() => pcall(pc1, pc1.createOffer))
+    .then(offer => pcall(pc1, pc1.setLocalDescription, offer))
+    .then(() => pcall(pc2, pc2.setRemoteDescription, pc1.localDescription))
+    .then(() => pcall(pc2, pc2.createAnswer))
+    .then(answer => pcall(pc2, pc2.setLocalDescription, answer))
+    .then(() => pcall(pc1, pc1.setRemoteDescription, pc2.localDescription))
+    .then(() => delivered)
+  //    .then(() => canPlayThrough)    // why doesn't this fire?
+    .then(() => waitUntil(() => v2.currentTime > 0))
+    .then(() => ok(v2.currentTime > 0, "v2.currentTime is moving (" + v2.currentTime + ")"))
+    .then(() => ok(true, "Connected."))
+    .then(() => { v1.pause(); v2.pause(); });
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_captureStream_canvas_2d.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_captureStream_canvas_2d.html
new file mode 100644
index 0000000000..db3a735008
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_captureStream_canvas_2d.html
@@ -0,0 +1,81 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1032848",
+  title: "Canvas(2D)::CaptureStream as video-only input to peerconnection",
+  visible: true
+});
+
+runNetworkTest(async () => {
+  // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+  if (navigator.userAgent.includes("Android")) {
+    await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                    ["media.webrtc.hw.h264.enabled", false]);
+  }
+
+  var test = new PeerConnectionTest();
+  var mediaElement;
+  var h = new CaptureStreamTestHelper2D();
+  var canvas = document.createElement('canvas');
+  var stream;
+  canvas.id = 'source_canvas';
+  canvas.width = canvas.height = 16;
+  document.getElementById('content').appendChild(canvas);
+
+  test.setMediaConstraints([{video: true}], []);
+  test.chain.replace("PC_LOCAL_GUM", [
+    function PC_LOCAL_CANVAS_CAPTURESTREAM(test) {
+      h.drawColor(canvas, h.green);
+      stream = canvas.captureStream(0);
+      test.pcLocal.attachLocalStream(stream);
+      stream.requestFrame();
+      var i = 0;
+      return setInterval(function() {
+        try {
+          info("draw " + i ? "green" : "red");
+          h.drawColor(canvas, i ? h.green : h.red);
+          i = 1 - i;
+          stream.requestFrame();
+        } catch (e) {
+          // ignore; stream might have shut down, and we don't bother clearing
+          // the setInterval.
+        }
+      }, 500);
+    }
+  ]);
+  test.chain.append([
+    function PC_REMOTE_WAIT_FOR_REMOTE_GREEN() {
+      mediaElement = test.pcRemote.remoteMediaElements[0];
+      ok(!!mediaElement, "Should have remote video element for pcRemote");
+      return h.pixelMustBecome(mediaElement, h.green, {
+        threshold: 128,
+        infoString: "pcRemote's remote should become green",
+      });
+    },
+    function PC_LOCAL_DRAW_LOCAL_RED() {
+      // After requesting a frame it will be captured at the time of next render.
+      // Next render will happen at next stable state, at the earliest,
+      // i.e., this order of `requestFrame(); draw();` should work.
+      stream.requestFrame();
+      h.drawColor(canvas, h.red);
+    },
+    function PC_REMOTE_WAIT_FOR_REMOTE_RED() {
+      return h.pixelMustBecome(mediaElement, h.red, {
+        threshold: 128,
+        infoString: "pcRemote's remote should become red",
+      });
+    }
+  ]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_captureStream_canvas_2d_noSSRC.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_captureStream_canvas_2d_noSSRC.html
new file mode 100644
index 0000000000..e33a7e8886
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_captureStream_canvas_2d_noSSRC.html
@@ -0,0 +1,83 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  title: "Canvas(2D)::CaptureStream as video-only input to peerconnection with no a=ssrc",
+  visible: true
+});
+
+var test;
+runNetworkTest(async (options) => {
+  // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+  if (navigator.userAgent.includes("Android")) {
+    await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                    ["media.webrtc.hw.h264.enabled", false]);
+  }
+
+  options = options || { };
+  options.ssrc = false;
+  test = new PeerConnectionTest(options);
+  var mediaElement;
+  var h = new CaptureStreamTestHelper2D();
+  var canvas = document.createElement('canvas');
+  var stream;
+  canvas.id = 'source_canvas';
+  canvas.width = canvas.height = 16;
+  document.getElementById('content').appendChild(canvas);
+
+  test.setMediaConstraints([{video: true}], []);
+  test.chain.replace("PC_LOCAL_GUM", [
+    function PC_LOCAL_CANVAS_CAPTURESTREAM(test) {
+      h.drawColor(canvas, h.green);
+      stream = canvas.captureStream(0);
+      test.pcLocal.attachLocalStream(stream);
+      stream.requestFrame();
+      var i = 0;
+      return setInterval(function() {
+        try {
+          info("draw " + i ? "green" : "red");
+          h.drawColor(canvas, i ? h.green : h.red);
+          i = 1 - i;
+          stream.requestFrame();
+        } catch (e) {
+          // ignore; stream might have shut down, and we don't bother clearing
+          // the setInterval.
+        }
+      }, 500);
+    }
+  ]);
+  test.chain.append([
+    function PC_REMOTE_WAIT_FOR_REMOTE_GREEN() {
+      mediaElement = test.pcRemote.remoteMediaElements[0];
+      ok(!!mediaElement, "Should have remote video element for pcRemote");
+      return h.pixelMustBecome(mediaElement, h.green, {
+        threshold: 128,
+        infoString: "pcRemote's remote should become green",
+      });
+    },
+    function PC_LOCAL_DRAW_LOCAL_RED() {
+      // After requesting a frame it will be captured at the time of next render.
+      // Next render will happen at next stable state, at the earliest,
+      // i.e., this order of `requestFrame(); draw();` should work.
+      stream.requestFrame();
+      h.drawColor(canvas, h.red);
+    },
+    function PC_REMOTE_WAIT_FOR_REMOTE_RED() {
+      return h.pixelMustBecome(mediaElement, h.red, {
+        threshold: 128,
+        infoString: "pcRemote's remote should become red",
+      });
+    }
+  ]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_captureStream_canvas_webgl.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_captureStream_canvas_webgl.html
new file mode 100644
index 0000000000..167379fb37
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_captureStream_canvas_webgl.html
@@ -0,0 +1,130 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/webgl-mochitest/webgl-util.js"></script>
+</head>
+<body>
+<pre id="test">
+<script id="v-shader" type="x-shader/x-vertex">
+  attribute vec2 aPosition;
+  void main() {
+   gl_Position = vec4(aPosition, 0, 1);
+}
+</script>
+<script id="f-shader" type="x-shader/x-fragment">
+  precision mediump float;
+  uniform vec4 uColor;
+  void main() { gl_FragColor = uColor; }
+</script>
+<script type="application/javascript">
+createHTML({
+  bug: "1032848",
+  title: "Canvas(WebGL)::CaptureStream as video-only input to peerconnection"
+});
+
+runNetworkTest(async () => {
+  // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+  if (navigator.userAgent.includes("Android")) {
+    await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                    ["media.webrtc.hw.h264.enabled", false]);
+  }
+
+  var test = new PeerConnectionTest();
+  var vremote;
+  var h = new CaptureStreamTestHelperWebGL();
+  var canvas = document.createElement('canvas');
+  canvas.id = 'source_canvas';
+  canvas.width = canvas.height = 16;
+  canvas.style.display = 'none';
+  document.getElementById('content').appendChild(canvas);
+
+  var gl = canvas.getContext('webgl');
+  if (!gl) {
+    todo(false, "WebGL unavailable.");
+    networkTestFinished();
+    return;
+  }
+
+  test.setMediaConstraints([{video: true}], []);
+  test.chain.replace("PC_LOCAL_GUM", [
+    function WEBGL_SETUP(test) {
+      var program = WebGLUtil.createProgramByIds(gl, 'v-shader', 'f-shader');
+
+      if (!program) {
+        ok(false, "Program should link");
+        return Promise.reject("Program should link");
+      }
+      gl.useProgram(program);
+
+      var uColorLocation = gl.getUniformLocation(program, "uColor");
+      h.setFragmentColorLocation(uColorLocation);
+
+      var squareBuffer = gl.createBuffer();
+      gl.bindBuffer(gl.ARRAY_BUFFER, squareBuffer);
+
+      var vertices = [ 0,  0,
+                      -1,  0,
+                       0,  1,
+                      -1,  1 ];
+      gl.bufferData(gl.ARRAY_BUFFER, new Float32Array(vertices), gl.STATIC_DRAW);
+      squareBuffer.itemSize = 2;
+      squareBuffer.numItems = 4;
+
+      program.aPosition = gl.getAttribLocation(program, "aPosition");
+      gl.enableVertexAttribArray(program.aPosition);
+      gl.vertexAttribPointer(program.aPosition, squareBuffer.itemSize, gl.FLOAT, false, 0, 0);
+    },
+    function PC_LOCAL_CANVAS_CAPTURESTREAM(test) {
+      h.drawColor(canvas, h.green);
+      test.pcLocal.canvasStream = canvas.captureStream(0.0);
+      is(test.pcLocal.canvasStream.canvas, canvas, "Canvas attribute is correct");
+      test.pcLocal.attachLocalStream(test.pcLocal.canvasStream);
+      var i = 0;
+      return setInterval(function() {
+        try {
+          info("draw " + i ? "green" : "red");
+          h.drawColor(canvas, i ? h.green : h.red);
+          i = 1 - i;
+          test.pcLocal.canvasStream.requestFrame();
+        } catch (e) {
+          // ignore; stream might have shut down, and we don't bother clearing
+          // the setInterval.
+        }
+      }, 500);
+    }
+  ]);
+  test.chain.append([
+    function FIND_REMOTE_VIDEO() {
+      vremote = test.pcRemote.remoteMediaElements[0];
+      ok(!!vremote, "Should have remote video element for pcRemote");
+    },
+    function WAIT_FOR_REMOTE_GREEN() {
+      return h.pixelMustBecome(vremote, h.green, {
+        threshold: 128,
+        infoString: "pcRemote's remote should become green",
+      });
+    },
+    function REQUEST_FRAME(test) {
+      // After requesting a frame it will be captured at the time of next render.
+      // Next render will happen at next stable state, at the earliest,
+      // i.e., this order of `requestFrame(); draw();` should work.
+      test.pcLocal.canvasStream.requestFrame();
+    },
+    function DRAW_LOCAL_RED() {
+      h.drawColor(canvas, h.red);
+    },
+    function WAIT_FOR_REMOTE_RED() {
+      return h.pixelMustBecome(vremote, h.red, {
+        threshold: 128,
+        infoString: "pcRemote's remote should become red",
+      });
+    }
+  ]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_capturedVideo.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_capturedVideo.html
new file mode 100644
index 0000000000..f6f48ba429
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_capturedVideo.html
@@ -0,0 +1,81 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script src="pc.js"></script>
+  <script src="../../../test/manifest.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+(async () => {
+  await createHTML({
+    bug: "1081409",
+    title: "Captured video-only over peer connection",
+    visible: true
+  });
+
+  // Run tests in sequence for log readability.
+  PARALLEL_TESTS = 1;
+  const manager = new MediaTestManager;
+
+  async function startTest(media, token) {
+    manager.started(token);
+    info(`Starting test for ${media.name}`);
+    const video = document.createElement('video');
+    video.id = "id_" + media.name;
+    video.width = 160;
+    video.height = 120;
+    video.muted = true;
+    video.controls = true;
+    video.preload = "metadata";
+    video.src = "../../../test/" + media.name;
+
+    document.getElementById("content").appendChild(video);
+
+    const onerror = new Promise(r => video.onerror = r).then(_ =>
+      new Error(`${media.name} failed in playback. code=${video.error.code}`));
+
+    await Promise.race([
+      new Promise(res => video.onloadedmetadata = res),
+      onerror,
+    ]);
+    onerror.catch(e => ok(false, e));
+    setupEnvironment();
+    await testConfigured;
+    const stream = video.mozCaptureStream();
+    const test = new PeerConnectionTest(
+      {
+        config_local: { label_suffix: media.name },
+        config_remote: { label_suffix: media.name },
+      }
+    );
+    test.setOfferOptions(
+      {
+        offerToReceiveVideo: false,
+        offerToReceiveAudio: false,
+      }
+    );
+    const hasVideo = !!stream.getVideoTracks().length;
+    const hasAudio = !!stream.getAudioTracks().length;
+    test.setMediaConstraints([{ video: hasVideo, audio: hasAudio }], []);
+    test.chain.replace("PC_LOCAL_GUM", [
+      function PC_LOCAL_CAPTUREVIDEO(test) {
+        test.pcLocal.attachLocalStream(stream);
+      },
+    ]);
+    test.chain.insertBefore("PC_LOCAL_WAIT_FOR_MEDIA_FLOW", [
+      function PC_LOCAL_START_MEDIA(test) {
+        video.play();
+      },
+    ]);
+    await test.run();
+    removeNodeAndSource(video);
+    manager.finished(token);
+  }
+
+  manager.runTests(getPlayableVideos(gLongerTests), startTest);
+})();
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_certificates.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_certificates.html
new file mode 100644
index 0000000000..561f285f60
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_certificates.html
@@ -0,0 +1,185 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1172785",
+    title: "Certificate management"
+  });
+
+  function badCertificate(config, expectedError, message) {
+    return RTCPeerConnection.generateCertificate(config)
+      .then(() => ok(false, message),
+            e => is(e.name, expectedError, message));
+  }
+
+  // Checks a handful of obviously bad options to RTCCertificate.create().  Most
+  // of the checking is done by the WebCrypto code underpinning this, hence the
+  // baffling error codes, but a sanity check is still in order.
+  function checkBadParameters() {
+    return Promise.all([
+      badCertificate({
+        name: "RSASSA-PKCS1-v1_5",
+        hash: "SHA-256",
+        modulusLength: 1023,
+        publicExponent: new Uint8Array([1, 0, 1])
+      }, "NotSupportedError", "1023-bit is too small to succeed"),
+
+      badCertificate({
+        name: "RSASSA-PKCS1-v1_5",
+          hash: "SHA-384",
+          modulusLength: 2048,
+          publicExponent: new Uint8Array([1, 0, 1])
+      }, "NotSupportedError", "SHA-384 isn't supported yet"),
+
+      // A SyntaxError happens in the "generate key operation" step, but
+      // webrtc-pc does not say to reject the promise if this step fails.
+      // It does say to throw NotSupportedError if we have passed "an
+      // algorithm that the user agent cannot or will not use to generate a
+      // certificate".
+      badCertificate({
+        name: "ECDH",
+        namedCurve: "P-256"
+      }, "NotSupportedError", "ECDH is rejected because the usage is neither \"deriveKey\" or \"deriveBits\""),
+
+      badCertificate({
+        name: "not a valid algorithm"
+      }, "NotSupportedError", "not a valid algorithm"),
+
+      badCertificate("ECDSA", "NotSupportedError", "a bare name is not enough"),
+
+      badCertificate({
+        name: "ECDSA",
+        namedCurve: "not a curve"
+      }, "NotSupportedError", "ECDSA with an unknown curve")
+    ]);
+  }
+
+  function createDB() {
+    var openDB = indexedDB.open("genericstore");
+    openDB.onupgradeneeded = e => {
+      var db = e.target.result;
+      db.createObjectStore("data");
+    };
+    return new Promise(resolve => {
+      openDB.onsuccess = e => resolve(e.target.result);
+    });
+  }
+
+  function resultPromise(tx, op) {
+    return new Promise((resolve, reject) => {
+      op.onsuccess = e => resolve(e.target.result);
+      op.onerror = () => reject(op.error);
+      tx.onabort = () => reject(tx.error);
+    });
+  }
+
+  function store(db, value) {
+    var tx = db.transaction("data", "readwrite");
+    var store = tx.objectStore("data");
+    return resultPromise(tx, store.put(value, "value"));
+  }
+
+  function retrieve(db) {
+    var tx = db.transaction("data", "readonly");
+    var store = tx.objectStore("data");
+    return resultPromise(tx, store.get("value"));
+  }
+
+  // Creates a database, stores a value, retrieves it.
+  function storeAndRetrieve(value) {
+    return createDB().then(db => {
+      return store(db, value)
+        .then(() => retrieve(db))
+        .then(retrieved => {
+          db.close();
+          return retrieved;
+        });
+    });
+  }
+
+  var test;
+  runNetworkTest(function (options) {
+    var expiredCert;
+    return Promise.resolve()
+      .then(() => RTCPeerConnection.generateCertificate({
+        name: "ECDSA",
+        namedCurve: "P-256",
+        expires: 1 // smallest possible expiration window
+      }))
+      .then(cert => {
+        ok(!isNaN(cert.expires), 'cert has expiration time');
+        info('Expires at ' + new Date(cert.expires));
+        expiredCert = cert;
+      })
+
+      .then(() => checkBadParameters())
+
+      .then(() => {
+        var delay = expiredCert.expires - Date.now();
+        // Hopefully this delay is never needed.
+        if (delay > 0) {
+          return new Promise(r => setTimeout(r, delay));
+        }
+      })
+      .then(() => {
+        ok(expiredCert.expires <= Date.now(), 'Cert should be at or past expiration');
+        try {
+          new RTCPeerConnection({ certificates: [expiredCert] });
+          ok(false, 'Constructing peer connection with an expired cert is not allowed');
+        } catch(e) {
+          is(e.name, 'InvalidAccessError',
+             'Constructing peer connection with an expired certs is not allowed');
+        }
+      })
+
+      .then(() => Promise.all([
+        RTCPeerConnection.generateCertificate({
+          name: "ECDSA",
+          namedCurve: "P-256"
+        }),
+        RTCPeerConnection.generateCertificate({
+          name: "RSASSA-PKCS1-v1_5",
+          hash: "SHA-256",
+          modulusLength: 2048,
+          publicExponent: new Uint8Array([1, 0, 1])
+        })
+      ]))
+
+    // A round trip through indexedDB should not do anything.
+      .then(storeAndRetrieve)
+      .then(certs => {
+        try {
+          new RTCPeerConnection({ certificates: certs });
+          ok(false, 'Constructing peer connection with multiple certs is not allowed');
+        } catch(e) {
+          is(e.name, 'NotSupportedError',
+             'Constructing peer connection with multiple certs is not allowed');
+        }
+        return certs;
+      })
+      .then(certs => {
+        test = new PeerConnectionTest({
+          config_local: {
+            certificates: [certs[0]]
+          },
+          config_remote: {
+            certificates: [certs[1]]
+          }
+        });
+        test.setMediaConstraints([{audio: true}], [{audio: true}]);
+        return test.run();
+      })
+      .catch(e => {
+        console.log('test failure', e);
+        ok(false, 'test failed: ' + e);
+      });
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_checkPacketDumpHook.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_checkPacketDumpHook.html
new file mode 100644
index 0000000000..248e102dd2
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_checkPacketDumpHook.html
@@ -0,0 +1,107 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1377299",
+    title: "Check that packet dump hooks generate callbacks"
+  });
+
+  function waitForPacket(pc, checkFunction) {
+    return new Promise(resolve => {
+      function onPacket(level, type, sending, packet) {
+        if (checkFunction(level, type, sending, packet)) {
+          SpecialPowers.wrap(pc).mozSetPacketCallback(() => {});
+          resolve();
+        }
+      }
+
+      SpecialPowers.wrap(pc).mozSetPacketCallback(onPacket);
+    }
+    );
+  }
+
+  async function waitForSendPacket(pc, type, level) {
+    await SpecialPowers.wrap(pc).mozEnablePacketDump(level, type, true);
+    await timeout(
+      waitForPacket(pc, (obsLevel, obsType, sending) => {
+        is(obsLevel, level, "Level for packet is " + level);
+        is(obsType, type, "Type for packet is " + type);
+        ok(sending, "This is a send packet");
+        return true;
+      }),
+      10000, "Timeout waiting for " + type + " send packet on level " + level);
+    await SpecialPowers.wrap(pc).mozDisablePacketDump(level, type, true);
+  }
+
+  async function waitForRecvPacket(pc, type, level) {
+    await SpecialPowers.wrap(pc).mozEnablePacketDump(level, type, false);
+    await timeout(
+      waitForPacket(pc, (obsLevel, obsType, sending) => {
+        is(obsLevel, level, "Level for packet is " + level);
+        is(obsType, type, "Type for packet is " + type);
+        ok(!sending, "This is a recv packet");
+        return true;
+      }),
+      10000, "Timeout waiting for " + type + " recv packet on level " + level);
+    await SpecialPowers.wrap(pc).mozDisablePacketDump(level, type, false);
+  }
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true, video: true}],
+                             [{audio: true, video: true}]);
+    // pc.js uses video elements by default, we want to test audio elements here
+    test.pcLocal.audioElementsOnly = true;
+
+    test.chain.insertBefore('PC_LOCAL_WAIT_FOR_MEDIA_FLOW',[
+        async function PC_LOCAL_CHECK_PACKET_DUMP_HOOKS() {
+          await waitForRecvPacket(test.pcLocal._pc, "rtp", 0);
+          await waitForRecvPacket(test.pcLocal._pc, "rtcp", 0);
+          await waitForRecvPacket(test.pcLocal._pc, "srtp", 0);
+          await waitForRecvPacket(test.pcLocal._pc, "srtcp", 0);
+          await waitForSendPacket(test.pcLocal._pc, "rtp", 0);
+          await waitForSendPacket(test.pcLocal._pc, "rtcp", 0);
+          await waitForSendPacket(test.pcLocal._pc, "srtp", 0);
+          await waitForSendPacket(test.pcLocal._pc, "srtcp", 0);
+
+          await waitForRecvPacket(test.pcLocal._pc, "rtp", 1);
+          await waitForRecvPacket(test.pcLocal._pc, "rtcp", 1);
+          await waitForRecvPacket(test.pcLocal._pc, "srtp", 1);
+          await waitForRecvPacket(test.pcLocal._pc, "srtcp", 1);
+          await waitForSendPacket(test.pcLocal._pc, "rtp", 1);
+          await waitForSendPacket(test.pcLocal._pc, "rtcp", 1);
+          await waitForSendPacket(test.pcLocal._pc, "srtp", 1);
+          await waitForSendPacket(test.pcLocal._pc, "srtcp", 1);
+        },
+        async function PC_REMOTE_CHECK_PACKET_DUMP_HOOKS() {
+          await waitForRecvPacket(test.pcRemote._pc, "rtp", 0);
+          await waitForRecvPacket(test.pcRemote._pc, "rtcp", 0);
+          await waitForRecvPacket(test.pcRemote._pc, "srtp", 0);
+          await waitForRecvPacket(test.pcRemote._pc, "srtcp", 0);
+          await waitForSendPacket(test.pcRemote._pc, "rtp", 0);
+          await waitForSendPacket(test.pcRemote._pc, "rtcp", 0);
+          await waitForSendPacket(test.pcRemote._pc, "srtp", 0);
+          await waitForSendPacket(test.pcRemote._pc, "srtcp", 0);
+
+          await waitForRecvPacket(test.pcRemote._pc, "rtp", 1);
+          await waitForRecvPacket(test.pcRemote._pc, "rtcp", 1);
+          await waitForRecvPacket(test.pcRemote._pc, "srtp", 1);
+          await waitForRecvPacket(test.pcRemote._pc, "srtcp", 1);
+          await waitForSendPacket(test.pcRemote._pc, "rtp", 1);
+          await waitForSendPacket(test.pcRemote._pc, "rtcp", 1);
+          await waitForSendPacket(test.pcRemote._pc, "srtp", 1);
+          await waitForSendPacket(test.pcRemote._pc, "srtcp", 1);
+        }
+    ]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_close.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_close.html
new file mode 100644
index 0000000000..3edf677203
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_close.html
@@ -0,0 +1,134 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "991877",
+    title: "Basic RTCPeerConnection.close() tests"
+  });
+
+  runNetworkTest(function () {
+    var pc = new RTCPeerConnection();
+    var sender = pc.addTrack(getSilentTrack(), new MediaStream());
+    var exception = null;
+    var eTimeout = null;
+
+    // everything should be in initial state
+    is(pc.signalingState, "stable", "Initial signalingState is 'stable'");
+    is(pc.iceConnectionState, "new", "Initial iceConnectionState is 'new'");
+    is(pc.iceGatheringState, "new", "Initial iceGatheringState is 'new'");
+
+    var finish;
+    var finished = new Promise(resolve => finish = resolve);
+
+    var mustNotSettle = (p, ms, msg) => Promise.race([
+      p.then(() => ok(false, msg + " must not settle"),
+             e => ok(false, msg + " must not settle. Got " + e.name)),
+      wait(ms).then(() => ok(true, msg + " must not settle"))
+    ]);
+
+    var silence = mustNotSettle(pc.createOffer(), 1000,
+                                "createOffer immediately followed by close");
+    try {
+      pc.close();
+    } catch (e) {
+      exception = e;
+    }
+    is(exception, null, "closing the connection raises no exception");
+    is(pc.signalingState, "closed", "Final signalingState is 'closed'");
+    is(pc.iceConnectionState, "closed", "Final iceConnectionState is 'closed'");
+
+    // test that pc is really closed (and doesn't crash, bug 1259728)
+    try {
+      pc.getLocalStreams();
+    } catch (e) {
+      exception = e;
+    }
+    is(exception && exception.name, "InvalidStateError",
+       "pc.getLocalStreams should throw when closed");
+    exception = null;
+
+    try {
+      pc.close();
+    } catch (e) {
+      exception = e;
+    }
+    is(exception, null, "A second close() should not raise an exception");
+    is(pc.signalingState, "closed", "Final signalingState stays at 'closed'");
+    is(pc.iceConnectionState, "closed", "Final iceConnectionState stays at 'closed'");
+
+    // Due to a limitation in our WebIDL compiler that prevents overloads with
+    // both Promise and non-Promise return types, legacy APIs with callbacks
+    // are unable to continue to throw exceptions. Luckily the spec uses
+    // exceptions solely for "programming errors" so this should not hinder
+    // working code from working, which is the point of the legacy API. All
+    // new code should use the promise API.
+    //
+    // The legacy methods that no longer throw on programming errors like
+    // "invalid-on-close" are:
+    // - createOffer
+    // - createAnswer
+    // - setLocalDescription
+    // - setRemoteDescription
+    // - addIceCandidate
+    // - getStats
+    //
+    // These legacy methods fire the error callback instead. This is not
+    // entirely to spec but is better than ignoring programming errors.
+
+    var offer = new RTCSessionDescription({ sdp: "sdp", type: "offer" });
+    var answer = new RTCSessionDescription({ sdp: "sdp", type: "answer" });
+    var candidate = new RTCIceCandidate({ candidate: "dummy",
+                                             sdpMid: "test",
+                                             sdpMLineIndex: 3 });
+
+    var doesFail = (p, msg) => p.then(generateErrorCallback(msg),
+                                      r => is(r.name, "InvalidStateError", msg));
+    Promise.all([
+      [pc.createOffer(), "createOffer"],
+      [pc.createOffer({offerToReceiveAudio: true}), "createOffer({offerToReceiveAudio: true})"],
+      [pc.createOffer({offerToReceiveAudio: false}), "createOffer({offerToReceiveAudio: false})"],
+      [pc.createOffer({offerToReceiveVideo: true}), "createOffer({offerToReceiveVideo: true})"],
+      [pc.createOffer({offerToReceiveVideo: false}), "createOffer({offerToReceiveVideo: false})"],
+      [pc.createAnswer(), "createAnswer"],
+      [pc.setLocalDescription(offer), "setLocalDescription"],
+      [pc.setRemoteDescription(answer), "setRemoteDescription"],
+      [pc.addIceCandidate(candidate), "addIceCandidate"],
+      [new Promise((y, n) => pc.createOffer(y, n)), "Legacy createOffer"],
+      [new Promise((y, n) => pc.createAnswer(y, n)), "Legacy createAnswer"],
+      [new Promise((y, n) => pc.setLocalDescription(offer, y, n)), "Legacy setLocalDescription"],
+      [new Promise((y, n) => pc.setRemoteDescription(answer, y, n)), "Legacy setRemoteDescription"],
+      [new Promise((y, n) => pc.addIceCandidate(candidate, y, n)), "Legacy addIceCandidate"],
+      [sender.replaceTrack(getSilentTrack()), "replaceTrack"],
+    ].map(([p, name]) => doesFail(p, name + " fails on close")))
+    .catch(reason => ok(false, "unexpected failure: " + reason))
+    .then(finish);
+
+    // Other methods are unaffected.
+
+    SimpleTest.doesThrow(function() {
+      pc.updateIce("Invalid RTC Configuration")},
+      "updateIce() on closed PC raised expected exception");
+
+    SimpleTest.doesThrow(function() {
+      pc.addStream("Invalid Media Stream")},
+      "addStream() on closed PC raised expected exception");
+
+    SimpleTest.doesThrow(function() {
+      pc.createDataChannel({})},
+      "createDataChannel() on closed PC raised expected exception");
+
+    SimpleTest.doesThrow(function() {
+      pc.setIdentityProvider("Invalid Provider")},
+      "setIdentityProvider() on closed PC raised expected exception");
+
+    return Promise.all([finished, silence]);
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_closeDuringIce.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_closeDuringIce.html
new file mode 100644
index 0000000000..db3a2922d5
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_closeDuringIce.html
@@ -0,0 +1,79 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1087629",
+    title: "Close PCs during ICE connectivity check"
+  });
+
+// Test closeDuringIce to simulate problems during peer connections
+
+
+function PC_LOCAL_SETUP_NULL_ICE_HANDLER(test) {
+  test.pcLocal.setupIceCandidateHandler(test, function() {}, function () {});
+}
+function PC_REMOTE_SETUP_NULL_ICE_HANDLER(test) {
+  test.pcRemote.setupIceCandidateHandler(test, function() {}, function () {});
+}
+function PC_REMOTE_ADD_FAKE_ICE_CANDIDATE(test) {
+  var cand = {"candidate":"candidate:0 1 UDP 2130379007 192.0.2.1 12345 typ host","sdpMid":"","sdpMLineIndex":0};
+  test.pcRemote.storeOrAddIceCandidate(cand);
+  info(test.pcRemote + " Stored fake candidate: " + JSON.stringify(cand));
+}
+function PC_LOCAL_ADD_FAKE_ICE_CANDIDATE(test) {
+  var cand = {"candidate":"candidate:0 1 UDP 2130379007 192.0.2.2 56789 typ host","sdpMid":"","sdpMLineIndex":0};
+    test.pcLocal.storeOrAddIceCandidate(cand);
+  info(test.pcLocal + " Stored fake candidate: " + JSON.stringify(cand));
+}
+function PC_LOCAL_CLOSE_DURING_ICE(test) {
+  return test.pcLocal.iceChecking.then(() => {
+    test.pcLocal.onsignalingstatechange = function () {};
+    test.pcLocal.close();
+    });
+}
+function PC_REMOTE_CLOSE_DURING_ICE(test) {
+  return test.pcRemote.iceChecking.then(() => {
+    test.pcRemote.onsignalingstatechange = function () {};
+    test.pcRemote.close();
+    });
+}
+function PC_LOCAL_WAIT_FOR_ICE_CHECKING(test) {
+  var resolveIceChecking;
+  test.pcLocal.iceChecking = new Promise(r => resolveIceChecking = r);
+  test.pcLocal.ice_connection_callbacks.checkIceStatus = () => {
+	if (test.pcLocal._pc.iceConnectionState === "checking") {
+	  resolveIceChecking();
+	}
+  }
+}
+function PC_REMOTE_WAIT_FOR_ICE_CHECKING(test) {
+  var resolveIceChecking;
+  test.pcRemote.iceChecking = new Promise(r => resolveIceChecking = r);
+  test.pcRemote.ice_connection_callbacks.checkIceStatus = () => {
+	if (test.pcRemote._pc.iceConnectionState === "checking") {
+	  resolveIceChecking();
+	}
+  }
+}
+
+runNetworkTest(() => {
+  var test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.chain.replace("PC_LOCAL_SETUP_ICE_HANDLER", PC_LOCAL_SETUP_NULL_ICE_HANDLER);
+  test.chain.replace("PC_REMOTE_SETUP_ICE_HANDLER", PC_REMOTE_SETUP_NULL_ICE_HANDLER);
+  test.chain.insertAfter("PC_REMOTE_SETUP_NULL_ICE_HANDLER", PC_LOCAL_WAIT_FOR_ICE_CHECKING);
+  test.chain.insertAfter("PC_LOCAL_WAIT_FOR_ICE_CHECKING", PC_REMOTE_WAIT_FOR_ICE_CHECKING);
+  test.chain.removeAfter("PC_LOCAL_SET_REMOTE_DESCRIPTION");
+  test.chain.append([PC_REMOTE_ADD_FAKE_ICE_CANDIDATE, PC_LOCAL_ADD_FAKE_ICE_CANDIDATE,
+	PC_LOCAL_CLOSE_DURING_ICE, PC_REMOTE_CLOSE_DURING_ICE]);
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_codecNegotiationFailure.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_codecNegotiationFailure.html
new file mode 100644
index 0000000000..819e13fe1b
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_codecNegotiationFailure.html
@@ -0,0 +1,111 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="iceTestUtils.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1683934",
+    title: "RTCPeerConnection check codec negotiation failure"
+  });
+
+  function makeWeirdCodecs(sdp) {
+    return sdp
+      .replaceAll('VP8', 'VEEEEEEEEP8')
+      .replaceAll('VP9', 'VEEEEEEEEP9')
+      .replaceAll('H264', 'HERP264');
+  }
+
+  const tests = [
+    async function offererWeirdCodecs() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({video: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      const offer = await pc1.createOffer();
+      offer.sdp = makeWeirdCodecs(offer.sdp);
+      // It is not an error to receive an offer with no codecs we support
+      await pc2.setRemoteDescription(offer);
+      await pc2.setLocalDescription();
+      await wait(2000);
+    },
+
+    async function answererWeirdCodecs() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({video: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      await pc1.setLocalDescription();
+      await pc2.setRemoteDescription(pc1.localDescription);
+      const answer = await pc2.createAnswer();
+      answer.sdp = makeWeirdCodecs(answer.sdp);
+      try {
+        await pc1.setRemoteDescription(answer);
+        ok(false, "Should have thrown");
+      } catch (e) {
+        ok(true, "Should have thrown");
+      }
+    },
+
+    async function reoffererWeirdCodecs() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({video: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      await connect(pc1, pc2, 32000, "Initial connection");
+
+      const offer = await pc1.createOffer();
+      offer.sdp = makeWeirdCodecs(offer.sdp);
+      // It is not an error to receive an offer with no codecs we support
+      await pc2.setRemoteDescription(offer);
+      await pc2.setLocalDescription();
+      await wait(2000);
+    },
+
+    async function reanswererWeirdCodecs() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({video: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      await connect(pc1, pc2, 32000, "Initial connection");
+      await pc1.setLocalDescription();
+      await pc2.setRemoteDescription(pc1.localDescription);
+      const answer = await pc2.createAnswer();
+      answer.sdp = makeWeirdCodecs(answer.sdp);
+      try {
+        await pc1.setRemoteDescription(answer);
+        ok(false, "Should have thrown");
+      } catch (e) {
+        ok(true, "Should have thrown");
+      }
+    },
+
+  ];
+
+  runNetworkTest(async () => {
+    for (const test of tests) {
+      info(`Running test: ${test.name}`);
+      await test();
+      info(`Done running test: ${test.name}`);
+    }
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_constructedStream.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_constructedStream.html
new file mode 100644
index 0000000000..8431b7534e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_constructedStream.html
@@ -0,0 +1,67 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1271669",
+  title: "Test that pc.addTrack() accepts any MediaStream",
+  visible: true
+});
+
+runNetworkTest(() => {
+  var test = new PeerConnectionTest();
+  var constructedStream;
+  var dummyStream = new MediaStream();
+  var dummyStreamTracks = [];
+
+  test.setMediaConstraints([ {audio: true, video: true}
+                           , {audio: true}
+                           , {video: true}
+                           ], []);
+  test.chain.replace("PC_LOCAL_GUM", [
+    function PC_LOCAL_GUM_CONSTRUCTED_STREAM(test) {
+      return getUserMedia(test.pcLocal.constraints[0]).then(stream => {
+        constructedStream = new MediaStream(stream.getTracks());
+        test.pcLocal.attachLocalStream(constructedStream);
+      });
+    },
+    function PC_LOCAL_GUM_DUMMY_STREAM(test) {
+      return getUserMedia(test.pcLocal.constraints[1])
+        .then(stream => dummyStreamTracks.push(...stream.getTracks()))
+        .then(() => getUserMedia(test.pcLocal.constraints[2]))
+        .then(stream => dummyStreamTracks.push(...stream.getTracks()))
+        .then(() => dummyStreamTracks.forEach(t =>
+          test.pcLocal.attachLocalTrack(t, dummyStream)));
+    },
+  ]);
+
+  let checkSentTracksReceived = (sentStreamId, sentTracks) => {
+    let receivedStream =
+      test.pcRemote._pc.getRemoteStreams().find(s => s.id == sentStreamId);
+    ok(receivedStream, "We should receive a stream with with the sent stream's id (" + sentStreamId + ")");
+    if (!receivedStream) {
+      return;
+    }
+
+    is(receivedStream.getTracks().length, sentTracks.length,
+       "Should receive same number of tracks as were sent");
+  };
+
+  test.chain.append([
+    function PC_REMOTE_CHECK_RECEIVED_CONSTRUCTED_STREAM() {
+      checkSentTracksReceived(constructedStream.id, constructedStream.getTracks());
+    },
+    function PC_REMOTE_CHECK_RECEIVED_DUMMY_STREAM() {
+      checkSentTracksReceived(dummyStream.id, dummyStreamTracks);
+    },
+  ]);
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_disabledVideoPreNegotiation.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_disabledVideoPreNegotiation.html
new file mode 100644
index 0000000000..4c06de792e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_disabledVideoPreNegotiation.html
@@ -0,0 +1,45 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1570673",
+    title: "Sending an initially disabled video track should be playable remotely",
+    visible: true,
+  });
+
+  var test;
+  runNetworkTest(async (options) => {
+    // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+    if (navigator.userAgent.includes("Android")) {
+      await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                      ["media.webrtc.hw.h264.enabled", false]);
+    }
+
+    test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{video: true}], []);
+    test.chain.insertAfter("PC_LOCAL_GUM", function PC_LOCAL_DISABLE_VIDEO() {
+      for (const {track} of test.pcLocal._pc.getSenders()) {
+        if (track.kind == "video") {
+          track.enabled = false;
+        }
+      }
+    });
+    test.chain.append(async function PC_REMOTE_RECEIVING_BLACK() {
+      const v = test.pcRemote.remoteMediaElements[0];
+      is(v.readyState, v.HAVE_ENOUGH_DATA, "video element should be playing");
+      const h = new CaptureStreamTestHelper2D();
+      await h.waitForPixel(test.pcRemote.remoteMediaElements[0],
+        px => h.isPixel(px, h.black, 128));
+    });
+    await test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_encodingsNegotiation.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_encodingsNegotiation.html
new file mode 100644
index 0000000000..f46d7eb0d2
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_encodingsNegotiation.html
@@ -0,0 +1,85 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="simulcast.js"></script>
+  <script type="application/javascript" src="helpers_from_wpt/sdp.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1401592",
+  title: "Simulcast negotiation tests",
+  visible: true
+});
+
+// simulcast negotiation is mostly tested in wpt, but we test a few
+// implementation-specific things here.
+const tests = [
+  async function checkVideoEncodingLimit() {
+    const pc1 = new RTCPeerConnection();
+    const pc2 = new RTCPeerConnection();
+
+    const stream = await navigator.mediaDevices.getUserMedia({video: true});
+    const sender = pc1.addTrack(stream.getTracks()[0]);
+    pc2.addTrack(stream.getTracks()[0]);
+
+    await doOfferToRecvSimulcast(pc2, pc1, ["1", "2", "3", "4"]);
+
+    const {encodings} = sender.getParameters();
+    const rids = encodings.map(({rid}) => rid);
+    isDeeply(rids, ["1", "2", "3"]);
+
+    pc1.close();
+    pc2.close();
+    stream.getTracks().forEach(track => track.stop());
+  },
+
+  // wpt currently does not assume support for 3 encodings, which limits the
+  // effectiveness of its powers-of-2 test (since it can test only for 1 and 2)
+  async function checkScaleResolutionDownByAutoFillPowersOf2() {
+    const pc1 = new RTCPeerConnection();
+    const pc2 = new RTCPeerConnection();
+    const stream = await navigator.mediaDevices.getUserMedia({video: true});
+    const sender = pc1.addTrack(stream.getTracks()[0]);
+    pc2.addTrack(stream.getTracks()[0]);
+
+    await doOfferToRecvSimulcast(pc2, pc1, ["1", "2", "3"]);
+
+    const {encodings} = sender.getParameters();
+    const scaleValues = encodings.map(({scaleResolutionDownBy}) => scaleResolutionDownBy);
+    isDeeply(scaleValues, [4, 2, 1]);
+  },
+
+  async function checkLibwebrtcRidLengthLimit() {
+    const pc1 = new RTCPeerConnection();
+    const pc2 = new RTCPeerConnection();
+
+    const stream = await navigator.mediaDevices.getUserMedia({video: true});
+    const sender = pc1.addTrack(stream.getTracks()[0]);
+    pc2.addTrack(stream.getTracks()[0]);
+
+    await doOfferToRecvSimulcast(pc2, pc1, ["foo", "wibblywobblyjeremybearimy"]);
+    const {encodings} = sender.getParameters();
+    const rids = encodings.map(({rid}) => rid);
+    isDeeply(rids, ["foo"]);
+
+    pc1.close();
+    pc2.close();
+    stream.getTracks().forEach(track => track.stop());
+  },
+];
+
+runNetworkTest(async () => {
+  for (const test of tests) {
+    info(`Running test: ${test.name}`);
+    await test();
+    info(`Done running test: ${test.name}`);
+  }
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_errorCallbacks.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_errorCallbacks.html
new file mode 100644
index 0000000000..851a256509
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_errorCallbacks.html
@@ -0,0 +1,55 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "834270",
+    title: "Align PeerConnection error handling with WebRTC specification"
+  });
+
+  function validateReason(reason) {
+    ok(reason.name.length, "Reason name = " + reason.name);
+    ok(reason.message.length, "Reason message = " + reason.message);
+  };
+
+  function testCreateAnswerError() {
+    var pc = new RTCPeerConnection();
+    info ("Testing createAnswer error");
+    return pc.createAnswer()
+    .then(generateErrorCallback("createAnswer before offer should fail"),
+          validateReason);
+  };
+
+  function testSetLocalDescriptionError() {
+    var pc = new RTCPeerConnection();
+    info ("Testing setLocalDescription error");
+    return pc.setLocalDescription({ sdp: "Picklechips!", type: "offer" })
+    .then(generateErrorCallback("setLocalDescription with nonsense SDP should fail"),
+          validateReason);
+  };
+
+  function testSetRemoteDescriptionError() {
+    var pc = new RTCPeerConnection();
+    info ("Testing setRemoteDescription error");
+    return pc.setRemoteDescription({ sdp: "Who?", type: "offer" })
+    .then(generateErrorCallback("setRemoteDescription with nonsense SDP should fail"),
+          validateReason);
+  };
+
+  // No test for createOffer errors -- there's nothing we can do at this
+  // level to evoke an error in createOffer.
+
+  runNetworkTest(function () {
+    return testCreateAnswerError()
+    .then(testSetLocalDescriptionError)
+    .then(testSetRemoteDescriptionError);
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_extmapRenegotiation.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_extmapRenegotiation.html
new file mode 100644
index 0000000000..78c6bb986c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_extmapRenegotiation.html
@@ -0,0 +1,325 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="iceTestUtils.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1799932",
+    title: "RTCPeerConnection check renegotiation of extmap"
+  });
+
+  function setExtmap(sdp, uri, id) {
+    const regex = new RegExp(`a=extmap:[0-9]+(\/[a-z]+)? ${uri}`, 'g');
+    if (id) {
+      return sdp.replaceAll(regex, `a=extmap:${id}$1 ${uri}`);
+    } else {
+      return sdp.replaceAll(regex, `a=unknownattr`);
+    }
+  }
+
+  function getExtmap(sdp, uri) {
+    const regex = new RegExp(`a=extmap:([0-9]+)(\/[a-z]+)? ${uri}`);
+    return sdp.match(regex)[1];
+  }
+
+  function replaceExtUri(sdp, oldUri, newUri) {
+    const regex = new RegExp(`(a=extmap:[0-9]+\/[a-z]+)? ${oldUri}`, 'g');
+    return sdp.replaceAll(regex, `$1 ${newUri}`);
+  }
+
+  const tests = [
+    async function checkAudioMidChange() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      await connect(pc1, pc2, 32000, "Initial connection");
+
+      // Sadly, there's no way to tell the offerer to change the extmap. Other
+      // types of endpoint could conceivably do this, so we at least don't want
+      // to crash.
+      // TODO: Would be nice to be able to test this with an endpoint that
+      // actually changes the ids it uses.
+      const reoffer = await pc1.createOffer();
+      reoffer.sdp = setExtmap(reoffer.sdp, "urn:ietf:params:rtp-hdrext:sdes:mid", 14);
+      info(`New reoffer: ${reoffer.sdp}`);
+      await pc2.setRemoteDescription(reoffer);
+      await pc2.setLocalDescription();
+      await wait(2000);
+    },
+
+    async function checkVideoMidChange() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({video: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      await connect(pc1, pc2, 32000, "Initial connection");
+
+      // Sadly, there's no way to tell the offerer to change the extmap. Other
+      // types of endpoint could conceivably do this, so we at least don't want
+      // to crash.
+      // TODO: Would be nice to be able to test this with an endpoint that
+      // actually changes the ids it uses.
+      const reoffer = await pc1.createOffer();
+      reoffer.sdp = setExtmap(reoffer.sdp, "urn:ietf:params:rtp-hdrext:sdes:mid", 14);
+      info(`New reoffer: ${reoffer.sdp}`);
+      await pc2.setRemoteDescription(reoffer);
+      await pc2.setLocalDescription();
+      await wait(2000);
+    },
+
+    async function checkAudioMidSwap() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      await connect(pc1, pc2, 32000, "Initial connection");
+
+      // Sadly, there's no way to tell the offerer to change the extmap. Other
+      // types of endpoint could conceivably do this, so we at least don't want
+      // to crash.
+      // TODO: Would be nice to be able to test this with an endpoint that
+      // actually changes the ids it uses.
+      const reoffer = await pc1.createOffer();
+      const midId = getExtmap(reoffer.sdp, "urn:ietf:params:rtp-hdrext:sdes:mid");
+      const ssrcLevelId = getExtmap(reoffer.sdp, "urn:ietf:params:rtp-hdrext:ssrc-audio-level");
+      reoffer.sdp = setExtmap(reoffer.sdp, "urn:ietf:params:rtp-hdrext:sdes:mid", ssrcLevelId);
+      reoffer.sdp = setExtmap(reoffer.sdp, "urn:ietf:params:rtp-hdrext:ssrc-audio-level", midId);
+      info(`New reoffer: ${reoffer.sdp}`);
+      try {
+        await pc2.setRemoteDescription(reoffer);
+        ok(false, "sRD should fail when it attempts extension id remapping");
+      } catch (e) {
+        ok(true, "sRD should fail when it attempts extension id remapping");
+      }
+    },
+
+    async function checkVideoMidSwap() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({video: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      await connect(pc1, pc2, 32000, "Initial connection");
+
+      // Sadly, there's no way to tell the offerer to change the extmap. Other
+      // types of endpoint could conceivably do this, so we at least don't want
+      // to crash.
+      // TODO: Would be nice to be able to test this with an endpoint that
+      // actually changes the ids it uses.
+      const reoffer = await pc1.createOffer();
+      const midId = getExtmap(reoffer.sdp, "urn:ietf:params:rtp-hdrext:sdes:mid");
+      const toffsetId = getExtmap(reoffer.sdp, "urn:ietf:params:rtp-hdrext:toffset");
+      reoffer.sdp = setExtmap(reoffer.sdp, "urn:ietf:params:rtp-hdrext:sdes:mid", toffsetId);
+      reoffer.sdp = setExtmap(reoffer.sdp, "urn:ietf:params:rtp-hdrext:toffset", midId);
+      info(`New reoffer: ${reoffer.sdp}`);
+      try {
+        await pc2.setRemoteDescription(reoffer);
+        ok(false, "sRD should fail when it attempts extension id remapping");
+      } catch (e) {
+        ok(true, "sRD should fail when it attempts extension id remapping");
+      }
+    },
+
+    async function checkAudioIdReuse() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      await connect(pc1, pc2, 32000, "Initial connection");
+
+      // Sadly, there's no way to tell the offerer to change the extmap. Other
+      // types of endpoint could conceivably do this, so we at least don't want
+      // to crash.
+      // TODO: Would be nice to be able to test this with an endpoint that
+      // actually changes the ids it uses.
+      const reoffer = await pc1.createOffer();
+      // Change uri, but not the id, so the id now refers to foo.
+      reoffer.sdp = replaceExtUri(reoffer.sdp, "urn:ietf:params:rtp-hdrext:ssrc-audio-level", "foo");
+      info(`New reoffer: ${reoffer.sdp}`);
+      try {
+        await pc2.setRemoteDescription(reoffer);
+        ok(false, "sRD should fail when it attempts extension id remapping");
+      } catch (e) {
+        ok(true, "sRD should fail when it attempts extension id remapping");
+      }
+    },
+
+    async function checkVideoIdReuse() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({video: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      await connect(pc1, pc2, 32000, "Initial connection");
+
+      // Sadly, there's no way to tell the offerer to change the extmap. Other
+      // types of endpoint could conceivably do this, so we at least don't want
+      // to crash.
+      // TODO: Would be nice to be able to test this with an endpoint that
+      // actually changes the ids it uses.
+      const reoffer = await pc1.createOffer();
+      // Change uri, but not the id, so the id now refers to foo.
+      reoffer.sdp = replaceExtUri(reoffer.sdp, "urn:ietf:params:rtp-hdrext:toffset", "foo");
+      info(`New reoffer: ${reoffer.sdp}`);
+      try {
+        await pc2.setRemoteDescription(reoffer);
+        ok(false, "sRD should fail when it attempts extension id remapping");
+      } catch (e) {
+        ok(true, "sRD should fail when it attempts extension id remapping");
+      }
+    },
+
+    // What happens when remote answer uses an extmap id, and then a remote
+    // reoffer tries to use the same id for something else?
+    async function checkAudioIdReuseOffererThenAnswerer() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      await connect(pc1, pc2, 32000, "Initial connection");
+
+      const reoffer = await pc2.createOffer();
+      // Change uri, but not the id, so the id now refers to foo.
+      reoffer.sdp = replaceExtUri(reoffer.sdp, "urn:ietf:params:rtp-hdrext:ssrc-audio-level", "foo");
+      info(`New reoffer: ${reoffer.sdp}`);
+      try {
+        await pc1.setRemoteDescription(reoffer);
+        ok(false, "sRD should fail when it attempts extension id remapping");
+      } catch (e) {
+        ok(true, "sRD should fail when it attempts extension id remapping");
+      }
+    },
+
+    // What happens when a remote offer uses a different extmap id than the
+    // default? Does the answerer remember the new id in reoffers?
+    async function checkAudioIdReuseOffererThenAnswerer() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      // Negotiate, but change id for ssrc-audio-level to something pc2 would
+      // not typically use.
+      await pc1.setLocalDescription();
+      const mungedOffer = setExtmap(pc1.localDescription.sdp, "urn:ietf:params:rtp-hdrext:ssrc-audio-level", 12);
+      await pc2.setRemoteDescription({sdp: mungedOffer, type: "offer"});
+      await pc2.setLocalDescription();
+
+      const reoffer = await pc2.createOffer();
+      is(getExtmap(reoffer.sdp, "urn:ietf:params:rtp-hdrext:ssrc-audio-level"), "12");
+    },
+
+    async function checkAudioUnnegotiatedIdReuse1() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      // Negotiate, but remove ssrc-audio-level from answer
+      await pc1.setLocalDescription();
+      const levelId = getExtmap(pc1.localDescription.sdp, "urn:ietf:params:rtp-hdrext:ssrc-audio-level");
+      await pc2.setRemoteDescription(pc1.localDescription);
+      await pc2.setLocalDescription();
+      const answerNoExt = setExtmap(pc2.localDescription.sdp, "urn:ietf:params:rtp-hdrext:ssrc-audio-level", undefined);
+      await pc1.setRemoteDescription({sdp: answerNoExt, type: "answer"});
+
+      // Renegotiate, and use the id that offerer used for ssrc-audio-level for
+      // something different (while making sure we don't use it twice)
+      await pc2.setLocalDescription();
+      const mungedReoffer = setExtmap(pc2.localDescription.sdp, "urn:ietf:params:rtp-hdrext:sdes:mid", levelId);
+      const twiceMungedReoffer = setExtmap(mungedReoffer, "urn:ietf:params:rtp-hdrext:ssrc-audio-level", undefined);
+      await pc1.setRemoteDescription({sdp: twiceMungedReoffer, type: "offer"});
+    },
+
+    async function checkAudioUnnegotiatedIdReuse2() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      // Negotiate, but remove ssrc-audio-level from offer. pc2 has never seen
+      // |levelId| in extmap yet, but internally probably wants to use that for
+      // ssrc-audio-level
+      await pc1.setLocalDescription();
+      const levelId = getExtmap(pc1.localDescription.sdp, "urn:ietf:params:rtp-hdrext:ssrc-audio-level");
+      const offerNoExt = setExtmap(pc1.localDescription.sdp, "urn:ietf:params:rtp-hdrext:ssrc-audio-level", undefined);
+      await pc2.setRemoteDescription({sdp: offerNoExt, type: "offer"});
+      await pc2.setLocalDescription();
+      await pc1.setRemoteDescription(pc2.localDescription);
+
+      // Renegotiate, but use |levelId| for something other than
+      // ssrc-audio-level. pc2 should not throw.
+      await pc1.setLocalDescription();
+      const mungedReoffer = setExtmap(pc1.localDescription.sdp, "urn:ietf:params:rtp-hdrext:sdes:mid", levelId);
+      const twiceMungedReoffer = setExtmap(mungedReoffer, "urn:ietf:params:rtp-hdrext:ssrc-audio-level", undefined);
+      await pc2.setRemoteDescription({sdp: twiceMungedReoffer, type: "offer"});
+    },
+
+    async function checkAudioUnnegotiatedIdReuse3() {
+      const pc1 = new RTCPeerConnection();
+      const pc2 = new RTCPeerConnection();
+
+      const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+      pc1.addTrack(stream.getTracks()[0]);
+      pc2.addTrack(stream.getTracks()[0]);
+
+      // Negotiate, but replace ssrc-audio-level with something pc2 won't
+      // support in offer.
+      await pc1.setLocalDescription();
+      const levelId = getExtmap(pc1.localDescription.sdp, "urn:ietf:params:rtp-hdrext:ssrc-audio-level");
+      const mungedOffer = replaceExtUri(pc1.localDescription.sdp, "urn:ietf:params:rtp-hdrext:ssrc-audio-level", "fooba");
+      await pc2.setRemoteDescription({sdp: mungedOffer, type: "offer"});
+      await pc2.setLocalDescription();
+      await pc1.setRemoteDescription(pc2.localDescription);
+
+      // Renegotiate, and use levelId for something pc2 _will_ support.
+      await pc1.setLocalDescription();
+      const mungedReoffer = setExtmap(pc1.localDescription.sdp, "urn:ietf:params:rtp-hdrext:sdes:mid", levelId);
+      const twiceMungedReoffer = setExtmap(mungedReoffer, "urn:ietf:params:rtp-hdrext:ssrc-audio-level", undefined);
+      await pc2.setRemoteDescription({sdp: twiceMungedReoffer, type: "offer"});
+    },
+
+  ];
+
+  runNetworkTest(async () => {
+    for (const test of tests) {
+      info(`Running test: ${test.name}`);
+      await test();
+      info(`Done running test: ${test.name}`);
+    }
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_forwarding_basicAudioVideoCombined.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_forwarding_basicAudioVideoCombined.html
new file mode 100644
index 0000000000..84b53a123b
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_forwarding_basicAudioVideoCombined.html
@@ -0,0 +1,41 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "931903",
+    title: "Forwarding a stream from a combined audio/video peerconnection to another"
+  });
+
+runNetworkTest(function() {
+  var gumTest = new PeerConnectionTest();
+
+  var forwardingOptions = { config_local: { label_suffix: "forwarded" },
+                            config_remote: { label_suffix: "forwarded" } };
+  var forwardingTest = new PeerConnectionTest(forwardingOptions);
+
+  gumTest.setMediaConstraints([{audio: true, video: true}], []);
+  forwardingTest.setMediaConstraints([{audio: true, video: true}], []);
+  forwardingTest.chain.replace("PC_LOCAL_GUM", [
+    function PC_FORWARDING_CAPTUREVIDEO(test) {
+      var streams = gumTest.pcRemote._pc.getRemoteStreams();
+      is(streams.length, 1, "One stream to forward");
+      is(streams[0].getTracks().length, 2, "Forwarded stream has 2 tracks");
+      forwardingTest.pcLocal.attachLocalStream(streams[0]);
+      return Promise.resolve();
+    }
+  ]);
+  gumTest.chain.removeAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW");
+  return gumTest.chain.execute()
+    .then(() => forwardingTest.chain.execute())
+    .then(() => gumTest.close())
+    .then(() => forwardingTest.close());
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_gatherWithSetConfiguration.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_gatherWithSetConfiguration.html
new file mode 100644
index 0000000000..6710e628aa
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_gatherWithSetConfiguration.html
@@ -0,0 +1,450 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="iceTestUtils.js"></script>
+  <script type="application/javascript" src="helpers_from_wpt/sdp.js"></script></head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1253706",
+    title: "Test ICE gathering when setConfiguration is used to change the ICE config"
+  });
+
+  const tests = [
+    async function baselineV4Cases() {
+      await checkSrflx([{urls:[`stun:${turnAddressV4}`]}]);
+      await checkRelayUdp([{urls:[`turn:${turnAddressV4}`], username, credential}]);
+      await checkRelayTcp([{urls:[`turn:${turnAddressV4}?transport=tcp`], username, credential}]);
+      await checkRelayUdpTcp([{urls:[`turn:${turnAddressV4}`, `turn:${turnAddressV4}?transport=tcp`], username, credential}]);
+      await checkNoSrflx();
+      await checkNoRelay();
+    },
+
+    async function addStunServerBeforeOffer() {
+      const pc = new RTCPeerConnection();
+      try {
+        pc.setConfiguration({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+        const candidates = await gatherWithTimeout(pc, 32000, `just a stun server`);
+        ok(candidates.some(c => c.candidate.includes("srflx")), "Should get at least one srflx candidate");
+        ok(!candidates.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function addTurnServerBeforeOffer() {
+      const pc = new RTCPeerConnection();
+      try {
+        pc.setConfiguration({iceServers: [{urls:[`turn:${turnAddressV4}`], username, credential}]});
+        const candidates = await gatherWithTimeout(pc, 32000, `a turn (udp) server`);
+        ok(candidates.some(c => c.candidate.includes("srflx")), "Should get at least one srflx candidate");
+        ok(candidates.some(c => c.candidate.includes("relay")), "Should get at least one relay candidate");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function addTurnTcpServerBeforeOffer() {
+      const pc = new RTCPeerConnection();
+      try {
+        pc.setConfiguration({iceServers: [{urls:[`turn:${turnAddressV4}?transport=tcp`], username, credential}]});
+        const candidates = await gatherWithTimeout(pc, 32000, `a turn (tcp) server`);
+        ok(!candidates.some(c => c.candidate.includes("srflx")), "Should not get any srflx candidates");
+        ok(candidates.some(c => c.candidate.includes("relay")), "Should get at least one relay candidate");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function addStunServerAfterOffer() {
+      const pc = new RTCPeerConnection();
+      try {
+        const candidates1 = await gatherWithTimeout(pc, 32000, `no ICE servers`);
+        ok(!candidates1.some(c => c.candidate.includes("srflx")), "Should not get any srflx candidates");
+        ok(!candidates1.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+        await pc.setLocalDescription({type: "rollback"});
+
+        pc.setConfiguration({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+        const candidates2 = await gatherWithTimeout(pc, 32000, `just a stun server`);
+        ok(candidates2.some(c => c.candidate.includes("srflx")), "Should get at least one srflx candidate");
+        ok(!candidates2.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function addTurnServerAfterOffer() {
+      const pc = new RTCPeerConnection();
+      try {
+        const candidates1 = await gatherWithTimeout(pc, 32000, `no ICE servers`);
+        ok(!candidates1.some(c => c.candidate.includes("srflx")), "Should not get any srflx candidates");
+        ok(!candidates1.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+        await pc.setLocalDescription({type: "rollback"});
+
+        pc.setConfiguration({iceServers: [{urls:[`turn:${turnAddressV4}`], username, credential}]});
+        const candidates2 = await gatherWithTimeout(pc, 32000, `a turn (udp) server`);
+        ok(candidates2.some(c => c.candidate.includes("srflx")), "Should get at least one srflx candidate");
+        ok(candidates2.some(c => c.candidate.includes("relay")), "Should get at least one relay candidate");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function addTurnTcpServerAfterOffer() {
+      const pc = new RTCPeerConnection();
+      try {
+        const candidates1 = await gatherWithTimeout(pc, 32000, `no ICE servers`);
+        ok(!candidates1.some(c => c.candidate.includes("srflx")), "Should not get any srflx candidates");
+        ok(!candidates1.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+        await pc.setLocalDescription({type: "rollback"});
+
+        pc.setConfiguration({iceServers: [{urls:[`turn:${turnAddressV4}?transport=tcp`], username, credential}]});
+        const candidates2 = await gatherWithTimeout(pc, 32000, `a turn (tcp) server`);
+        ok(!candidates2.some(c => c.candidate.includes("srflx")), "Should get no srflx candidates");
+        ok(candidates2.some(c => c.candidate.includes("relay")), "Should get at least one relay candidate");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function removeStunServerBeforeOffer() {
+      const pc = new RTCPeerConnection({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+      try {
+        pc.setConfiguration({});
+        const candidates = await gatherWithTimeout(pc, 32000, `no ICE servers`);
+        ok(!candidates.some(c => c.candidate.includes("srflx")), "Should not get any srflx candidates");
+        ok(!candidates.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function removeTurnServerBeforeOffer() {
+      const pc = new RTCPeerConnection({iceServers: [{urls:[`turn:${turnAddressV4}`], username, credential}]});
+      try {
+        pc.setConfiguration({});
+        const candidates = await gatherWithTimeout(pc, 32000, `no ICE servers`);
+        ok(!candidates.some(c => c.candidate.includes("srflx")), "Should not get any srflx candidates");
+        ok(!candidates.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function removeTurnTcpServerBeforeOffer() {
+      const pc = new RTCPeerConnection({iceServers: [{urls:[`turn:${turnAddressV4}?transport=tcp`], username, credential}]});
+      try {
+        pc.setConfiguration({});
+        const candidates = await gatherWithTimeout(pc, 32000, `no ICE servers`);
+        ok(!candidates.some(c => c.candidate.includes("srflx")), "Should not get any srflx candidates");
+        ok(!candidates.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function removeStunServerAfterOffer() {
+      const pc = new RTCPeerConnection({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+      try {
+        const candidates1 = await gatherWithTimeout(pc, 32000, `just a stun server`);
+        ok(candidates1.some(c => c.candidate.includes("srflx")), "Should get at least one srflx candidate");
+        ok(!candidates1.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+        await pc.setLocalDescription({type: "rollback"});
+
+        pc.setConfiguration({});
+        const candidates2 = await gatherWithTimeout(pc, 32000, `no ICE servers`);
+        ok(!candidates2.some(c => c.candidate.includes("srflx")), "Should not get any srflx candidates");
+        ok(!candidates2.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function removeTurnServerAfterOffer() {
+      const pc = new RTCPeerConnection({iceServers: [{urls:[`turn:${turnAddressV4}`], username, credential}]});
+      try {
+        const candidates1 = await gatherWithTimeout(pc, 32000, `a turn (udp) server`);
+        ok(candidates1.some(c => c.candidate.includes("srflx")), "Should get at least one srflx candidate");
+        ok(candidates1.some(c => c.candidate.includes("relay")), "Should get at least one relay candidate");
+        await pc.setLocalDescription({type: "rollback"});
+
+        pc.setConfiguration({});
+        const candidates2 = await gatherWithTimeout(pc, 32000, `no ICE servers`);
+        ok(!candidates2.some(c => c.candidate.includes("srflx")), "Should not get any srflx candidates");
+        ok(!candidates2.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function removeTurnTcpServerAfterOffer() {
+      const pc = new RTCPeerConnection({iceServers: [{urls:[`turn:${turnAddressV4}?transport=tcp`], username, credential}]});
+      try {
+        const candidates1 = await gatherWithTimeout(pc, 32000, `a turn (tcp) server`);
+        ok(!candidates1.some(c => c.candidate.includes("srflx")), "Should get no srflx candidates");
+        ok(candidates1.some(c => c.candidate.includes("relay")), "Should get at least one relay candidate");
+        await pc.setLocalDescription({type: "rollback"});
+
+        pc.setConfiguration({});
+        const candidates2 = await gatherWithTimeout(pc, 32000, `no ICE servers`);
+        ok(!candidates2.some(c => c.candidate.includes("srflx")), "Should not get any srflx candidates");
+        ok(!candidates2.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function addStunServerAfterNegotiation() {
+      const offerer = new RTCPeerConnection();
+      const answerer = new RTCPeerConnection({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+      try {
+        const candidatePromise = trickleIce(offerer);
+        await connect(offerer, answerer, 32000, `no ICE servers`);
+        const candidates = await candidatePromise;
+        const ufrags = Array.from(new Set(candidates.map(c => c.usernameFragment)));
+        is(ufrags.length, 1, "Should have one ufrag in candidate set");
+
+        offerer.setConfiguration({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+        const candidates2 = await gatherWithTimeout(offerer, 32000, `just a stun server`);
+        ok(candidates2.some(c => c.candidate.includes("srflx")), "Should get at least one srflx candidate");
+        ok(!candidates2.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+        const ufrags2 = Array.from(new Set(candidates2.map(c => c.usernameFragment)));
+        is(ufrags2.length, 1, "Should have one ufrag in candidate set");
+        isnot(ufrags[0], ufrags2[0], "ufrag should change, because setConfiguration should have triggered an ICE restart");
+      } finally {
+        offerer.close();
+        answerer.close();
+      }
+    },
+
+    async function addTurnServerAfterNegotiation() {
+      const offerer = new RTCPeerConnection();
+      const answerer = new RTCPeerConnection({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+      try {
+        const candidatePromise = trickleIce(offerer);
+        await connect(offerer, answerer, 32000, `no ICE servers`);
+        const candidates = await candidatePromise;
+        const ufrags = Array.from(new Set(candidates.map(c => c.usernameFragment)));
+        is(ufrags.length, 1, "Should have one ufrag in candidate set");
+
+        offerer.setConfiguration({iceServers: [{urls:[`turn:${turnAddressV4}`], username, credential}]});
+        const candidates2 = await gatherWithTimeout(offerer, 32000, `a turn (udp) server`);
+        ok(candidates2.some(c => c.candidate.includes("srflx")), "Should get at least one srflx candidate");
+        ok(candidates2.some(c => c.candidate.includes("relay")), "Should get at least one relay candidate");
+        const ufrags2 = Array.from(new Set(candidates2.map(c => c.usernameFragment)));
+        is(ufrags2.length, 1, "Should have one ufrag in candidate set");
+        isnot(ufrags[0], ufrags2[0], "ufrag should change, because setConfiguration should have triggered an ICE restart");
+      } finally {
+        offerer.close();
+        answerer.close();
+      }
+    },
+
+    async function addTurnTcpServerAfterNegotiation() {
+      const offerer = new RTCPeerConnection();
+      const answerer = new RTCPeerConnection({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+      try {
+        const candidatePromise = trickleIce(offerer);
+        await connect(offerer, answerer, 32000, `no ICE servers`);
+        const candidates = await candidatePromise;
+        const ufrags = Array.from(new Set(candidates.map(c => c.usernameFragment)));
+        is(ufrags.length, 1, "Should have one ufrag in candidate set");
+
+        offerer.setConfiguration({iceServers: [{urls:[`turn:${turnAddressV4}?transport=tcp`], username, credential}]});
+        const candidates2 = await gatherWithTimeout(offerer, 32000, `a turn (tcp) server`);
+        ok(!candidates2.some(c => c.candidate.includes("srflx")), "Should not get any srflx candidates");
+        ok(candidates2.some(c => c.candidate.includes("relay")), "Should get at least one relay candidate");
+        const ufrags2 = Array.from(new Set(candidates2.map(c => c.usernameFragment)));
+        is(ufrags2.length, 1, "Should have one ufrag in candidate set");
+        isnot(ufrags[0], ufrags2[0], "ufrag should change, because setConfiguration should have triggered an ICE restart");
+      } finally {
+        offerer.close();
+        answerer.close();
+      }
+    },
+
+    async function addStunServerBeforeCreateAnswer() {
+      const offerer = new RTCPeerConnection();
+      const answerer = new RTCPeerConnection();
+
+      try {
+        await answerer.setRemoteDescription(await offerer.createOffer({offerToReceiveAudio: true}));
+
+        answerer.setConfiguration({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+        const candidates = await gatherWithTimeout(answerer, 32000, `just a stun server`);
+        ok(candidates.some(c => c.candidate.includes("srflx")), "Should get at least one srflx candidate");
+        ok(!candidates.some(c => c.candidate.includes("relay")), "Should not get any relay candidates");
+      } finally {
+        offerer.close();
+        answerer.close();
+      }
+    },
+
+    async function addTurnServerBeforeCreateAnswer() {
+      const offerer = new RTCPeerConnection();
+      const answerer = new RTCPeerConnection();
+
+      try {
+        await answerer.setRemoteDescription(await offerer.createOffer({offerToReceiveAudio: true}));
+
+        answerer.setConfiguration({iceServers: [{urls:[`turn:${turnAddressV4}`], username, credential}]});
+        const candidates = await gatherWithTimeout(answerer, 32000, `a turn (udp) server`);
+        ok(candidates.some(c => c.candidate.includes("srflx")), "Should get at least one srflx candidate");
+        ok(candidates.some(c => c.candidate.includes("relay")), "Should get at least one relay candidate");
+      } finally {
+        offerer.close();
+        answerer.close();
+      }
+    },
+
+    async function addTurnTcpServerBeforeCreateAnswer() {
+      const offerer = new RTCPeerConnection();
+      const answerer = new RTCPeerConnection();
+
+      try {
+        await answerer.setRemoteDescription(await offerer.createOffer({offerToReceiveAudio: true}));
+
+        answerer.setConfiguration({iceServers: [{urls:[`turn:${turnAddressV4}?transport=tcp`], username, credential}]});
+        const candidates = await gatherWithTimeout(answerer, 32000, `a turn (tcp) server`);
+        ok(!candidates.some(c => c.candidate.includes("srflx")), "Should not get any srflx candidates");
+        ok(candidates.some(c => c.candidate.includes("relay")), "Should get at least one relay candidate");
+      } finally {
+        offerer.close();
+        answerer.close();
+      }
+    },
+
+    async function relayPolicyPreventsSrflx() {
+      const pc = new RTCPeerConnection();
+      try {
+        pc.setConfiguration({iceServers: [{urls:[`turn:${turnAddressV4}`], username, credential}], iceTransportPolicy: "relay"});
+        const candidates = await gatherWithTimeout(pc, 32000, `a turn (udp) server`);
+        ok(!candidates.some(c => c.candidate.includes("srflx")), "Should not get a srflx candidate");
+        ok(candidates.some(c => c.candidate.includes("relay")), "Should get at least one relay candidate");
+      } finally {
+        pc.close();
+      }
+    },
+
+    async function addOffererStunServerAllowsIceToConnect() {
+      const offerer = new RTCPeerConnection();
+      const answerer = new RTCPeerConnection();
+
+      try {
+        try {
+          // Both ends are behind a simulated endpoint-independent NAT, which
+          // requires at least one side to have a srflx candidate to work.
+          await connect(offerer, answerer, 2000, `no ICE servers`);
+          ok(false, "ICE should either have failed, or timed out!");
+        } catch (e) {
+          if (!(e instanceof Error)) throw e;
+          ok(true, "ICE should either have failed, or timed out!");
+        }
+
+        offerer.setConfiguration({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+        await connect(offerer, answerer, 32000, `just a STUN server`);
+      } finally {
+        offerer.close();
+        answerer.close();
+      }
+    },
+
+    async function addAnswererStunServerDoesNotAllowIceToConnect() {
+      const offerer = new RTCPeerConnection();
+      const answerer = new RTCPeerConnection();
+
+      try {
+        try {
+          // Both ends are behind a simulated endpoint-independent NAT, which
+          // requires at least one side to have a srflx candidate to work.
+          await connect(offerer, answerer, 2000, `no ICE servers`);
+          ok(false, "ICE should either have failed, or timed out!");
+        } catch (e) {
+          if (!(e instanceof Error)) throw e;
+          ok(true, "ICE should either have failed, or timed out!");
+        }
+
+        // This _won't_ help, because the answerer does not get to decide to
+        // trigger an ICE restart.
+        answerer.setConfiguration({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+        try {
+          await connectNoTrickleWait(offerer, answerer, 2000, `no ICE servers`);
+          ok(false, "ICE should either have failed, or timed out!");
+        } catch (e) {
+          if (!(e instanceof Error)) throw e;
+          ok(true, "ICE should either have failed, or timed out!");
+        }
+      } finally {
+        offerer.close();
+        answerer.close();
+      }
+    },
+
+    async function addOffererTurnServerAllowsIceToConnect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.filtering_type', 'PORT_DEPENDENT'],
+          ['media.peerconnection.nat_simulator.mapping_type', 'PORT_DEPENDENT']);
+
+      const offerer = new RTCPeerConnection({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+      const answerer = new RTCPeerConnection({iceServers: [{urls:[`stun:${turnAddressV4}`]}]});
+
+      try {
+        try {
+          // Both ends are behind a simulated port-dependent NAT, which
+          // requires at least one side to have a relay candidate to work.
+          await connect(offerer, answerer, 2000, `just a STUN server`);
+          ok(false, "ICE should either have failed, or timed out!");
+        } catch (e) {
+          if (!(e instanceof Error)) throw e;
+          ok(true, "ICE should either have failed, or timed out!");
+        }
+
+        offerer.setConfiguration({iceServers: [{urls:[`turn:${turnAddressV4}`], username, credential}]});
+        await connect(offerer, answerer, 32000, `a TURN (udp) server`);
+      } finally {
+        offerer.close();
+        answerer.close();
+        await SpecialPowers.popPrefEnv();
+      }
+    },
+
+  ];
+
+  runNetworkTest(async () => {
+    const turnServer = iceServersArray.find(server => "username" in server);
+    username = turnServer.username;
+    credential = turnServer.credential;
+    // Just use the first url. It might make sense to look for TURNS first,
+    // since that will always use a hostname, but on CI we don't have TURNS
+    // support anyway (see bug 1323439).
+    const turnHostname = getTurnHostname(turnServer.urls[0]);
+    turnAddressV4 = await dnsLookupV4(turnHostname);
+
+    await pushPrefs(
+        ['media.peerconnection.ice.obfuscate_host_addresses', false],
+        ['media.peerconnection.nat_simulator.filtering_type', 'ENDPOINT_INDEPENDENT'],
+        ['media.peerconnection.nat_simulator.mapping_type', 'ENDPOINT_INDEPENDENT'],
+        ['media.peerconnection.ice.loopback', true],
+        ['media.getusermedia.insecure.enabled', true]);
+
+    for (const test of tests) {
+      info(`Running test: ${test.name}`);
+      try {
+        await test();
+      } catch (e) {
+        ok(false, `Caught ${e.name}: ${e.message} ${e.stack}`);
+      }
+      info(`Done running test: ${test.name}`);
+      // Make sure we don't build up a pile of GC work, and also get PCImpl to
+      // print their timecards.
+      await new Promise(r => SpecialPowers.exactGC(r));
+    }
+
+    await SpecialPowers.popPrefEnv();
+  }, { useIceServer: true });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_gatherWithStun300.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_gatherWithStun300.html
new file mode 100644
index 0000000000..50bc4a6553
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_gatherWithStun300.html
@@ -0,0 +1,269 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="iceTestUtils.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "857668",
+    title: "RTCPeerConnection check STUN gathering with STUN/300 responses"
+  });
+
+  /* This is pretty hairy, so some background:
+   * Spec is here: https://datatracker.ietf.org/doc/html/rfc8489#section-10
+   * STUN/300 responses allow a server to redirect STUN requests to one or
+     more other servers, as ALTERNATE-SERVER attributes.
+   * The server specifies the IP address, IP version, and port for each
+     ALTERNATE-SERVER.
+   * The spec allows multiple rounds of redirects, and requires the client to
+     remember the servers it has already tried to avoid redirect loops.
+   * For TURNS, the TURN server can also supply an ALTERNATE-DOMAIN attribute,
+     which the client MUST use for the TLS handshake on the new target. The
+     client does _not_ use this as an FQDN; it always uses the address in the
+     ALTERNATE-SERVER. ALTERNATE-DOMAIN is meaningless in the non-TLS case.
+   * STUN/300 with ALTERNATE-SERVER is only defined for the TURN Allocate
+     message type (at least in the context of ICE). Clients are supposed to
+     treat STUN/300 as an unrecoverable error in all other cases. The TURN spec
+     does _not_ spell out how a client should handle multiple ALTERNATE-SERVERs.
+     We just take the first one that we have not already tried, and that is the
+     same IP version that we started with. This is because switching the IP
+     version is problematic for ICE.
+   * The test TURN server opens extra ports that will respond with redirects to
+     the _real_ ports, but the address remains the same. This is because we
+     cannot know ahead of time whether the machine we're running on has more
+     than one IP address of each version. This means the test TURN server is not
+     useful for testing cases where the address changes. Also, the test TURN
+     server does not currently know how to respond with multiple
+     ALTERNATE-SERVERs.
+   * To test cases where the _address_ changes, we instead use a feature in the
+     NAT simulator to respond with fake redirects when the destination address
+     matches an address that we configure with a pref. This feature can add
+     multiple ALTERNATE-SERVERs.
+   * The test TURN server's STUN/300 responses have a proper MESSAGE-INTEGRITY,
+     but the NAT simulator's do _not_. For now, we want both cases to work,
+     because some servers respond with STUN/300 without including
+     MESSAGE-INTEGRITY. This is a spec violation, even though the spec
+     contradicts itself in non-normative language elsewhere.
+   * Right now, neither the NAT simulator nor the test TURN server support
+     ALTERNATE-DOMAIN.
+  */
+
+  // These are the ports the test TURN server will respond with redirects on.
+  // The test TURN server tells us what these are in the JSON it spits out when
+  // we start it.
+  let turnRedirectPort;
+  let turnsRedirectPort;
+
+  // These are the addresses that we will configure the NAT simulator to
+  // redirect to. We do DNS lookups of the host in iceServersArray (provided
+  // by the test TURN server), and put the results here. On some platforms this
+  // will be 127.0.0.1 and ::1, but on others we may use a real address.
+  let redirectTargetV4;
+
+  // Test TURN server tells us these in the JSON it spits out when we start it
+  let username;
+  let credential;
+
+  // This is the address we will configure the NAT simulator to respond with
+  // redirects for. We use an address from TEST-NET since it is really unlikely
+  // we'll see that on a real machine, and also because we do not have
+  // special-case code in nICEr for TEST-NET (like we do for link-local, for
+  // example).
+  const redirectAddressV4 = '198.51.100.1';
+
+  const tests = [
+    async function baselineV4Cases() {
+      await checkSrflx([{urls:[`stun:${redirectTargetV4}`]}]);
+      await checkRelayUdp([{urls:[`turn:${redirectTargetV4}`], username, credential}]);
+      await checkRelayTcp([{urls:[`turn:${redirectTargetV4}?transport=tcp`], username, credential}]);
+      await checkRelayUdpTcp([{urls:[`turn:${redirectTargetV4}`, `turn:${redirectTargetV4}?transport=tcp`], username, credential}]);
+    },
+
+    async function stunV4Redirect() {
+      // This test uses the test TURN server, because nICEr drops responses
+      // without MESSAGE-INTEGRITY on the floor _unless_ they are a STUN/300 to
+      // an Allocate request. If we tried to use the NAT simulator for this, we
+      // would have to wait for nICEr to time out, since the NAT simulator does
+      // not know how to do MESSAGE-INTEGRITY.
+      await checkNoSrflx(
+          [{urls:[`stun:${redirectTargetV4}:${turnRedirectPort}`]}]);
+    },
+
+    async function turnV4UdpPortRedirect() {
+      await checkRelayUdp([{urls:[`turn:${redirectTargetV4}:${turnRedirectPort}`], username, credential}]);
+    },
+
+    async function turnV4TcpPortRedirect() {
+      await checkRelayTcp([{urls:[`turn:${redirectTargetV4}:${turnRedirectPort}?transport=tcp`], username, credential}]);
+    },
+
+    async function turnV4UdpTcpPortRedirect() {
+      await checkRelayUdpTcp([{urls:[`turn:${redirectTargetV4}:${turnRedirectPort}`, `turn:${redirectTargetV4}:${turnRedirectPort}?transport=tcp`], username, credential}]);
+    },
+
+    async function turnV4UdpAddressRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectTargetV4}`]);
+      await checkRelayUdp([{urls:[`turn:${redirectAddressV4}`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4TcpAddressRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectTargetV4}`]);
+      await checkRelayTcp([{urls:[`turn:${redirectAddressV4}?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4UdpTcpAddressRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectTargetV4}`]);
+      await checkRelayUdpTcp([{urls:[`turn:${redirectAddressV4}`, `turn:${redirectAddressV4}?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4UdpEmptyRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', '']);
+      await checkNoRelay([{urls:[`turn:${redirectAddressV4}`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4TcpEmptyRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', '']);
+      await checkNoRelay([{urls:[`turn:${redirectAddressV4}?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4UdpTcpEmptyRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', '']);
+      await checkNoRelay([{urls:[`turn:${redirectAddressV4}`, `turn:${redirectAddressV4}?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4UdpAddressAndPortRedirect() {
+      // This should result in two rounds of redirection; the first is by
+      // address, the second is by port.
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectTargetV4}`]);
+      await checkRelayUdp([{urls:[`turn:${redirectAddressV4}:${turnRedirectPort}`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4TcpAddressAndPortRedirect() {
+      // This should result in two rounds of redirection; the first is by
+      // address, the second is by port.
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectTargetV4}`]);
+      await checkRelayTcp([{urls:[`turn:${redirectAddressV4}:${turnRedirectPort}?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4UdpTcpAddressAndPortRedirect() {
+      // This should result in two rounds of redirection; the first is by
+      // address, the second is by port.
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectTargetV4}`]);
+      await checkRelayUdpTcp([{urls:[`turn:${redirectAddressV4}:${turnRedirectPort}`, `turn:${redirectAddressV4}:${turnRedirectPort}?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4UdpRedirectLoop() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectAddressV4}`]);
+      // If we don't detect the loop, gathering will not finish
+      await checkNoRelay([{urls:[`turn:${redirectAddressV4}`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4TcpRedirectLoop() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectAddressV4}`]);
+      // If we don't detect the loop, gathering will not finish
+      await checkNoRelay([{urls:[`turn:${redirectAddressV4}?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4UdpTcpRedirectLoop() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectAddressV4}`]);
+      // If we don't detect the loop, gathering will not finish
+      await checkNoRelay([{urls:[`turn:${redirectAddressV4}`, `turn:${redirectAddressV4}?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4UdpMultipleAddressRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectAddressV4},${redirectTargetV4}`]);
+      await checkRelayUdp([{urls:[`turn:${redirectAddressV4}`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4TcpMultipleAddressRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectAddressV4},${redirectTargetV4}`]);
+      await checkRelayTcp([{urls:[`turn:${redirectAddressV4}?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV4UdpTcpMultipleAddressRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV4}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectAddressV4},${redirectTargetV4}`]);
+      await checkRelayUdpTcp([{urls:[`turn:${redirectAddressV4}`, `turn:${redirectAddressV4}?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+  ];
+
+  runNetworkTest(async () => {
+    const turnServer = iceServersArray.find(server => "username" in server);
+    username = turnServer.username;
+    credential = turnServer.credential;
+    // Special props, non-standard
+    turnRedirectPort = turnServer.turn_redirect_port;
+    turnsRedirectPort = turnServer.turns_redirect_port;
+    // Just use the first url. It might make sense to look for TURNS first,
+    // since that will always use a hostname, but on CI we don't have TURNS
+    // support anyway (see bug 1323439).
+    const turnHostname = getTurnHostname(turnServer.urls[0]);
+    redirectTargetV4 = await dnsLookupV4(turnHostname);
+
+    await pushPrefs(
+        ['media.peerconnection.ice.obfuscate_host_addresses', false],
+        ['media.peerconnection.nat_simulator.filtering_type', 'ENDPOINT_INDEPENDENT'],
+        ['media.peerconnection.nat_simulator.mapping_type', 'ENDPOINT_INDEPENDENT'],
+        ['media.peerconnection.ice.loopback', true],
+        ['media.getusermedia.insecure.enabled', true]);
+
+    for (const test of tests) {
+      info(`Running test: ${test.name}`);
+      await test();
+      info(`Done running test: ${test.name}`);
+    }
+
+    await SpecialPowers.popPrefEnv();
+  }, { useIceServer: true });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_gatherWithStun300IPv6.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_gatherWithStun300IPv6.html
new file mode 100644
index 0000000000..16f8f39978
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_gatherWithStun300IPv6.html
@@ -0,0 +1,283 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="iceTestUtils.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "857668",
+    title: "RTCPeerConnection check STUN gathering with STUN/300 responses (IPv6)"
+  });
+
+  /* This is pretty hairy, so some background:
+   * Spec is here: https://datatracker.ietf.org/doc/html/rfc8489#section-10
+   * STUN/300 responses allow a server to redirect STUN requests to one or
+     more other servers, as ALTERNATE-SERVER attributes.
+   * The server specifies the IP address, IP version, and port for each
+     ALTERNATE-SERVER.
+   * The spec allows multiple rounds of redirects, and requires the client to
+     remember the servers it has already tried to avoid redirect loops.
+   * For TURNS, the TURN server can also supply an ALTERNATE-DOMAIN attribute,
+     which the client MUST use for the TLS handshake on the new target. The
+     client does _not_ use this as an FQDN; it always uses the address in the
+     ALTERNATE-SERVER. ALTERNATE-DOMAIN is meaningless in the non-TLS case.
+   * STUN/300 with ALTERNATE-SERVER is only defined for the TURN Allocate
+     message type (at least in the context of ICE). Clients are supposed to
+     treat STUN/300 as an unrecoverable error in all other cases. The TURN spec
+     does _not_ spell out how a client should handle multiple ALTERNATE-SERVERs.
+     We just take the first one that we have not already tried, and that is the
+     same IP version that we started with. This is because switching the IP
+     version is problematic for ICE.
+   * The test TURN server opens extra ports that will respond with redirects to
+     the _real_ ports, but the address remains the same. This is because we
+     cannot know ahead of time whether the machine we're running on has more
+     than one IP address of each version. This means the test TURN server is not
+     useful for testing cases where the address changes. Also, the test TURN
+     server does not currently know how to respond with multiple
+     ALTERNATE-SERVERs.
+   * To test cases where the _address_ changes, we instead use a feature in the
+     NAT simulator to respond with fake redirects when the destination address
+     matches an address that we configure with a pref. This feature can add
+     multiple ALTERNATE-SERVERs.
+   * The test TURN server's STUN/300 responses have a proper MESSAGE-INTEGRITY,
+     but the NAT simulator's do _not_. For now, we want both cases to work,
+     because some servers respond with STUN/300 without including
+     MESSAGE-INTEGRITY. This is a spec violation, even though the spec
+     contradicts itself in non-normative language elsewhere.
+   * Right now, neither the NAT simulator nor the test TURN server support
+     ALTERNATE-DOMAIN.
+  */
+
+  // These are the ports the test TURN server will respond with redirects on.
+  // The test TURN server tells us what these are in the JSON it spits out when
+  // we start it.
+  let turnRedirectPort;
+  let turnsRedirectPort;
+
+  // These are the addresses that we will configure the NAT simulator to
+  // redirect to. We do DNS lookups of the host in iceServersArray (provided
+  // by the test TURN server), and put the results here. On some platforms this
+  // will be 127.0.0.1 and ::1, but on others we may use a real address.
+  let redirectTargetV6;
+
+  // Test TURN server tells us these in the JSON it spits out when we start it
+  let username;
+  let credential;
+
+  // This is the address we will configure the NAT simulator to respond with
+  // redirects for. We use an address from TEST-NET since it is really unlikely
+  // we'll see that on a real machine, and also because we do not have
+  // special-case code in nICEr for TEST-NET (like we do for link-local, for
+  // example).
+  const redirectAddressV6 = '::ffff:198.51.100.1';
+
+  const tests = [
+    async function baselineV6Cases() {
+      await checkSrflx([{urls:[`stun:[${redirectTargetV6}]`]}]);
+      await checkRelayUdp([{urls:[`turn:[${redirectTargetV6}]`], username, credential}]);
+      await checkRelayTcp([{urls:[`turn:[${redirectTargetV6}]?transport=tcp`], username, credential}]);
+      await checkRelayUdpTcp([{urls:[`turn:[${redirectTargetV6}]`, `turn:[${redirectTargetV6}]?transport=tcp`], username, credential}]);
+    },
+
+    async function stunV6Redirect() {
+      // This test uses the test TURN server, because nICEr drops responses
+      // without MESSAGE-INTEGRITY on the floor _unless_ they are a STUN/300 to
+      // an Allocate request. If we tried to use the NAT simulator for this, we
+      // would have to wait for nICEr to time out, since the NAT simulator does
+      // not know how to do MESSAGE-INTEGRITY.
+      await checkNoSrflx(
+          [{urls:[`stun:[${redirectTargetV6}]:${turnRedirectPort}`]}]);
+    },
+
+    async function turnV6UdpPortRedirect() {
+      await checkRelayUdp([{urls:[`turn:[${redirectTargetV6}]:${turnRedirectPort}`], username, credential}]);
+    },
+
+    async function turnV6TcpPortRedirect() {
+      await checkRelayTcp([{urls:[`turn:[${redirectTargetV6}]:${turnRedirectPort}?transport=tcp`], username, credential}]);
+    },
+
+    async function turnV6UdpTcpPortRedirect() {
+      await checkRelayUdpTcp([{urls:[`turn:[${redirectTargetV6}]:${turnRedirectPort}`, `turn:[${redirectTargetV6}]:${turnRedirectPort}?transport=tcp`], username, credential}]);
+    },
+
+    async function turnV6UdpAddressRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectTargetV6}`]);
+      await checkRelayUdp([{urls:[`turn:[${redirectAddressV6}]`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6TcpAddressRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectTargetV6}`]);
+      await checkRelayTcp([{urls:[`turn:[${redirectAddressV6}]?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6UdpTcpAddressRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectTargetV6}`]);
+      await checkRelayUdpTcp([{urls:[`turn:[${redirectAddressV6}]`, `turn:[${redirectAddressV6}]?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6UdpEmptyRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', '']);
+      await checkNoRelay([{urls:[`turn:[${redirectAddressV6}]`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6TcpEmptyRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', '']);
+      await checkNoRelay([{urls:[`turn:[${redirectAddressV6}]?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6UdpTcpEmptyRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', '']);
+      await checkNoRelay([{urls:[`turn:[${redirectAddressV6}]`, `turn:[${redirectAddressV6}]?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6UdpAddressAndPortRedirect() {
+      // This should result in two rounds of redirection; the first is by
+      // address, the second is by port.
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectTargetV6}`]);
+      await checkRelayUdp([{urls:[`turn:[${redirectAddressV6}]:${turnRedirectPort}`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6TcpAddressAndPortRedirect() {
+      // This should result in two rounds of redirection; the first is by
+      // address, the second is by port.
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectTargetV6}`]);
+      await checkRelayTcp([{urls:[`turn:[${redirectAddressV6}]:${turnRedirectPort}?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6UdpTcpAddressAndPortRedirect() {
+      // This should result in two rounds of redirection; the first is by
+      // address, the second is by port.
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectTargetV6}`]);
+      await checkRelayUdpTcp([{urls:[`turn:[${redirectAddressV6}]:${turnRedirectPort}`, `turn:[${redirectAddressV6}]:${turnRedirectPort}?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6UdpRedirectLoop() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectAddressV6}`]);
+      // If we don't detect the loop, gathering will not finish
+      await checkNoRelay([{urls:[`turn:[${redirectAddressV6}]`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6TcpRedirectLoop() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectAddressV6}`]);
+      // If we don't detect the loop, gathering will not finish
+      await checkNoRelay([{urls:[`turn:[${redirectAddressV6}]?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6UdpTcpRedirectLoop() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectAddressV6}`]);
+      // If we don't detect the loop, gathering will not finish
+      await checkNoRelay([{urls:[`turn:[${redirectAddressV6}]`, `turn:[${redirectAddressV6}]?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6UdpMultipleAddressRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectAddressV6},${redirectTargetV6}`]);
+      await checkRelayUdp([{urls:[`turn:[${redirectAddressV6}]`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6TcpMultipleAddressRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectAddressV6},${redirectTargetV6}`]);
+      await checkRelayTcp([{urls:[`turn:[${redirectAddressV6}]?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+
+    async function turnV6UdpTcpMultipleAddressRedirect() {
+      await pushPrefs(
+          ['media.peerconnection.nat_simulator.redirect_address', `${redirectAddressV6}`],
+          ['media.peerconnection.nat_simulator.redirect_targets', `${redirectAddressV6},${redirectTargetV6}`]);
+      await checkRelayUdpTcp([{urls:[`turn:[${redirectAddressV6}]`, `turn:[${redirectAddressV6}]?transport=tcp`], username, credential}]);
+      await SpecialPowers.popPrefEnv();
+    },
+  ];
+
+  runNetworkTest(async () => {
+    const turnServer = iceServersArray.find(server => "username" in server);
+    username = turnServer.username;
+    credential = turnServer.credential;
+    // Special props, non-standard
+    turnRedirectPort = turnServer.turn_redirect_port;
+    turnsRedirectPort = turnServer.turns_redirect_port;
+    // Just use the first url. It might make sense to look for TURNS first,
+    // since that will always use a hostname, but on CI we don't have TURNS
+    // support anyway (see bug 1323439).
+    const turnHostname = getTurnHostname(turnServer.urls[0]);
+
+    await pushPrefs(
+        ['media.peerconnection.ice.obfuscate_host_addresses', false],
+        ['media.peerconnection.nat_simulator.filtering_type', 'ENDPOINT_INDEPENDENT'],
+        ['media.peerconnection.nat_simulator.mapping_type', 'ENDPOINT_INDEPENDENT'],
+        ['media.peerconnection.ice.loopback', true],
+        ['media.getusermedia.insecure.enabled', true]);
+
+    if (await ipv6Supported()) {
+      redirectTargetV6 = await dnsLookupV6(turnHostname);
+      if (redirectTargetV6 == '' && turnHostname == 'localhost') {
+        // Our testers don't seem to have IPv6 DNS resolution for localhost
+        // set up...
+        redirectTargetV6 = '::1';
+      }
+
+      if (redirectTargetV6 != '') {
+        for (const test of tests) {
+          info(`Running test: ${test.name}`);
+          await test();
+          info(`Done running test: ${test.name}`);
+        }
+      } else {
+        ok(false, `This machine has an IPv6 address, but ${turnHostname} does not resolve to an IPv6 address`);
+      }
+    } else {
+      ok(false, 'This machine appears to not have an IPv6 address');
+    }
+
+    await SpecialPowers.popPrefEnv();
+  }, { useIceServer: true });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_glean.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_glean.html
new file mode 100644
index 0000000000..d5d6026ee5
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_glean.html
@@ -0,0 +1,488 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1401592",
+  title: "Test that glean is recording stats as expected",
+  visible: true
+});
+
+const { AppConstants } = SpecialPowers.ChromeUtils.import(
+  "resource://gre/modules/AppConstants.jsm"
+);
+
+async function getAllWarningRates() {
+  return {
+    warnNoGetparameters:
+    await GleanTest.rtcrtpsenderSetparameters.warnNoGetparameters.testGetValue(),
+    warnLengthChanged:
+    await GleanTest.rtcrtpsenderSetparameters.warnLengthChanged.testGetValue(),
+    warnRidChanged:
+    await GleanTest.rtcrtpsenderSetparameters.warnRidChanged.testGetValue(),
+    warnNoTransactionid:
+    await GleanTest.rtcrtpsenderSetparameters.warnNoTransactionid.testGetValue(),
+    warnStaleTransactionid:
+    await GleanTest.rtcrtpsenderSetparameters.warnStaleTransactionid.testGetValue(),
+  };
+}
+
+const tests = [
+  async function checkRTCRtpSenderCount() {
+    const pc = new RTCPeerConnection();
+    const oldCount = await GleanTest.rtcrtpsender.count.testGetValue() ?? 0;
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+    const countDiff = await GleanTest.rtcrtpsender.count.testGetValue() - oldCount;
+    is(countDiff, 1, "Glean should have recorded the creation of a single RTCRtpSender");
+  },
+
+  async function checkRTCRtpSenderSetParametersCompatCount() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc = new RTCPeerConnection();
+    const oldCount = await GleanTest.rtcrtpsender.countSetparametersCompat.testGetValue() ?? 0;
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+    const countDiff = await GleanTest.rtcrtpsender.countSetparametersCompat.testGetValue() - oldCount;
+    is(countDiff, 1, "Glean should have recorded the creation of a single RTCRtpSender that uses the setParameters compat mode");
+  },
+
+  async function checkSendEncodings() {
+    const pc = new RTCPeerConnection();
+    const oldRate = await GleanTest.rtcrtpsender.usedSendencodings.testGetValue();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+    const newRate = await GleanTest.rtcrtpsender.usedSendencodings.testGetValue();
+    is(newRate.denominator, oldRate.denominator + 1, "Glean should have recorded the creation of a single RTCRtpSender");
+    is(newRate.numerator, oldRate.numerator + 1, "Glean should have recorded the use of sendEncodings");
+  },
+
+  async function checkAddTransceiverNoSendEncodings() {
+    const pc = new RTCPeerConnection();
+    const oldRate = await GleanTest.rtcrtpsender.usedSendencodings.testGetValue();
+    const {sender} = pc.addTransceiver('video');
+    const newRate = await GleanTest.rtcrtpsender.usedSendencodings.testGetValue();
+    is(newRate.denominator, oldRate.denominator + 1, "Glean should have recorded the creation of a single RTCRtpSender");
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded a use of sendEncodings");
+  },
+
+  async function checkAddTrack() {
+    const pc = new RTCPeerConnection();
+    const oldRate = await GleanTest.rtcrtpsender.usedSendencodings.testGetValue();
+    const stream = await navigator.mediaDevices.getUserMedia({video: true});
+    const sender = pc.addTrack(stream.getTracks()[0]);
+    const newRate = await GleanTest.rtcrtpsender.usedSendencodings.testGetValue();
+    is(newRate.denominator, oldRate.denominator + 1, "Glean should have recorded the creation of a single RTCRtpSender");
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded a use of sendEncodings");
+  },
+
+  async function checkGoodSetParametersCompatMode() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+    const oldWarningRates = await getAllWarningRates();
+    await sender.setParameters(sender.getParameters());
+    const newWarningRates = await getAllWarningRates();
+    isDeeply(oldWarningRates, newWarningRates);
+  },
+
+  async function checkBadSetParametersNoGetParametersWarning() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+
+    let oldRate = await GleanTest.rtcrtpsenderSetparameters.warnNoGetparameters.testGetValue();
+    let oldBlameCount = await GleanTest.rtcrtpsenderSetparameters.blameNoGetparameters["example.com"].testGetValue() || 0;
+
+    await sender.setParameters({encodings: [{rid: "0"},{rid: "1"},{rid: "2"}]});
+
+    let newRate = await GleanTest.rtcrtpsenderSetparameters.warnNoGetparameters.testGetValue();
+    let newBlameCount = await GleanTest.rtcrtpsenderSetparameters.blameNoGetparameters["example.com"].testGetValue() || 0;
+
+    is(newRate.numerator, oldRate.numerator + 1, "Glean should have recorded a warning in setParameters due to lack of a getParameters call");
+
+    if (AppConstants.EARLY_BETA_OR_EARLIER) {
+      is(newBlameCount, oldBlameCount + 1, "Glean should have recorded that example.com encountered this warning");
+    } else {
+      is(newBlameCount, oldBlameCount, "Glean should not have recorded that example.com encountered this warning, because we're running this test on a stable channel");
+    }
+
+    oldRate = newRate;
+    oldBlameCount = newBlameCount;
+
+    // Glean should only record the warning once per sender!
+    await sender.setParameters({encodings: [{rid: "0"},{rid: "1"},{rid: "2"}]});
+
+    newRate = await GleanTest.rtcrtpsenderSetparameters.warnNoGetparameters.testGetValue();
+    newBlameCount = await GleanTest.rtcrtpsenderSetparameters.blameNoGetparameters["example.com"].testGetValue() || 0;
+
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded another warning in setParameters due to lack of a getParameters call");
+    is(newBlameCount, oldBlameCount, "Glean should not have recorded that example.com encountered this warning a second time, since this is the same sender");
+  },
+
+  async function checkBadSetParametersLengthChangedWarning() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+
+    let oldRate = await GleanTest.rtcrtpsenderSetparameters.warnLengthChanged.testGetValue();
+    let oldBlameCount = await GleanTest.rtcrtpsenderSetparameters.blameLengthChanged["example.com"].testGetValue() || 0;
+
+    let params = sender.getParameters();
+    params.encodings.pop();
+    await sender.setParameters(params);
+
+    let newRate = await GleanTest.rtcrtpsenderSetparameters.warnLengthChanged.testGetValue();
+    let newBlameCount = await GleanTest.rtcrtpsenderSetparameters.blameLengthChanged["example.com"].testGetValue() || 0;
+
+    is(newRate.numerator, oldRate.numerator + 1, "Glean should have recorded a warning due to a length change in encodings");
+
+    if (AppConstants.EARLY_BETA_OR_EARLIER) {
+      is(newBlameCount, oldBlameCount + 1, "Glean should have recorded that example.com encountered this warning");
+    } else {
+      is(newBlameCount, oldBlameCount, "Glean should not have recorded that example.com encountered this warning, because we're running this test on a stable channel");
+    }
+
+    oldRate = newRate;
+    oldBlameCount = newBlameCount;
+
+    // Glean should only record the warning once per sender!
+    params = sender.getParameters();
+    params.encodings.pop();
+    await sender.setParameters(params);
+
+    newRate = await GleanTest.rtcrtpsenderSetparameters.warnLengthChanged.testGetValue();
+    newBlameCount = await GleanTest.rtcrtpsenderSetparameters.blameLengthChanged["example.com"].testGetValue() || 0;
+
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded another warning due to a length change in encodings");
+    is(newBlameCount, oldBlameCount, "Glean should not have recorded that example.com encountered this warning a second time, since this is the same sender");
+  },
+
+  async function checkBadSetParametersRidChangedWarning() {
+    // This pref does not let rid change errors slide anymore
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+
+    let oldRate = await GleanTest.rtcrtpsenderSetparameters.failRidChanged.testGetValue();
+    let oldWarnRate = await GleanTest.rtcrtpsenderSetparameters.warnRidChanged.testGetValue();
+
+    let params = sender.getParameters();
+    params.encodings[1].rid = "foo";
+    try {
+      await sender.setParameters(params);
+    } catch (e) {
+    }
+    let newRate = await GleanTest.rtcrtpsenderSetparameters.failRidChanged.testGetValue();
+    let newWarnRate = await GleanTest.rtcrtpsenderSetparameters.warnRidChanged.testGetValue();
+    is(newRate.numerator, oldRate.numerator + 1, "Glean should have recorded an error due to a rid change in encodings");
+    is(newWarnRate.numerator, oldWarnRate.numerator, "Glean should not have recorded a warning due to a rid change in encodings");
+
+    // Glean should only record the error once per sender!
+    params = sender.getParameters();
+    params.encodings[1].rid = "bar";
+    oldRate = newRate;
+    try {
+      await sender.setParameters(params);
+    } catch (e) {
+    }
+    newRate = await GleanTest.rtcrtpsenderSetparameters.failRidChanged.testGetValue();
+    newWarnRate = await GleanTest.rtcrtpsenderSetparameters.warnRidChanged.testGetValue();
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded another error due to a rid change in encodings");
+    is(newWarnRate.numerator, oldWarnRate.numerator, "Glean should not have recorded a warning due to a rid change in encodings");
+   },
+
+  async function checkBadSetParametersNoTransactionIdWarning() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+
+    let oldRate = await GleanTest.rtcrtpsenderSetparameters.warnNoTransactionid.testGetValue();
+    let oldBlameCount = await GleanTest.rtcrtpsenderSetparameters.blameNoTransactionid["example.com"].testGetValue() || 0;
+
+    await sender.setParameters({encodings: [{rid: "0"},{rid: "1"},{rid: "2"}]});
+
+    let newRate = await GleanTest.rtcrtpsenderSetparameters.warnNoTransactionid.testGetValue();
+    let newBlameCount = await GleanTest.rtcrtpsenderSetparameters.blameNoTransactionid["example.com"].testGetValue() || 0;
+
+    is(newRate.numerator, oldRate.numerator + 1, "Glean should have recorded a warning due to missing transactionId in setParameters");
+
+    if (AppConstants.EARLY_BETA_OR_EARLIER) {
+      is(newBlameCount, oldBlameCount + 1, "Glean should have recorded that example.com encountered this warning");
+    } else {
+      is(newBlameCount, oldBlameCount, "Glean should not have recorded that example.com encountered this warning, because we're running this test on a stable channel");
+    }
+
+    oldRate = newRate;
+    oldBlameCount = newBlameCount;
+
+    // Glean should only record the warning once per sender!
+    await sender.setParameters({encodings: [{rid: "0"},{rid: "1"},{rid: "2"}]});
+
+    newRate = await GleanTest.rtcrtpsenderSetparameters.warnNoTransactionid.testGetValue();
+    newBlameCount = await GleanTest.rtcrtpsenderSetparameters.blameNoTransactionid["example.com"].testGetValue() || 0;
+
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded another warning due to missing transactionId in setParameters");
+    is(newBlameCount, oldBlameCount, "Glean should not have recorded that example.com encountered this warning a second time, since this is the same sender");
+  },
+
+  async function checkBadSetParametersStaleTransactionIdWarning() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+
+    let oldRate = await GleanTest.rtcrtpsenderSetparameters.warnStaleTransactionid.testGetValue();
+    let oldBlameCount = await GleanTest.rtcrtpsenderSetparameters.blameStaleTransactionid["example.com"].testGetValue() || 0;
+
+    let params = sender.getParameters();
+    // Cause transactionId to be stale
+    await pc.createOffer();
+    // ...but make sure there is a recent getParameters call
+    sender.getParameters();
+    await sender.setParameters(params);
+
+    let newRate = await GleanTest.rtcrtpsenderSetparameters.warnStaleTransactionid.testGetValue();
+    let newBlameCount = await GleanTest.rtcrtpsenderSetparameters.blameStaleTransactionid["example.com"].testGetValue() || 0;
+
+    is(newRate.numerator, oldRate.numerator + 1, "Glean should have recorded a warning due to stale transactionId in setParameters");
+
+    if (AppConstants.EARLY_BETA_OR_EARLIER) {
+      is(newBlameCount, oldBlameCount + 1, "Glean should have recorded that example.com encountered this warning");
+    } else {
+      is(newBlameCount, oldBlameCount, "Glean should not have recorded that example.com encountered this warning, because we're running this test on a stable channel");
+    }
+
+    oldRate = newRate;
+    oldBlameCount = newBlameCount;
+
+    // Glean should only record the warning once per sender!
+    params = sender.getParameters();
+    // Cause transactionId to be stale
+    await pc.createOffer();
+    // ...but make sure there is a recent getParameters call
+    sender.getParameters();
+    await sender.setParameters(params);
+
+    newRate = await GleanTest.rtcrtpsenderSetparameters.warnStaleTransactionid.testGetValue();
+    newBlameCount = await GleanTest.rtcrtpsenderSetparameters.blameStaleTransactionid["example.com"].testGetValue() || 0;
+
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded another warning due to stale transactionId in setParameters");
+    is(newBlameCount, oldBlameCount, "Glean should not have recorded that example.com encountered this warning a second time, since this is the same sender");
+  },
+
+  async function checkBadSetParametersLengthChangedError() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", false]);
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+    let oldRate = await GleanTest.rtcrtpsenderSetparameters.failLengthChanged.testGetValue();
+    let params = sender.getParameters();
+    params.encodings.pop();
+    try {
+      await sender.setParameters(params);
+    } catch(e) {
+    }
+    let newRate = await GleanTest.rtcrtpsenderSetparameters.failLengthChanged.testGetValue();
+    is(newRate.numerator, oldRate.numerator + 1, "Glean should have recorded an error due to a length change in encodings");
+
+    // Glean should only record the error once per sender!
+    params = sender.getParameters();
+    params.encodings.pop();
+    oldRate = newRate;
+    try {
+      await sender.setParameters(params);
+    } catch (e) {
+    }
+    newRate = await GleanTest.rtcrtpsenderSetparameters.failLengthChanged.testGetValue();
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded another error due to a length change in encodings");
+  },
+
+  async function checkBadSetParametersRidChangedError() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", false]);
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+    let oldRate = await GleanTest.rtcrtpsenderSetparameters.failRidChanged.testGetValue();
+    let params = sender.getParameters();
+    params.encodings[1].rid = "foo";
+    try {
+      await sender.setParameters(params);
+    } catch (e) {
+    }
+    let newRate = await GleanTest.rtcrtpsenderSetparameters.failRidChanged.testGetValue();
+    is(newRate.numerator, oldRate.numerator + 1, "Glean should have recorded an error due to a rid change in encodings");
+
+    // Glean should only record the error once per sender!
+    params = sender.getParameters();
+    params.encodings[1].rid = "bar";
+    oldRate = newRate;
+    try {
+      await sender.setParameters(params);
+    } catch (e) {
+    }
+    newRate = await GleanTest.rtcrtpsenderSetparameters.failRidChanged.testGetValue();
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded another error due to a rid change in encodings");
+  },
+
+  async function checkBadSetParametersNoGetParametersError() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", false]);
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+    let oldRate = await GleanTest.rtcrtpsenderSetparameters.failNoGetparameters.testGetValue();
+    try {
+      await sender.setParameters({encodings: [{rid: "0"},{rid: "1"},{rid: "2"}]});
+    } catch (e) {
+    }
+    let newRate = await GleanTest.rtcrtpsenderSetparameters.failNoGetparameters.testGetValue();
+    is(newRate.numerator, oldRate.numerator + 1, "Glean should have recorded an error in setParameters due to lack of a getParameters call");
+
+    // Glean should only record the error once per sender!
+    oldRate = newRate;
+    try {
+      await sender.setParameters({encodings: [{rid: "0"},{rid: "1"},{rid: "2"}]});
+    } catch (e) {
+    }
+    newRate = await GleanTest.rtcrtpsenderSetparameters.failNoGetparameters.testGetValue();
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded another error in setParameters due to lack of a getParameters call");
+  },
+
+  async function checkBadSetParametersStaleTransactionIdError() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", false]);
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+    let oldRate = await GleanTest.rtcrtpsenderSetparameters.failStaleTransactionid.testGetValue();
+    let params = sender.getParameters();
+    // Cause transactionId to be stale
+    await pc.createOffer();
+    // ...but make sure there is a recent getParameters call
+    sender.getParameters();
+    try {
+      await sender.setParameters(params);
+    } catch (e) {
+    }
+    let newRate = await GleanTest.rtcrtpsenderSetparameters.failStaleTransactionid.testGetValue();
+    is(newRate.numerator, oldRate.numerator + 1, "Glean should have recorded an error due to stale transactionId in setParameters");
+
+    // Glean should only record the error once per sender!
+    oldRate = newRate;
+    params = sender.getParameters();
+    // Cause transactionId to be stale
+    await pc.createOffer();
+    // ...but make sure there is a recent getParameters call
+    sender.getParameters();
+    try {
+      await sender.setParameters(params);
+    } catch (e) {
+    }
+    newRate = await GleanTest.rtcrtpsenderSetparameters.failStaleTransactionid.testGetValue();
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded another error due to stale transactionId in setParameters");
+  },
+
+  async function checkBadSetParametersNoEncodingsError() {
+    // If we do not allow the old setParameters, this will fail the length check
+    // instead.
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+    let oldRate = await GleanTest.rtcrtpsenderSetparameters.failNoEncodings.testGetValue();
+    let params = sender.getParameters();
+    params.encodings = [];
+    try {
+      await sender.setParameters(params);
+    } catch (e) {
+    }
+    let newRate = await GleanTest.rtcrtpsenderSetparameters.failNoEncodings.testGetValue();
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded an error due to empty encodings in setParameters");
+
+    // Glean should only record the error once per sender!
+    oldRate = newRate;
+    params = sender.getParameters();
+    params.encodings = [];
+    try {
+      await sender.setParameters(params);
+    } catch (e) {
+    }
+    newRate = await GleanTest.rtcrtpsenderSetparameters.failNoEncodings.testGetValue();
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded an error due empty encodings in setParameters");
+  },
+
+  async function checkBadSetParametersOtherError() {
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+    let oldRate = await GleanTest.rtcrtpsenderSetparameters.failOther.testGetValue();
+    let params = sender.getParameters();
+    params.encodings[0].scaleResolutionDownBy = 0.5;
+    try {
+      await sender.setParameters(params);
+    } catch (e) {
+    }
+    let newRate = await GleanTest.rtcrtpsenderSetparameters.failOther.testGetValue();
+    is(newRate.numerator, oldRate.numerator + 1, "Glean should have recorded an error due to some other failure");
+
+    // Glean should only record the error once per sender!
+    oldRate = newRate;
+    params = sender.getParameters();
+    params.encodings[0].scaleResolutionDownBy = 0.5;
+    try {
+      await sender.setParameters(params);
+    } catch (e) {
+    }
+    newRate = await GleanTest.rtcrtpsenderSetparameters.failOther.testGetValue();
+    is(newRate.numerator, oldRate.numerator, "Glean should not have recorded another error due to some other failure");
+  },
+
+];
+
+runNetworkTest(async () => {
+  for (const test of tests) {
+    info(`Running test: ${test.name}`);
+    await test();
+    info(`Done running test: ${test.name}`);
+  }
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_iceFailure.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_iceFailure.html
new file mode 100644
index 0000000000..1b82473997
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_iceFailure.html
@@ -0,0 +1,84 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1087629",
+    title: "Wait for ICE failure"
+  });
+
+// Test iceFailure
+
+function PC_LOCAL_SETUP_NULL_ICE_HANDLER(test) {
+  test.pcLocal.setupIceCandidateHandler(test, function() {}, function () {});
+}
+function PC_REMOTE_SETUP_NULL_ICE_HANDLER(test) {
+  test.pcRemote.setupIceCandidateHandler(test, function() {}, function () {});
+}
+function PC_REMOTE_ADD_FAKE_ICE_CANDIDATE(test) {
+  var cand = {"candidate":"candidate:0 1 UDP 2130379007 192.0.2.1 12345 typ host","sdpMid":"","sdpMLineIndex":0};
+  test.pcRemote.storeOrAddIceCandidate(cand);
+  info(test.pcRemote + " Stored fake candidate: " + JSON.stringify(cand));
+}
+function PC_LOCAL_ADD_FAKE_ICE_CANDIDATE(test) {
+  var cand = {"candidate":"candidate:0 1 UDP 2130379007 192.0.2.2 56789 typ host","sdpMid":"","sdpMLineIndex":0};
+    test.pcLocal.storeOrAddIceCandidate(cand);
+  info(test.pcLocal + " Stored fake candidate: " + JSON.stringify(cand));
+}
+function PC_LOCAL_WAIT_FOR_ICE_FAILURE(test) {
+  return test.pcLocal.iceFailed.then(() => {
+    ok(true, this.pcLocal + " Ice Failure Reached.");
+	});
+}
+function PC_REMOTE_WAIT_FOR_ICE_FAILURE(test) {
+  return test.pcRemote.iceFailed.then(() => {
+    ok(true, this.pcRemote + " Ice Failure Reached.");
+    });
+}
+function PC_LOCAL_WAIT_FOR_ICE_FAILED(test) {
+  var resolveIceFailed;
+  test.pcLocal.iceFailed = new Promise(r => resolveIceFailed = r);
+  test.pcLocal.ice_connection_callbacks.checkIceStatus = () => {
+	if (test.pcLocal._pc.iceConnectionState === "failed") {
+	  resolveIceFailed();
+	}
+  }
+}
+function PC_REMOTE_WAIT_FOR_ICE_FAILED(test) {
+  var resolveIceFailed;
+  test.pcRemote.iceFailed = new Promise(r => resolveIceFailed = r);
+  test.pcRemote.ice_connection_callbacks.checkIceStatus = () => {
+	if (test.pcRemote._pc.iceConnectionState === "failed") {
+	  resolveIceFailed();
+	}
+  }
+}
+
+runNetworkTest(async () => {
+  await pushPrefs(
+    ['media.peerconnection.ice.stun_client_maximum_transmits', 3],
+    ['media.peerconnection.ice.trickle_grace_period', 3000],
+  );
+  var test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.chain.replace("PC_LOCAL_SETUP_ICE_HANDLER", PC_LOCAL_SETUP_NULL_ICE_HANDLER);
+  test.chain.replace("PC_REMOTE_SETUP_ICE_HANDLER", PC_REMOTE_SETUP_NULL_ICE_HANDLER);
+  test.chain.insertAfter("PC_REMOTE_SETUP_NULL_ICE_HANDLER", PC_LOCAL_WAIT_FOR_ICE_FAILED);
+  test.chain.insertAfter("PC_LOCAL_WAIT_FOR_ICE_FAILED", PC_REMOTE_WAIT_FOR_ICE_FAILED);
+  test.chain.removeAfter("PC_LOCAL_SET_REMOTE_DESCRIPTION");
+  test.chain.append([
+    PC_REMOTE_ADD_FAKE_ICE_CANDIDATE,
+    PC_LOCAL_ADD_FAKE_ICE_CANDIDATE,
+    PC_LOCAL_WAIT_FOR_ICE_FAILURE,
+    PC_REMOTE_WAIT_FOR_ICE_FAILURE
+  ]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_insertDTMF.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_insertDTMF.html
new file mode 100644
index 0000000000..ca8b866a6d
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_insertDTMF.html
@@ -0,0 +1,76 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1291715",
+  title: "Test insertDTMF on sender",
+  visible: true
+});
+
+function insertdtmftest(pc) {
+  ok(pc.getSenders().length, "have senders");
+  var sender = pc.getSenders()[0];
+  ok(sender.dtmf, "sender has dtmf object");
+
+  ok(sender.dtmf.toneBuffer === "", "sender should start with empty tonebuffer");
+
+  // These will trigger assertions on debug builds if we do not enforce the
+  // specified minimums and maximums for duration and interToneGap.
+  sender.dtmf.insertDTMF("A", 10);
+  sender.dtmf.insertDTMF("A", 10000);
+  sender.dtmf.insertDTMF("A", 70, 10);
+
+  var threw = false;
+  try {
+    sender.dtmf.insertDTMF("bad tones");
+  } catch (ex) {
+    threw = true;
+    is(ex.code, DOMException.INVALID_CHARACTER_ERR, "Expected InvalidCharacterError");
+  }
+  ok(threw, "Expected exception");
+
+  sender.dtmf.insertDTMF("A");
+  sender.dtmf.insertDTMF("B");
+  ok(!sender.dtmf.toneBuffer.includes("A"), "calling insertDTMF should replace current characters");
+
+  sender.dtmf.insertDTMF("aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa");
+  ok(sender.dtmf.toneBuffer.includes("A"), "lowercase characters should be normalized");
+
+  pc.removeTrack(sender);
+  threw = false;
+  try {
+    sender.dtmf.insertDTMF("AAA");
+  } catch (ex) {
+    threw = true;
+    is(ex.code, DOMException.INVALID_STATE_ERR, "Expected InvalidStateError");
+  }
+  ok(threw, "Expected exception");
+}
+
+runNetworkTest(() => {
+  test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.chain.removeAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW");
+
+  // Test sender dtmf.
+  test.chain.append([
+    function PC_LOCAL_INSERT_DTMF(test) {
+      // We want to call removeTrack
+      test.pcLocal.expectNegotiationNeeded();
+      return insertdtmftest(test.pcLocal._pc);
+    }
+  ]);
+
+  return pushPrefs(['media.peerconnection.dtmf.enabled', true])
+    .then(() => test.run());
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_localReofferRollback.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_localReofferRollback.html
new file mode 100644
index 0000000000..16406ece6e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_localReofferRollback.html
@@ -0,0 +1,44 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "952145",
+    title: "Rollback local reoffer"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    addRenegotiation(test.chain, [
+        function PC_LOCAL_ADD_SECOND_STREAM(test) {
+          test.setMediaConstraints([{audio: true}, {audio: true}],
+                                   [{audio: true}]);
+          return test.pcLocal.getAllUserMediaAndAddStreams([{audio: true}]);
+        },
+
+        function PC_LOCAL_CREATE_AND_SET_OFFER(test) {
+          return test.createOffer(test.pcLocal).then(offer => {
+            return test.setLocalDescription(test.pcLocal, offer, HAVE_LOCAL_OFFER);
+          });
+        },
+
+        function PC_LOCAL_ROLLBACK(test) {
+          // the negotiationNeeded slot should have been true both before and
+          // after this SLD, so the event should fire again.
+          test.pcLocal.expectNegotiationNeeded();
+          return test.setLocalDescription(test.pcLocal,
+                                          { type: "rollback", sdp: "" },
+                                          STABLE);
+        },
+    ]);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_localRollback.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_localRollback.html
new file mode 100644
index 0000000000..5bdc8cc029
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_localRollback.html
@@ -0,0 +1,47 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "952145",
+    title: "Rollback local offer"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    test.chain.insertBefore('PC_LOCAL_CREATE_OFFER', [
+        function PC_REMOTE_CREATE_AND_SET_OFFER(test) {
+          return test.createOffer(test.pcRemote).then(offer => {
+            return test.setLocalDescription(test.pcRemote, offer, HAVE_LOCAL_OFFER);
+          });
+        },
+
+        function PC_REMOTE_ROLLBACK(test) {
+          // the negotiationNeeded slot should have been true both before and
+          // after this SLD, so the event should fire again.
+          test.pcRemote.expectNegotiationNeeded();
+          return test.setLocalDescription(test.pcRemote,
+                                          { type: "rollback", sdp: "" },
+                                          STABLE);
+        },
+
+        // Rolling back should shut down gathering
+        function PC_REMOTE_WAIT_FOR_END_OF_TRICKLE(test) {
+          return test.pcRemote.endOfTrickleIce;
+        },
+
+        function PC_REMOTE_SETUP_ICE_HANDLER(test) {
+          test.pcRemote.setupIceCandidateHandler(test);
+        },
+    ]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_maxFsConstraint.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_maxFsConstraint.html
new file mode 100644
index 0000000000..a2f2555020
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_maxFsConstraint.html
@@ -0,0 +1,112 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1393687",
+    title: "Enforce max-fs constraint on a PeerConnection",
+    visible: true
+  });
+
+  var mustRejectWith = (msg, reason, f) =>
+    f().then(() => ok(false, msg),
+             e => is(e.name, reason, msg));
+
+  var removeAllButCodec = (d, codec) =>
+    (d.sdp = d.sdp.replace(/m=video (\w) UDP\/TLS\/RTP\/SAVPF \w.*\r\n/,
+                           "m=video $1 UDP/TLS/RTP/SAVPF " + codec + "\r\n"), d);
+
+  var mungeSDP = (d, forceH264) => {
+    if (forceH264) {
+      removeAllButCodec(d, 126);
+      d.sdp = d.sdp.replace(/a=fmtp:126 (.*);packetization-mode=1/, "a=fmtp:126 $1;packetization-mode=1;max-fs=100");
+    } else {
+      d.sdp = d.sdp.replace(/max-fs=\d+/, "max-fs=100");
+    }
+    return d;
+  };
+
+  const checkForH264Support = async () => {
+    const pc = new RTCPeerConnection();
+    const offer = await pc.createOffer({offerToReceiveVideo: true});
+    return offer.sdp.match(/a=rtpmap:[1-9][0-9]* H264/);
+  };
+
+  let resolutionAlignment = 1;
+
+  function testScale(codec) {
+    var v1 = createMediaElement('video', 'v1');
+    var v2 = createMediaElement('video', 'v2');
+
+    var pc1 = new RTCPeerConnection();
+    var pc2 = new RTCPeerConnection();
+
+    var add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    pc1.onicecandidate = e => add(pc2, e.candidate, generateErrorCallback());
+    pc2.onicecandidate = e => add(pc1, e.candidate, generateErrorCallback());
+
+    info("testing max-fs with" + codec);
+
+    pc1.onnegotiationneeded = e =>
+      pc1.createOffer()
+      .then(d => pc1.setLocalDescription(mungeSDP(d, codec == "H264")))
+      .then(() => pc2.setRemoteDescription(pc1.localDescription))
+      .then(() => pc2.createAnswer()).then(d => pc2.setLocalDescription(mungeSDP(d, codec =="H264")))
+      .then(() => pc1.setRemoteDescription(pc2.localDescription))
+      .catch(generateErrorCallback());
+
+    pc2.ontrack = e => {
+      v2.srcObject = e.streams[0];
+    };
+
+    var stream;
+
+    return navigator.mediaDevices.getUserMedia({ video: true })
+      .then(s => {
+        stream = s;
+        v1.srcObject = stream;
+        let track = stream.getVideoTracks()[0];
+        let sender = pc1.addTrack(track, stream);
+        is(v2.currentTime, 0, "v2.currentTime is zero at outset");
+      })
+      .then(() => wait(5000))
+      .then(() => {
+        if (v2.videoWidth == 0 && v2.videoHeight == 0) {
+          info("Skipping test, insufficient time for video to start.");
+        } else {
+          const expectedWidth = 184 - 184 % resolutionAlignment;
+          const expectedHeight = 138 - 138 % resolutionAlignment;
+          is(v2.videoWidth, expectedWidth,
+             `sink width should be ${expectedWidth} for ${codec}`);
+          is(v2.videoHeight, expectedHeight,
+             `sink height should be ${expectedHeight} for ${codec}`);
+        }})
+      .then(() => {
+        stream.getTracks().forEach(track => track.stop());
+        v1.srcObject = v2.srcObject = null;
+      }).catch(generateErrorCallback());
+  }
+
+  runNetworkTest(async () => {
+    await pushPrefs(['media.peerconnection.video.lock_scaling', true]);
+    if (await checkForH264Support()) {
+      if (navigator.userAgent.includes("Android")) {
+        // Android only has a hw encoder for h264
+        resolutionAlignment = 16;
+      }
+      await matchPlatformH264CodecPrefs();
+      await testScale("H264");
+    }
+
+    // Disable h264 hardware support, to ensure it is not prioritized over VP8
+    await pushPrefs(["media.webrtc.hw.h264.enabled", false]);
+    await testScale("VP8");
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_multiple_captureStream_canvas_2d.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_multiple_captureStream_canvas_2d.html
new file mode 100644
index 0000000000..9ad25e7852
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_multiple_captureStream_canvas_2d.html
@@ -0,0 +1,115 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1166832",
+  title: "Canvas(2D)::Multiple CaptureStream as video-only input to peerconnection",
+  visible: true
+});
+
+/**
+ * Test to verify using multiple capture streams concurrently.
+ */
+runNetworkTest(async () => {
+  // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+  if (navigator.userAgent.includes("Android")) {
+    await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false]);
+    await pushPrefs(["media.webrtc.hw.h264.enabled", false]);
+  }
+
+  var test = new PeerConnectionTest();
+  var h = new CaptureStreamTestHelper2D(50, 50);
+
+  var vremote1;
+  var stream1;
+  var canvas1 = h.createAndAppendElement('canvas', 'source_canvas1');
+
+  var vremote2;
+  var stream2;
+  var canvas2 = h.createAndAppendElement('canvas', 'source_canvas2');
+
+  const threshold = 128;
+
+  test.setMediaConstraints([{video: true}, {video: true}], []);
+  test.chain.replace("PC_LOCAL_GUM", [
+    function PC_LOCAL_CANVAS_CAPTURESTREAM(test) {
+      h.drawColor(canvas1, h.green);
+      h.drawColor(canvas2, h.blue);
+      stream1 = canvas1.captureStream(0); // fps = 0 to capture single frame
+      test.pcLocal.attachLocalStream(stream1);
+      stream2 = canvas2.captureStream(0); // fps = 0 to capture single frame
+      test.pcLocal.attachLocalStream(stream2);
+      var i = 0;
+      return setInterval(function() {
+        try {
+          info("draw " + i ? "green" : "red/blue");
+          h.drawColor(canvas1, i ? h.green : h.red);
+          h.drawColor(canvas2, i ? h.green : h.blue);
+          i = 1 - i;
+          stream1.requestFrame();
+          stream2.requestFrame();
+        } catch (e) {
+          // ignore; stream might have shut down, and we don't bother clearing
+          // the setInterval.
+        }
+      }, 500);
+    }
+  ]);
+
+  test.chain.append([
+    function CHECK_REMOTE_VIDEO() {
+      is(test.pcRemote.remoteMediaElements.length, 2, "pcRemote Should have 2 remote media elements");
+      vremote1 = test.pcRemote.remoteMediaElements[0];
+      vremote2 = test.pcRemote.remoteMediaElements[1];
+
+      // since we don't know which remote video is created first, we don't know
+      // which should be blue or red, but this will make sure that one is
+      // green and one is blue
+      return Promise.race([
+               Promise.all([
+                 h.pixelMustBecome(vremote1, h.red, {
+                   threshold,
+                   infoString: "pcRemote's remote1 should become red",
+                 }),
+                 h.pixelMustBecome(vremote2, h.blue, {
+                   threshold,
+                   infoString: "pcRemote's remote2 should become blue",
+                 }),
+               ]),
+               Promise.all([
+                 h.pixelMustBecome(vremote2, h.red, {
+                   threshold,
+                   infoString: "pcRemote's remote2 should become red",
+                 }),
+                 h.pixelMustBecome(vremote1, h.blue, {
+                   threshold,
+                   infoString: "pcRemote's remote1 should become blue",
+                 }),
+               ])
+             ]);
+    },
+    function WAIT_FOR_REMOTE_BOTH_GREEN() {
+      return Promise.all([
+               h.pixelMustBecome(vremote1, h.green, {
+                   threshold,
+                   infoString: "pcRemote's remote1 should become green",
+               }),
+               h.pixelMustBecome(vremote2, h.green, {
+                   threshold,
+                   infoString: "pcRemote's remote2 should become green",
+               }),
+             ])
+    },
+  ]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_noTrickleAnswer.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_noTrickleAnswer.html
new file mode 100644
index 0000000000..7e3fd78430
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_noTrickleAnswer.html
@@ -0,0 +1,25 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="nonTrickleIce.js"></script>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1060102",
+    title: "Basic audio only SDP answer without trickle ICE"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    makeAnswererNonTrickle(test.chain);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_noTrickleOffer.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_noTrickleOffer.html
new file mode 100644
index 0000000000..12b2a95596
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_noTrickleOffer.html
@@ -0,0 +1,25 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="nonTrickleIce.js"></script>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1060102",
+    title: "Basic audio only SDP offer without trickle ICE"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    makeOffererNonTrickle(test.chain);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_noTrickleOfferAnswer.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_noTrickleOfferAnswer.html
new file mode 100644
index 0000000000..554750e975
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_noTrickleOfferAnswer.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="nonTrickleIce.js"></script>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1060102",
+    title: "Basic audio only SDP offer and answer without trickle ICE"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    makeOffererNonTrickle(test.chain);
+    makeAnswererNonTrickle(test.chain);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_nonDefaultRate.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_nonDefaultRate.html
new file mode 100644
index 0000000000..ad9414cef2
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_nonDefaultRate.html
@@ -0,0 +1,200 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({ title: "PeerConnection feed to a graph with non default rate", bug: "1387454" });
+  /**
+   * Run a test to verify that when we use the streem with nonDefault rate to/from a PC
+   * the connection fails. (PC is always on default rate).
+   */
+
+  let pc1;
+  let pc2;
+
+  const offerOptions = {
+    offerToReceiveAudio: 1,
+  };
+
+  function getName(pc) {
+    return (pc === pc1) ? 'pc1' : 'pc2';
+  }
+
+  function getOtherPc(pc) {
+    return (pc === pc1) ? pc2 : pc1;
+  }
+
+  function onAddIceCandidateSuccess(pc) {
+    ok(true, getName(pc) + ' addIceCandidate success');
+  }
+
+  function onAddIceCandidateError(pc, error) {
+    ok(false, getName(pc) + ' failed to add ICE Candidate: ' + error.toString());
+  }
+
+  function onIceCandidate(pc, event, done) {
+    if (!event.candidate) {
+      ok(pc.iceGatheringState === 'complete', getName(pc) + " ICE gathering state has reached complete");
+      done();
+      return;
+    }
+    getOtherPc(pc).addIceCandidate(event.candidate)
+    .then(() => {
+        onAddIceCandidateSuccess(pc);
+      },
+      (err) => {
+        onAddIceCandidateError(pc, err);
+      });
+    info(getName(pc) + ' ICE candidate: ' + event.candidate.candidate);
+  }
+
+  function onIceStateChange(pc, event) {
+    if (pc) {
+      info(getName(pc) + ' ICE state: ' + pc.iceConnectionState);
+      info('ICE state change event: ', event);
+    }
+  }
+
+  function onCreateOfferSuccess(desc) {
+    info('Offer from pc1\n' + desc.sdp);
+    info('pc1 setLocalDescription start');
+
+    pc1.setLocalDescription(desc)
+    .then(() => {
+      onSetLocalSuccess(pc1);
+    },
+    onSetSessionDescriptionError);
+
+    info('pc2 setRemoteDescription start');
+    pc2.setRemoteDescription(desc).then(() => {
+      onSetRemoteSuccess(pc2);
+    },
+    onSetSessionDescriptionError);
+
+    info('pc2 createAnswer start');
+
+    // Since the 'remote' side has no media stream we need
+    // to pass in the right constraints in order for it to
+    // accept the incoming offer of audio and video.
+    pc2.createAnswer()
+    .then(onCreateAnswerSuccess, onCreateSessionDescriptionError);
+  }
+
+  function onSetSessionDescriptionError(error) {
+    ok(false, 'Failed to set session description: ' + error.toString());
+  }
+
+  function onSetLocalSuccess(pc) {
+    ok(true, getName(pc) + ' setLocalDescription complete');
+  }
+
+  function onCreateSessionDescriptionError(error) {
+    ok(false, 'Failed to create session description: ' + error.toString());
+  }
+
+  function onSetRemoteSuccess(pc) {
+    ok(true, getName(pc) + ' setRemoteDescription complete');
+  }
+
+  function onCreateAnswerSuccess(desc) {
+    info('Answer from pc2:\n' + desc.sdp);
+    info('pc2 setLocalDescription start');
+    pc2.setLocalDescription(desc).then(() => {
+      onSetLocalSuccess(pc2);
+    },
+    onSetSessionDescriptionError);
+    info('pc1 setRemoteDescription start');
+    pc1.setRemoteDescription(desc).then(() => {
+        onSetRemoteSuccess(pc1);
+    },
+    onSetSessionDescriptionError);
+  }
+
+  async function getRemoteStream(localStream) {
+    info("got local stream")
+    const audioTracks = localStream.getAudioTracks();
+
+    const servers = null;
+
+    pc1 = new RTCPeerConnection(servers);
+    info('Created local peer connection object pc1');
+    const iceComplete1 = new Promise((resolve, reject) => {
+      pc1.onicecandidate = (e) => {
+        onIceCandidate(pc1, e, resolve);
+      };
+    });
+
+    pc2 = new RTCPeerConnection(servers);
+    info('Created remote peer connection object pc2');
+    const iceComplete2 = new Promise((resolve, reject) => {
+      pc2.onicecandidate = (e) => {
+        onIceCandidate(pc2, e, resolve);
+      };
+    });
+
+    pc1.oniceconnectionstatechange = (e) => {
+      onIceStateChange(pc1, e);
+    };
+    pc2.oniceconnectionstatechange = (e) => {
+      onIceStateChange(pc2, e);
+    };
+
+    const remoteStreamPromise = new Promise((resolve, reject) => {
+      pc2.ontrack = (e) => {
+        info('pc2 received remote stream ' + e.streams[0]);
+        resolve(e.streams[0]);
+      };
+    });
+
+    localStream.getTracks().forEach((track) => {
+      pc1.addTrack(track, localStream);
+    });
+    info('Added local stream to pc1');
+
+    info('pc1 createOffer start');
+    pc1.createOffer(offerOptions)
+      .then(onCreateOfferSuccess,onCreateSessionDescriptionError);
+
+    let promise_arr = await Promise.all([remoteStreamPromise, iceComplete1, iceComplete2]);
+    return promise_arr[0];
+  }
+
+  runTest(async () => {
+    // Local stream operates at non default rate (32000)
+    const nonDefaultRate = 32000;
+    const nonDefault_ctx = new AudioContext({sampleRate: nonDefaultRate});
+    oscillator = nonDefault_ctx.createOscillator();
+    const dest = nonDefault_ctx.createMediaStreamDestination();
+    oscillator.connect(dest);
+    oscillator.start();
+
+    // Wait for remote stream
+    const remoteStream = await getRemoteStream(dest.stream)
+    ok(true, 'Got remote stream ' + remoteStream);
+
+    // remoteStream now comes from PC so operates at default
+    // rates. Verify that by adding to a default context
+    const ac = new AudioContext;
+    const source_default_rate = ac.createMediaStreamSource(remoteStream);
+
+    // Now try to add the remoteStream on a non default context
+    mustThrowWith(
+      "Connect stream with graph of different sample rate",
+      "NotSupportedError", () => {
+        nonDefault_ctx.createMediaStreamSource(remoteStream);
+      }
+    );
+
+    // Close peer connections to make sure we don't get error:
+    // "logged result after SimpleTest.finish(): pc1 addIceCandidate success"
+    // See Bug 1626814.
+    pc1.close();
+    pc2.close();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_offerRequiresReceiveAudio.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_offerRequiresReceiveAudio.html
new file mode 100644
index 0000000000..1f936714f1
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_offerRequiresReceiveAudio.html
@@ -0,0 +1,23 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "850275",
+    title: "Simple offer media constraint test with audio"
+  });
+
+  runNetworkTest(function() {
+    var test = new PeerConnectionTest();
+    test.setMediaConstraints([], [{audio: true}]);
+    test.setOfferOptions({ offerToReceiveAudio: true });
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_offerRequiresReceiveVideo.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_offerRequiresReceiveVideo.html
new file mode 100644
index 0000000000..c5afbb5c1f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_offerRequiresReceiveVideo.html
@@ -0,0 +1,23 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "850275",
+    title: "Simple offer media constraint test with video"
+  });
+
+  runNetworkTest(function() {
+    var test = new PeerConnectionTest();
+    test.setMediaConstraints([], [{video: true}]);
+    test.setOfferOptions({ offerToReceiveVideo: true });
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_offerRequiresReceiveVideoAudio.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_offerRequiresReceiveVideoAudio.html
new file mode 100644
index 0000000000..d7bc29c6d3
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_offerRequiresReceiveVideoAudio.html
@@ -0,0 +1,23 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "850275",
+    title: "Simple offer media constraint test with video/audio"
+  });
+
+  runNetworkTest(function() {
+    var test = new PeerConnectionTest();
+    test.setMediaConstraints([], [{audio: true, video: true}]);
+    test.setOfferOptions({ offerToReceiveVideo: true, offerToReceiveAudio: true });
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_portRestrictions.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_portRestrictions.html
new file mode 100644
index 0000000000..7cd695ff54
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_portRestrictions.html
@@ -0,0 +1,63 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1677046",
+    title: "RTCPeerConnection check restricted ports"
+  });
+
+var makePC = (config, expected_error) => {
+  var exception;
+  try {
+    new RTCPeerConnection(config).close();
+  } catch (e) {
+    exception = e;
+  }
+  is((exception? exception.name : "success"), expected_error || "success",
+     "RTCPeerConnection(" + JSON.stringify(config) + ")");
+};
+
+// This is a test of the iceServers parsing code + readable errors
+runNetworkTest(() => {
+  var exception = null;
+
+  // check various ports on the blocklist
+  makePC({ iceServers: [
+    { urls:"turn:[::1]:6666", username:"p", credential:"p" }] }, "NS_ERROR_UNEXPECTED");
+  makePC({ iceServers: [
+    { urls:"turns:localhost:6667?transport=udp", username:"p", credential:"p" }] },
+    "NS_ERROR_UNEXPECTED");
+  makePC({ iceServers: [
+    { urls:"stun:localhost:21", foo:"" }] }, "NS_ERROR_UNEXPECTED");
+  makePC({ iceServers: [
+    { urls:"stun:[::1]:22", foo:"" }] }, "NS_ERROR_UNEXPECTED");
+  makePC({ iceServers: [
+    { urls:"turn:localhost:5060", username:"p", credential:"p" }] },
+    "NS_ERROR_UNEXPECTED");
+
+  // check various ports on the good list for webrtc (or default port)
+  makePC({ iceServers: [
+    { urls:"turn:[::1]:53", username:"p", credential:"p" },
+    { urls:"turn:[::1]:5349", username:"p", credential:"p" },
+    { urls:"turn:[::1]:3478", username:"p", credential:"p" },
+    { urls:"turn:[::1]", username:"p", credential:"p" },
+    { urls:"turn:localhost:53?transport=udp", username:"p", credential:"p" },
+    { urls:"turn:localhost:3478?transport=udp", username:"p", credential:"p" },
+    { urls:"turn:localhost:53?transport=tcp", username:"p", credential:"p" },
+    { urls:"turn:localhost:3478?transport=tcp", username:"p", credential:"p" },
+    { urls:"turns:localhost:3478?transport=udp", username:"p", credential:"p" },
+    { urls:"stun:localhost", foo:"" }
+  ]});
+
+  // not in the known good ports and not on the generic block list
+  makePC({ iceServers: [{ urls:"turn:localhost:6664", username:"p", credential:"p" }] });
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_promiseSendOnly.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_promiseSendOnly.html
new file mode 100644
index 0000000000..a3fbb5753c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_promiseSendOnly.html
@@ -0,0 +1,61 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1091898",
+    title: "PeerConnection with promises (sendonly)",
+    visible: true
+  });
+
+  var pc1 = new RTCPeerConnection();
+  var pc2 = new RTCPeerConnection();
+
+  var add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+  pc1.onicecandidate = e => add(pc2, e.candidate, generateErrorCallback());
+  pc2.onicecandidate = e => add(pc1, e.candidate, generateErrorCallback());
+
+  var v1, v2;
+  var delivered = new Promise(resolve => pc2.ontrack = e => {
+    // Test RTCTrackEvent here.
+    ok(e.streams.length, "has streams");
+    ok(e.streams[0].getTrackById(e.track.id), "has track");
+    ok(pc2.getReceivers().some(receiver => receiver == e.receiver), "has receiver");
+    if (e.streams[0].getTracks().length == 2) {
+      // Test RTCTrackEvent required args here.
+      mustThrowWith("RTCTrackEvent wo/required args",
+                    "TypeError", () => new RTCTrackEvent("track", {}));
+      v2.srcObject = e.streams[0];
+      resolve();
+    }
+  });
+
+  runNetworkTest(function() {
+    v1 = createMediaElement('video', 'v1');
+    v2 = createMediaElement('video', 'v2');
+    var canPlayThrough = new Promise(resolve => v2.canplaythrough = e => resolve());
+
+    is(v2.currentTime, 0, "v2.currentTime is zero at outset");
+
+    return navigator.mediaDevices.getUserMedia({ video: true, audio: true })
+    .then(stream => (v1.srcObject = stream).getTracks().forEach(t => pc1.addTrack(t, stream)))
+    .then(() => pc1.createOffer({})) // check that createOffer accepts arg.
+    .then(offer => pc1.setLocalDescription(offer))
+    .then(() => pc2.setRemoteDescription(pc1.localDescription))
+    .then(() => pc2.createAnswer({}))  // check that createAnswer accepts arg.
+    .then(answer => pc2.setLocalDescription(answer))
+    .then(() => pc1.setRemoteDescription(pc2.localDescription))
+    .then(() => delivered)
+//    .then(() => canPlayThrough)    // why doesn't this fire?
+    .then(() => waitUntil(() => v2.currentTime > 0))
+    .then(() => ok(v2.currentTime > 0, "v2.currentTime is moving (" + v2.currentTime + ")"))
+    .then(() => ok(true, "Connected."));
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_recordReceiveTrack.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_recordReceiveTrack.html
new file mode 100644
index 0000000000..d5cb91b048
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_recordReceiveTrack.html
@@ -0,0 +1,101 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+<script src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script>
+createHTML({
+  bug: "1212237",
+  title: "Recording a fresh receive track should not throw",
+  visible: true,
+});
+
+/**
+ * Called when a fresh track is available, and tests that the track can be
+ * recorded until it ends without any thrown errors or fired error events.
+ */
+let generation = 0;
+async function testTrackAccessible(track) {
+  const id = ++generation;
+  info(`Testing accessibility for ${track.kind} track ${id}`);
+  const recorder = new MediaRecorder(new MediaStream([track]));
+  recorder.start();
+  let haveError = new Promise((_, rej) => recorder.onerror = e => rej(e.error));
+  await Promise.race([
+    new Promise(r => recorder.onstart = r),
+    haveError,
+  ]);
+  info(`Recording of ${track.kind} track ${id} started`);
+
+  const {data} = await Promise.race([
+    new Promise(r => recorder.ondataavailable = r),
+    haveError,
+  ]);
+  info(`Recording of ${track.kind} track ${id} finished at size ${data.size}`);
+
+  await Promise.race([
+    new Promise(r => recorder.onstop = r),
+    haveError,
+  ]);
+  info(`Recording of ${track.kind} track ${id} stopped`);
+
+  const element = createMediaElement(track.kind, `recording_${track.id}`);
+  const url = URL.createObjectURL(data);
+  try {
+    element.src = url;
+    element.preload = "metadata";
+    haveError = new Promise(
+      (_, rej) => element.onerror = e => rej(element.error));
+    await Promise.race([
+      new Promise(r => element.onloadeddata = r),
+      haveError,
+    ]);
+    info(`Playback of recording of ${track.kind} track ${id} loaded data`);
+
+    element.play();
+    await Promise.race([
+      new Promise(r => element.onended = r),
+      haveError,
+    ]);
+    info(`Playback of recording of ${track.kind} track ${id} ended`);
+  } finally {
+    URL.revokeObjectURL(data);
+  }
+}
+
+runNetworkTest(async options => {
+  // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+  if (navigator.userAgent.includes("Android")) {
+    await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                    ["media.webrtc.hw.h264.enabled", false]);
+  }
+  const test = new PeerConnectionTest(options);
+  test.setMediaConstraints([{video: true}], [{audio: true}]);
+  test.setOfferOptions({offerToReceiveAudio: true});
+  const freshVideoTrackIsAccessible = new Promise(
+      r => test.pcRemote._pc.addEventListener("track", r, {once: true})
+    ).then(({track}) => testTrackAccessible(track));
+  const freshAudioTrackIsAccessible = new Promise(
+      r => test.pcLocal._pc.addEventListener("track", r, {once: true})
+    ).then(({track}) => testTrackAccessible(track));
+  test.chain.append([
+    function PC_CLOSE_TO_END_TRACKS() {
+      return test.close();
+    },
+    async function FRESH_VIDEO_TRACK_IS_ACCESSIBLE() {
+      await freshVideoTrackIsAccessible;
+      ok(true, "A freshly received video track is accessible by MediaRecorder");
+    },
+    async function FRESH_AUDIO_TRACK_IS_ACCESSIBLE() {
+      await freshAudioTrackIsAccessible;
+      ok(true, "A freshly received audio track is accessible by MediaRecorder");
+    },
+  ]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_relayOnly.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_relayOnly.html
new file mode 100644
index 0000000000..3b07783c04
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_relayOnly.html
@@ -0,0 +1,60 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1187775",
+  title: "peer connection ICE fails on relay-only without TURN"
+});
+
+function PC_LOCAL_NO_CANDIDATES(test) {
+  var isnt = can => is(can, null, "No candidates: " + JSON.stringify(can));
+  test.pcLocal._pc.addEventListener("icecandidate", e => isnt(e.candidate));
+}
+
+function PC_BOTH_WAIT_FOR_ICE_FAILED(test) {
+  var isFail = (f, reason, msg) =>
+    f().then(() => { throw new Error(msg + " must fail"); },
+             e => is(e.message, reason, msg + " must fail with: " + e.message));
+
+  return Promise.all([
+    isFail(() => test.pcLocal.waitForIceConnected(), "ICE failed", "Local ICE"),
+    isFail(() => test.pcRemote.waitForIceConnected(), "ICE failed", "Remote ICE")
+  ])
+  .then(() => ok(true, "ICE on both sides must fail."));
+}
+
+runNetworkTest(async options => {
+  await pushPrefs(
+    ['media.peerconnection.ice.stun_client_maximum_transmits', 3],
+    ['media.peerconnection.ice.trickle_grace_period', 5000]
+  );
+  options = options || {};
+  options.config_local = options.config_local || {};
+  const servers = options.config_local.iceServers || [];
+  // remove any turn servers
+  options.config_local.iceServers = servers.filter(server =>
+      server.urls.every(u => !u.toLowerCase().startsWith('turn')));
+
+  // Here's the setting we're testing. Comment out and this test should fail:
+  options.config_local.iceTransportPolicy = "relay";
+
+  const test = new PeerConnectionTest(options);
+  test.setMediaConstraints([{audio: true}, {video: true}],
+                           [{audio: true}, {video: true}]);
+  test.chain.remove("PC_LOCAL_SETUP_ICE_LOGGER");  // Needed to suppress failing
+  test.chain.remove("PC_REMOTE_SETUP_ICE_LOGGER"); // on ICE-failure.
+  test.chain.insertAfter("PC_LOCAL_SETUP_ICE_HANDLER", PC_LOCAL_NO_CANDIDATES);
+  test.chain.replace("PC_LOCAL_WAIT_FOR_ICE_CONNECTED", PC_BOTH_WAIT_FOR_ICE_FAILED);
+  test.chain.removeAfter("PC_BOTH_WAIT_FOR_ICE_FAILED");
+  await test.run();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_remoteReofferRollback.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_remoteReofferRollback.html
new file mode 100644
index 0000000000..80aa30beaa
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_remoteReofferRollback.html
@@ -0,0 +1,50 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "952145",
+    title: "Rollback remote reoffer"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    addRenegotiation(test.chain,
+      [
+        function PC_LOCAL_ADD_SECOND_STREAM(test) {
+          test.setMediaConstraints([{audio: true}, {audio: true}],
+                                   [{audio: true}]);
+          return test.pcLocal.getAllUserMediaAndAddStreams([{audio: true}]);
+        },
+      ]
+    );
+    test.chain.replaceAfter('PC_REMOTE_SET_REMOTE_DESCRIPTION',
+      [
+        function PC_REMOTE_ROLLBACK(test) {
+          return test.setRemoteDescription(test.pcRemote, { type: "rollback" },
+                                           STABLE);
+        },
+
+        function PC_LOCAL_ROLLBACK(test) {
+          // We haven't negotiated the new stream yet.
+          test.pcLocal.expectNegotiationNeeded();
+          return test.setLocalDescription(
+              test.pcLocal,
+              new RTCSessionDescription({ type: "rollback", sdp: ""}),
+              STABLE);
+        },
+      ],
+      1 // Second PC_REMOTE_SET_REMOTE_DESCRIPTION
+    );
+    test.chain.append(commandsPeerConnectionOfferAnswer);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_remoteRollback.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_remoteRollback.html
new file mode 100644
index 0000000000..827646b0de
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_remoteRollback.html
@@ -0,0 +1,51 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "952145",
+    title: "Rollback remote offer"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    test.chain.removeAfter('PC_REMOTE_CHECK_CAN_TRICKLE_SYNC');
+    test.chain.append([
+      function PC_REMOTE_ROLLBACK(test) {
+        // We still haven't negotiated the tracks
+        test.pcRemote.expectNegotiationNeeded();
+        return test.setRemoteDescription(test.pcRemote, { type: "rollback" },
+                                         STABLE);
+      },
+
+      function PC_REMOTE_CHECK_CAN_TRICKLE_REVERT_SYNC(test) {
+        is(test.pcRemote._pc.canTrickleIceCandidates, null,
+           "Remote canTrickleIceCandidates is reverted to null");
+      },
+
+      function PC_LOCAL_ROLLBACK(test) {
+        // We still haven't negotiated the tracks
+        test.pcLocal.expectNegotiationNeeded();
+        return test.setLocalDescription(
+          test.pcLocal,
+          new RTCSessionDescription({ type: "rollback", sdp: ""}),
+          STABLE);
+      },
+
+      // Rolling back should shut down gathering
+      function PC_LOCAL_WAIT_FOR_END_OF_TRICKLE(test) {
+        return test.pcLocal.endOfTrickleIce;
+      },
+    ]);
+    test.chain.append(commandsPeerConnectionOfferAnswer);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_removeAudioTrack.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_removeAudioTrack.html
new file mode 100644
index 0000000000..e1e99b38c9
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_removeAudioTrack.html
@@ -0,0 +1,57 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: remove audio track"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    let receivedTrack, analyser, freq;
+    addRenegotiation(test.chain,
+      [
+        function PC_REMOTE_SETUP_ANALYSER(test) {
+          is(test.pcRemote._pc.getReceivers().length, 1,
+             "pcRemote should have one receiver before renegotiation");
+
+          receivedTrack = test.pcRemote._pc.getReceivers()[0].track;
+          is(receivedTrack.readyState, "live",
+             "The received track should be live");
+
+          analyser = new AudioStreamAnalyser(
+              new AudioContext(), new MediaStream([receivedTrack]));
+          freq = analyser.binIndexForFrequency(TEST_AUDIO_FREQ);
+
+          return analyser.waitForAnalysisSuccess(arr => arr[freq] > 200);
+        },
+        function PC_LOCAL_REMOVE_AUDIO_TRACK(test) {
+          test.setOfferOptions({ offerToReceiveAudio: true });
+          return test.pcLocal.removeSender(0);
+        },
+      ],
+      [
+        function PC_REMOTE_CHECK_FLOW_STOPPED(test) {
+          // Simply removing a track is not enough to cause it to be
+          // signaled as ended. Spec may change though.
+          // TODO: One last check of the spec is in order
+          is(receivedTrack.readyState, "live",
+             "The received track should not have ended");
+
+          return analyser.waitForAnalysisSuccess(arr => arr[freq] < 50);
+        },
+      ]
+    );
+
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_removeThenAddAudioTrack.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_removeThenAddAudioTrack.html
new file mode 100644
index 0000000000..28b76e3b43
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_removeThenAddAudioTrack.html
@@ -0,0 +1,87 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: remove then add audio track"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    let originalTrack;
+    let haveMuteEvent = new Promise(() => {});
+    let haveUnmuteEvent = new Promise(() => {});
+    addRenegotiation(test.chain,
+      [
+        function PC_REMOTE_FIND_RECEIVER(test) {
+          is(test.pcRemote._pc.getReceivers().length, 1,
+             "pcRemote should have one receiver");
+          originalTrack = test.pcRemote._pc.getReceivers()[0].track;
+        },
+        function PC_LOCAL_REMOVE_AUDIO_TRACK(test) {
+          return test.pcLocal.removeSender(0);
+        },
+        function PC_LOCAL_ADD_AUDIO_TRACK(test) {
+          // The new track's pipeline will start with a packet count of
+          // 0, but the remote side will keep its old pipeline and packet
+          // count.
+          test.pcLocal.disableRtpCountChecking = true;
+          return test.pcLocal.getAllUserMediaAndAddStreams([{audio: true}]);
+        },
+      ],
+      [
+        function PC_REMOTE_WAIT_FOR_UNMUTE() {
+          return haveUnmuteEvent;
+        },
+        function PC_REMOTE_CHECK_ADDED_TRACK(test) {
+          is(test.pcRemote._pc.getTransceivers().length, 2,
+              "pcRemote should have two transceivers");
+          const track = test.pcRemote._pc.getTransceivers()[1].receiver.track;
+
+          const analyser = new AudioStreamAnalyser(
+              new AudioContext(), new MediaStream([track]));
+          const freq = analyser.binIndexForFrequency(TEST_AUDIO_FREQ);
+          return analyser.waitForAnalysisSuccess(arr => arr[freq] > 200);
+        },
+        function PC_REMOTE_WAIT_FOR_MUTE() {
+          return haveMuteEvent;
+        },
+        function PC_REMOTE_CHECK_REMOVED_TRACK(test) {
+          is(test.pcRemote._pc.getTransceivers().length, 2,
+              "pcRemote should have two transceivers");
+          const track = test.pcRemote._pc.getTransceivers()[0].receiver.track;
+
+          const analyser = new AudioStreamAnalyser(
+              new AudioContext(), new MediaStream([track]));
+          const freq = analyser.binIndexForFrequency(TEST_AUDIO_FREQ);
+          return analyser.waitForAnalysisSuccess(arr => arr[freq] < 50);
+        }
+      ]
+    );
+
+    // The first track should mute when the connection is closed.
+    test.chain.insertBefore("PC_REMOTE_SET_REMOTE_DESCRIPTION", [
+      function PC_REMOTE_SETUP_ONMUTE(test) {
+        haveMuteEvent = haveEvent(test.pcRemote._pc.getReceivers()[0].track, "mute");
+      }
+    ]);
+
+    // Second negotiation should cause the second track to unmute.
+    test.chain.insertAfter("PC_REMOTE_SET_REMOTE_DESCRIPTION", [
+      function PC_REMOTE_SETUP_ONUNMUTE(test) {
+        haveUnmuteEvent = haveEvent(test.pcRemote._pc.getReceivers()[1].track, "unmute");
+      }
+    ], false, 1);
+
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_removeThenAddAudioTrackNoBundle.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_removeThenAddAudioTrackNoBundle.html
new file mode 100644
index 0000000000..cff424e12c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_removeThenAddAudioTrackNoBundle.html
@@ -0,0 +1,76 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: remove then add audio track"
+  });
+
+  runNetworkTest(function (options) {
+    options = options || { };
+    options.bundle = false;
+    const test = new PeerConnectionTest(options);
+    let originalTrack;
+    addRenegotiation(test.chain,
+      [
+        function PC_REMOTE_FIND_RECEIVER(test) {
+          is(test.pcRemote._pc.getReceivers().length, 1,
+             "pcRemote should have one receiver");
+          originalTrack = test.pcRemote._pc.getReceivers()[0].track;
+        },
+        function PC_LOCAL_REMOVE_AUDIO_TRACK(test) {
+          // The new track's pipeline will start with a packet count of
+          // 0, but the remote side will keep its old pipeline and packet
+          // count.
+          test.pcLocal.disableRtpCountChecking = true;
+          return test.pcLocal.removeSender(0);
+        },
+        function PC_LOCAL_ADD_AUDIO_TRACK(test) {
+          return test.pcLocal.getAllUserMediaAndAddStreams([{audio: true}]);
+        },
+        function PC_LOCAL_EXPECT_ICE_CHECKING(test) {
+          test.pcLocal.expectIceChecking();
+        },
+        function PC_REMOTE_EXPECT_ICE_CHECKING(test) {
+          test.pcRemote.expectIceChecking();
+        },
+      ],
+      [
+        function PC_REMOTE_CHECK_ADDED_TRACK(test) {
+          is(test.pcRemote._pc.getTransceivers().length, 2,
+              "pcRemote should have two transceivers");
+          const track = test.pcRemote._pc.getTransceivers()[1].receiver.track;
+
+          const analyser = new AudioStreamAnalyser(
+              new AudioContext(), new MediaStream([track]));
+          const freq = analyser.binIndexForFrequency(TEST_AUDIO_FREQ);
+          return analyser.waitForAnalysisSuccess(arr => arr[freq] > 200);
+        },
+        function PC_REMOTE_CHECK_REMOVED_TRACK(test) {
+          is(test.pcRemote._pc.getTransceivers().length, 2,
+              "pcRemote should have two transceivers");
+          const track = test.pcRemote._pc.getTransceivers()[0].receiver.track;
+
+          const analyser = new AudioStreamAnalyser(
+              new AudioContext(), new MediaStream([track]));
+          const freq = analyser.binIndexForFrequency(TEST_AUDIO_FREQ);
+          return analyser.waitForAnalysisSuccess(arr => arr[freq] < 50);
+        }
+      ]
+    );
+
+    test.chain.insertAfterEach('PC_LOCAL_CREATE_OFFER',
+                               PC_LOCAL_REMOVE_BUNDLE_FROM_OFFER);
+
+    test.setMediaConstraints([{audio: true}], [{audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_removeThenAddVideoTrack.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_removeThenAddVideoTrack.html
new file mode 100644
index 0000000000..b1be690e5b
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_removeThenAddVideoTrack.html
@@ -0,0 +1,98 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: remove then add video track"
+  });
+
+  runNetworkTest(async function (options) {
+    // Use fake video here since the native fake device on linux doesn't
+    // change color as needed by checkVideoPlaying() below.
+    await pushPrefs(
+      ['media.video_loopback_dev', ''],
+      ['media.navigator.streams.fake', true]);
+    // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+    if (navigator.userAgent.includes("Android")) {
+      await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                      ["media.webrtc.hw.h264.enabled", false]);
+    }
+
+    const test = new PeerConnectionTest(options);
+    const helper = new VideoStreamHelper();
+    var originalTrack;
+    let haveMuteEvent = new Promise(() => {});
+    let haveUnmuteEvent = new Promise(() => {});
+    addRenegotiation(test.chain,
+      [
+        function PC_REMOTE_FIND_RECEIVER(test) {
+          is(test.pcRemote._pc.getReceivers().length, 1,
+             "pcRemote should have one receiver");
+          originalTrack = test.pcRemote._pc.getReceivers()[0].track;
+        },
+        function PC_LOCAL_REMOVE_VIDEO_TRACK(test) {
+          // The new track's pipeline will start with a packet count of
+          // 0, but the remote side will keep its old pipeline and packet
+          // count.
+          test.pcLocal.disableRtpCountChecking = true;
+          return test.pcLocal.removeSender(0);
+        },
+        function PC_LOCAL_ADD_VIDEO_TRACK(test) {
+          return test.pcLocal.getAllUserMediaAndAddStreams([{video: true}]);
+        },
+      ],
+      [
+        function PC_REMOTE_WAIT_FOR_UNMUTE() {
+          return haveUnmuteEvent;
+        },
+        function PC_REMOTE_CHECK_ADDED_TRACK(test) {
+          is(test.pcRemote._pc.getTransceivers().length, 2,
+              "pcRemote should have two transceivers");
+          const track = test.pcRemote._pc.getTransceivers()[1].receiver.track;
+
+          const vAdded = test.pcRemote.remoteMediaElements.find(
+              elem => elem.id.includes(track.id));
+          return helper.checkVideoPlaying(vAdded);
+        },
+        function PC_REMOTE_WAIT_FOR_MUTE() {
+          return haveMuteEvent;
+        },
+        function PC_REMOTE_CHECK_REMOVED_TRACK(test) {
+          is(test.pcRemote._pc.getTransceivers().length, 2,
+              "pcRemote should have two transceivers");
+          const track = test.pcRemote._pc.getTransceivers()[0].receiver.track;
+
+          const vAdded = test.pcRemote.remoteMediaElements.find(
+              elem => elem.id.includes(track.id));
+          return helper.checkVideoPaused(vAdded, 10, 10, 16, 5000);
+        }
+      ]
+    );
+
+    // The first track should mute when the connection is closed.
+    test.chain.insertBefore("PC_REMOTE_SET_REMOTE_DESCRIPTION", [
+      function PC_REMOTE_SETUP_ONMUTE(test) {
+        haveMuteEvent = haveEvent(test.pcRemote._pc.getReceivers()[0].track, "mute");
+      }
+    ]);
+
+    // Second negotiation should cause the second track to unmute.
+    test.chain.insertAfter("PC_REMOTE_SET_REMOTE_DESCRIPTION", [
+      function PC_REMOTE_SETUP_ONUNMUTE(test) {
+        haveUnmuteEvent = haveEvent(test.pcRemote._pc.getReceivers()[1].track, "unmute");
+      }
+    ], false, 1);
+
+    test.setMediaConstraints([{video: true}], [{video: true}]);
+    await test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_removeThenAddVideoTrackNoBundle.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_removeThenAddVideoTrackNoBundle.html
new file mode 100644
index 0000000000..dcaf7943e2
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_removeThenAddVideoTrackNoBundle.html
@@ -0,0 +1,89 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: remove then add video track, no bundle"
+  });
+
+  runNetworkTest(async function (options) {
+    // Use fake video here since the native fake device on linux doesn't
+    // change color as needed by checkVideoPlaying() below.
+    await pushPrefs(
+      ['media.video_loopback_dev', ''],
+      ['media.navigator.streams.fake', true]);
+    // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+    if (navigator.userAgent.includes("Android")) {
+      await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                      ["media.webrtc.hw.h264.enabled", false]);
+    }
+
+    options = options || { };
+    options.bundle = false;
+    const test = new PeerConnectionTest(options);
+    const helper = new VideoStreamHelper();
+    var originalTrack;
+    addRenegotiation(test.chain,
+      [
+        function PC_REMOTE_FIND_RECEIVER(test) {
+          is(test.pcRemote._pc.getReceivers().length, 1,
+             "pcRemote should have one receiver");
+          originalTrack = test.pcRemote._pc.getReceivers()[0].track;
+        },
+        function PC_LOCAL_REMOVE_VIDEO_TRACK(test) {
+          // The new track's pipeline will start with a packet count of
+          // 0, but the remote side will keep its old pipeline and packet
+          // count.
+          test.pcLocal.disableRtpCountChecking = true;
+          return test.pcLocal.removeSender(0);
+        },
+        function PC_LOCAL_ADD_VIDEO_TRACK(test) {
+          // Use fake:true here since the native fake device on linux doesn't
+          // change color as needed by checkVideoPlaying() below.
+          return test.pcLocal.getAllUserMediaAndAddStreams([{video: true}]);
+        },
+        function PC_LOCAL_EXPECT_ICE_CHECKING(test) {
+          test.pcLocal.expectIceChecking();
+        },
+        function PC_REMOTE_EXPECT_ICE_CHECKING(test) {
+          test.pcRemote.expectIceChecking();
+        },
+      ],
+      [
+        function PC_REMOTE_CHECK_ADDED_TRACK(test) {
+          is(test.pcRemote._pc.getTransceivers().length, 2,
+              "pcRemote should have two transceivers");
+          const track = test.pcRemote._pc.getTransceivers()[1].receiver.track;
+
+          const vAdded = test.pcRemote.remoteMediaElements.find(
+              elem => elem.id.includes(track.id));
+          return helper.checkVideoPlaying(vAdded);
+        },
+        function PC_REMOTE_CHECK_REMOVED_TRACK(test) {
+          is(test.pcRemote._pc.getTransceivers().length, 2,
+              "pcRemote should have two transceivers");
+          const track = test.pcRemote._pc.getTransceivers()[0].receiver.track;
+
+          const vAdded = test.pcRemote.remoteMediaElements.find(
+              elem => elem.id.includes(track.id));
+          return helper.checkVideoPaused(vAdded, 10, 10, 16, 5000);
+        },
+      ]
+    );
+
+    test.chain.insertAfterEach('PC_LOCAL_CREATE_OFFER',
+                               PC_LOCAL_REMOVE_BUNDLE_FROM_OFFER);
+
+    test.setMediaConstraints([{video: true}], [{video: true}]);
+    await test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_removeVideoTrack.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_removeVideoTrack.html
new file mode 100644
index 0000000000..4c4e7905e1
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_removeVideoTrack.html
@@ -0,0 +1,64 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: remove video track"
+  });
+
+  runNetworkTest(async (options) => {
+    // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+    if (navigator.userAgent.includes("Android")) {
+      await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                      ["media.webrtc.hw.h264.enabled", false]);
+    }
+
+    const test = new PeerConnectionTest(options);
+    let receivedTrack, element;
+    addRenegotiation(test.chain,
+      [
+        function PC_REMOTE_SETUP_HELPER(test) {
+          is(test.pcRemote._pc.getReceivers().length, 1,
+             "pcRemote should have one receiver before renegotiation");
+
+          receivedTrack = test.pcRemote._pc.getReceivers()[0].track;
+          is(receivedTrack.readyState, "live",
+             "The received track should be live");
+
+          element = createMediaElement("video", "pcRemoteReceivedVideo");
+          element.srcObject = new MediaStream([receivedTrack]);
+          return haveEvent(element, "loadeddata");
+        },
+        function PC_LOCAL_REMOVE_VIDEO_TRACK(test) {
+          test.setOfferOptions({ offerToReceiveVideo: true });
+          test.setMediaConstraints([], [{video: true}]);
+          return test.pcLocal.removeSender(0);
+        },
+      ],
+      [
+        function PC_REMOTE_CHECK_FLOW_STOPPED(test) {
+          is(test.pcRemote._pc.getTransceivers().length, 1,
+              "pcRemote should have one transceiver");
+          const track = test.pcRemote._pc.getTransceivers()[0].receiver.track;
+
+          const vAdded = test.pcRemote.remoteMediaElements.find(
+              elem => elem.id.includes(track.id));
+          const helper = new VideoStreamHelper();
+          return helper.checkVideoPaused(vAdded, 10, 10, 16, 5000);
+        },
+      ]
+    );
+
+    test.setMediaConstraints([{video: true}], [{video: true}]);
+    await test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_renderAfterRenegotiation.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_renderAfterRenegotiation.html
new file mode 100644
index 0000000000..c8091d7a9e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_renderAfterRenegotiation.html
@@ -0,0 +1,89 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1273652",
+    title: "Video receiver still renders after renegotiation",
+    visible: true
+  });
+
+  var pc1 = new RTCPeerConnection();
+  var pc2 = new RTCPeerConnection();
+
+  var add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+  pc1.onicecandidate = e => add(pc2, e.candidate, generateErrorCallback());
+  pc2.onicecandidate = e => add(pc1, e.candidate, generateErrorCallback());
+
+  var v1, v2;
+  var delivered = new Promise(resolve => pc2.ontrack = e => {
+    // Test RTCTrackEvent here.
+    ok(e.streams.length, "has streams");
+    ok(e.streams[0].getTrackById(e.track.id), "has track");
+    ok(pc2.getReceivers().some(receiver => receiver == e.receiver), "has receiver");
+    if (e.streams[0].getTracks().length == 1) {
+      // Test RTCTrackEvent required args here.
+      mustThrowWith("RTCTrackEvent wo/required args",
+                    "TypeError", () => new RTCTrackEvent("track", {}));
+      v2.srcObject = e.streams[0];
+      resolve();
+    }
+  });
+
+  runNetworkTest(async () => {
+    // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+    if (navigator.userAgent.includes("Android")) {
+      await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false]);
+      await pushPrefs(["media.webrtc.hw.h264.enabled", false]);
+    }
+
+    v2 = createMediaElement('video', 'v2');
+    is(v2.currentTime, 0, "v2.currentTime is zero at outset");
+
+    const emitter = new VideoFrameEmitter(CaptureStreamTestHelper.prototype.blue,
+                                          CaptureStreamTestHelper.prototype.green,
+                                          16, 16);
+    emitter.start();
+    emitter.stream().getTracks().forEach(t => pc1.addTrack(t, emitter.stream()));
+    let h = emitter.helper();
+
+    let offer = await pc1.createOffer({});
+    await pc1.setLocalDescription(offer);
+    await pc2.setRemoteDescription(pc1.localDescription);
+    // check that createAnswer accepts arg.
+    let answer = await pc2.createAnswer({});
+    await pc2.setLocalDescription(answer);
+    await pc1.setRemoteDescription(pc2.localDescription);
+
+    // re-negotiate to trigger the race condition in the jitter buffer
+    offer = await pc1.createOffer({}); // check that createOffer accepts arg.
+    await pc1.setLocalDescription(offer);
+    await pc2.setRemoteDescription(pc1.localDescription);
+    answer = await pc2.createAnswer({});
+    await pc2.setLocalDescription(answer);
+    await pc1.setRemoteDescription(pc2.localDescription);
+    await delivered;
+
+    // now verify that actually something gets rendered into the remote video
+    // element.
+    await h.pixelMustBecome(v2, h.blue, {
+      threshold: 128,
+      infoString: "pcRemote's video should become blue",
+    });
+    // This will verify that new changes to the canvas propagate through
+    // the peerconnection
+    emitter.colors(h.red, h.green)
+    await h.pixelMustBecome(v2, h.red, {
+      threshold: 128,
+      infoString: "pcRemote's video should become red",
+    });
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceNullTrackThenRenegotiateAudio.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceNullTrackThenRenegotiateAudio.html
new file mode 100644
index 0000000000..2253b87672
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceNullTrackThenRenegotiateAudio.html
@@ -0,0 +1,53 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1763832",
+    title: "Renegotiation (audio): Start with no track and recvonly, then replace and set direction to sendrecv, then renegotiate"
+  });
+
+  runNetworkTest(async () => {
+    await pushPrefs(
+      ['media.audio_loopback_dev', ''],
+      ['media.navigator.streams.fake', true]);
+    const offerer = new RTCPeerConnection();
+    const answerer = new RTCPeerConnection();
+    const transceiverSend = offerer.addTransceiver('audio', {direction: 'recvonly'});
+
+    const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    offerer.onicecandidate = e => add(answerer, e.candidate, generateErrorCallback());
+    answerer.onicecandidate = e => add(offerer, e.candidate, generateErrorCallback());
+
+    await offerer.setLocalDescription();
+    await answerer.setRemoteDescription(offerer.localDescription);
+    await answerer.setLocalDescription();
+    await offerer.setRemoteDescription(answerer.localDescription);
+
+    // add audio with replaceTrack, set send bit, and renegotiate
+    const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+    const [track] = stream.getAudioTracks();
+    transceiverSend.sender.replaceTrack(track);
+    transceiverSend.direction = "sendrecv";
+    const remoteStreamAvailable = new Promise(r => {
+      answerer.ontrack = ({track}) => r(new MediaStream([track]));
+    });
+
+    await offerer.setLocalDescription();
+    await answerer.setRemoteDescription(offerer.localDescription);
+    await answerer.setLocalDescription();
+    await offerer.setRemoteDescription(answerer.localDescription);
+
+    const remoteStream = await remoteStreamAvailable;
+    const h = new AudioStreamHelper();
+    await h.checkAudioFlowing(remoteStream);
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceNullTrackThenRenegotiateVideo.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceNullTrackThenRenegotiateVideo.html
new file mode 100644
index 0000000000..d7bd6d8a37
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceNullTrackThenRenegotiateVideo.html
@@ -0,0 +1,63 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="simulcast.js"></script></head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1763832",
+    title: "Renegotiation (video): Start with no track and recvonly, then replace and set direction to sendrecv, then renegotiate"
+  });
+
+  runNetworkTest(async () => {
+    await pushPrefs(
+      ['media.video_loopback_dev', ''],
+      ['media.navigator.streams.fake', true]);
+    // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+    if (navigator.userAgent.includes("Android")) {
+      await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                      ["media.webrtc.hw.h264.enabled", false]);
+    }
+
+    const offerer = new RTCPeerConnection();
+    const answerer = new RTCPeerConnection();
+    const transceiverSend = offerer.addTransceiver('video', {direction: 'recvonly'});
+
+    const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    offerer.onicecandidate = e => add(answerer, e.candidate, generateErrorCallback());
+    answerer.onicecandidate = e => add(offerer, e.candidate, generateErrorCallback());
+
+    await offerer.setLocalDescription();
+    await answerer.setRemoteDescription(offerer.localDescription);
+    await answerer.setLocalDescription();
+    await offerer.setRemoteDescription(answerer.localDescription);
+
+    // add video with replaceTrack, set send bit, and renegotiate
+    const stream = await navigator.mediaDevices.getUserMedia({video: true});
+    const [track] = stream.getVideoTracks();
+    transceiverSend.sender.replaceTrack(track);
+    transceiverSend.direction = "sendrecv";
+    const metadataToBeLoaded = [];
+    answerer.ontrack = (e) => {
+      metadataToBeLoaded.push(getPlaybackWithLoadedMetadata(e.track));
+    };
+
+    await offerer.setLocalDescription();
+    await answerer.setRemoteDescription(offerer.localDescription);
+    await answerer.setLocalDescription();
+    await offerer.setRemoteDescription(answerer.localDescription);
+
+    const elems = await Promise.all(metadataToBeLoaded);
+    is(elems.length, 1, "Should have one video element");
+
+    const helper = new VideoStreamHelper();
+    await helper.checkVideoPlaying(elems[0]);
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceTrack.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceTrack.html
new file mode 100644
index 0000000000..9befc5c564
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceTrack.html
@@ -0,0 +1,187 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1032839",
+    title: "Replace video and audio (with WebAudio) tracks",
+    visible: true
+  });
+
+  function allLocalStreamsHaveSender(pc) {
+    return pc.getLocalStreams()
+      .every(s => s.getTracks()        // Every local stream,
+        .some(t => pc.getSenders()     // should have some track,
+          .some(sn => sn.track == t))) // that's being sent over |pc|.
+  }
+
+  function allRemoteStreamsHaveReceiver(pc) {
+    return pc.getRemoteStreams()
+      .every(s => s.getTracks()        // Every remote stream,
+        .some(t => pc.getReceivers()   // should have some track,
+          .some(sn => sn.track == t))) // that's being received over |pc|.
+  }
+
+  function replacetest(wrapper) {
+    var pc = wrapper._pc;
+    var oldSenderCount = pc.getSenders().length;
+    var sender = pc.getSenders().find(sn => sn.track.kind == "video");
+    var oldTrack = sender.track;
+    ok(sender, "We have a sender for video");
+    ok(allLocalStreamsHaveSender(pc),
+       "Shouldn't have any local streams without a corresponding sender");
+    ok(allRemoteStreamsHaveReceiver(pc),
+       "Shouldn't have any remote streams without a corresponding receiver");
+
+    var newTrack;
+    var audiotrack;
+    return getUserMedia({video:true, audio:true})
+      .then(newStream => {
+        window.grip = newStream;
+        newTrack = newStream.getVideoTracks()[0];
+        audiotrack = newStream.getAudioTracks()[0];
+        isnot(newTrack, sender.track, "replacing with a different track");
+        ok(!pc.getLocalStreams().some(s => s == newStream),
+           "from a different stream");
+        // Use wrapper function, since it updates expected tracks
+        return wrapper.senderReplaceTrack(sender, newTrack, newStream);
+      })
+      .then(() => {
+        is(pc.getSenders().length, oldSenderCount, "same sender count");
+        is(sender.track, newTrack, "sender.track has been replaced");
+        ok(!pc.getSenders().map(sn => sn.track).some(t => t == oldTrack),
+           "old track not among senders");
+        // Spec does not say we add this new track to any stream
+        ok(!pc.getLocalStreams().some(s => s.getTracks()
+                                           .some(t => t == sender.track)),
+           "track does not exist among pc's local streams");
+        return sender.replaceTrack(audiotrack)
+          .then(() => ok(false, "replacing with different kind should fail"),
+                e => is(e.name, "TypeError",
+                        "replacing with different kind should fail"));
+      });
+  }
+
+  runNetworkTest(function () {
+    test = new PeerConnectionTest();
+    test.audioCtx = new AudioContext();
+    test.setMediaConstraints([{video: true, audio: true}], [{video: true}]);
+    test.chain.removeAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW");
+
+    // Test replaceTrack on pcRemote separately since it's video only.
+    test.chain.append([
+      function PC_REMOTE_VIDEOONLY_REPLACE_VIDEOTRACK(test) {
+        return replacetest(test.pcRemote);
+      },
+      function PC_LOCAL_NEW_VIDEOTRACK_WAIT_FOR_MEDIA_FLOW(test) {
+        return test.pcLocal.waitForMediaFlow();
+      }
+    ]);
+
+    // Replace video twice on pcLocal to make sure it still works
+    // (does audio twice too, but hey)
+    test.chain.append([
+      function PC_LOCAL_AUDIOVIDEO_REPLACE_VIDEOTRACK_1(test) {
+        return replacetest(test.pcLocal);
+      },
+      function PC_REMOTE_NEW_VIDEOTRACK_WAIT_FOR_MEDIA_FLOW_1(test) {
+        return test.pcRemote.waitForMediaFlow();
+      },
+      function PC_LOCAL_AUDIOVIDEO_REPLACE_VIDEOTRACK_2(test) {
+        return replacetest(test.pcLocal);
+      },
+      function PC_REMOTE_NEW_VIDEOTRACK_WAIT_FOR_MEDIA_FLOW_2(test) {
+        return test.pcRemote.waitForMediaFlow();
+      }
+    ]);
+
+    test.chain.append([
+      function PC_LOCAL_AUDIOVIDEO_REPLACE_VIDEOTRACK_WITHSAME(test) {
+        var pc = test.pcLocal._pc;
+        var sender = pc.getSenders().find(sn => sn.track.kind == "video");
+        ok(sender, "should still have a sender of video");
+        return sender.replaceTrack(sender.track)
+          .then(() => ok(true, "replacing with itself should succeed"));
+      },
+      function PC_REMOTE_NEW_SAME_VIDEOTRACK_WAIT_FOR_MEDIA_FLOW(test) {
+        return test.pcRemote.waitForMediaFlow();
+      }
+    ]);
+
+    // Replace the gUM audio track on pcLocal with a WebAudio track.
+    test.chain.append([
+      function PC_LOCAL_AUDIOVIDEO_REPLACE_AUDIOTRACK_WEBAUDIO(test) {
+        var pc = test.pcLocal._pc;
+        var sender = pc.getSenders().find(sn => sn.track.kind == "audio");
+        ok(sender, "track has a sender");
+        var oldSenderCount = pc.getSenders().length;
+        var oldTrack = sender.track;
+
+        var sourceNode = test.audioCtx.createOscillator();
+        sourceNode.type = 'sine';
+        // We need a frequency not too close to the fake audio track
+        // (440Hz for loopback devices, 1kHz for fake tracks).
+        sourceNode.frequency.value = 2000;
+        sourceNode.start();
+
+        var destNode = test.audioCtx.createMediaStreamDestination();
+        sourceNode.connect(destNode);
+        var newTrack = destNode.stream.getAudioTracks()[0];
+
+        return test.pcLocal.senderReplaceTrack(
+            sender, newTrack, destNode.stream)
+          .then(() => {
+            is(pc.getSenders().length, oldSenderCount, "same sender count");
+            ok(!pc.getSenders().some(sn => sn.track == oldTrack),
+               "Replaced track should be removed from senders");
+            // TODO: Should PC remove local streams when there are no senders
+            // associated with it? getLocalStreams() isn't in the spec anymore,
+            // so I guess it is pretty arbitrary?
+            is(sender.track, newTrack, "sender.track has been replaced");
+            // Spec does not say we add this new track to any stream
+            ok(!pc.getLocalStreams().some(s => s.getTracks()
+                                               .some(t => t == sender.track)),
+               "track exists among pc's local streams");
+          });
+      }
+    ]);
+    test.chain.append([
+      function PC_LOCAL_CHECK_WEBAUDIO_FLOW_PRESENT(test) {
+        return test.pcRemote.checkReceivingToneFrom(test.audioCtx, test.pcLocal);
+      }
+    ]);
+    test.chain.append([
+      function PC_LOCAL_INVALID_ADD_VIDEOTRACKS(test) {
+        let videoTransceivers = test.pcLocal._pc.getTransceivers()
+          .filter(transceiver => {
+            return !transceiver.stopped &&
+                   transceiver.receiver.track.kind == "video" &&
+                   transceiver.sender.track;
+          });
+
+        ok(videoTransceivers.length,
+           "There is at least one non-stopped video transceiver with a track.");
+
+        videoTransceivers.forEach(transceiver => {
+            var stream = test.pcLocal._pc.getLocalStreams()[0];;
+            var track = transceiver.sender.track;
+            try {
+              test.pcLocal._pc.addTrack(track, stream);
+              ok(false, "addTrack existing track should fail");
+            } catch (e) {
+              is(e.name, "InvalidAccessError",
+                 "addTrack existing track should fail");
+            }
+          });
+      }
+    ]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceTrack_camera.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceTrack_camera.html
new file mode 100644
index 0000000000..356517e79f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceTrack_camera.html
@@ -0,0 +1,48 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+
+<video id="video" width="160" height="120" autoplay></video>
+
+<script type="application/javascript">
+  createHTML({
+    bug: "1709481",
+    title: "replaceTrack (null -> camera) test",
+    visible: true
+  });
+
+  runNetworkTest(async () => {
+    // Make sure we use the fake video device, and not loopback
+    await pushPrefs(
+      ['media.video_loopback_dev', ''],
+      ['media.navigator.streams.fake', true]);
+    const pc1 = new RTCPeerConnection(), pc2 = new RTCPeerConnection();
+    pc1.onicecandidate = e => pc2.addIceCandidate(e.candidate);
+    pc2.onicecandidate = e => pc1.addIceCandidate(e.candidate);
+    pc2.ontrack = ({track}) => video.srcObject = new MediaStream([track]);
+    pc1.addTransceiver("audio");
+    const tc1 = pc1.addTransceiver("video");
+    const stream = await navigator.mediaDevices.getUserMedia({video: true});
+    const [track] = stream.getVideoTracks();
+    await pc1.setLocalDescription();
+    await pc2.setRemoteDescription(pc1.localDescription);
+    await pc2.setLocalDescription();
+    await pc1.setRemoteDescription(pc2.localDescription);
+    await wait(100);
+    await tc1.sender.replaceTrack(track);
+    const h = new VideoStreamHelper();
+    await h.checkVideoPlaying(video);
+    pc1.close();
+    pc2.close();
+    await SpecialPowers.popPrefEnv();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceTrack_disabled.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceTrack_disabled.html
new file mode 100644
index 0000000000..11b2762d96
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceTrack_disabled.html
@@ -0,0 +1,60 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+<script src="pc.js"></script>
+<script src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script>
+createHTML({
+  bug: "1576771",
+  title: "Replace a disabled video track with an enabled one",
+  visible: true,
+});
+
+runNetworkTest(() => {
+  const helper = new CaptureStreamTestHelper2D(240, 160);
+  const emitter = new VideoFrameEmitter(helper.green, helper.green, 240, 160);
+  const test = new PeerConnectionTest();
+  test.setMediaConstraints([{video: true}], []);
+  test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW", [
+    function PC_LOCAL_DISABLE_SENDTRACK(test) {
+      test.pcLocal._pc.getSenders()[0].track.enabled = false;
+    },
+    function PC_REMOTE_WAIT_FOR_BLACK(test) {
+      return helper.pixelMustBecome(
+        test.pcRemote.remoteMediaElements[0], helper.black, {
+          threshold: 128,
+          infoString: "Remote disabled track becomes black",
+          cancel: wait(10000).then(
+            () => new Error("Timeout waiting for black"))});
+    },
+    function PC_LOCAL_REPLACETRACK_WITH_ENABLED_TRACK(test) {
+      emitter.start();
+      test.pcLocal._pc.getSenders()[0].replaceTrack(
+        emitter.stream().getTracks()[0]);
+    },
+  ]);
+  test.chain.append([
+    function PC_REMOTE_WAIT_FOR_GREEN(test) {
+      return helper.pixelMustBecome(
+        test.pcRemote.remoteMediaElements[0], helper.green, {
+          threshold: 128,
+          infoString: "Remote disabled track becomes green",
+          cancel: wait(10000).then(
+            () => new Error("Timeout waiting for green"))});
+    },
+    function CLEANUP(test) {
+      emitter.stop();
+      for (const track of emitter.stream().getTracks()) {
+        track.stop();
+      }
+    },
+  ]);
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceTrack_microphone.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceTrack_microphone.html
new file mode 100644
index 0000000000..5886caf6a4
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceTrack_microphone.html
@@ -0,0 +1,46 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+
+<script type="application/javascript">
+  createHTML({
+    bug: "1709481",
+    title: "replaceTrack (null -> microphone) test",
+    visible: true
+  });
+
+  runNetworkTest(async () => {
+    // Make sure we use the fake audio device, and not loopback
+    await pushPrefs(
+      ['media.audio_loopback_dev', ''],
+      ['media.navigator.streams.fake', true]);
+    const pc1 = new RTCPeerConnection(), pc2 = new RTCPeerConnection();
+    pc1.onicecandidate = e => pc2.addIceCandidate(e.candidate);
+    pc2.onicecandidate = e => pc1.addIceCandidate(e.candidate);
+    let remoteStream;
+    pc2.ontrack = ({track}) => remoteStream = new MediaStream([track]);
+    const tc1 = pc1.addTransceiver("audio");
+    const stream = await navigator.mediaDevices.getUserMedia({audio: true});
+    const [track] = stream.getAudioTracks();
+    await pc1.setLocalDescription();
+    await pc2.setRemoteDescription(pc1.localDescription);
+    await pc2.setLocalDescription();
+    await pc1.setRemoteDescription(pc2.localDescription);
+    await wait(100);
+    await tc1.sender.replaceTrack(track);
+    const h = new AudioStreamHelper();
+    await h.checkAudioFlowing(remoteStream);
+    pc1.close();
+    pc2.close();
+    await SpecialPowers.popPrefEnv();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceVideoThenRenegotiate.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceVideoThenRenegotiate.html
new file mode 100644
index 0000000000..070cb42fcb
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_replaceVideoThenRenegotiate.html
@@ -0,0 +1,74 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1017888",
+    title: "Renegotiation: replaceTrack followed by adding a second video stream"
+  });
+
+  runNetworkTest(async (options) => {
+    await pushPrefs(['media.peerconnection.video.min_bitrate_estimate', 180*1000]);
+    // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+    if (navigator.userAgent.includes("Android")) {
+      await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                      ["media.webrtc.hw.h264.enabled", false]);
+    }
+
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{video:true}], [{video:true}]);
+    const helper = new VideoStreamHelper();
+    const emitter1 = new VideoFrameEmitter(CaptureStreamTestHelper.prototype.red,
+                                           CaptureStreamTestHelper.prototype.green);
+    const emitter2 = new VideoFrameEmitter(CaptureStreamTestHelper.prototype.blue,
+                                           CaptureStreamTestHelper.prototype.grey);
+    test.chain.replace("PC_LOCAL_GUM", [
+      function PC_LOCAL_ADDTRACK(test) {
+        test.pcLocal.attachLocalStream(emitter1.stream());
+        emitter1.start();
+      },
+    ]);
+    addRenegotiation(test.chain,
+      [
+        function PC_LOCAL_REPLACE_VIDEO_TRACK_THEN_ADD_SECOND_STREAM(test) {
+          emitter1.stop();
+          emitter2.start();
+          const newstream = emitter2.stream();
+          const newtrack = newstream.getVideoTracks()[0];
+          var sender = test.pcLocal._pc.getSenders()[0];
+          return test.pcLocal.senderReplaceTrack(sender, newtrack, newstream)
+            .then(() => {
+              test.setMediaConstraints([{video: true}, {video: true}],
+                                       [{video: true}]);
+            });
+        },
+      ],
+      [
+        function PC_REMOTE_CHECK_ORIGINAL_TRACK_NOT_ENDED(test) {
+          is(test.pcRemote._pc.getTransceivers().length, 1,
+              "pcRemote should have one transceiver");
+          const track = test.pcRemote._pc.getTransceivers()[0].receiver.track;
+
+          const vremote = test.pcRemote.remoteMediaElements.find(
+              elem => elem.id.includes(track.id));
+          if (!vremote) {
+            return Promise.reject(new Error("Couldn't find video element"));
+          }
+          ok(!vremote.ended, "Original track should not have ended after renegotiation (replaceTrack is not signalled!)");
+          return helper.checkVideoPlaying(vremote);
+        }
+      ]
+    );
+
+    await test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIce.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIce.html
new file mode 100644
index 0000000000..d94bb084b7
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIce.html
@@ -0,0 +1,41 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "906986",
+    title: "Renegotiation: restart ice"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+
+    addRenegotiation(test.chain,
+      [
+        // causes a full, normal ice restart
+        function PC_LOCAL_SET_OFFER_OPTION(test) {
+          test.setOfferOptions({ iceRestart: true });
+        },
+        function PC_LOCAL_EXPECT_ICE_CHECKING(test) {
+          test.pcLocal.expectIceChecking();
+        },
+        function PC_REMOTE_EXPECT_ICE_CHECKING(test) {
+          test.pcRemote.expectIceChecking();
+        }
+      ]
+    );
+
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             [{audio: true}, {video: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceBadAnswer.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceBadAnswer.html
new file mode 100644
index 0000000000..b71001b0db
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceBadAnswer.html
@@ -0,0 +1,58 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1413709",
+    title: "Renegotiation: bad answer ICE credentials"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+
+    addRenegotiation(test.chain,
+      [
+        function PC_LOCAL_ADD_SECOND_STREAM(test) {
+          test.setMediaConstraints([{audio: true}],
+                                   []);
+          return test.pcLocal.getAllUserMedia([{audio: true}]);
+        },
+      ]
+    );
+
+    // If the offerer hasn't indicated ICE restart, then an answer
+    // arriving during renegotiation that has modified ICE credentials
+    // should cause an error
+    test.chain.replaceAfter("PC_LOCAL_GET_ANSWER",
+      [
+        function PC_LOCAL_REWRITE_REMOTE_SDP_ICE_CREDS(test) {
+          test._remote_answer.sdp =
+            test._remote_answer.sdp.replace(/a=ice-pwd:.*\r\n/g,
+                                            "a=ice-pwd:bad-pwd\r\n")
+                                   .replace(/a=ice-ufrag:.*\r\n/g,
+                                            "a=ice-ufrag:bad-ufrag\r\n");
+        },
+
+        function PC_LOCAL_EXPECT_SET_REMOTE_DESCRIPTION_FAIL(test) {
+          return test.setRemoteDescription(test.pcLocal,
+                                           test._remote_answer,
+                                           STABLE)
+           .then(() => ok(false, "setRemoteDescription must fail"),
+                 e => is(e.name, "InvalidAccessError",
+                         "setRemoteDescription must fail and did"));
+         }
+      ], 1 // replace after the second PC_LOCAL_GET_ANSWER
+    );
+
+    test.setMediaConstraints([{audio: true}], []);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceLocalAndRemoteRollback.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceLocalAndRemoteRollback.html
new file mode 100644
index 0000000000..6bbf9440fc
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceLocalAndRemoteRollback.html
@@ -0,0 +1,82 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "906986",
+    title: "Renegotiation: restart ice, local and remote rollback"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+
+    addRenegotiation(test.chain,
+      [
+        async function PC_LOCAL_SETUP_ICE_HANDLER(test) {
+          await test.pcLocal.endOfTrickleIce;
+          test.pcLocal.setupIceCandidateHandler(test);
+        },
+
+        // causes a full, normal ice restart
+        function PC_LOCAL_SET_OFFER_OPTION(test) {
+          test.setOfferOptions({ iceRestart: true });
+        }
+      ]
+    );
+
+    test.chain.replaceAfter('PC_REMOTE_CREATE_ANSWER',
+      [
+        function PC_LOCAL_EXPECT_ICE_CONNECTED(test) {
+          test.pcLocal.iceCheckingIceRollbackExpected = true;
+        },
+
+        function PC_REMOTE_ROLLBACK(test) {
+          return test.setRemoteDescription(test.pcRemote, { type: "rollback" },
+                                           STABLE);
+        },
+
+        async function PC_LOCAL_ROLLBACK(test) {
+          await test.pcLocal.endOfTrickleIce;
+          // We haven't negotiated the new stream yet.
+          test.pcLocal.expectNegotiationNeeded();
+          return test.setLocalDescription(
+              test.pcLocal,
+              new RTCSessionDescription({ type: "rollback", sdp: ""}),
+              STABLE);
+        },
+
+        // Rolling back should shut down gathering for the offerer,
+        // but because the answerer never set a local description, no ICE
+        // gathering has happened yet, so there's no changes to ICE gathering
+        // state
+        function PC_LOCAL_WAIT_FOR_END_OF_TRICKLE(test) {
+          return test.pcLocal.endOfTrickleIce;
+        },
+
+        function PC_LOCAL_EXPECT_ICE_CHECKING(test) {
+          test.pcLocal.expectIceChecking();
+        },
+        function PC_REMOTE_EXPECT_ICE_CHECKING(test) {
+          test.pcRemote.expectIceChecking();
+        }
+      ],
+      1 // Replaces after second PC_REMOTE_CREATE_ANSWER
+    );
+    test.chain.append(commandsPeerConnectionOfferAnswer);
+
+    // for now, only use one stream, because rollback doesn't seem to
+    // like multiple streams.  See bug 1259465.
+    test.setMediaConstraints([{audio: true}],
+                             [{audio: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceLocalAndRemoteRollbackNoSubsequentRestart.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceLocalAndRemoteRollbackNoSubsequentRestart.html
new file mode 100644
index 0000000000..37b0fc68fc
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceLocalAndRemoteRollbackNoSubsequentRestart.html
@@ -0,0 +1,77 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "906986",
+    title: "Renegotiation: restart ice, local and remote rollback, without a subsequent ICE restart"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+
+    addRenegotiation(test.chain,
+      [
+        function PC_LOCAL_SETUP_ICE_HANDLER(test) {
+          test.pcLocal.setupIceCandidateHandler(test);
+        },
+        function PC_REMOTE_SETUP_ICE_HANDLER(test) {
+          test.pcRemote.setupIceCandidateHandler(test);
+        },
+
+        // causes a full, normal ice restart
+        function PC_LOCAL_SET_OFFER_OPTION(test) {
+          test.setOfferOptions({ iceRestart: true });
+        }
+      ]
+    );
+
+    test.chain.replaceAfter('PC_REMOTE_CREATE_ANSWER',
+      [
+        function PC_REMOTE_ROLLBACK(test) {
+          return test.setRemoteDescription(test.pcRemote, { type: "rollback" },
+                                           STABLE);
+        },
+
+        async function PC_LOCAL_ROLLBACK(test) {
+          await test.pcLocal.endOfTrickleIce;
+          // We haven't negotiated the new stream yet.
+          test.pcLocal.expectNegotiationNeeded();
+          return test.setLocalDescription(
+              test.pcLocal,
+              new RTCSessionDescription({ type: "rollback", sdp: ""}),
+              STABLE);
+        },
+
+        // Rolling back should shut down gathering for the offerer,
+        // but because the answerer never set a local description, no ICE
+        // gathering has happened yet, so there's no changes to ICE gathering
+        // state
+        function PC_LOCAL_WAIT_FOR_END_OF_TRICKLE(test) {
+          return test.pcLocal.endOfTrickleIce;
+        },
+
+        function PC_LOCAL_SET_OFFER_OPTION(test) {
+          test.setOfferOptions({ iceRestart: false });
+        }
+      ],
+      1 // Replaces after second PC_REMOTE_CREATE_ANSWER
+    );
+    test.chain.append(commandsPeerConnectionOfferAnswer);
+
+    // for now, only use one stream, because rollback doesn't seem to
+    // like multiple streams.  See bug 1259465.
+    test.setMediaConstraints([{audio: true}],
+                             [{audio: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceLocalRollback.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceLocalRollback.html
new file mode 100644
index 0000000000..f5f9a1f220
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceLocalRollback.html
@@ -0,0 +1,76 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "906986",
+    title: "Renegotiation: restart ice, local rollback"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+
+    addRenegotiation(test.chain,
+      [
+        // causes a full, normal ice restart
+        function PC_LOCAL_SET_OFFER_OPTION(test) {
+          test.setOfferOptions({ iceRestart: true });
+        },
+        // causes an ice restart and then rolls it back
+        // (does not result in sending an offer)
+        function PC_LOCAL_SETUP_ICE_HANDLER(test) {
+          test.pcLocal.setupIceCandidateHandler(test, () => {});
+        },
+        function PC_LOCAL_CREATE_AND_SET_OFFER(test) {
+          return test.createOffer(test.pcLocal).then(offer => {
+            return test.setLocalDescription(test.pcLocal,
+                                            offer,
+                                            HAVE_LOCAL_OFFER);
+          });
+        },
+        function PC_LOCAL_EXPECT_ICE_CONNECTED(test) {
+          test.pcLocal.iceCheckingIceRollbackExpected = true;
+        },
+        function PC_LOCAL_WAIT_FOR_GATHERING(test) {
+          return new Promise(r => {
+            test.pcLocal._pc.addEventListener("icegatheringstatechange", () => {
+              if (test.pcLocal._pc.iceGatheringState == "gathering") {
+                r();
+              }
+            });
+          });
+        },
+        function PC_LOCAL_ROLLBACK(test) {
+          return test.setLocalDescription(test.pcLocal,
+                                          { type: "rollback", sdp: ""},
+                                          STABLE);
+        },
+        // Rolling back should shut down gathering
+        function PC_LOCAL_WAIT_FOR_END_OF_TRICKLE(test) {
+          return test.pcLocal.endOfTrickleIce;
+        },
+        function PC_LOCAL_EXPECT_ICE_CHECKING(test) {
+          test.pcLocal.expectIceChecking();
+        },
+        function PC_REMOTE_EXPECT_ICE_CHECKING(test) {
+          test.pcRemote.expectIceChecking();
+        }
+      ]
+    );
+
+    // for now, only use one stream, because rollback doesn't seem to
+    // like multiple streams.  See bug 1259465.
+    test.setMediaConstraints([{audio: true}],
+                             [{audio: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceLocalRollbackNoSubsequentRestart.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceLocalRollbackNoSubsequentRestart.html
new file mode 100644
index 0000000000..8e27864aae
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceLocalRollbackNoSubsequentRestart.html
@@ -0,0 +1,60 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "906986",
+    title: "Renegotiation: restart ice, local rollback, then renegotiation without ICE restart"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+
+    addRenegotiation(test.chain,
+      [
+        // causes a full, normal ice restart
+        function PC_LOCAL_SET_OFFER_OPTION(test) {
+          test.setOfferOptions({ iceRestart: true });
+        },
+        // causes an ice restart and then rolls it back
+        // (does not result in sending an offer)
+        function PC_LOCAL_SETUP_ICE_HANDLER(test) {
+          test.pcLocal.setupIceCandidateHandler(test, () => {});
+        },
+        function PC_LOCAL_CREATE_AND_SET_OFFER(test) {
+          return test.createOffer(test.pcLocal).then(offer => {
+            return test.setLocalDescription(test.pcLocal,
+                                            offer,
+                                            HAVE_LOCAL_OFFER);
+          });
+        },
+        function PC_LOCAL_WAIT_FOR_END_OF_TRICKLE(test) {
+          return test.pcLocal.endOfTrickleIce;
+        },
+        function PC_LOCAL_ROLLBACK(test) {
+          return test.setLocalDescription(test.pcLocal,
+                                          { type: "rollback", sdp: ""},
+                                          STABLE);
+        },
+        function PC_LOCAL_SET_OFFER_OPTION(test) {
+          test.setOfferOptions({ iceRestart: false });
+        }
+      ]
+    );
+
+    // for now, only use one stream, because rollback doesn't seem to
+    // like multiple streams.  See bug 1259465.
+    test.setMediaConstraints([{audio: true}],
+                             [{audio: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceNoBundle.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceNoBundle.html
new file mode 100644
index 0000000000..134fa97cc0
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceNoBundle.html
@@ -0,0 +1,43 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "906986",
+    title: "Renegotiation: restart ice, no bundle"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    options = options || { };
+    options.bundle = false;
+    test = new PeerConnectionTest(options);
+
+    addRenegotiation(test.chain,
+      [
+        // causes a full, normal ice restart
+        function PC_LOCAL_SET_OFFER_OPTION(test) {
+          test.setOfferOptions({ iceRestart: true });
+        },
+        function PC_LOCAL_EXPECT_ICE_CHECKING(test) {
+          test.pcLocal.expectIceChecking();
+        },
+        function PC_REMOTE_EXPECT_ICE_CHECKING(test) {
+          test.pcRemote.expectIceChecking();
+        }
+      ]
+    );
+
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             [{audio: true}, {video: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceNoBundleNoRtcpMux.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceNoBundleNoRtcpMux.html
new file mode 100644
index 0000000000..06a3a3c980
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceNoBundleNoRtcpMux.html
@@ -0,0 +1,44 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "906986",
+    title: "Renegotiation: restart ice, no bundle and disabled RTCP-Mux"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    options = options || { };
+    options.bundle = false;
+    options.rtcpmux = false;
+    test = new PeerConnectionTest(options);
+
+    addRenegotiation(test.chain,
+      [
+        // causes a full, normal ice restart
+        function PC_LOCAL_SET_OFFER_OPTION(test) {
+          test.setOfferOptions({ iceRestart: true });
+        },
+        function PC_LOCAL_EXPECT_ICE_CHECKING(test) {
+          test.pcLocal.expectIceChecking();
+        },
+        function PC_REMOTE_EXPECT_ICE_CHECKING(test) {
+          test.pcRemote.expectIceChecking();
+        }
+      ]
+    );
+
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             [{audio: true}, {video: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceNoRtcpMux.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceNoRtcpMux.html
new file mode 100644
index 0000000000..5d4780211a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_restartIceNoRtcpMux.html
@@ -0,0 +1,43 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "906986",
+    title: "Renegotiation: restart ice, with disabled RTCP-Mux"
+  });
+
+  var test;
+  runNetworkTest(function (options) {
+    options = options || { };
+    options.rtcpmux = false;
+    test = new PeerConnectionTest(options);
+
+    addRenegotiation(test.chain,
+      [
+        // causes a full, normal ice restart
+        function PC_LOCAL_SET_OFFER_OPTION(test) {
+          test.setOfferOptions({ iceRestart: true });
+        },
+        function PC_LOCAL_EXPECT_ICE_CHECKING(test) {
+          test.pcLocal.expectIceChecking();
+        },
+        function PC_REMOTE_EXPECT_ICE_CHECKING(test) {
+          test.pcRemote.expectIceChecking();
+        }
+      ]
+    );
+
+    test.setMediaConstraints([{audio: true}, {video: true}],
+                             [{audio: true}, {video: true}]);
+    return test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_restrictBandwidthTargetBitrate.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_restrictBandwidthTargetBitrate.html
new file mode 100644
index 0000000000..ff9fb1fc22
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_restrictBandwidthTargetBitrate.html
@@ -0,0 +1,29 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "Bug 1404250",
+    title: "Extremely bitrate restricted video-only peer connection"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{video: true}], [{video: true}]);
+    test.chain.insertAfter('PC_REMOTE_GET_OFFER', [
+      function PC_REMOTE_ADD_TIAS(test) {
+        test._local_offer.sdp = sdputils.addTiasBps(
+          test._local_offer.sdp, 25000);
+        info("Offer with TIAS: " + JSON.stringify(test._local_offer));
+      }
+    ]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_restrictBandwidthWithTias.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_restrictBandwidthWithTias.html
new file mode 100644
index 0000000000..85b831e9a8
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_restrictBandwidthWithTias.html
@@ -0,0 +1,30 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1359854",
+    title: "500Kb restricted video-only peer connection"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{video: true}], [{video: true}]);
+    test.chain.insertAfter('PC_REMOTE_GET_OFFER', [
+      function PC_REMOTE_ADD_TIAS(test) {
+        test._local_offer.sdp = sdputils.addTiasBps(
+          test._local_offer.sdp, 250000);
+        info("Offer with TIAS: " + JSON.stringify(test._local_offer));
+      }
+    ]);
+    // TODO it would be nice to verify the used bandwidth
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_rtcp_rsize.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_rtcp_rsize.html
new file mode 100644
index 0000000000..25270984ea
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_rtcp_rsize.html
@@ -0,0 +1,81 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+  <script type="application/javascript" src="sdpUtils.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1279153",
+    title: "rtcp-rsize",
+    visible: true
+  });
+
+  // 0) Use webrtc-sdp
+  // 1) ADD RTCP-RISZE to all video m-sections
+  // 2) Check for RTCP-RSIZE in ANSWER
+  // 3) Wait for media to flow
+  // 4) Wait for RTCP stats
+
+  runNetworkTest(async function (options) {
+    const test = new PeerConnectionTest(options);
+
+    let mSectionsAltered = 0;
+
+    test.chain.insertAfter("PC_LOCAL_CREATE_OFFER", [
+      function PC_LOCAL_ADD_RTCP_RSIZE(test) {
+        const lines = test.originalOffer.sdp.split("\r\n");
+        info(`SDP before rtcp-rsize: ${lines.join('\n')}`);
+        // Insert an rtcp-rsize for each m section
+        const rsizeAdded = lines.flatMap(line => {
+          if (line.startsWith("m=video")) {
+            mSectionsAltered = mSectionsAltered + 1;
+            return [line, "a=rtcp-rsize"];
+          }
+          return [line];
+        });
+        test.originalOffer.sdp = rsizeAdded.join("\r\n");
+        info(`SDP with rtcp-rsize: ${rsizeAdded.join("\n")}`);
+        is(mSectionsAltered, 1, "We only altered 1 msection")
+    }]);
+
+    // Check that the rtcp-rsize makes into the answer
+    test.chain.insertAfter("PC_LOCAL_SET_REMOTE_DESCRIPTION", [
+      function PC_LOCAL_CHECK_RTCP_RSIZE(test) {
+        const msections = sdputils.getMSections(test.pcLocal._pc.currentRemoteDescription.sdp);
+        var alteredMSectionsFound = 0;
+        for (msection of msections) {
+          if (msection.startsWith("m=video")) {
+            ok(msection.includes("\r\na=rtcp-rsize\r\n"), "video m-section includes RTCP-RSIZE");
+            alteredMSectionsFound = alteredMSectionsFound + 1;
+          } else {
+            ok(!msection.includes("\r\na=rtcp-rsize\r\n"), "audio m-section does not include RTCP-RSIZE");
+          }
+        }
+        is(alteredMSectionsFound, mSectionsAltered, "correct number of msections found");
+      }
+    ]);
+
+    // Make sure that we are still getting RTCP stats
+    test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW",
+      async function PC_LOCAL_AND_REMOTE_CHECK_FOR_RTCP_STATS(test) {
+        await Promise.all([
+          waitForSyncedRtcp(test.pcLocal._pc),
+          waitForSyncedRtcp(test.pcRemote._pc),
+        ]);
+        // The work is done by waitForSyncedRtcp which will throw if
+        // RTCP stats are not received.
+        info("RTCP stats received!");
+      },
+    );
+    test.setMediaConstraints([{audio: true}, {video: true}], []);
+    await test.run();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_scaleResolution.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_scaleResolution.html
new file mode 100644
index 0000000000..4be6873fa6
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_scaleResolution.html
@@ -0,0 +1,119 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1244913",
+    title: "Scale resolution down on a PeerConnection",
+    visible: true
+  });
+
+  async function checkForH264Support() {
+    const pc = new RTCPeerConnection();
+    const offer = await pc.createOffer({offerToReceiveVideo: true});
+    return offer.sdp.match(/a=rtpmap:[1-9][0-9]* H264/);
+  }
+
+  let resolutionAlignment = 1;
+
+  var mustRejectWith = (msg, reason, f) =>
+    f().then(() => ok(false, msg),
+             e => is(e.name, reason, msg));
+
+  async function testScale(codec) {
+    var pc1 = new RTCPeerConnection();
+    var pc2 = new RTCPeerConnection();
+
+    var add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    pc1.onicecandidate = e => add(pc2, e.candidate, generateErrorCallback());
+    pc2.onicecandidate = e => add(pc1, e.candidate, generateErrorCallback());
+
+    info("testing scaling with " + codec);
+
+    let stream = await navigator.mediaDevices.getUserMedia({ video: true });
+
+    var v1 = createMediaElement('video', 'v1');
+    var v2 = createMediaElement('video', 'v2');
+
+    var ontrackfired = new Promise(resolve => pc2.ontrack = e => resolve(e));
+    var v2loadedmetadata = new Promise(resolve => v2.onloadedmetadata = resolve);
+
+    is(v2.currentTime, 0, "v2.currentTime is zero at outset");
+
+    v1.srcObject = stream;
+    var sender = pc1.addTrack(stream.getVideoTracks()[0], stream);
+    let parameters = sender.getParameters();
+    is(parameters.encodings.length, 1, "Default number of encodings should be 1");
+    parameters.encodings[0].scaleResolutionDownBy = 0.5;
+
+    await mustRejectWith(
+        "Invalid scaleResolutionDownBy must reject", "RangeError",
+        () => sender.setParameters(parameters)
+    );
+
+    parameters = sender.getParameters();
+    parameters.encodings[0].scaleResolutionDownBy = 2;
+    parameters.encodings[0].maxBitrate = 60000;
+
+    await sender.setParameters(parameters);
+
+    parameters = sender.getParameters();
+    is(parameters.encodings[0].scaleResolutionDownBy, 2, "Should be able to set scaleResolutionDownBy");
+    is(parameters.encodings[0].maxBitrate, 60000, "Should be able to set maxBitrate");
+
+    let offer = await pc1.createOffer();
+    if (codec == "VP8") {
+      offer.sdp = sdputils.removeAllButPayloadType(offer.sdp, 126);
+    }
+    await pc1.setLocalDescription(offer);
+    await pc2.setRemoteDescription(pc1.localDescription);
+
+    let answer = await pc2.createAnswer();
+    await pc2.setLocalDescription(answer);
+    await pc1.setRemoteDescription(pc2.localDescription);
+    let trackevent = await ontrackfired;
+
+    v2.srcObject = trackevent.streams[0];
+
+    await v2loadedmetadata;
+
+    await waitUntil(() => v2.currentTime > 0);
+    ok(v2.currentTime > 0, "v2.currentTime is moving (" + v2.currentTime + ")");
+
+    ok(v1.videoWidth > 0, "source width is positive");
+    ok(v1.videoHeight > 0, "source height is positive");
+    const expectedWidth =
+      v1.videoWidth / 2 - (v1.videoWidth / 2 % resolutionAlignment);
+    const expectedHeight =
+      v1.videoHeight / 2 - (v1.videoHeight / 2 % resolutionAlignment);
+    is(v2.videoWidth, expectedWidth,
+       "sink is half the width of the source");
+    is(v2.videoHeight, expectedHeight,
+       "sink is half the height of the source");
+    stream.getTracks().forEach(track => track.stop());
+    v1.srcObject = v2.srcObject = null;
+    pc1.close()
+    pc2.close()
+  }
+
+  runNetworkTest(async () => {
+    await matchPlatformH264CodecPrefs();
+    const hasH264 = await checkForH264Support();
+    if (hasH264 && navigator.userAgent.includes("Android")) {
+      // Android only has a hw encoder for h264
+      resolutionAlignment = 16;
+    }
+    await pushPrefs(['media.peerconnection.video.lock_scaling', true]);
+    await testScale("VP8");
+    if (hasH264) {
+      await testScale("H264");
+    }
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_scaleResolution_oldSetParameters.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_scaleResolution_oldSetParameters.html
new file mode 100644
index 0000000000..85a989ba32
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_scaleResolution_oldSetParameters.html
@@ -0,0 +1,122 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1244913",
+    title: "Scale resolution down on a PeerConnection",
+    visible: true
+  });
+
+  async function checkForH264Support() {
+    const pc = new RTCPeerConnection();
+    const offer = await pc.createOffer({offerToReceiveVideo: true});
+    return offer.sdp.match(/a=rtpmap:[1-9][0-9]* H264/);
+  }
+
+  let resolutionAlignment = 1;
+
+  var mustRejectWith = (msg, reason, f) =>
+    f().then(() => ok(false, msg),
+             e => is(e.name, reason, msg));
+
+  async function testScale(codec) {
+    var pc1 = new RTCPeerConnection();
+    var pc2 = new RTCPeerConnection();
+
+    var add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    pc1.onicecandidate = e => add(pc2, e.candidate, generateErrorCallback());
+    pc2.onicecandidate = e => add(pc1, e.candidate, generateErrorCallback());
+
+    info("testing scaling with " + codec);
+
+    let stream = await navigator.mediaDevices.getUserMedia({ video: true });
+
+    var v1 = createMediaElement('video', 'v1');
+    var v2 = createMediaElement('video', 'v2');
+
+    var ontrackfired = new Promise(resolve => pc2.ontrack = e => resolve(e));
+    var v2loadedmetadata = new Promise(resolve => v2.onloadedmetadata = resolve);
+
+    is(v2.currentTime, 0, "v2.currentTime is zero at outset");
+
+    v1.srcObject = stream;
+    var sender = pc1.addTrack(stream.getVideoTracks()[0], stream);
+
+    const otherErrorStart = await GleanTest.rtcrtpsenderSetparameters.failOther.testGetValue();
+    const noTransactionIdWarningStart = await GleanTest.rtcrtpsenderSetparameters.warnNoTransactionid.testGetValue();
+
+    await mustRejectWith(
+        "Invalid scaleResolutionDownBy must reject", "RangeError",
+        () => sender.setParameters(
+            { encodings:[{ scaleResolutionDownBy: 0.5 } ] })
+    );
+
+    const otherErrorEnd = await GleanTest.rtcrtpsenderSetparameters.failOther.testGetValue();
+    const noTransactionIdWarningEnd = await GleanTest.rtcrtpsenderSetparameters.warnNoTransactionid.testGetValue();
+
+    // Make sure Glean is recording these statistics
+    is(otherErrorEnd.denominator, otherErrorStart.denominator, "No new RTCRtpSenders were created during this time");
+    is(otherErrorEnd.numerator, otherErrorStart.numerator + 1, "RTCRtpSender.setParameters reported a failure via Glean");
+    is(noTransactionIdWarningEnd.denominator, noTransactionIdWarningStart.denominator, "No new RTCRtpSenders were created during this time");
+    is(noTransactionIdWarningEnd.numerator, noTransactionIdWarningStart.numerator + 1, "Glean should have recorded a warning due to missing transactionId");
+
+    await sender.setParameters({ encodings: [{ maxBitrate: 60000,
+                                               scaleResolutionDownBy: 2 }] });
+
+    let offer = await pc1.createOffer();
+    if (codec == "VP8") {
+      offer.sdp = sdputils.removeAllButPayloadType(offer.sdp, 126);
+    }
+    await pc1.setLocalDescription(offer);
+    await pc2.setRemoteDescription(pc1.localDescription);
+
+    let answer = await pc2.createAnswer();
+    await pc2.setLocalDescription(answer);
+    await pc1.setRemoteDescription(pc2.localDescription);
+    let trackevent = await ontrackfired;
+
+    v2.srcObject = trackevent.streams[0];
+
+    await v2loadedmetadata;
+
+    await waitUntil(() => v2.currentTime > 0);
+    ok(v2.currentTime > 0, "v2.currentTime is moving (" + v2.currentTime + ")");
+
+    ok(v1.videoWidth > 0, "source width is positive");
+    ok(v1.videoHeight > 0, "source height is positive");
+    const expectedWidth =
+      v1.videoWidth / 2 - (v1.videoWidth / 2 % resolutionAlignment);
+    const expectedHeight =
+      v1.videoHeight / 2 - (v1.videoHeight / 2 % resolutionAlignment);
+    is(v2.videoWidth, expectedWidth,
+       "sink is half the width of the source");
+    is(v2.videoHeight, expectedHeight,
+       "sink is half the height of the source");
+    stream.getTracks().forEach(track => track.stop());
+    v1.srcObject = v2.srcObject = null;
+    pc1.close()
+    pc2.close()
+  }
+
+  runNetworkTest(async () => {
+    await matchPlatformH264CodecPrefs();
+    const hasH264 = await checkForH264Support();
+    if (hasH264 && navigator.userAgent.includes("Android")) {
+      // Android only has a hw encoder for h264
+      resolutionAlignment = 16;
+    }
+    await pushPrefs(['media.peerconnection.video.lock_scaling', true]);
+    await testScale("VP8");
+    if (hasH264) {
+      await testScale("H264");
+    }
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_sender_and_receiver_stats.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_sender_and_receiver_stats.html
new file mode 100644
index 0000000000..72749e8c50
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_sender_and_receiver_stats.html
@@ -0,0 +1,73 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1355220",
+    title: "RTCRtpSender.getStats() and RTCRtpReceiver.getStats()",
+    visible: true
+  });
+
+  const checkStats = (sndReport, rcvReport, kind) => {
+    ok(sndReport instanceof window.RTCStatsReport, "sender stats are a RTCStatsReport");
+    ok(rcvReport instanceof window.RTCStatsReport, "receiver stats are a RTCStatsReport");
+    // Returns SSRCs and performs some checks
+    let getSsrcs = (report, kind) => {
+      return [...report.values()]
+        .filter(stat => stat.type.endsWith("bound-rtp")).map(stat =>{
+          isnot(parseInt(stat.id, 16), NaN,
+            `id ${stat.id} is an opaque (hex) number`);
+          is(stat.kind, kind, "kind of " + stat.id
+              + " is expected type " + kind);
+          return stat.ssrc;
+      }).sort().join("|");
+    };
+    let sndSsrcs = getSsrcs(sndReport, kind);
+    let rcvSsrcs = getSsrcs(rcvReport, kind);
+    ok(sndSsrcs, "sender SSRCs is not empty");
+    ok(rcvSsrcs, "receiver SSRCs is not empty");
+    is(sndSsrcs, rcvSsrcs, "sender SSRCs match receiver SSRCs");
+  };
+
+  // This MUST be run after PC_*_WAIT_FOR_MEDIA_FLOW to ensure that we have RTP
+  // before checking for RTCP.
+  // It will throw UnsyncedRtcpError if it times out waiting for sync.
+
+  runNetworkTest(function (options) {
+    test = new PeerConnectionTest(options);
+    test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW",
+      async function PC_LOCAL_AND_REMOTE_CHECK_SENDER_RECEIVER_STATS(test) {
+        await Promise.all([
+          waitForSyncedRtcp(test.pcLocal._pc),
+          waitForSyncedRtcp(test.pcRemote._pc),
+        ]);
+        let senders = test.pcLocal.getSenders();
+        let receivers = test.pcRemote.getReceivers();
+        is(senders.length, 2, "Have exactly two senders.");
+        is(receivers.length, 2, "Have exactly two receivers.");
+        for(let kind of ["audio", "video"]) {
+          let senderStats =
+              await senders.find(s => s.track.kind == kind).getStats();
+          is(senders.filter(s => s.track.kind == kind).length, 1,
+              "Exactly 1 sender of kind " + kind);
+          let receiverStats =
+              await receivers.find(r => r.track.kind == kind).getStats();
+          is(receivers.filter(r => r.track.kind == kind).length, 1,
+              "Exactly 1 receiver of kind " + kind);
+
+          checkStats(senderStats, receiverStats, kind);
+        }
+      }
+    );
+    test.setMediaConstraints([{audio: true}, {video: true}], []);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_setLocalAnswerInHaveLocalOffer.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_setLocalAnswerInHaveLocalOffer.html
new file mode 100644
index 0000000000..07cdd7d6bd
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_setLocalAnswerInHaveLocalOffer.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "784519",
+    title: "setLocalDescription (answer) in 'have-local-offer'"
+  });
+
+runNetworkTest(function () {
+  const test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.chain.removeAfter("PC_LOCAL_SET_LOCAL_DESCRIPTION");
+
+  test.chain.append([
+    function PC_LOCAL_SET_LOCAL_ANSWER(test) {
+      test.pcLocal._latest_offer.type = "answer";
+      return test.pcLocal.setLocalDescriptionAndFail(test.pcLocal._latest_offer)
+        .then(err => {
+          is(err.name, "InvalidModificationError", "Error is InvalidModificationError");
+        });
+    }
+  ]);
+
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_setLocalAnswerInStable.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_setLocalAnswerInStable.html
new file mode 100644
index 0000000000..e57c0640f4
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_setLocalAnswerInStable.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "784519",
+    title: "setLocalDescription (answer) in 'stable'"
+  });
+
+runNetworkTest(function () {
+  const test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.chain.removeAfter("PC_LOCAL_CREATE_OFFER");
+
+  test.chain.append([
+    function PC_LOCAL_SET_LOCAL_ANSWER(test) {
+      test.pcLocal._latest_offer.type = "answer";
+      return test.pcLocal.setLocalDescriptionAndFail(test.pcLocal._latest_offer)
+        .then(err => {
+          is(err.name, "InvalidModificationError", "Error is InvalidModificationError");
+        });
+    }
+  ]);
+
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_setLocalOfferInHaveRemoteOffer.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_setLocalOfferInHaveRemoteOffer.html
new file mode 100644
index 0000000000..bd98a83635
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_setLocalOfferInHaveRemoteOffer.html
@@ -0,0 +1,31 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "784519",
+    title: "setLocalDescription (offer) in 'have-remote-offer'"
+  });
+
+runNetworkTest(function () {
+  const test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.chain.removeAfter("PC_REMOTE_SET_REMOTE_DESCRIPTION");
+
+  test.chain.append([
+    async function PC_REMOTE_SET_LOCAL_OFFER(test) {
+      const err = await test.pcRemote.setLocalDescriptionAndFail(test.pcLocal._latest_offer);
+      is(err.name, "InvalidModificationError", "Error is InvalidModificationError");
+    }
+  ]);
+
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters.html
new file mode 100644
index 0000000000..5df97e39f5
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters.html
@@ -0,0 +1,470 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1230184",
+  title: "Set parameters on sender",
+  visible: true
+});
+
+const simulcastOffer = `v=0
+o=- 3840232462471583827 0 IN IP4 127.0.0.1
+s=-
+t=0 0
+a=group:BUNDLE 0
+a=msid-semantic: WMS
+m=video 9 UDP/TLS/RTP/SAVPF 96
+c=IN IP4 0.0.0.0
+a=rtcp:9 IN IP4 0.0.0.0
+a=ice-ufrag:Li6+
+a=ice-pwd:3C05CTZBRQVmGCAq7hVasHlT
+a=ice-options:trickle
+a=fingerprint:sha-256 5B:D3:8E:66:0E:7D:D3:F3:8E:E6:80:28:19:FC:55:AD:58:5D:B9:3D:A8:DE:45:4A:E7:87:02:F8:3C:0B:3B:B3
+a=setup:actpass
+a=mid:0
+a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
+a=recvonly
+a=rtcp-mux
+a=rtpmap:96 VP8/90000
+a=rtcp-fb:96 goog-remb
+a=rtcp-fb:96 transport-cc
+a=rtcp-fb:96 ccm fir
+a=rid:foo recv
+a=rid:bar recv
+a=simulcast:recv foo;bar
+`;
+
+function buildMaximumSendEncodings() {
+  const sendEncodings = [];
+  while (true) {
+    // isDeeply does not see identical string primitives and String objects
+    // as the same, so we make this a string primitive.
+    sendEncodings.push({rid: `${sendEncodings.length}`});
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {sendEncodings});
+    const {encodings} = sender.getParameters();
+    if (encodings.length < sendEncodings.length) {
+      sendEncodings.pop();
+      return sendEncodings;
+    }
+  }
+}
+
+async function queueAWebrtcTask() {
+  const pc = new RTCPeerConnection();
+  pc.addTransceiver('audio');
+  await new Promise(r => pc.onnegotiationneeded = r);
+  pc.close();
+}
+
+// setParameters is mostly tested in wpt, but we test a few
+// implementation-specific things here. Other mochitests check whether the
+// set parameters actually have the desired effect on the media streams.
+const tests = [
+
+  // wpt currently does not assume support for 3 encodings, which limits the
+  // effectiveness of its powers-of-2 test (since it can test only for 1 and 2)
+  async function checkScaleResolutionDownByAutoFillPowersOf2() {
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [{rid: "0"},{rid: "1"},{rid: "2"}]
+    });
+    const {encodings} = sender.getParameters();
+    const scaleValues = encodings.map(({scaleResolutionDownBy}) => scaleResolutionDownBy);
+    isDeeply(scaleValues, [4, 2, 1]);
+  },
+
+  // wpt currently does not assume support for 3 encodings, which limits the
+  // effectiveness of its fill-with-1 test
+  async function checkScaleResolutionDownByAutoFillWith1() {
+    const pc = new RTCPeerConnection();
+    const {sender} = pc.addTransceiver('video', {
+      sendEncodings: [
+        {rid: "0"},{rid: "1", scaleResolutionDownBy: 3},{rid: "2"}
+      ]
+    });
+    const {encodings} = sender.getParameters();
+    const scaleValues = encodings.map(({scaleResolutionDownBy}) => scaleResolutionDownBy);
+    isDeeply(scaleValues, [1, 3, 1]);
+  },
+
+  async function checkVideoEncodingLimit() {
+    const pc = new RTCPeerConnection();
+    const maxSendEncodings = buildMaximumSendEncodings();
+    const sendEncodings = maxSendEncodings.concat({rid: "a"});
+    const {sender} = pc.addTransceiver('video', {sendEncodings});
+    const {encodings} = sender.getParameters();
+
+    const rids = encodings.map(({rid}) => rid);
+    const expectedRids = maxSendEncodings.map(({rid}) => rid);
+    isDeeply(rids, expectedRids);
+
+    const scaleValues = encodings.map(({scaleResolutionDownBy}) => scaleResolutionDownBy);
+    const expectedScaleValues = [];
+    let scale = 1;
+    while (expectedScaleValues.length < maxSendEncodings.length) {
+      expectedScaleValues.push(scale);
+      scale *= 2;
+    }
+    isDeeply(scaleValues, expectedScaleValues.reverse());
+  },
+
+  async function checkScaleDownByInTrimmedEncoding() {
+    const pc = new RTCPeerConnection();
+    const maxSendEncodings = buildMaximumSendEncodings();
+    const sendEncodings = maxSendEncodings.concat({rid: "a", scaleResolutionDownBy: 3});
+    const {sender} = pc.addTransceiver('video', {sendEncodings});
+    const {encodings} = sender.getParameters();
+    const rids = encodings.map(({rid}) => rid);
+    const expectedRids = maxSendEncodings.map(({rid}) => rid);
+    isDeeply(rids, expectedRids);
+    const scaleValues = encodings.map(({scaleResolutionDownBy}) => scaleResolutionDownBy);
+    const expectedScaleValues = maxSendEncodings.map(() => 1);
+    isDeeply(scaleValues, expectedScaleValues);
+  },
+
+  async function checkLibwebrtcRidLengthLimit() {
+    const pc = new RTCPeerConnection();
+    try {
+      pc.addTransceiver('video', {
+        sendEncodings: [{rid: "wibblywobblyjeremybearimy"}]}
+      );
+      ok(false, "Rid should be too long for libwebrtc!");
+    } catch (e) {
+      is(e.name, "TypeError",
+        "Rid that is too long for libwebrtc should result in a TypeError");
+    }
+  },
+
+  async function checkErrorsInTrimmedEncodings() {
+    const pc = new RTCPeerConnection();
+    const maxSendEncodings = buildMaximumSendEncodings();
+    try {
+      const sendEncodings = maxSendEncodings.concat({rid: "foo-bar"});
+      pc.addTransceiver('video', { sendEncodings });
+      ok(false, "Should throw due to invalid rid characters");
+    } catch (e) {
+      is(e.name, "TypeError")
+    }
+    try {
+      const sendEncodings = maxSendEncodings.concat({rid: "wibblywobblyjeremybearimy"});
+      pc.addTransceiver('video', { sendEncodings });
+      ok(false, "Should throw because rid too long");
+    } catch (e) {
+      is(e.name, "TypeError")
+    }
+    try {
+      const sendEncodings = maxSendEncodings.concat({scaleResolutionDownBy: 2});
+      pc.addTransceiver('video', { sendEncodings });
+      ok(false, "Should throw due to missing rid");
+    } catch (e) {
+      is(e.name, "TypeError")
+    }
+    try {
+      const sendEncodings = maxSendEncodings.concat(maxSendEncodings[0]);
+      pc.addTransceiver('video', { sendEncodings });
+      ok(false, "Should throw due to duplicate rid");
+    } catch (e) {
+      is(e.name, "TypeError")
+    }
+    try {
+      const sendEncodings = maxSendEncodings.concat({rid: maxSendEncodings.length, scaleResolutionDownBy: 0});
+      pc.addTransceiver('video', { sendEncodings });
+      ok(false, "Should throw due to invalid scaleResolutionDownBy");
+    } catch (e) {
+      is(e.name, "RangeError")
+    }
+    try {
+      const sendEncodings = maxSendEncodings.concat({rid: maxSendEncodings.length, maxFramerate: -1});
+      pc.addTransceiver('video', { sendEncodings });
+      ok(false, "Should throw due to invalid maxFramerate");
+    } catch (e) {
+      is(e.name, "RangeError")
+    }
+  },
+
+  async function checkCompatModeUnicastSetParametersAllowsSimulcastOffer() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc1 = new RTCPeerConnection();
+    const stream = await navigator.mediaDevices.getUserMedia({video: true});
+    const sender = pc1.addTrack(stream.getTracks()[0]);
+    const parameters = sender.getParameters();
+    parameters.encodings[0].scaleResolutionDownBy = 3.0;
+    await sender.setParameters(parameters);
+
+    await pc1.setRemoteDescription({type: "offer", sdp: simulcastOffer});
+
+    const {encodings} = sender.getParameters();
+    const rids = encodings.map(({rid}) => rid);
+    isDeeply(rids, ["foo", "bar"]);
+    is(encodings[0].scaleResolutionDownBy, 2.0);
+    is(encodings[1].scaleResolutionDownBy, 1.0);
+  },
+
+  async function checkCompatModeUnicastSetParametersInterruptAllowsSimulcastOffer() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc1 = new RTCPeerConnection();
+    const stream = await navigator.mediaDevices.getUserMedia({video: true});
+    const sender = pc1.addTrack(stream.getTracks()[0]);
+    const parameters = sender.getParameters();
+    parameters.encodings[0].scaleResolutionDownBy = 3.0;
+
+    const offerDone = pc1.setRemoteDescription({type: "offer", sdp: simulcastOffer});
+    await sender.setParameters(parameters);
+    await offerDone;
+
+    const {encodings} = sender.getParameters();
+    const rids = encodings.map(({rid}) => rid);
+    isDeeply(rids, ["foo", "bar"]);
+    is(encodings[0].scaleResolutionDownBy, 2.0);
+    is(encodings[1].scaleResolutionDownBy, 1.0);
+  },
+
+  async function checkCompatModeSimulcastSetParametersSetsSimulcastEnvelope() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc1 = new RTCPeerConnection();
+    const stream = await navigator.mediaDevices.getUserMedia({video: true});
+    const sender = pc1.addTrack(stream.getTracks()[0]);
+    const parameters = sender.getParameters();
+    parameters.encodings[0].rid = "1";
+    parameters.encodings.push({rid: "2"});
+    await sender.setParameters(parameters);
+
+    await pc1.setRemoteDescription({type: "offer", sdp: simulcastOffer});
+
+    const {encodings} = sender.getParameters();
+    const rids = encodings.map(({rid}) => rid);
+    // No overlap in rids -> unicast
+    isDeeply(rids, [undefined]);
+  },
+
+  async function checkCompatModeSimulcastSetParametersRacesLocalUnicastOffer() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc1 = new RTCPeerConnection();
+    const pc2 = new RTCPeerConnection();
+    const stream = await navigator.mediaDevices.getUserMedia({video: true});
+    const sender = pc1.addTrack(stream.getTracks()[0]);
+    // unicast offer
+    const offer = await pc1.createOffer();
+    const aTask = queueAWebrtcTask();
+    const sldPromise =  pc1.setLocalDescription(offer);
+
+    // Right now, we have aTask queued. The success task for sLD is not queued
+    // yet, because Firefox performs the initial steps on the microtask queue,
+    // which we have not allowed to run yet. Awaiting aTask will first clear
+    // the microtask queue, then run the task queue until aTask is finished.
+    // That _should_ result in the success task for sLD(offer) being queued.
+    await aTask;
+
+    const parameters = sender.getParameters();
+    parameters.encodings[0].rid = "foo";
+    parameters.encodings.push({rid: "bar"});
+    // simulcast setParameters; the task to update [[SendEncodings]] should be
+    // queued after the success task for sLD(offer)
+    await sender.setParameters(parameters);
+    await sldPromise;
+
+    const {encodings} = sender.getParameters();
+    const rids = encodings.map(({rid}) => rid);
+    // Compat mode lets this slide, but won't try to negotiate it since we've
+    // already applied a unicast local offer.
+    isDeeply(rids, ["foo", "bar"]);
+
+    // Let negotiation finish, so we can generate a new offer
+    await pc2.setRemoteDescription(pc1.localDescription);
+    await pc2.setLocalDescription();
+    await pc1.setRemoteDescription(pc2.localDescription);
+
+    const reoffer = await pc1.createOffer();
+    ok(!reoffer.sdp.includes("a=simulcast"), "reoffer should be unicast");
+  },
+
+  async function checkCompatModeSimulcastSetParametersRacesRemoteOffer() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc1 = new RTCPeerConnection();
+    const stream = await navigator.mediaDevices.getUserMedia({video: true});
+    const sender = pc1.addTrack(stream.getTracks()[0]);
+    const parameters = sender.getParameters();
+    parameters.encodings[0].rid = "foo";
+    parameters.encodings.push({rid: "bar"});
+    const p = sender.setParameters(parameters);
+    await pc1.setRemoteDescription({type: "offer", sdp: simulcastOffer});
+    await p;
+    const answer = await pc1.createAnswer();
+
+    const {encodings} = sender.getParameters();
+    const rids = encodings.map(({rid}) => rid);
+    isDeeply(rids, ["foo", "bar"]);
+    ok(answer.sdp.includes("a=simulcast:send foo;bar"), "answer should be simulcast");
+  },
+
+  async function checkCompatModeSimulcastSetParametersRacesLocalAnswer() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    // We do an initial negotiation, and while the local answer is pending,
+    // perform a setParameters on a not-yet-negotiated video sender. The intent
+    // here is to have the success task for sLD(answer) run while the
+    // setParameters is pending.
+    const pc1 = new RTCPeerConnection();
+    const pc2 = new RTCPeerConnection();
+
+    const audioStream = await navigator.mediaDevices.getUserMedia({audio: true});
+    // We use this later on, but set it up now so we don't inadvertently
+    // crank the event loop more than we intend below.
+    const videoStream = await navigator.mediaDevices.getUserMedia({video: true});
+    pc2.addTrack(audioStream.getTracks()[0]);
+    await pc2.setLocalDescription();
+    await pc1.setRemoteDescription(pc2.localDescription);
+    const answer = await pc1.createAnswer();
+    const aTask = queueAWebrtcTask();
+    const sldPromise = pc1.setLocalDescription(answer);
+
+    // Right now, we have aTask queued. The success task for sLD is not queued
+    // yet, because Firefox performs the initial steps on the microtask queue,
+    // which we have not allowed to run yet. Awaiting aTask will first clear
+    // the microtask queue, then run the task queue until aTask is finished.
+    // That _should_ result in the success task for sLD(answer) being queued.
+    await aTask;
+
+    // The success task for sLD(answer) should be queued now. Don't relinquish
+    // the event loop!
+
+    // New sender that has nothing to do with the negotiation in progress.
+    const sender = pc1.addTrack(videoStream.getTracks()[0]);
+    const parameters = sender.getParameters();
+    parameters.encodings[0].rid = "foo";
+    parameters.encodings.push({rid: "bar"});
+
+    // We have not relinquished the event loop, so the sLD(answer) should still
+    // be queued. The task that updates [[SendEncodings]] (from setParameters)
+    // should be queued behind it. Let them both run.
+    await sender.setParameters(parameters);
+    await sldPromise;
+
+    const offer = await pc1.createOffer();
+
+    const {encodings} = sender.getParameters();
+    const rids = encodings.map(({rid}) => rid);
+    isDeeply(rids, ["foo", "bar"]);
+    ok(offer.sdp.includes("a=simulcast:send foo;bar"), "offer should be simulcast");
+  },
+
+  async function checkCompatModeSimulcastSetParametersRacesRemoteAnswer() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    // We do an initial negotiation, and while the remote answer is pending,
+    // perform a setParameters on a not-yet-negotiated video sender. The intent
+    // here is to have the success task for sRD(answer) run while the
+    // setParameters is pending.
+    const pc1 = new RTCPeerConnection();
+    const pc2 = new RTCPeerConnection();
+
+    const audioStream = await navigator.mediaDevices.getUserMedia({audio: true});
+    // We use this later on, but set it up now so we don't inadvertently
+    // crank the event loop more than we intend below.
+    const videoStream = await navigator.mediaDevices.getUserMedia({video: true});
+    pc1.addTrack(audioStream.getTracks()[0]);
+    await pc1.setLocalDescription();
+    await pc2.setRemoteDescription(pc1.localDescription);
+    await pc2.setLocalDescription();
+    const aTask = queueAWebrtcTask();
+    const srdPromise = pc1.setRemoteDescription(pc2.localDescription);
+
+    // Right now, we have aTask queued. The success task for sRD is not queued
+    // yet, because Firefox performs the initial steps on the microtask queue,
+    // which we have not allowed to run yet. Awaiting aTask will first clear
+    // the microtask queue, then run the task queue until aTask is finished.
+    // That _should_ result in the success task for sRD(answer) being queued.
+    await aTask;
+
+    // The success task for sRD(answer) should be queued now. Don't relinquish
+    // the event loop!
+
+    const sender = pc1.addTrack(videoStream.getTracks()[0]);
+    const parameters = sender.getParameters();
+    parameters.encodings[0].rid = "foo";
+    parameters.encodings.push({rid: "bar"});
+
+    // We have not relinquished the event loop, so the sRD(answer) should still
+    // be queued. The task that updates [[SendEncodings]] (from setParameters)
+    // should be queued behind it. Let them both run.
+    await sender.setParameters(parameters);
+    await srdPromise;
+
+    const offer = await pc1.createOffer();
+
+    const {encodings} = sender.getParameters();
+    const rids = encodings.map(({rid}) => rid);
+    isDeeply(rids, ["foo", "bar"]);
+    ok(offer.sdp.includes("a=simulcast:send foo;bar"), "offer should be simulcast");
+  },
+
+  async function checkCompatModeSimulcastRidlessSetParametersRacesLocalOffer() {
+    await pushPrefs(
+      ["media.peerconnection.allow_old_setParameters", true]);
+    const pc1 = new RTCPeerConnection();
+    const pc2 = new RTCPeerConnection();
+    const stream = await navigator.mediaDevices.getUserMedia({video: true});
+    const sender = pc1.addTrack(stream.getTracks()[0]);
+    // unicast offer
+    const aTask = queueAWebrtcTask();
+    const sldPromise =  pc1.setLocalDescription();
+
+    // Right now, we have aTask queued. The success task for sLD is not queued
+    // yet, because Firefox performs the initial steps on the microtask queue,
+    // which we have not allowed to run yet. Awaiting aTask will first clear
+    // the microtask queue, then run the task queue until aTask is finished.
+    // That _should_ result in the success task for sLD(offer) being queued.
+    await aTask;
+
+    // simulcast setParameters; the task to update [[SendEncodings]] should be
+    // queued after the success task for sLD(offer)
+    try {
+      await sender.setParameters({"encodings": [{}, {}]});
+      ok(false, "setParameters with two ridless encodings should fail");
+    } catch (e) {
+      ok(true, "setParameters with two ridless encodings should fail");
+    }
+    await sldPromise;
+
+    const {encodings} = sender.getParameters();
+    const rids = encodings.map(({rid}) => rid);
+    // Compat mode lets this slide, but won't try to negotiate it since we've
+    // already applied a unicast local offer.
+    isDeeply(rids, [undefined]);
+
+    // Let negotiation finish, so we can generate a new offer
+    await pc2.setRemoteDescription(pc1.localDescription);
+    await pc2.setLocalDescription();
+    await pc1.setRemoteDescription(pc2.localDescription);
+
+    const reoffer = await pc1.createOffer();
+    ok(!reoffer.sdp.includes("a=simulcast"), "reoffer should be unicast");
+  },
+
+];
+
+runNetworkTest(async () => {
+  await pushPrefs(
+    ["media.peerconnection.allow_old_setParameters", false]);
+  for (const test of tests) {
+    info(`Running test: ${test.name}`);
+    await test();
+    info(`Done running test: ${test.name}`);
+  }
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_maxFramerate.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_maxFramerate.html
new file mode 100644
index 0000000000..8047719775
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_maxFramerate.html
@@ -0,0 +1,63 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1611957",
+  title: "Live-updating maxFramerate"
+});
+
+let sender, receiver;
+
+async function checkMaxFrameRate(rate) {
+  const parameters = sender.getParameters();
+  parameters.encodings[0].maxFramerate = rate;
+  await sender.setParameters(parameters);
+  await wait(2000);
+  const stats = Array.from((await receiver.getStats()).values());
+  const inboundRtp = stats.find(stat => stat.type == "inbound-rtp");
+  info(`inbound-rtp stats: ${JSON.stringify(inboundRtp)}`);
+  const fps = inboundRtp.framesPerSecond;
+  ok(fps <= (rate * 1.1) + 1,
+    `fps is an appropriate value (${fps}) for rate (${rate})`);
+}
+
+runNetworkTest(async function (options) {
+  let test = new PeerConnectionTest(options);
+  test.setMediaConstraints([{video: true}], []);
+  test.chain.append([
+    function CHECK_PRECONDITIONS() {
+      is(test.pcLocal._pc.getSenders().length, 1,
+        "Should have 1 local sender");
+      is(test.pcRemote._pc.getReceivers().length, 1,
+        "Should have 1 remote receiver");
+
+      sender = test.pcLocal._pc.getSenders()[0];
+      receiver = test.pcRemote._pc.getReceivers()[0];
+    },
+    function PC_LOCAL_SET_MAX_FRAMERATE_2() {
+      return checkMaxFrameRate(2);
+    },
+    function PC_LOCAL_SET_MAX_FRAMERATE_4() {
+      return checkMaxFrameRate(4);
+    },
+    function PC_LOCAL_SET_MAX_FRAMERATE_15() {
+      return checkMaxFrameRate(15);
+    },
+    function PC_LOCAL_SET_MAX_FRAMERATE_8() {
+      return checkMaxFrameRate(8);
+    },
+    function PC_LOCAL_SET_MAX_FRAMERATE_1() {
+      return checkMaxFrameRate(1);
+    },
+  ]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_maxFramerate_oldSetParameters.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_maxFramerate_oldSetParameters.html
new file mode 100644
index 0000000000..9c68a31c0a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_maxFramerate_oldSetParameters.html
@@ -0,0 +1,60 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1611957",
+  title: "Live-updating maxFramerate"
+});
+
+let sender, receiver;
+
+async function checkMaxFrameRate(rate) {
+  sender.setParameters({ encodings: [{ maxFramerate: rate }] });
+  await wait(2000);
+  const stats = Array.from((await receiver.getStats()).values());
+  const inboundRtp = stats.find(stat => stat.type == "inbound-rtp");
+  info(`inbound-rtp stats: ${JSON.stringify(inboundRtp)}`);
+  const fps = inboundRtp.framesPerSecond;
+  ok(fps <= (rate * 1.1) + 1, `fps is an appropriate value (${fps}) for rate (${rate})`);
+}
+
+runNetworkTest(async function (options) {
+  let test = new PeerConnectionTest(options);
+  test.setMediaConstraints([{video: true}], []);
+  test.chain.append([
+    function CHECK_PRECONDITIONS() {
+      is(test.pcLocal._pc.getSenders().length, 1,
+          "Should have 1 local sender");
+      is(test.pcRemote._pc.getReceivers().length, 1,
+          "Should have 1 remote receiver");
+
+      sender = test.pcLocal._pc.getSenders()[0];
+      receiver = test.pcRemote._pc.getReceivers()[0];
+    },
+    function PC_LOCAL_SET_MAX_FRAMERATE_2() {
+      return checkMaxFrameRate(2);
+    },
+    function PC_LOCAL_SET_MAX_FRAMERATE_4() {
+      return checkMaxFrameRate(4);
+    },
+    function PC_LOCAL_SET_MAX_FRAMERATE_15() {
+      return checkMaxFrameRate(15);
+    },
+    function PC_LOCAL_SET_MAX_FRAMERATE_8() {
+      return checkMaxFrameRate(8);
+    },
+    function PC_LOCAL_SET_MAX_FRAMERATE_1() {
+      return checkMaxFrameRate(1);
+    },
+  ]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_oldSetParameters.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_oldSetParameters.html
new file mode 100644
index 0000000000..2b55ec46e6
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_oldSetParameters.html
@@ -0,0 +1,86 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1230184",
+  title: "Set parameters on sender",
+  visible: true
+});
+
+function parameterstest(pc) {
+  ok(pc.getSenders().length, "have senders");
+  var sender = pc.getSenders()[0];
+
+  var testParameters = (params, errorName, errorMsg) => {
+    info("Trying to set " + JSON.stringify(params));
+
+    var validateParameters = (a, b) => {
+      var validateEncoding = (a, b) => {
+        is(a.rid, b.rid, "same rid");
+        is(a.maxBitrate, b.maxBitrate, "same maxBitrate");
+        is(a.maxFramerate, b.maxFramerate, "same maxFramerate");
+        is(a.scaleResolutionDownBy, b.scaleResolutionDownBy,
+           "same scaleResolutionDownBy");
+      };
+      is(a.encodings.length, (b.encodings || []).length, "same encodings");
+      a.encodings.forEach((en, i) => validateEncoding(en, b.encodings[i]));
+    };
+
+    var before = JSON.stringify(sender.getParameters());
+    isnot(JSON.stringify(params), before, "starting condition");
+
+    var p = sender.setParameters(params)
+    .then(() => {
+      isnot(JSON.stringify(sender.getParameters()), before, "parameters changed");
+      validateParameters(sender.getParameters(), params);
+      is(null, errorName || null, "is success expected");
+    }, e => {
+      is(e.name, errorName, "correct error name");
+      is(e.message, errorMsg, "correct error message");
+    });
+    is(JSON.stringify(sender.getParameters()), before, "parameters not set yet");
+    return p;
+  };
+
+  return [
+    [{ encodings: [ { rid: "foo", maxBitrate: 40000, scaleResolutionDownBy: 2 },
+                    { rid: "bar", maxBitrate: 10000, scaleResolutionDownBy: 4 }]
+    }],
+    [{ encodings: [{ maxBitrate: 10000, scaleResolutionDownBy: 4 }]}],
+    [{ encodings: [{ maxFramerate: 0.0, scaleResolutionDownBy: 1 }]}],
+    [{ encodings: [{ maxFramerate: 30.5, scaleResolutionDownBy: 1 }]}],
+    [{ encodings: [{ maxFramerate: -1, scaleResolutionDownBy: 1 }]}, "RangeError", "maxFramerate must be non-negative"],
+    [{ encodings: [{ maxBitrate: 40000 },
+                   { rid: "bar", maxBitrate: 10000 }] }, "TypeError", "Missing rid"],
+    [{ encodings: [{ rid: "foo", maxBitrate: 40000 },
+                   { rid: "bar", maxBitrate: 10000 },
+                   { rid: "bar", maxBitrate: 20000 }] }, "TypeError", "Duplicate rid"],
+    [{}, "TypeError", `RTCRtpSender.setParameters: Missing required 'encodings' member of RTCRtpSendParameters.`]
+  ].reduce((p, args) => p.then(() => testParameters.apply(this, args)),
+           Promise.resolve());
+}
+
+runNetworkTest(() => {
+  const test = new PeerConnectionTest();
+  test.setMediaConstraints([{video: true}], [{video: true}]);
+  test.chain.removeAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW");
+
+  // Test sender parameters.
+  test.chain.append([
+    function PC_LOCAL_SET_PARAMETERS(test) {
+      return parameterstest(test.pcLocal._pc);
+    }
+  ]);
+
+  return test.run();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_scaleResolutionDownBy.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_scaleResolutionDownBy.html
new file mode 100644
index 0000000000..d1275d6523
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_scaleResolutionDownBy.html
@@ -0,0 +1,98 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1253499",
+  title: "Live-updating scaleResolutionDownBy"
+});
+
+async function checkForH264Support() {
+  const pc = new RTCPeerConnection();
+  const offer = await pc.createOffer({offerToReceiveVideo: true});
+  return offer.sdp.match(/a=rtpmap:[1-9][0-9]* H264/);
+}
+
+let sender, localElem, remoteElem;
+let originalWidth, originalHeight;
+let resolutionAlignment = 1;
+
+async function checkScaleDownBy(scale) {
+  const parameters = sender.getParameters();
+  parameters.encodings[0].scaleResolutionDownBy = scale;
+  await sender.setParameters(parameters);
+  await haveEvent(remoteElem, "resize", wait(5000, new Error("Timeout")));
+
+  // Find the expected resolution. Internally we floor the exact scaling, then
+  // shrink each dimension to the alignment requested by the encoder.
+  let expectedWidth =
+    originalWidth / scale - (originalWidth / scale % resolutionAlignment);
+  let expectedHeight =
+    originalHeight / scale - (originalHeight / scale % resolutionAlignment);
+
+  is(remoteElem.videoWidth, expectedWidth,
+    `Width should have scaled down by ${scale}`);
+  is(remoteElem.videoHeight, expectedHeight,
+    `Height should have scaled down by ${scale}`);
+}
+
+runNetworkTest(async function (options) {
+  await pushPrefs(['media.peerconnection.video.lock_scaling', true]);
+  // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+  if (navigator.userAgent.includes("Android")) {
+    if (await checkForH264Support()) {
+      // Android only has h264 in hw, so now we know it will use vp8 in hw too.
+      resolutionAlignment = 16;
+    }
+    await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                    ["media.webrtc.hw.h264.enabled", false]);
+  }
+
+  let test = new PeerConnectionTest(options);
+  test.setMediaConstraints([{video: true}], []);
+  test.chain.append([
+    function CHECK_PRECONDITIONS() {
+      is(test.pcLocal._pc.getSenders().length, 1,
+          "Should have 1 local sender");
+      is(test.pcLocal.localMediaElements.length, 1,
+          "Should have 1 local sending media element");
+      is(test.pcRemote.remoteMediaElements.length, 1,
+          "Should have 1 remote media element");
+
+      sender = test.pcLocal._pc.getSenders()[0];
+      localElem = test.pcLocal.localMediaElements[0];
+      remoteElem = test.pcRemote.remoteMediaElements[0];
+
+      remoteElem.addEventListener("resize", () =>
+        info(`Video resized to ${remoteElem.videoWidth}x${remoteElem.videoHeight}`));
+
+      originalWidth = localElem.videoWidth;
+      originalHeight = localElem.videoHeight;
+      info(`Original width is ${originalWidth}`);
+    },
+    function PC_LOCAL_SCALEDOWNBY_2() {
+      return checkScaleDownBy(2);
+    },
+    function PC_LOCAL_SCALEDOWNBY_4() {
+      return checkScaleDownBy(4);
+    },
+    function PC_LOCAL_SCALEDOWNBY_15() {
+      return checkScaleDownBy(15);
+    },
+    function PC_LOCAL_SCALEDOWNBY_8() {
+      return checkScaleDownBy(8);
+    },
+    function PC_LOCAL_SCALEDOWNBY_1() {
+      return checkScaleDownBy(1);
+    },
+  ]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_scaleResolutionDownBy_oldSetParameters.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_scaleResolutionDownBy_oldSetParameters.html
new file mode 100644
index 0000000000..4d515bd5c1
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_setParameters_scaleResolutionDownBy_oldSetParameters.html
@@ -0,0 +1,96 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1253499",
+  title: "Live-updating scaleResolutionDownBy"
+});
+
+async function checkForH264Support() {
+  const pc = new RTCPeerConnection();
+  const offer = await pc.createOffer({offerToReceiveVideo: true});
+  return offer.sdp.match(/a=rtpmap:[1-9][0-9]* H264/);
+}
+
+let sender, localElem, remoteElem;
+let originalWidth, originalHeight;
+let resolutionAlignment = 1;
+
+async function checkScaleDownBy(scale) {
+  sender.setParameters({ encodings: [{ scaleResolutionDownBy: scale }] });
+  await haveEvent(remoteElem, "resize", wait(5000, new Error("Timeout")));
+
+  // Find the expected resolution. Internally we floor the exact scaling, then
+  // shrink each dimension to the alignment requested by the encoder.
+  let expectedWidth =
+    originalWidth / scale - (originalWidth / scale % resolutionAlignment);
+  let expectedHeight =
+    originalHeight / scale - (originalHeight / scale % resolutionAlignment);
+
+  is(remoteElem.videoWidth, expectedWidth,
+    `Width should have scaled down by ${scale}`);
+  is(remoteElem.videoHeight, expectedHeight,
+    `Height should have scaled down by ${scale}`);
+}
+
+runNetworkTest(async function (options) {
+  await pushPrefs(['media.peerconnection.video.lock_scaling', true]);
+  // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+  if (navigator.userAgent.includes("Android")) {
+    if (await checkForH264Support()) {
+      // Android only has h264 in hw, so now we know it will use vp8 in hw too.
+      resolutionAlignment = 16;
+    }
+    await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                    ["media.webrtc.hw.h264.enabled", false]);
+  }
+
+  let test = new PeerConnectionTest(options);
+  test.setMediaConstraints([{video: true}], []);
+  test.chain.append([
+    function CHECK_PRECONDITIONS() {
+      is(test.pcLocal._pc.getSenders().length, 1,
+          "Should have 1 local sender");
+      is(test.pcLocal.localMediaElements.length, 1,
+          "Should have 1 local sending media element");
+      is(test.pcRemote.remoteMediaElements.length, 1,
+          "Should have 1 remote media element");
+
+      sender = test.pcLocal._pc.getSenders()[0];
+      localElem = test.pcLocal.localMediaElements[0];
+      remoteElem = test.pcRemote.remoteMediaElements[0];
+
+      remoteElem.addEventListener("resize", () =>
+        info(`Video resized to ${remoteElem.videoWidth}x${remoteElem.videoHeight}`));
+
+      originalWidth = localElem.videoWidth;
+      originalHeight = localElem.videoHeight;
+      info(`Original width is ${originalWidth}`);
+    },
+    function PC_LOCAL_SCALEDOWNBY_2() {
+      return checkScaleDownBy(2);
+    },
+    function PC_LOCAL_SCALEDOWNBY_4() {
+      return checkScaleDownBy(4);
+    },
+    function PC_LOCAL_SCALEDOWNBY_15() {
+      return checkScaleDownBy(15);
+    },
+    function PC_LOCAL_SCALEDOWNBY_8() {
+      return checkScaleDownBy(8);
+    },
+    function PC_LOCAL_SCALEDOWNBY_1() {
+      return checkScaleDownBy(1);
+    },
+  ]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_setRemoteAnswerInHaveRemoteOffer.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_setRemoteAnswerInHaveRemoteOffer.html
new file mode 100644
index 0000000000..1912835160
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_setRemoteAnswerInHaveRemoteOffer.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "784519",
+    title: "setRemoteDescription (answer) in 'have-remote-offer'"
+  });
+
+runNetworkTest(function () {
+  const test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.chain.removeAfter("PC_REMOTE_SET_REMOTE_DESCRIPTION");
+
+  test.chain.append([
+    function PC_REMOTE_SET_REMOTE_ANSWER(test) {
+      test.pcLocal._latest_offer.type = "answer";
+      test.pcRemote._pc.setRemoteDescription(test.pcLocal._latest_offer)
+        .then(generateErrorCallback('setRemoteDescription should fail'),
+              err =>
+              is(err.name, "InvalidStateError", "Error is InvalidStateError"));
+    }
+  ]);
+
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_setRemoteAnswerInStable.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_setRemoteAnswerInStable.html
new file mode 100644
index 0000000000..6208fdea3e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_setRemoteAnswerInStable.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "784519",
+    title: "setRemoteDescription (answer) in 'stable'"
+  });
+
+runNetworkTest(function () {
+  const test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.chain.removeAfter("PC_LOCAL_CREATE_OFFER");
+
+  test.chain.append([
+    function PC_LOCAL_SET_REMOTE_ANSWER(test) {
+      test.pcLocal._latest_offer.type = "answer";
+      test.pcLocal._pc.setRemoteDescription(test.pcLocal._latest_offer)
+        .then(generateErrorCallback('setRemoteDescription should fail'),
+              err =>
+              is(err.name, "InvalidStateError", "Error is InvalidStateError"));
+    }
+  ]);
+
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_setRemoteOfferInHaveLocalOffer.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_setRemoteOfferInHaveLocalOffer.html
new file mode 100644
index 0000000000..20236f442c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_setRemoteOfferInHaveLocalOffer.html
@@ -0,0 +1,37 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "784519",
+    title: "setRemoteDescription (offer) in 'have-local-offer'"
+  });
+
+runNetworkTest(function () {
+  const test = new PeerConnectionTest();
+  test.setMediaConstraints([{audio: true}], [{audio: true}]);
+  test.chain.removeAfter("PC_LOCAL_SET_LOCAL_DESCRIPTION");
+
+  test.chain.append([
+    async function PC_LOCAL_SET_REMOTE_OFFER(test) {
+      const p = test.pcLocal._pc.setRemoteDescription(test.pcLocal._latest_offer);
+      await new Promise(r => test.pcLocal.onsignalingstatechange = r);
+      is(test.pcLocal._pc.signalingState, 'stable', 'should fire stable');
+      await new Promise(r => test.pcLocal.onsignalingstatechange = r);
+      is(test.pcLocal._pc.signalingState, 'have-remote-offer',
+         'should fire have-remote-offer');
+      await p;
+      ok(true, 'setRemoteDescription should succeed');
+    }
+  ]);
+
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastAnswer.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastAnswer.html
new file mode 100644
index 0000000000..ba75c72022
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastAnswer.html
@@ -0,0 +1,121 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="helpers_from_wpt/sdp.js"></script>
+  <script type="application/javascript" src="simulcast.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1231507",
+    title: "Basic video-only peer connection with Simulcast answer",
+    visible: true
+  });
+
+  runNetworkTest(async () => {
+    await pushPrefs(
+        // 180Kbps was determined empirically, set well-higher than
+        // the 80Kbps+overhead needed for the two simulcast streams.
+        // 100Kbps was apparently too low.
+        ['media.peerconnection.video.min_bitrate_estimate', 180*1000]);
+
+
+    const offerer = new RTCPeerConnection();
+    const answerer = new RTCPeerConnection();
+
+    const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    offerer.onicecandidate = e => add(answerer, e.candidate, generateErrorCallback());
+    answerer.onicecandidate = e => add(offerer, e.candidate, generateErrorCallback());
+
+    const metadataToBeLoaded = [];
+    offerer.ontrack = (e) => {
+      metadataToBeLoaded.push(getPlaybackWithLoadedMetadata(e.track));
+    };
+
+    // Two recv transceivers, one for each simulcast stream
+    offerer.addTransceiver('video', { direction: 'recvonly' });
+    offerer.addTransceiver('video', { direction: 'recvonly' });
+
+    const offer = await offerer.createOffer();
+
+    const mungedOffer = midToRid(offer);
+    info(`Transformed recv offer to simulcast: ${offer.sdp} to ${mungedOffer}`);
+
+    await answerer.setRemoteDescription({type: 'offer', sdp: mungedOffer});
+
+    // One send transceiver, that will be used to send both simulcast streams
+    const emitter = new VideoFrameEmitter();
+    const videoStream = emitter.stream();
+    const sender = answerer.addTrack(videoStream.getVideoTracks()[0], videoStream);
+    let parameters = sender.getParameters();
+    is(parameters.encodings.length, 2);
+    is(answerer.getSenders().length, 1);
+    emitter.start();
+
+    await offerer.setLocalDescription(offer);
+
+    const rids = offerer.getTransceivers().map(t => t.mid);
+    is(rids.length, 2, 'Should have 2 mids in offer');
+    ok(rids[0] != '', 'First mid should be non-empty');
+    ok(rids[1] != '', 'Second mid should be non-empty');
+    info(`rids: ${JSON.stringify(rids)}`);
+
+    parameters = sender.getParameters();
+    info(`parameters: ${JSON.stringify(parameters)}`);
+    const observedRids = parameters.encodings.map(({rid}) => rid);
+    isDeeply(observedRids, rids);
+    parameters.encodings[0].maxBitrate = 40000;
+    parameters.encodings[0].scaleResolutionDownBy = 1;
+    parameters.encodings[1].maxBitrate = 40000;
+    parameters.encodings[1].scaleResolutionDownBy = 2;
+    await sender.setParameters(parameters);
+
+    const answer = await answerer.createAnswer();
+
+    const mungedAnswer = ridToMid(answer);
+    info(`Transformed send simulcast answer to multiple m-sections: ${answer.sdp} to ${mungedAnswer}`);
+    await offerer.setRemoteDescription({type: 'answer', sdp: mungedAnswer});
+    await answerer.setLocalDescription(answer);
+
+    is(metadataToBeLoaded.length, 2, 'Offerer should have gotten 2 ontrack events');
+    info('Waiting for 2 loadedmetadata events');
+    const videoElems = await Promise.all(metadataToBeLoaded);
+
+    const statsReady =
+      Promise.all([waitForSyncedRtcp(offerer), waitForSyncedRtcp(answerer)]);
+
+    const helper = new VideoStreamHelper();
+    info('Waiting for first video element to start playing');
+    await helper.checkVideoPlaying(videoElems[0]);
+    info('Waiting for second video element to start playing');
+    await helper.checkVideoPlaying(videoElems[1]);
+
+    is(videoElems[0].videoWidth, 50,
+       "sink is same width as source, modulo our cropping algorithm");
+    is(videoElems[0].videoHeight, 50,
+       "sink is same height as source, modulo our cropping algorithm");
+    is(videoElems[1].videoWidth, 25,
+       "sink is 1/2 width of source, modulo our cropping algorithm");
+    is(videoElems[1].videoHeight, 25,
+       "sink is 1/2 height of source, modulo our cropping algorithm");
+
+    await statsReady;
+    info("Stats ready");
+    const senderStats = await sender.getStats();
+    checkSenderStats(senderStats, 2);
+    checkExpectedFields(senderStats);
+    pedanticChecks(senderStats);
+
+    emitter.stop();
+    videoStream.getVideoTracks()[0].stop();
+    offerer.close();
+    answerer.close();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastAnswer_lowResFirst.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastAnswer_lowResFirst.html
new file mode 100644
index 0000000000..00c6e4ad3a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastAnswer_lowResFirst.html
@@ -0,0 +1,113 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="helpers_from_wpt/sdp.js"></script>
+  <script type="application/javascript" src="simulcast.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1231507",
+    title: "Basic video-only peer connection with Simulcast answer, first rid has lowest resolution",
+    visible: true
+  });
+
+  runNetworkTest(async () => {
+    await pushPrefs(
+        // 180Kbps was determined empirically, set well-higher than
+        // the 80Kbps+overhead needed for the two simulcast streams.
+        // 100Kbps was apparently too low.
+        ['media.peerconnection.video.min_bitrate_estimate', 180*1000]);
+
+
+    const offerer = new RTCPeerConnection();
+    const answerer = new RTCPeerConnection();
+
+    const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    offerer.onicecandidate = e => add(answerer, e.candidate, generateErrorCallback());
+    answerer.onicecandidate = e => add(offerer, e.candidate, generateErrorCallback());
+
+    const metadataToBeLoaded = [];
+    offerer.ontrack = (e) => {
+      metadataToBeLoaded.push(getPlaybackWithLoadedMetadata(e.track));
+    };
+
+    // Two recv transceivers, one for each simulcast stream
+    offerer.addTransceiver('video', { direction: 'recvonly' });
+    offerer.addTransceiver('video', { direction: 'recvonly' });
+
+    // One send transceiver, that will be used to send both simulcast streams
+    const emitter = new VideoFrameEmitter();
+    const videoStream = emitter.stream();
+    answerer.addTrack(videoStream.getVideoTracks()[0], videoStream);
+    emitter.start();
+
+    const offer = await offerer.createOffer();
+
+    const mungedOffer = midToRid(offer);
+    info(`Transformed recv offer to simulcast: ${offer.sdp} to ${mungedOffer}`);
+
+    await answerer.setRemoteDescription({type: 'offer', sdp: mungedOffer});
+    await offerer.setLocalDescription(offer);
+
+    const rids = offerer.getTransceivers().map(t => t.mid);
+    is(rids.length, 2, 'Should have 2 mids in offer');
+    ok(rids[0] != '', 'First mid should be non-empty');
+    ok(rids[1] != '', 'Second mid should be non-empty');
+    info(`rids: ${JSON.stringify(rids)}`);
+
+    const sender = answerer.getSenders()[0];
+    const parameters = sender.getParameters();
+    parameters.encodings[0].maxBitrate = 40000;
+    parameters.encodings[0].scaleResolutionDownBy = 2;
+    parameters.encodings[1].maxBitrate = 40000;
+    await sender.setParameters(parameters);
+
+    const answer = await answerer.createAnswer();
+
+    const mungedAnswer = ridToMid(answer);
+    info(`Transformed send simulcast answer to multiple m-sections: ${answer.sdp} to ${mungedAnswer}`);
+    await offerer.setRemoteDescription({type: 'answer', sdp: mungedAnswer});
+    await answerer.setLocalDescription(answer);
+
+    is(metadataToBeLoaded.length, 2, 'Offerer should have gotten 2 ontrack events');
+    info('Waiting for 2 loadedmetadata events');
+    const videoElems = await Promise.all(metadataToBeLoaded);
+
+    const statsReady =
+      Promise.all([waitForSyncedRtcp(offerer), waitForSyncedRtcp(answerer)]);
+
+    const helper = new VideoStreamHelper();
+    info('Waiting for first video element to start playing');
+    await helper.checkVideoPlaying(videoElems[0]);
+    info('Waiting for second video element to start playing');
+    await helper.checkVideoPlaying(videoElems[1]);
+
+    is(videoElems[1].videoWidth, 50,
+       "sink is same width as source, modulo our cropping algorithm");
+    is(videoElems[1].videoHeight, 50,
+       "sink is same height as source, modulo our cropping algorithm");
+    is(videoElems[0].videoWidth, 25,
+       "sink is 1/2 width of source, modulo our cropping algorithm");
+    is(videoElems[0].videoHeight, 25,
+       "sink is 1/2 height of source, modulo our cropping algorithm");
+
+    await statsReady;
+    const senderStats = await sender.getStats();
+    checkSenderStats(senderStats, 2);
+    checkExpectedFields(senderStats);
+    pedanticChecks(senderStats);
+
+    emitter.stop();
+    videoStream.getVideoTracks()[0].stop();
+    offerer.close();
+    answerer.close();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastAnswer_lowResFirst_oldSetParameters.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastAnswer_lowResFirst_oldSetParameters.html
new file mode 100644
index 0000000000..c2aafc4575
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastAnswer_lowResFirst_oldSetParameters.html
@@ -0,0 +1,115 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="helpers_from_wpt/sdp.js"></script>
+  <script type="application/javascript" src="simulcast.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1231507",
+    title: "Basic video-only peer connection with Simulcast answer, first rid has lowest resolution",
+    visible: true
+  });
+
+  runNetworkTest(async () => {
+    await pushPrefs(
+        ['media.peerconnection.simulcast', true],
+        // 180Kbps was determined empirically, set well-higher than
+        // the 80Kbps+overhead needed for the two simulcast streams.
+        // 100Kbps was apparently too low.
+        ['media.peerconnection.video.min_bitrate_estimate', 180*1000]);
+
+
+    const offerer = new RTCPeerConnection();
+    const answerer = new RTCPeerConnection();
+
+    const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    offerer.onicecandidate = e => add(answerer, e.candidate, generateErrorCallback());
+    answerer.onicecandidate = e => add(offerer, e.candidate, generateErrorCallback());
+
+    const metadataToBeLoaded = [];
+    offerer.ontrack = (e) => {
+      metadataToBeLoaded.push(getPlaybackWithLoadedMetadata(e.track));
+    };
+
+    // Two recv transceivers, one for each simulcast stream
+    offerer.addTransceiver('video', { direction: 'recvonly' });
+    offerer.addTransceiver('video', { direction: 'recvonly' });
+
+    // One send transceiver, that will be used to send both simulcast streams
+    const emitter = new VideoFrameEmitter();
+    const videoStream = emitter.stream();
+    answerer.addTrack(videoStream.getVideoTracks()[0], videoStream);
+    emitter.start();
+
+    const offer = await offerer.createOffer();
+
+    const mungedOffer = midToRid(offer);
+    info(`Transformed recv offer to simulcast: ${offer.sdp} to ${mungedOffer}`);
+
+    await answerer.setRemoteDescription({type: "offer", sdp: mungedOffer});
+    await offerer.setLocalDescription(offer);
+
+    const rids = offerer.getTransceivers().map(t => t.mid);
+    is(rids.length, 2, 'Should have 2 mids in offer');
+    ok(rids[0] != '', 'First mid should be non-empty');
+    ok(rids[1] != '', 'Second mid should be non-empty');
+    info(`rids: ${JSON.stringify(rids)}`);
+
+    const sender = answerer.getSenders()[0];
+    sender.setParameters({
+        encodings: [
+            { rid: rids[0], maxBitrate: 40000, scaleResolutionDownBy: 2 },
+            { rid: rids[1], maxBitrate: 40000 }
+          ]
+    });
+
+    const answer = await answerer.createAnswer();
+
+    const mungedAnswer = ridToMid(answer);
+    info(`Transformed send simulcast answer to multiple m-sections: ${answer.sdp} to ${mungedAnswer}`);
+    await offerer.setRemoteDescription({type: "answer", sdp: mungedAnswer});
+    await answerer.setLocalDescription(answer);
+
+    is(metadataToBeLoaded.length, 2, 'Offerer should have gotten 2 ontrack events');
+    info('Waiting for 2 loadedmetadata events');
+    const videoElems = await Promise.all(metadataToBeLoaded);
+
+    const statsReady =
+      Promise.all([waitForSyncedRtcp(offerer), waitForSyncedRtcp(answerer)]);
+
+    const helper = new VideoStreamHelper();
+    info('Waiting for first video element to start playing');
+    await helper.checkVideoPlaying(videoElems[0]);
+    info('Waiting for second video element to start playing');
+    await helper.checkVideoPlaying(videoElems[1]);
+
+    is(videoElems[1].videoWidth, 50,
+       "sink is same width as source, modulo our cropping algorithm");
+    is(videoElems[1].videoHeight, 50,
+       "sink is same height as source, modulo our cropping algorithm");
+    is(videoElems[0].videoWidth, 25,
+       "sink is 1/2 width of source, modulo our cropping algorithm");
+    is(videoElems[0].videoHeight, 25,
+       "sink is 1/2 height of source, modulo our cropping algorithm");
+
+    await statsReady;
+    const senderStats = await sender.getStats();
+    checkSenderStats(senderStats, 2);
+    checkExpectedFields(senderStats);
+    pedanticChecks(senderStats);
+
+    emitter.stop();
+    videoStream.getVideoTracks()[0].stop();
+    offerer.close();
+    answerer.close();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastAnswer_oldSetParameters.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastAnswer_oldSetParameters.html
new file mode 100644
index 0000000000..bc0b9f71cc
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastAnswer_oldSetParameters.html
@@ -0,0 +1,115 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="helpers_from_wpt/sdp.js"></script>
+  <script type="application/javascript" src="simulcast.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1231507",
+    title: "Basic video-only peer connection with Simulcast answer",
+    visible: true
+  });
+
+  runNetworkTest(async () => {
+    await pushPrefs(
+        ['media.peerconnection.simulcast', true],
+        // 180Kbps was determined empirically, set well-higher than
+        // the 80Kbps+overhead needed for the two simulcast streams.
+        // 100Kbps was apparently too low.
+        ['media.peerconnection.video.min_bitrate_estimate', 180*1000]);
+
+
+    const offerer = new RTCPeerConnection();
+    const answerer = new RTCPeerConnection();
+
+    const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    offerer.onicecandidate = e => add(answerer, e.candidate, generateErrorCallback());
+    answerer.onicecandidate = e => add(offerer, e.candidate, generateErrorCallback());
+
+    const metadataToBeLoaded = [];
+    offerer.ontrack = (e) => {
+      metadataToBeLoaded.push(getPlaybackWithLoadedMetadata(e.track));
+    };
+
+    // Two recv transceivers, one for each simulcast stream
+    offerer.addTransceiver('video', { direction: 'recvonly' });
+    offerer.addTransceiver('video', { direction: 'recvonly' });
+
+    // One send transceiver, that will be used to send both simulcast streams
+    const emitter = new VideoFrameEmitter();
+    const videoStream = emitter.stream();
+    answerer.addTrack(videoStream.getVideoTracks()[0], videoStream);
+    emitter.start();
+
+    const offer = await offerer.createOffer();
+
+    const mungedOffer = midToRid(offer);
+    info(`Transformed recv offer to simulcast: ${offer.sdp} to ${mungedOffer}`);
+
+    await answerer.setRemoteDescription({type: "offer", sdp: mungedOffer});
+    await offerer.setLocalDescription(offer);
+
+    const rids = offerer.getTransceivers().map(t => t.mid);
+    is(rids.length, 2, 'Should have 2 mids in offer');
+    ok(rids[0] != '', 'First mid should be non-empty');
+    ok(rids[1] != '', 'Second mid should be non-empty');
+    info(`rids: ${JSON.stringify(rids)}`);
+
+    const sender = answerer.getSenders()[0];
+    await sender.setParameters({
+        encodings: [
+            { rid: rids[0], maxBitrate: 40000 },
+            { rid: rids[1], maxBitrate: 40000, scaleResolutionDownBy: 2 }
+          ]
+    });
+
+    const answer = await answerer.createAnswer();
+
+    const mungedAnswer = ridToMid(answer);
+    info(`Transformed send simulcast answer to multiple m-sections: ${answer.sdp} to ${mungedAnswer}`);
+    await offerer.setRemoteDescription({type: "answer", sdp: mungedAnswer});
+    await answerer.setLocalDescription(answer);
+
+    is(metadataToBeLoaded.length, 2, 'Offerer should have gotten 2 ontrack events');
+    info('Waiting for 2 loadedmetadata events');
+    const videoElems = await Promise.all(metadataToBeLoaded);
+
+    const statsReady =
+      Promise.all([waitForSyncedRtcp(offerer), waitForSyncedRtcp(answerer)]);
+
+    const helper = new VideoStreamHelper();
+    info('Waiting for first video element to start playing');
+    await helper.checkVideoPlaying(videoElems[0]);
+    info('Waiting for second video element to start playing');
+    await helper.checkVideoPlaying(videoElems[1]);
+
+    is(videoElems[0].videoWidth, 50,
+       "sink is same width as source, modulo our cropping algorithm");
+    is(videoElems[0].videoHeight, 50,
+       "sink is same height as source, modulo our cropping algorithm");
+    is(videoElems[1].videoWidth, 25,
+       "sink is 1/2 width of source, modulo our cropping algorithm");
+    is(videoElems[1].videoHeight, 25,
+       "sink is 1/2 height of source, modulo our cropping algorithm");
+
+    await statsReady;
+    const senderStats = await sender.getStats();
+    checkSenderStats(senderStats, 2);
+    checkExpectedFields(senderStats);
+    pedanticChecks(senderStats);
+
+    emitter.stop();
+    videoStream.getVideoTracks()[0].stop();
+    offerer.close();
+    answerer.close();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOddResolution.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOddResolution.html
new file mode 100644
index 0000000000..c380b34f1a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOddResolution.html
@@ -0,0 +1,183 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="helpers_from_wpt/sdp.js"></script>
+  <script type="application/javascript" src="simulcast.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1432793",
+    title: "Simulcast with odd resolution",
+    visible: true
+  });
+
+  runNetworkTest(async () => {
+    const helper = new VideoStreamHelper();
+    const emitter = new VideoFrameEmitter(helper.green, helper.red, 705, 528);
+
+    async function checkVideoElement(senderElement, receiverElement, encoding) {
+      info(`Waiting for receiver video element ${encoding.rid} to start playing`);
+      await helper.checkVideoPlaying(receiverElement);
+      const srcWidth = senderElement.videoWidth;
+      const srcHeight = senderElement.videoHeight;
+      info(`Source resolution is ${srcWidth}x${srcHeight}`);
+
+      const scaleDownBy = encoding.scaleResolutionDownBy;
+      const expectedWidth = srcWidth / scaleDownBy;
+      const expectedHeight = srcHeight / scaleDownBy;
+      const margin = srcWidth * 0.1;
+      const width = receiverElement.videoWidth;
+      const height = receiverElement.videoHeight;
+      const rid = encoding.rid;
+      ok(width >= expectedWidth - margin && width <= expectedWidth + margin,
+        `Width ${width} should be within 10% of ${expectedWidth} for rid '${rid}'`);
+      ok(height >= expectedHeight - margin && height <= expectedHeight + margin,
+        `Height ${height} should be within 10% of ${expectedHeight} for rid '${rid}'`);
+    }
+
+    async function checkVideoElements(senderElement, receiverElements, encodings) {
+      is(receiverElements.length, encodings.length, 'Number of video elements should match number of encodings');
+      info('Waiting for sender video element to start playing');
+      await helper.checkVideoPlaying(senderElement);
+      for (let i = 0; i < encodings.length; i++) {
+        await checkVideoElement(senderElement, receiverElements[i], encodings[i]);
+      }
+    }
+
+    const sendEncodings = [{ rid: "0", maxBitrate: 40000, scaleResolutionDownBy: 1.9 },
+                           { rid: "1", maxBitrate: 40000, scaleResolutionDownBy: 3.5 },
+                           { rid: "2", maxBitrate: 40000, scaleResolutionDownBy: 17.8 }];
+
+    async function checkSenderStats(sender) {
+      const senderStats = await sender.getStats();
+      checkSenderStats(senderStats, sendEncodings.length);
+      checkExpectedFields(senderStats);
+      pedanticChecks(senderStats);
+    }
+
+    async function waitForResizeEvents(elements) {
+      return Promise.all(elements.map(elem => haveEvent(elem, 'resize')));
+    }
+
+    await pushPrefs(
+        // 180Kbps was determined empirically, set well-higher than
+        // the 80Kbps+overhead needed for the two simulcast streams.
+        // 100Kbps was apparently too low.
+        ['media.peerconnection.video.min_bitrate_estimate', 180*1000]);
+
+
+    const offerer = new RTCPeerConnection();
+    const answerer = new RTCPeerConnection();
+
+    const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    offerer.onicecandidate = e => add(answerer, e.candidate, generateErrorCallback());
+    answerer.onicecandidate = e => add(offerer, e.candidate, generateErrorCallback());
+
+    const metadataToBeLoaded = [];
+    answerer.ontrack = (e) => {
+      metadataToBeLoaded.push(getPlaybackWithLoadedMetadata(e.track));
+    };
+
+    // One send transceiver, that will be used to send both simulcast streams
+    const videoStream = emitter.stream();
+    offerer.addTransceiver(videoStream.getVideoTracks()[0], {sendEncodings});
+    const senderElement = document.createElement('video');
+    senderElement.autoplay = true;
+    senderElement.srcObject = videoStream;
+    senderElement.id = videoStream.id
+
+    const sender = offerer.getSenders()[0];
+    let parameters = sender.getParameters();
+    is(parameters.encodings[0].maxBitrate, sendEncodings[0].maxBitrate);
+    isfuzzy(parameters.encodings[0].scaleResolutionDownBy,
+      sendEncodings[0].scaleResolutionDownBy, 0.01);
+    is(parameters.encodings[1].maxBitrate, sendEncodings[1].maxBitrate);
+    isfuzzy(parameters.encodings[1].scaleResolutionDownBy,
+      sendEncodings[1].scaleResolutionDownBy, 0.01);
+    is(parameters.encodings[2].maxBitrate, sendEncodings[2].maxBitrate);
+    isfuzzy(parameters.encodings[2].scaleResolutionDownBy,
+      sendEncodings[2].scaleResolutionDownBy, 0.01);
+
+    const offer = await offerer.createOffer();
+
+    const mungedOffer = ridToMid(offer);
+    info(`Transformed send simulcast offer to multiple m-sections: ${offer.sdp} to ${mungedOffer}`);
+
+    await answerer.setRemoteDescription({type: 'offer', sdp: mungedOffer});
+    await offerer.setLocalDescription(offer);
+
+    const rids = answerer.getTransceivers().map(t => t.mid);
+    is(rids.length, 3, 'Should have 3 mids in offer');
+    ok(rids[0], 'First mid should be non-empty');
+    ok(rids[1], 'Second mid should be non-empty');
+    ok(rids[2], 'Third mid should be non-empty');
+    info(`rids: ${JSON.stringify(rids)}`);
+
+    const answer = await answerer.createAnswer();
+
+    const mungedAnswer = midToRid(answer);
+    info(`Transformed recv answer to simulcast: ${answer.sdp} to ${mungedAnswer}`);
+    await offerer.setRemoteDescription({type: 'answer', sdp: mungedAnswer});
+    await answerer.setLocalDescription(answer);
+
+    is(metadataToBeLoaded.length, 3, 'Offerer should have gotten 3 ontrack events');
+    emitter.start();
+    info('Waiting for 3 loadedmetadata events');
+    const videoElems = await Promise.all(metadataToBeLoaded);
+    await checkVideoElements(senderElement, videoElems, parameters.encodings);
+    emitter.stop();
+
+    await Promise.all([waitForSyncedRtcp(offerer), waitForSyncedRtcp(answerer)]);
+
+    info(`Changing source resolution to 1280x720`);
+    emitter.size(1280, 720);
+    emitter.start();
+    await waitForResizeEvents([senderElement, ...videoElems]);
+    await checkVideoElements(senderElement, videoElems, parameters.encodings);
+    await checkSenderStats(sender);
+
+    parameters = sender.getParameters();
+    parameters.encodings[0].scaleResolutionDownBy = 1;
+    parameters.encodings[1].scaleResolutionDownBy = 2;
+    parameters.encodings[2].scaleResolutionDownBy = 3;
+    info(`Changing encodings to ${JSON.stringify(parameters.encodings)}`);
+    await sender.setParameters(parameters);
+    await waitForResizeEvents(videoElems);
+    await checkVideoElements(senderElement, videoElems, parameters.encodings);
+    await checkSenderStats(sender);
+
+    parameters = sender.getParameters();
+    parameters.encodings[0].scaleResolutionDownBy = 6;
+    parameters.encodings[1].scaleResolutionDownBy = 5;
+    parameters.encodings[2].scaleResolutionDownBy = 4;
+    info(`Changing encodings to ${JSON.stringify(parameters.encodings)}`);
+    await sender.setParameters(parameters);
+    await waitForResizeEvents(videoElems);
+    await checkVideoElements(senderElement, videoElems, parameters.encodings);
+    await checkSenderStats(sender);
+
+    parameters = sender.getParameters();
+    parameters.encodings[0].scaleResolutionDownBy = 4;
+    parameters.encodings[1].scaleResolutionDownBy = 1;
+    parameters.encodings[2].scaleResolutionDownBy = 2;
+    info(`Changing encodings to ${JSON.stringify(parameters.encodings)}`);
+    await sender.setParameters(parameters);
+    await waitForResizeEvents(videoElems);
+    await checkVideoElements(senderElement, videoElems, parameters.encodings);
+    await checkSenderStats(sender);
+
+    emitter.stop();
+    videoStream.getVideoTracks()[0].stop();
+    offerer.close();
+    answerer.close();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOddResolution_oldSetParameters.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOddResolution_oldSetParameters.html
new file mode 100644
index 0000000000..0f6d3c8520
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOddResolution_oldSetParameters.html
@@ -0,0 +1,172 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="helpers_from_wpt/sdp.js"></script>
+  <script type="application/javascript" src="simulcast.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1432793",
+    title: "Simulcast with odd resolution",
+    visible: true
+  });
+
+  runNetworkTest(async () => {
+    const helper = new VideoStreamHelper();
+    const emitter = new VideoFrameEmitter(helper.green, helper.red, 705, 528);
+
+    async function checkVideoElement(senderElement, receiverElement, encoding) {
+      info(`Waiting for receiver video element ${encoding.rid} to start playing`);
+      await helper.checkVideoPlaying(receiverElement);
+      const srcWidth = senderElement.videoWidth;
+      const srcHeight = senderElement.videoHeight;
+      info(`Source resolution is ${srcWidth}x${srcHeight}`);
+
+      const scaleDownBy = encoding.scaleResolutionDownBy;
+      const expectedWidth = srcWidth / scaleDownBy;
+      const expectedHeight = srcHeight / scaleDownBy;
+      const margin = srcWidth * 0.1;
+      const width = receiverElement.videoWidth;
+      const height = receiverElement.videoHeight;
+      const rid = encoding.rid;
+      ok(width >= expectedWidth - margin && width <= expectedWidth + margin,
+        `Width ${width} should be within 10% of ${expectedWidth} for rid '${rid}'`);
+      ok(height >= expectedHeight - margin && height <= expectedHeight + margin,
+        `Height ${height} should be within 10% of ${expectedHeight} for rid '${rid}'`);
+    }
+
+    async function checkVideoElements(senderElement, receiverElements, encodings) {
+      is(receiverElements.length, encodings.length, 'Number of video elements should match number of encodings');
+      info('Waiting for sender video element to start playing');
+      await helper.checkVideoPlaying(senderElement);
+      for (let i = 0; i < encodings.length; i++) {
+        await checkVideoElement(senderElement, receiverElements[i], encodings[i]);
+      }
+    }
+
+    async function checkSenderStats(sender) {
+      const senderStats = await sender.getStats();
+      checkSenderStats(senderStats, encodings.length);
+      checkExpectedFields(senderStats);
+      pedanticChecks(senderStats);
+    }
+
+    async function waitForResizeEvents(elements) {
+      return Promise.all(elements.map(elem => haveEvent(elem, 'resize')));
+    }
+
+    const encodings = [{ rid: "0", maxBitrate: 40000, scaleResolutionDownBy: 1.9 },
+                       { rid: "1", maxBitrate: 40000, scaleResolutionDownBy: 3.5 },
+                       { rid: "2", maxBitrate: 40000, scaleResolutionDownBy: 17.8 }];
+
+    await pushPrefs(
+        ['media.peerconnection.simulcast', true],
+        // 180Kbps was determined empirically, set well-higher than
+        // the 80Kbps+overhead needed for the two simulcast streams.
+        // 100Kbps was apparently too low.
+        ['media.peerconnection.video.min_bitrate_estimate', 180*1000]);
+
+
+    const offerer = new RTCPeerConnection();
+    const answerer = new RTCPeerConnection();
+
+    const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    offerer.onicecandidate = e => add(answerer, e.candidate, generateErrorCallback());
+    answerer.onicecandidate = e => add(offerer, e.candidate, generateErrorCallback());
+
+    const metadataToBeLoaded = [];
+    answerer.ontrack = (e) => {
+      metadataToBeLoaded.push(getPlaybackWithLoadedMetadata(e.track));
+    };
+
+    // One send transceiver, that will be used to send both simulcast streams
+    const videoStream = emitter.stream();
+    offerer.addTrack(videoStream.getVideoTracks()[0], videoStream);
+    const senderElement = document.createElement('video');
+    senderElement.autoplay = true;
+    senderElement.srcObject = videoStream;
+    senderElement.id = videoStream.id
+
+    const sender = offerer.getSenders()[0];
+    sender.setParameters({encodings});
+
+    const offer = await offerer.createOffer();
+
+    const mungedOffer = ridToMid(offer);
+    info(`Transformed send simulcast offer to multiple m-sections: ${offer.sdp} to ${mungedOffer}`);
+
+    await answerer.setRemoteDescription({type: "offer", sdp: mungedOffer});
+    await offerer.setLocalDescription(offer);
+
+    const rids = answerer.getTransceivers().map(t => t.mid);
+    is(rids.length, 3, 'Should have 3 mids in offer');
+    ok(rids[0], 'First mid should be non-empty');
+    ok(rids[1], 'Second mid should be non-empty');
+    ok(rids[2], 'Third mid should be non-empty');
+    info(`rids: ${JSON.stringify(rids)}`);
+
+    const answer = await answerer.createAnswer();
+
+    const mungedAnswer = midToRid(answer);
+    info(`Transformed recv answer to simulcast: ${answer.sdp} to ${mungedAnswer}`);
+    await offerer.setRemoteDescription({type: "answer", sdp: mungedAnswer});
+    await answerer.setLocalDescription(answer);
+
+    is(metadataToBeLoaded.length, 3, 'Offerer should have gotten 3 ontrack events');
+    emitter.start();
+    info('Waiting for 3 loadedmetadata events');
+    const videoElems = await Promise.all(metadataToBeLoaded);
+    await checkVideoElements(senderElement, videoElems, encodings);
+    emitter.stop();
+
+    await Promise.all([waitForSyncedRtcp(offerer), waitForSyncedRtcp(answerer)]);
+
+    info(`Changing source resolution to 1280x720`);
+    emitter.size(1280, 720);
+    emitter.start();
+    await waitForResizeEvents([senderElement, ...videoElems]);
+    await checkVideoElements(senderElement, videoElems, encodings);
+    await checkSenderStats(sender);
+
+    encodings[0].scaleResolutionDownBy = 1;
+    encodings[1].scaleResolutionDownBy = 2;
+    encodings[2].scaleResolutionDownBy = 3;
+    info(`Changing encodings to ${JSON.stringify(encodings)}`);
+    await sender.setParameters({encodings});
+    await waitForResizeEvents(videoElems);
+    await checkVideoElements(senderElement, videoElems, encodings);
+    await checkSenderStats(sender);
+
+    encodings[0].scaleResolutionDownBy = 6;
+    encodings[1].scaleResolutionDownBy = 5;
+    encodings[2].scaleResolutionDownBy = 4;
+    info(`Changing encodings to ${JSON.stringify(encodings)}`);
+    await sender.setParameters({encodings});
+    await waitForResizeEvents(videoElems);
+    await checkVideoElements(senderElement, videoElems, encodings);
+    await checkSenderStats(sender);
+
+    encodings[0].scaleResolutionDownBy = 4;
+    encodings[1].scaleResolutionDownBy = 1;
+    encodings[2].scaleResolutionDownBy = 2;
+    info(`Changing encodings to ${JSON.stringify(encodings)}`);
+    await sender.setParameters({encodings});
+    await waitForResizeEvents(videoElems);
+    await checkVideoElements(senderElement, videoElems, encodings);
+    await checkSenderStats(sender);
+
+    emitter.stop();
+    videoStream.getVideoTracks()[0].stop();
+    offerer.close();
+    answerer.close();
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOffer.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOffer.html
new file mode 100644
index 0000000000..cb7c13a0d1
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOffer.html
@@ -0,0 +1,109 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="parser_rtp.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="helpers_from_wpt/sdp.js"></script>
+  <script type="application/javascript" src="simulcast.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1231507",
+    title: "Basic video-only peer connection with Simulcast offer",
+    visible: true
+  });
+
+  runNetworkTest(async () => {
+    await pushPrefs(
+        // 180Kbps was determined empirically, set well-higher than
+        // the 80Kbps+overhead needed for the two simulcast streams.
+        // 100Kbps was apparently too low.
+        ['media.peerconnection.video.min_bitrate_estimate', 180*1000]);
+
+
+    const offerer = new RTCPeerConnection();
+    const answerer = new RTCPeerConnection();
+
+    const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    offerer.onicecandidate = e => add(answerer, e.candidate, generateErrorCallback());
+    answerer.onicecandidate = e => add(offerer, e.candidate, generateErrorCallback());
+
+    const metadataToBeLoaded = [];
+    answerer.ontrack = (e) => {
+      metadataToBeLoaded.push(getPlaybackWithLoadedMetadata(e.track));
+    };
+
+    // One send transceiver, that will be used to send both simulcast streams
+    const emitter = new VideoFrameEmitter();
+    const videoStream = emitter.stream();
+    const sendEncodings = [
+            { rid: '0', maxBitrate: 40000 },
+            { rid: '1', maxBitrate: 40000, scaleResolutionDownBy: 2 }
+          ];
+    offerer.addTransceiver(videoStream.getVideoTracks()[0], {sendEncodings});
+    emitter.start();
+
+    const sender = offerer.getSenders()[0];
+
+    const offer = await offerer.createOffer();
+
+    const mungedOffer = ridToMid(offer);
+    info(`Transformed send simulcast offer to multiple m-sections: ${offer.sdp} to ${mungedOffer}`);
+
+    await answerer.setRemoteDescription({type: 'offer', sdp: mungedOffer});
+    await offerer.setLocalDescription(offer);
+
+    const rids = answerer.getTransceivers().map(t => t.mid);
+    is(rids.length, 2, 'Should have 2 mids in offer');
+    ok(rids[0] != '', 'First mid should be non-empty');
+    ok(rids[1] != '', 'Second mid should be non-empty');
+    info(`rids: ${JSON.stringify(rids)}`);
+
+    const answer = await answerer.createAnswer();
+
+    const mungedAnswer = midToRid(answer);
+    info(`Transformed recv answer to simulcast: ${answer.sdp} to ${mungedAnswer}`);
+    await offerer.setRemoteDescription({type: 'answer', sdp: mungedAnswer});
+    await answerer.setLocalDescription(answer);
+
+    is(metadataToBeLoaded.length, 2, 'Offerer should have gotten 2 ontrack events');
+    info('Waiting for 2 loadedmetadata events');
+    const videoElems = await Promise.all(metadataToBeLoaded);
+
+    const statsReady =
+      Promise.all([waitForSyncedRtcp(offerer), waitForSyncedRtcp(answerer)]);
+
+    const helper = new VideoStreamHelper();
+    info('Waiting for first video element to start playing');
+    await helper.checkVideoPlaying(videoElems[0]);
+    info('Waiting for second video element to start playing');
+    await helper.checkVideoPlaying(videoElems[1]);
+
+    is(videoElems[0].videoWidth, 50,
+       "sink is same width as source, modulo our cropping algorithm");
+    is(videoElems[0].videoHeight, 50,
+       "sink is same height as source, modulo our cropping algorithm");
+    is(videoElems[1].videoWidth, 25,
+       "sink is 1/2 width of source, modulo our cropping algorithm");
+    is(videoElems[1].videoHeight, 25,
+       "sink is 1/2 height of source, modulo our cropping algorithm");
+
+    await statsReady;
+    const senderStats = await sender.getStats();
+    checkSenderStats(senderStats, 2);
+    checkExpectedFields(senderStats);
+    pedanticChecks(senderStats);
+
+    emitter.stop();
+    videoStream.getVideoTracks()[0].stop();
+    offerer.close();
+    answerer.close();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOffer_lowResFirst.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOffer_lowResFirst.html
new file mode 100644
index 0000000000..93141311f1
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOffer_lowResFirst.html
@@ -0,0 +1,109 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="parser_rtp.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="helpers_from_wpt/sdp.js"></script>
+  <script type="application/javascript" src="simulcast.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1231507",
+    title: "Basic video-only peer connection with Simulcast offer, first rid has lowest resolution",
+    visible: true
+  });
+
+  runNetworkTest(async () => {
+    await pushPrefs(
+        // 180Kbps was determined empirically, set well-higher than
+        // the 80Kbps+overhead needed for the two simulcast streams.
+        // 100Kbps was apparently too low.
+        ['media.peerconnection.video.min_bitrate_estimate', 180*1000]);
+
+
+    const offerer = new RTCPeerConnection();
+    const answerer = new RTCPeerConnection();
+
+    const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    offerer.onicecandidate = e => add(answerer, e.candidate, generateErrorCallback());
+    answerer.onicecandidate = e => add(offerer, e.candidate, generateErrorCallback());
+
+    const metadataToBeLoaded = [];
+    answerer.ontrack = (e) => {
+      metadataToBeLoaded.push(getPlaybackWithLoadedMetadata(e.track));
+    };
+
+    // One send transceiver, that will be used to send both simulcast streams
+    const emitter = new VideoFrameEmitter();
+    const videoStream = emitter.stream();
+    const sendEncodings = [
+      { rid: '0', maxBitrate: 40000, scaleResolutionDownBy: 2 },
+      { rid: '1', maxBitrate: 40000 }
+    ];
+    offerer.addTransceiver(videoStream.getVideoTracks()[0], {sendEncodings});
+    emitter.start();
+
+    const sender = offerer.getSenders()[0];
+
+    const offer = await offerer.createOffer();
+
+    const mungedOffer = ridToMid(offer);
+    info(`Transformed send simulcast offer to multiple m-sections: ${offer.sdp} to ${mungedOffer}`);
+
+    await answerer.setRemoteDescription({type: 'offer', sdp: mungedOffer});
+    await offerer.setLocalDescription(offer);
+
+    const rids = answerer.getTransceivers().map(t => t.mid);
+    is(rids.length, 2, 'Should have 2 mids in offer');
+    ok(rids[0] != '', 'First mid should be non-empty');
+    ok(rids[1] != '', 'Second mid should be non-empty');
+    info(`rids: ${JSON.stringify(rids)}`);
+
+    const answer = await answerer.createAnswer();
+
+    const mungedAnswer = midToRid(answer);
+    info(`Transformed recv answer to simulcast: ${answer.sdp} to ${mungedAnswer}`);
+    await offerer.setRemoteDescription({type: 'answer', sdp: mungedAnswer});
+    await answerer.setLocalDescription(answer);
+
+    is(metadataToBeLoaded.length, 2, 'Offerer should have gotten 2 ontrack events');
+    info('Waiting for 2 loadedmetadata events');
+    const videoElems = await Promise.all(metadataToBeLoaded);
+
+    const statsReady =
+      Promise.all([waitForSyncedRtcp(offerer), waitForSyncedRtcp(answerer)]);
+
+    const helper = new VideoStreamHelper();
+    info('Waiting for first video element to start playing');
+    await helper.checkVideoPlaying(videoElems[0]);
+    info('Waiting for second video element to start playing');
+    await helper.checkVideoPlaying(videoElems[1]);
+
+    is(videoElems[1].videoWidth, 50,
+       "sink is same width as source, modulo our cropping algorithm");
+    is(videoElems[1].videoHeight, 50,
+       "sink is same height as source, modulo our cropping algorithm");
+    is(videoElems[0].videoWidth, 25,
+       "sink is 1/2 width of source, modulo our cropping algorithm");
+    is(videoElems[0].videoHeight, 25,
+       "sink is 1/2 height of source, modulo our cropping algorithm");
+
+    await statsReady;
+    const senderStats = await sender.getStats();
+    checkSenderStats(senderStats, 2);
+    checkExpectedFields(senderStats);
+    pedanticChecks(senderStats);
+
+    emitter.stop();
+    videoStream.getVideoTracks()[0].stop();
+    offerer.close();
+    answerer.close();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOffer_lowResFirst_oldSetParameters.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOffer_lowResFirst_oldSetParameters.html
new file mode 100644
index 0000000000..73e2d38eb2
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOffer_lowResFirst_oldSetParameters.html
@@ -0,0 +1,112 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="parser_rtp.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="helpers_from_wpt/sdp.js"></script>
+  <script type="application/javascript" src="simulcast.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1231507",
+    title: "Basic video-only peer connection with Simulcast offer, first rid has lowest resolution",
+    visible: true
+  });
+
+  runNetworkTest(async () => {
+    await pushPrefs(
+        ['media.peerconnection.simulcast', true],
+        // 180Kbps was determined empirically, set well-higher than
+        // the 80Kbps+overhead needed for the two simulcast streams.
+        // 100Kbps was apparently too low.
+        ['media.peerconnection.video.min_bitrate_estimate', 180*1000]);
+
+
+    const offerer = new RTCPeerConnection();
+    const answerer = new RTCPeerConnection();
+
+    const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    offerer.onicecandidate = e => add(answerer, e.candidate, generateErrorCallback());
+    answerer.onicecandidate = e => add(offerer, e.candidate, generateErrorCallback());
+
+    const metadataToBeLoaded = [];
+    answerer.ontrack = (e) => {
+      metadataToBeLoaded.push(getPlaybackWithLoadedMetadata(e.track));
+    };
+
+    // One send transceiver, that will be used to send both simulcast streams
+    const emitter = new VideoFrameEmitter();
+    const videoStream = emitter.stream();
+    offerer.addTrack(videoStream.getVideoTracks()[0], videoStream);
+    emitter.start();
+
+    const sender = offerer.getSenders()[0];
+    sender.setParameters({
+        encodings: [
+            { rid: '0', maxBitrate: 40000, scaleResolutionDownBy: 2 },
+            { rid: '1', maxBitrate: 40000 }
+          ]
+    });
+
+    const offer = await offerer.createOffer();
+
+    const mungedOffer = ridToMid(offer);
+    info(`Transformed send simulcast offer to multiple m-sections: ${offer.sdp} to ${mungedOffer}`);
+
+    await answerer.setRemoteDescription({type: 'offer', sdp: mungedOffer});
+    await offerer.setLocalDescription(offer);
+
+    const rids = answerer.getTransceivers().map(t => t.mid);
+    is(rids.length, 2, 'Should have 2 mids in offer');
+    ok(rids[0] != '', 'First mid should be non-empty');
+    ok(rids[1] != '', 'Second mid should be non-empty');
+    info(`rids: ${JSON.stringify(rids)}`);
+
+    const answer = await answerer.createAnswer();
+
+    const mungedAnswer = midToRid(answer);
+    info(`Transformed recv answer to simulcast: ${answer.sdp} to ${mungedAnswer}`);
+    await offerer.setRemoteDescription({type: 'answer', sdp: mungedAnswer});
+    await answerer.setLocalDescription(answer);
+
+    is(metadataToBeLoaded.length, 2, 'Offerer should have gotten 2 ontrack events');
+    info('Waiting for 2 loadedmetadata events');
+    const videoElems = await Promise.all(metadataToBeLoaded);
+
+    const statsReady =
+      Promise.all([waitForSyncedRtcp(offerer), waitForSyncedRtcp(answerer)]);
+
+    const helper = new VideoStreamHelper();
+    info('Waiting for first video element to start playing');
+    await helper.checkVideoPlaying(videoElems[0]);
+    info('Waiting for second video element to start playing');
+    await helper.checkVideoPlaying(videoElems[1]);
+
+    is(videoElems[1].videoWidth, 50,
+       "sink is same width as source, modulo our cropping algorithm");
+    is(videoElems[1].videoHeight, 50,
+       "sink is same height as source, modulo our cropping algorithm");
+    is(videoElems[0].videoWidth, 25,
+       "sink is 1/2 width of source, modulo our cropping algorithm");
+    is(videoElems[0].videoHeight, 25,
+       "sink is 1/2 height of source, modulo our cropping algorithm");
+
+    await statsReady;
+    const senderStats = await sender.getStats();
+    checkSenderStats(senderStats, 2);
+    checkExpectedFields(senderStats);
+    pedanticChecks(senderStats);
+
+    emitter.stop();
+    videoStream.getVideoTracks()[0].stop();
+    offerer.close();
+    answerer.close();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOffer_oldSetParameters.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOffer_oldSetParameters.html
new file mode 100644
index 0000000000..551273af5e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_simulcastOffer_oldSetParameters.html
@@ -0,0 +1,112 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="parser_rtp.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+  <script type="application/javascript" src="helpers_from_wpt/sdp.js"></script>
+  <script type="application/javascript" src="simulcast.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1231507",
+    title: "Basic video-only peer connection with Simulcast offer",
+    visible: true
+  });
+
+  runNetworkTest(async () => {
+    await pushPrefs(
+        ['media.peerconnection.simulcast', true],
+        // 180Kbps was determined empirically, set well-higher than
+        // the 80Kbps+overhead needed for the two simulcast streams.
+        // 100Kbps was apparently too low.
+        ['media.peerconnection.video.min_bitrate_estimate', 180*1000]);
+
+
+    const offerer = new RTCPeerConnection();
+    const answerer = new RTCPeerConnection();
+
+    const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+    offerer.onicecandidate = e => add(answerer, e.candidate, generateErrorCallback());
+    answerer.onicecandidate = e => add(offerer, e.candidate, generateErrorCallback());
+
+    const metadataToBeLoaded = [];
+    answerer.ontrack = (e) => {
+      metadataToBeLoaded.push(getPlaybackWithLoadedMetadata(e.track));
+    };
+
+    // One send transceiver, that will be used to send both simulcast streams
+    const emitter = new VideoFrameEmitter();
+    const videoStream = emitter.stream();
+    offerer.addTrack(videoStream.getVideoTracks()[0], videoStream);
+    emitter.start();
+
+    const sender = offerer.getSenders()[0];
+    sender.setParameters({
+        encodings: [
+            { rid: '0', maxBitrate: 40000 },
+            { rid: '1', maxBitrate: 40000, scaleResolutionDownBy: 2 }
+          ]
+    });
+
+    const offer = await offerer.createOffer();
+
+    const mungedOffer = ridToMid(offer);
+    info(`Transformed send simulcast offer to multiple m-sections: ${offer.sdp} to ${mungedOffer}`);
+
+    await answerer.setRemoteDescription({type: "offer", sdp: mungedOffer});
+    await offerer.setLocalDescription(offer);
+
+    const rids = answerer.getTransceivers().map(t => t.mid);
+    is(rids.length, 2, 'Should have 2 mids in offer');
+    ok(rids[0] != '', 'First mid should be non-empty');
+    ok(rids[1] != '', 'Second mid should be non-empty');
+    info(`rids: ${JSON.stringify(rids)}`);
+
+    const answer = await answerer.createAnswer();
+
+    const mungedAnswer = midToRid(answer);
+    info(`Transformed recv answer to simulcast: ${answer.sdp} to ${mungedAnswer}`);
+    await offerer.setRemoteDescription({type: "answer", sdp: mungedAnswer});
+    await answerer.setLocalDescription(answer);
+
+    is(metadataToBeLoaded.length, 2, 'Offerer should have gotten 2 ontrack events');
+    info('Waiting for 2 loadedmetadata events');
+    const videoElems = await Promise.all(metadataToBeLoaded);
+
+    const statsReady =
+      Promise.all([waitForSyncedRtcp(offerer), waitForSyncedRtcp(answerer)]);
+
+    const helper = new VideoStreamHelper();
+    info('Waiting for first video element to start playing');
+    await helper.checkVideoPlaying(videoElems[0]);
+    info('Waiting for second video element to start playing');
+    await helper.checkVideoPlaying(videoElems[1]);
+
+    is(videoElems[0].videoWidth, 50,
+       "sink is same width as source, modulo our cropping algorithm");
+    is(videoElems[0].videoHeight, 50,
+       "sink is same height as source, modulo our cropping algorithm");
+    is(videoElems[1].videoWidth, 25,
+       "sink is 1/2 width of source, modulo our cropping algorithm");
+    is(videoElems[1].videoHeight, 25,
+       "sink is 1/2 height of source, modulo our cropping algorithm");
+
+    await statsReady;
+    const senderStats = await sender.getStats();
+    checkSenderStats(senderStats, 2);
+    checkExpectedFields(senderStats);
+    pedanticChecks(senderStats);
+
+    emitter.stop();
+    videoStream.getVideoTracks()[0].stop();
+    offerer.close();
+    answerer.close();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_stats.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_stats.html
new file mode 100644
index 0000000000..2ef98dc9c8
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_stats.html
@@ -0,0 +1,42 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1337525",
+  title: "webRtc Stats composition and sanity"
+});
+
+runNetworkTest(async function (options) {
+  // We don't know how to get QP value when using Android system codecs.
+  if (navigator.userAgent.includes("Android")) {
+    await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                    ["media.webrtc.hw.h264.enabled", false]);
+  }
+
+  // For accurate comparisons of `remoteTimestamp` (not using reduced precision)
+  // to `timestamp` (using reduced precision).
+  await pushPrefs(["privacy.resistFingerprinting.reduceTimerPrecision.jitter",
+                   false]);
+
+  const test = new PeerConnectionTest(options);
+
+  test.chain.insertAfter("PC_LOCAL_WAIT_FOR_MEDIA_FLOW",
+    [PC_LOCAL_TEST_LOCAL_STATS]);
+
+  test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW",
+    [PC_REMOTE_TEST_REMOTE_STATS]);
+
+  test.setMediaConstraints([{audio: true}, {video: true}],
+                           [{audio: true}, {video: true}]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_stats_jitter.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_stats_jitter.html
new file mode 100644
index 0000000000..6e1ef698b4
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_stats_jitter.html
@@ -0,0 +1,58 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1672590",
+    title: "Jitter sanity check"
+  });
+
+const checkJitter = stats => {
+  stats.forEach((stat, mapKey) => {
+    if (stat.type == "remote-inbound-rtp") {
+      // This should be much lower for audio, TODO: Bug 1330575
+      const expectedJitter = stat.kind == "video" ? 0.5 : 1;
+
+      ok(stat.jitter < expectedJitter,
+          stat.type + ".jitter is sane number for a local only test. value="
+          + stat.jitter);
+    }
+  });
+};
+
+const PC_LOCAL_TEST_LOCAL_JITTER = async test => {
+  checkJitter(await waitForSyncedRtcp(test.pcLocal._pc));
+}
+
+const PC_REMOTE_TEST_REMOTE_JITTER = async test => {
+  checkJitter(await waitForSyncedRtcp(test.pcRemote._pc));
+}
+
+runNetworkTest(async function (options) {
+  // We don't know how to get QP value when using Android system codecs.
+  if (navigator.userAgent.includes("Android")) {
+    await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                    ["media.webrtc.hw.h264.enabled", false]);
+  }
+
+  const test = new PeerConnectionTest(options);
+
+  test.chain.insertAfter("PC_LOCAL_WAIT_FOR_MEDIA_FLOW",
+    [PC_LOCAL_TEST_LOCAL_JITTER]);
+
+  test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW",
+    [PC_REMOTE_TEST_REMOTE_JITTER]);
+
+  test.setMediaConstraints([{audio: true}, {video: true}],
+                           [{audio: true}, {video: true}]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_stats_oneway.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_stats_oneway.html
new file mode 100644
index 0000000000..02ace530a9
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_stats_oneway.html
@@ -0,0 +1,65 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1225722",
+  title: "WebRTC Stats composition and sanity for a one-way peer connection"
+});
+
+runNetworkTest(async function (options) {
+  // We don't know how to get QP value when using Android system codecs.
+  if (navigator.userAgent.includes("Android")) {
+    await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                    ["media.webrtc.hw.h264.enabled", false]);
+  }
+
+  // For accurate comparisons of `remoteTimestamp` (not using reduced precision)
+  // to `timestamp` (using reduced precision).
+  await pushPrefs(["privacy.resistFingerprinting.reduceTimerPrecision.jitter",
+                   false]);
+
+  const test = new PeerConnectionTest(options);
+
+  test.chain.insertAfter("PC_LOCAL_WAIT_FOR_MEDIA_FLOW",
+    [PC_LOCAL_TEST_LOCAL_STATS]);
+
+  test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW",
+    [PC_REMOTE_TEST_REMOTE_STATS]);
+
+  const testOneWayStats = (stats, codecType) => {
+    const codecs = [];
+    stats.forEach(stat => {
+      if (stat.type == "codec") {
+        codecs.push(stat);
+        is(stat.codecType, codecType, "One-way codec has specific codecType");
+      }
+    });
+    is(codecs.length, 2, "One audio and one video codec");
+    if (codecs.length == 2) {
+      isnot(codecs[0].mimeType.slice(0, 5), codecs[1].mimeType.slice(0, 5),
+        "Different media type for audio vs video mime types");
+    }
+  };
+
+  test.chain.append([
+    async function PC_LOCAL_TEST_CODECTYPE_ENCODE(test) {
+      testOneWayStats(await test.pcLocal._pc.getStats(), "encode");
+    },
+    async function PC_REMOTE_TEST_CODECTYPE_DECODE(test) {
+      testOneWayStats(await test.pcRemote._pc.getStats(), "decode");
+    },
+  ]);
+
+  test.setMediaConstraints([{audio: true}, {video: true}], []);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_stats_relayProtocol.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_stats_relayProtocol.html
new file mode 100644
index 0000000000..cdc328fd2b
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_stats_relayProtocol.html
@@ -0,0 +1,58 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="nonTrickleIce.js"></script>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1435789",
+  title: "WebRTC local-candidate relayProtocol stats attribute"
+});
+
+// This test uses the NAT simulator in order to get srflx candidates.
+// It doesn't work in https, so we turn on getUserMedia in http, which requires
+// a reload.
+if (!("mediaDevices" in navigator)) {
+  SpecialPowers.pushPrefEnv({set: [['media.devices.insecure.enabled', true]]},
+                            () => location.reload());
+} else {
+  runNetworkTest(async (options = {}) => {
+    await pushPrefs(
+        ['media.peerconnection.nat_simulator.filtering_type', 'ENDPOINT_INDEPENDENT'],
+        ['media.peerconnection.nat_simulator.mapping_type', 'ENDPOINT_INDEPENDENT'],
+        ['media.getusermedia.insecure.enabled', true]);
+    const test = new PeerConnectionTest(options);
+    makeOffererNonTrickle(test.chain);
+    makeAnswererNonTrickle(test.chain);
+
+    test.chain.removeAfter("PC_LOCAL_WAIT_FOR_MEDIA_FLOW");
+    test.chain.append([PC_LOCAL_TEST_LOCAL_STATS_RELAYCANDIDATE]);
+
+    test.setMediaConstraints([{ audio: true }], [{ audio: true }]);
+    await test.run();
+  }, { useIceServer: true });
+}
+
+const PC_LOCAL_TEST_LOCAL_STATS_RELAYCANDIDATE = test => {
+  return test.pcLocal.getStats().then(stats => {
+    let haveRelayProtocol = {};
+    for (let [k, v] of stats) {
+      if (v.type == "local-candidate") {
+        haveRelayProtocol[v.candidateType + "-" + v.relayProtocol] = v.relayProtocol;
+      }
+    }
+    is(haveRelayProtocol["host-undefined"], undefined, "relayProtocol not set for host candidates");
+    is(haveRelayProtocol["srflx-undefined"], undefined, "relayProtocol not set for server reflexive candidates");
+    ok(haveRelayProtocol["relay-udp"], "Has UDP relay candidate");
+    ok(haveRelayProtocol["relay-tcp"], "Has TCP relay candidate");
+    ok(haveRelayProtocol["relay-tls"], "Has TLS relay candidate");
+    is(Object.keys(haveRelayProtocol).length, 5, "All candidate types are accounted for");
+  });
+}
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_stereoFmtpPref.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_stereoFmtpPref.html
new file mode 100644
index 0000000000..ab7811fe82
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_stereoFmtpPref.html
@@ -0,0 +1,61 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1793776",
+    title: "Test that media.peerconnection.sdp.disable_stereo_fmtp works"
+  });
+
+  const tests = [
+    async function testStereo() {
+      const offerer = new RTCPeerConnection();
+      offerer.addTransceiver('audio');
+      const answerer = new RTCPeerConnection();
+      await offerer.setLocalDescription();
+      ok(offerer.localDescription.sdp.includes('stereo=1'),
+          'Offer uses stereo=1 when media.peerconnection.sdp.disable_stereo_fmtp is not set');
+      await answerer.setRemoteDescription(offerer.localDescription);
+      const {sdp} = await answerer.createAnswer();
+      ok(sdp.includes('stereo=1'), 'Answer uses stereo=1 when media.peerconnection.sdp.disable_stereo_fmtp is not set');
+    },
+
+    async function testNoStereo() {
+      await pushPrefs(
+          ['media.peerconnection.sdp.disable_stereo_fmtp', true]);
+
+      const offerer = new RTCPeerConnection();
+      offerer.addTransceiver('audio');
+      const answerer = new RTCPeerConnection();
+      await offerer.setLocalDescription();
+      ok(offerer.localDescription.sdp.includes('stereo=0'),
+          'Offer uses stereo=0 when media.peerconnection.sdp.disable_stereo_fmtp is set');
+      await answerer.setRemoteDescription(offerer.localDescription);
+      const {sdp} = await answerer.createAnswer();
+      ok(sdp.includes('stereo=0'), 'Answer uses stereo=0 when media.peerconnection.sdp.disable_stereo_fmtp is set');
+    },
+  ];
+
+  runNetworkTest(async () => {
+    for (const test of tests) {
+      info(`Running test: ${test.name}`);
+      try {
+        await test();
+      } catch (e) {
+        ok(false, `Caught ${e.name}: ${e.message} ${e.stack}`);
+      }
+      info(`Done running test: ${test.name}`);
+      // Make sure we don't build up a pile of GC work, and also get PCImpl to
+      // print their timecards.
+      await new Promise(r => SpecialPowers.exactGC(r));
+    }
+
+    await SpecialPowers.popPrefEnv();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_syncSetDescription.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_syncSetDescription.html
new file mode 100644
index 0000000000..98f0de1b4a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_syncSetDescription.html
@@ -0,0 +1,53 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1063971",
+    title: "Legacy sync setDescription calls",
+    visible: true
+  });
+
+// Test setDescription without callbacks, which many webrtc examples still do
+
+function PC_LOCAL_SET_LOCAL_DESCRIPTION_SYNC(test) {
+  test.pcLocal.onsignalingstatechange = function() {};
+  test.pcLocal._pc.setLocalDescription(test.originalOffer);
+}
+
+function PC_REMOTE_SET_REMOTE_DESCRIPTION_SYNC(test) {
+  test.pcRemote.onsignalingstatechange = function() {};
+  test.pcRemote._pc.setRemoteDescription(test._local_offer,
+    test.pcRemote.releaseIceCandidates,
+    generateErrorCallback("pcRemote._pc.setRemoteDescription() sync failed"));
+}
+function PC_REMOTE_SET_LOCAL_DESCRIPTION_SYNC(test) {
+  test.pcRemote.onsignalingstatechange = function() {};
+  test.pcRemote._pc.setLocalDescription(test.originalAnswer);
+}
+function PC_LOCAL_SET_REMOTE_DESCRIPTION_SYNC(test) {
+  test.pcLocal.onsignalingstatechange = function() {};
+  test.pcLocal._pc.setRemoteDescription(test._remote_answer,
+    test.pcLocal.releaseIceCandidates,
+    generateErrorCallback("pcLocal._pc.setRemoteDescription() sync failed"));
+}
+
+runNetworkTest(() => {
+  const test = new PeerConnectionTest();
+  test.setMediaConstraints([{video: true}], [{video: true}]);
+  test.chain.replace("PC_LOCAL_SET_LOCAL_DESCRIPTION", PC_LOCAL_SET_LOCAL_DESCRIPTION_SYNC);
+  test.chain.replace("PC_REMOTE_SET_REMOTE_DESCRIPTION", PC_REMOTE_SET_REMOTE_DESCRIPTION_SYNC);
+  test.chain.remove("PC_REMOTE_CHECK_CAN_TRICKLE_SYNC");
+  test.chain.replace("PC_REMOTE_SET_LOCAL_DESCRIPTION", PC_REMOTE_SET_LOCAL_DESCRIPTION_SYNC);
+  test.chain.replace("PC_LOCAL_SET_REMOTE_DESCRIPTION", PC_LOCAL_SET_REMOTE_DESCRIPTION_SYNC);
+  test.chain.remove("PC_LOCAL_CHECK_CAN_TRICKLE_SYNC");
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_telephoneEventFirst.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_telephoneEventFirst.html
new file mode 100644
index 0000000000..bde51c1fd0
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_telephoneEventFirst.html
@@ -0,0 +1,56 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  title: "RTCPeerConnection with telephone-event codec first in SDP",
+  bug: "1581898",
+  visible: true
+});
+
+const test = async () => {
+  const pc1 = new RTCPeerConnection();
+  const pc2 = new RTCPeerConnection();
+  const stream = await navigator.mediaDevices.getUserMedia({audio:true});
+  pc1.addTrack(stream.getAudioTracks()[0], stream);
+  pc2.addTrack(stream.getAudioTracks()[0], stream);
+
+  const offer = await pc1.createOffer();
+  await pc1.setLocalDescription(offer);
+
+  const regex = /^(m=audio \d+ [^ ]+) (.*) 101(.*)$/m;
+
+  // Rewrite offer so payload type 101 comes first
+  offer.sdp = offer.sdp.replace(regex, '$1 101 $2 $3');
+
+  ok(offer.sdp.match(/^m=audio \d+ [^ ]+ 101 /m),
+     "Payload type 101 should be first on the m-line");
+
+  await pc2.setRemoteDescription(offer);
+  const answer = await pc2.createAnswer();
+
+  pc1.onicecandidate = e => { pc2.addIceCandidate(e.candidate); }
+  pc2.onicecandidate = e => { pc1.addIceCandidate(e.candidate); }
+
+  await pc1.setRemoteDescription(answer);
+  await pc2.setLocalDescription(answer);
+  await new Promise(resolve => {
+    pc1.oniceconnectionstatechange = e => {
+      if (pc1.iceConnectionState == "connected") {
+        resolve();
+      }
+    };
+  });
+  await wait(1000);
+};
+
+runNetworkTest(test);
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_threeUnbundledConnections.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_threeUnbundledConnections.html
new file mode 100644
index 0000000000..75f0d12463
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_threeUnbundledConnections.html
@@ -0,0 +1,134 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1342579",
+    title: "Unbundled PC connects to two different PCs",
+    visible: true
+  });
+
+  const fakeFingerPrint = "a=fingerprint:sha-256 11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11:11";
+
+  const pc1 = new RTCPeerConnection();
+  const pc2 = new RTCPeerConnection();
+  const pc3 = new RTCPeerConnection();
+
+  const add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+  pc1.onicecandidate = e => {
+    if (e.candidate) {
+      if (e.candidate.sdpMid === "1") {
+        add(pc2, e.candidate, generateErrorCallback())
+      } else {
+        add(pc3, e.candidate, generateErrorCallback())
+      }
+    }
+  };
+  pc2.onicecandidate = e => add(pc1, e.candidate, generateErrorCallback());
+  pc3.onicecandidate = e => add(pc1, e.candidate, generateErrorCallback());
+
+  let ice1Finished, ice2Finished, ice3Finished;
+  const ice1Done = new Promise(r => ice1Finished = r);
+  const ice2Done = new Promise(r => ice2Finished = r);
+  const ice3Done = new Promise(r => ice3Finished = r);
+
+  const icsc = (pc, str, resolve) => {
+    const state = pc.iceConnectionState;
+    info(str + " ICE connection state is: " + state);
+    if (state == "connected") {
+      ok(true, str + " ICE connected");
+      resolve();
+    } else if (state == "failed") {
+      ok(false, str + " ICE failed")
+      resolve();
+    }
+  };
+
+  pc1.oniceconnectionstatechange = e => icsc(pc1, "PC1", ice1Finished);
+  pc2.oniceconnectionstatechange = e => icsc(pc2, "PC2", ice2Finished);
+  pc3.oniceconnectionstatechange = e => icsc(pc3, "PC3", ice3Finished);
+
+
+  function combineAnswer(origAnswer, answer) {
+    const sdplines = origAnswer.sdp.split('\r\n');
+    const fpIndex = sdplines.findIndex(l => l.match('^a=fingerprint'));
+    const FP = sdplines[fpIndex];
+    const audioIndex = sdplines.findIndex(l => l.match(/^m=audio [1-9]/));
+    const videoIndex = sdplines.findIndex(l => l.match(/^m=video [1-9]/));
+    if (audioIndex > -1) {
+      var ss = sdplines.slice(0, audioIndex);
+      ss.splice(fpIndex, 1);
+      answer.sessionSection = ss;
+      const rejectedVideoIndex = sdplines.findIndex(l => l.match('m=video 0'));
+      var ams = sdplines.slice(audioIndex, rejectedVideoIndex);
+      ams.push(FP);
+      ams.push(fakeFingerPrint);
+      answer.audioMsection = ams;
+    }
+    if (videoIndex > -1) {
+      var vms = sdplines.slice(videoIndex, sdplines.length -1);
+      vms.push(fakeFingerPrint);
+      vms.push(FP);
+      answer.videoMsection = vms;
+    }
+    return answer;
+  }
+
+runNetworkTest(async () => {
+    const v1 = createMediaElement('video', 'v1');
+    const v2 = createMediaElement('video', 'v2');
+    const v3 = createMediaElement('video', 'v3');
+
+    const stream = await navigator.mediaDevices.getUserMedia({ video: true, audio: true });
+    (v1.srcObject = stream).getTracks().forEach(t => pc1.addTrack(t, stream));
+
+    const stream2 = await navigator.mediaDevices.getUserMedia({ video: true });
+    (v2.srcObject = stream2).getTracks().forEach(t => pc2.addTrack(t, stream2));
+
+    const stream3 = await navigator.mediaDevices.getUserMedia({ audio: true });
+    (v3.srcObject = stream3).getTracks().forEach(t => pc3.addTrack(t, stream3));
+
+    const offer = await pc1.createOffer();
+    await pc1.setLocalDescription(offer);
+
+    //info("Original OFFER: " + JSON.stringify(offer));
+    offer.sdp = sdputils.removeBundle(offer.sdp);
+    //info("OFFER w/o BUNDLE: " + JSON.stringify(offer));
+    const offerAudio = new RTCSessionDescription(JSON.parse(JSON.stringify(offer)));
+    offerAudio.sdp = offerAudio.sdp.replace('m=video 9', 'm=video 0');
+    //info("offerAudio: " + JSON.stringify(offerAudio));
+    const offerVideo = new RTCSessionDescription(JSON.parse(JSON.stringify(offer)));
+    offerVideo.sdp = offerVideo.sdp.replace('m=audio 9', 'm=audio 0');
+    //info("offerVideo: " + JSON.stringify(offerVideo));
+
+    // We need to do these in parallel, otherwise pc1 will start firing
+    // icecandidate events before pc3 is ready.
+    await Promise.all([pc2.setRemoteDescription(offerVideo), pc3.setRemoteDescription(offerAudio)]);
+
+    const answerVideo = await pc2.createAnswer();
+    const answerAudio = await pc3.createAnswer();
+
+    const answer = combineAnswer(answerAudio, combineAnswer(answerVideo, {}));
+    const fakeAnswer = answer.sessionSection.concat(answer.audioMsection, answer.videoMsection).join('\r\n');
+    info("ANSWER: " + fakeAnswer);
+
+    // We want to do these in parallel, because if we do them seqentially, by the
+    // time pc3.sLD completes pc2 could have fired icecandidate events, when we
+    // haven't called pc1.sRD yet.
+    await Promise.all(
+      [pc2.setLocalDescription(answerVideo),
+       pc3.setLocalDescription(answerAudio),
+       pc1.setRemoteDescription({type: 'answer', sdp: fakeAnswer})]);
+
+    await Promise.all([ice1Done, ice2Done, ice3Done]);
+
+    ok(true, "Connected.");
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_throwInCallbacks.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_throwInCallbacks.html
new file mode 100644
index 0000000000..5a3872c120
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_throwInCallbacks.html
@@ -0,0 +1,83 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "857765",
+    title: "Throw in PeerConnection callbacks"
+  });
+
+runNetworkTest(function () {
+  let finish;
+  const onfinished = new Promise(r => finish = async () => {
+    window.onerror = oldOnError;
+    is(error_count, 7, "Seven expected errors verified.");
+    r();
+  });
+
+  function getFail() {
+    return err => {
+      window.onerror = oldOnError;
+      generateErrorCallback()(err);
+    };
+  }
+
+  let error_count = 0;
+  let oldOnError = window.onerror;
+  window.onerror = (errorMsg, url, lineNumber) => {
+    if (!errorMsg.includes("Expected")) {
+      getFail()(errorMsg);
+    }
+    error_count += 1;
+    info("onerror " + error_count + ": " + errorMsg);
+    if (error_count == 7) {
+      finish();
+    }
+    throw new Error("window.onerror may throw");
+    return false;
+  }
+
+  let pc0, pc1, pc2;
+  // Test failure callbacks (limited to 1 for now)
+  pc0 = new RTCPeerConnection();
+  pc0.close();
+  pc0.createOffer(getFail(), function(err) {
+    pc1 = new RTCPeerConnection();
+    pc2 = new RTCPeerConnection();
+
+    // Test success callbacks (happy path)
+    navigator.mozGetUserMedia({video:true}, function(video1) {
+      pc1.addStream(video1);
+      pc1.createOffer(function(offer) {
+        pc1.setLocalDescription(offer, function() {
+          pc2.setRemoteDescription(offer, function() {
+            pc2.createAnswer(function(answer) {
+              pc2.setLocalDescription(answer, function() {
+                pc1.setRemoteDescription(answer, function() {
+                  throw new Error("Expected");
+                }, getFail());
+                throw new Error("Expected");
+              }, getFail());
+              throw new Error("Expected");
+            }, getFail());
+            throw new Error("Expected");
+          }, getFail());
+          throw new Error("Expected");
+        }, getFail());
+        throw new Error("Expected");
+      }, getFail());
+    }, getFail());
+    throw new Error("Expected");
+  });
+
+  return onfinished;
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_toJSON.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_toJSON.html
new file mode 100644
index 0000000000..96c2c42b78
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_toJSON.html
@@ -0,0 +1,39 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "928304",
+    title: "test toJSON() on RTCSessionDescription and RTCIceCandidate"
+  });
+
+  runNetworkTest(function () {
+    /** Test for Bug 872377 **/
+
+    var rtcSession = new RTCSessionDescription({ sdp: "Picklechips!",
+                                                 type: "offer" });
+    var jsonCopy = JSON.parse(JSON.stringify(rtcSession));
+    for (key in rtcSession) {
+      if (typeof(rtcSession[key]) == "function") continue;
+      is(rtcSession[key], jsonCopy[key], "key " + key + " should match.");
+    }
+
+    /** Test for Bug 928304 **/
+
+    var rtcIceCandidate = new RTCIceCandidate({ candidate: "dummy",
+                                                   sdpMid: "test",
+                                                   sdpMLineIndex: 3 });
+    jsonCopy = JSON.parse(JSON.stringify(rtcIceCandidate));
+    for (key in rtcIceCandidate) {
+      if (typeof(rtcIceCandidate[key]) == "function") continue;
+      is(rtcIceCandidate[key], jsonCopy[key], "key " + key + " should match.");
+    }
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_trackDisabling.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_trackDisabling.html
new file mode 100644
index 0000000000..73323cf007
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_trackDisabling.html
@@ -0,0 +1,108 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1219711",
+  title: "Disabling locally should be reflected remotely"
+});
+
+runNetworkTest(async () => {
+  const test = new PeerConnectionTest();
+
+  await pushPrefs(
+    ["media.getusermedia.camera.stop_on_disable.enabled", true],
+    ["media.getusermedia.camera.stop_on_disable.delay_ms", 0],
+    ["media.getusermedia.microphone.stop_on_disable.enabled", true],
+    ["media.getusermedia.microphone.stop_on_disable.delay_ms", 0],
+    // Always use fake tracks since we depend on video to be somewhat green and
+    // audio to have a large 1000Hz component (or 440Hz if using fake devices).
+    ['media.audio_loopback_dev', ''],
+    ['media.video_loopback_dev', ''],
+    ['media.navigator.streams.fake', true]);
+
+  test.setMediaConstraints([{audio: true, video: true}], []);
+  test.chain.append([
+    function CHECK_ASSUMPTIONS() {
+      is(test.pcLocal.localMediaElements.length, 2,
+         "pcLocal should have one media element");
+      is(test.pcRemote.remoteMediaElements.length, 2,
+         "pcRemote should have one media element");
+      is(test.pcLocal._pc.getLocalStreams().length, 1,
+         "pcLocal should have one stream");
+      is(test.pcRemote._pc.getRemoteStreams().length, 1,
+         "pcRemote should have one stream");
+    },
+    async function CHECK_VIDEO() {
+      const h = new CaptureStreamTestHelper2D();
+      const localVideo = test.pcLocal.localMediaElements
+        .find(e => e instanceof HTMLVideoElement);
+      const remoteVideo = test.pcRemote.remoteMediaElements
+        .find(e => e instanceof HTMLVideoElement);
+      // We check a pixel somewhere away from the top left corner since
+      // MediaEngineFake puts semi-transparent time indicators there.
+      const offsetX = 50;
+      const offsetY = 50;
+      const threshold = 128;
+
+      // We're regarding black as disabled here, and we're setting the alpha
+      // channel of the pixel to 255 to disregard alpha when testing.
+      const checkVideoEnabled = video => h.waitForPixel(video,
+        px => (px[3] = 255, h.isPixelNot(px, h.black, threshold)),
+        { offsetX, offsetY }
+      );
+      const checkVideoDisabled = video => h.waitForPixel(video,
+        px => (px[3] = 255, h.isPixel(px, h.black, threshold)),
+        { offsetX, offsetY }
+      );
+
+      info("Checking local video enabled");
+      await checkVideoEnabled(localVideo);
+      info("Checking remote video enabled");
+      await checkVideoEnabled(remoteVideo);
+
+      info("Disabling original");
+      test.pcLocal._pc.getLocalStreams()[0].getVideoTracks()[0].enabled = false;
+
+      info("Checking local video disabled");
+      await checkVideoDisabled(localVideo);
+      info("Checking remote video disabled");
+      await checkVideoDisabled(remoteVideo);
+    },
+    async function CHECK_AUDIO() {
+      const ac = new AudioContext();
+      const localAnalyser = new AudioStreamAnalyser(ac, test.pcLocal._pc.getLocalStreams()[0]);
+      const remoteAnalyser = new AudioStreamAnalyser(ac, test.pcRemote._pc.getRemoteStreams()[0]);
+
+      const checkAudio = (analyser, fun) => analyser.waitForAnalysisSuccess(fun);
+
+      const freq = localAnalyser.binIndexForFrequency(TEST_AUDIO_FREQ);
+      const checkAudioEnabled = analyser =>
+        checkAudio(analyser, array => array[freq] > 200);
+      const checkAudioDisabled = analyser =>
+        checkAudio(analyser, array => array[freq] < 50);
+
+      info("Checking local audio enabled");
+      await checkAudioEnabled(localAnalyser);
+      info("Checking remote audio enabled");
+      await checkAudioEnabled(remoteAnalyser);
+
+      test.pcLocal._pc.getLocalStreams()[0].getAudioTracks()[0].enabled = false;
+
+      info("Checking local audio disabled");
+      await checkAudioDisabled(localAnalyser);
+      info("Checking remote audio disabled");
+      await checkAudioDisabled(remoteAnalyser);
+    },
+  ]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_trackDisabling_clones.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_trackDisabling_clones.html
new file mode 100644
index 0000000000..ae7647fa1a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_trackDisabling_clones.html
@@ -0,0 +1,162 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1219711",
+  title: "Disabling locally should be reflected remotely, individually for clones"
+});
+
+runNetworkTest(async () => {
+  var test = new PeerConnectionTest();
+
+  await pushPrefs(
+    ["media.getusermedia.camera.stop_on_disable.enabled", true],
+    ["media.getusermedia.camera.stop_on_disable.delay_ms", 0],
+    ["media.getusermedia.microphone.stop_on_disable.enabled", true],
+    ["media.getusermedia.microphone.stop_on_disable.delay_ms", 0],
+    // Always use fake tracks since we depend on audio to have a large 1000Hz
+    // component.
+    ['media.audio_loopback_dev', ''],
+    ['media.navigator.streams.fake', true]);
+    // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+    if (navigator.userAgent.includes("Android")) {
+      await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                      ["media.webrtc.hw.h264.enabled", false]);
+    }
+
+  var originalStream;
+  var localVideoOriginal;
+
+  test.setMediaConstraints([{audio: true, video: true}], []);
+  test.chain.replace("PC_LOCAL_GUM", [
+    function PC_LOCAL_GUM_CLONE() {
+      return getUserMedia(test.pcLocal.constraints[0]).then(stream => {
+        originalStream = stream;
+        localVideoOriginal =
+          createMediaElement("video", "local-original");
+        localVideoOriginal.srcObject = stream;
+        test.pcLocal.attachLocalStream(originalStream.clone());
+      });
+    }
+  ]);
+  test.chain.append([
+    function CHECK_ASSUMPTIONS() {
+      is(test.pcLocal.localMediaElements.length, 2,
+         "pcLocal should have one media element");
+      is(test.pcRemote.remoteMediaElements.length, 2,
+         "pcRemote should have one media element");
+      is(test.pcLocal._pc.getLocalStreams().length, 1,
+         "pcLocal should have one stream");
+      is(test.pcRemote._pc.getRemoteStreams().length, 1,
+         "pcRemote should have one stream");
+    },
+    async function CHECK_VIDEO() {
+      info("Checking video");
+      var h = new CaptureStreamTestHelper2D();
+      var localVideoClone = test.pcLocal.localMediaElements
+        .find(e => e instanceof HTMLVideoElement);
+      var remoteVideoClone = test.pcRemote.remoteMediaElements
+        .find(e => e instanceof HTMLVideoElement);
+
+      // We check a pixel somewhere away from the top left corner since
+      // MediaEngineFake puts semi-transparent time indicators there.
+      const offsetX = 50;
+      const offsetY = 50;
+      const threshold = 128;
+      const remoteDisabledColor = h.black;
+
+      // We're regarding black as disabled here, and we're setting the alpha
+      // channel of the pixel to 255 to disregard alpha when testing.
+      var checkVideoEnabled = video => h.waitForPixel(video,
+        px => (px[3] = 255, h.isPixelNot(px, h.black, threshold)),
+        { offsetX, offsetY }
+      );
+      var checkVideoDisabled = video => h.waitForPixel(video,
+        px => (px[3] = 255, h.isPixel(px, h.black, threshold)),
+        { offsetX, offsetY }
+      );
+
+      info("Checking local original enabled");
+      await checkVideoEnabled(localVideoOriginal);
+      info("Checking local clone enabled");
+      await checkVideoEnabled(localVideoClone);
+      info("Checking remote clone enabled");
+      await checkVideoEnabled(remoteVideoClone);
+
+      info("Disabling original");
+      originalStream.getVideoTracks()[0].enabled = false;
+
+      info("Checking local original disabled");
+      await checkVideoDisabled(localVideoOriginal);
+      info("Checking local clone enabled");
+      await checkVideoEnabled(localVideoClone);
+      info("Checking remote clone enabled");
+      await checkVideoEnabled(remoteVideoClone);
+
+      info("Re-enabling original; disabling clone");
+      originalStream.getVideoTracks()[0].enabled = true;
+      test.pcLocal._pc.getLocalStreams()[0].getVideoTracks()[0].enabled = false;
+
+      info("Checking local original enabled");
+      await checkVideoEnabled(localVideoOriginal);
+      info("Checking local clone disabled");
+      await checkVideoDisabled(localVideoClone);
+      info("Checking remote clone disabled");
+      await checkVideoDisabled(remoteVideoClone);
+    },
+    async function CHECK_AUDIO() {
+      info("Checking audio");
+      var ac = new AudioContext();
+      var localAnalyserOriginal = new AudioStreamAnalyser(ac, originalStream);
+      var localAnalyserClone =
+        new AudioStreamAnalyser(ac, test.pcLocal._pc.getLocalStreams()[0]);
+      var remoteAnalyserClone =
+        new AudioStreamAnalyser(ac, test.pcRemote._pc.getRemoteStreams()[0]);
+
+      var freq = localAnalyserOriginal.binIndexForFrequency(TEST_AUDIO_FREQ);
+      var checkAudioEnabled = analyser =>
+        analyser.waitForAnalysisSuccess(array => array[freq] > 200);
+      var checkAudioDisabled = analyser =>
+        analyser.waitForAnalysisSuccess(array => array[freq] < 50);
+
+      info("Checking local original enabled");
+      await checkAudioEnabled(localAnalyserOriginal);
+      info("Checking local clone enabled");
+      await checkAudioEnabled(localAnalyserClone);
+      info("Checking remote clone enabled");
+      await checkAudioEnabled(remoteAnalyserClone);
+
+      info("Disabling original");
+      originalStream.getAudioTracks()[0].enabled = false;
+
+      info("Checking local original disabled");
+      await checkAudioDisabled(localAnalyserOriginal);
+      info("Checking local clone enabled");
+      await checkAudioEnabled(localAnalyserClone);
+      info("Checking remote clone enabled");
+      await checkAudioEnabled(remoteAnalyserClone);
+
+      info("Re-enabling original; disabling clone");
+      originalStream.getAudioTracks()[0].enabled = true;
+      test.pcLocal._pc.getLocalStreams()[0].getAudioTracks()[0].enabled = false;
+
+      info("Checking local original enabled");
+      await checkAudioEnabled(localAnalyserOriginal);
+      info("Checking local clone disabled");
+      await checkAudioDisabled(localAnalyserClone);
+      info("Checking remote clone disabled");
+      await checkAudioDisabled(remoteAnalyserClone);
+    },
+  ]);
+  await test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_trackless_sender_stats.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_trackless_sender_stats.html
new file mode 100644
index 0000000000..f0356f5655
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_trackless_sender_stats.html
@@ -0,0 +1,56 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1452673",
+    title: "Trackless RTCRtpSender.getStats()",
+    visible: true
+  });
+
+  // Calling getstats() on a trackless RTCRtpSender should yield an empty
+  // stats report.  When track stats are added in the future, the stats
+  // for the removed tracks should continue to appear.
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    test.chain.removeAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW");
+    test.chain.append(
+      async function PC_LOCAL_AND_REMOTE_TRACKLESS_SENDER_STATS(test) {
+        await Promise.all([
+          waitForSyncedRtcp(test.pcLocal._pc),
+          waitForSyncedRtcp(test.pcRemote._pc),
+        ]);
+        let senders = test.pcLocal.getSenders();
+        let receivers = test.pcRemote.getReceivers();
+        is(senders.length, 2, "Have exactly two senders.");
+        is(receivers.length, 2, "Have exactly two receivers.");
+        for(let kind of ["audio", "video"]) {
+          is(senders.filter(s => s.track.kind == kind).length, 1,
+              "Exactly 1 sender of kind " + kind);
+          is(receivers.filter(r => r.track.kind == kind).length, 1,
+              "Exactly 1 receiver of kind " + kind);
+        }
+        // Remove tracks from senders
+        for (const sender of senders) {
+          await sender.replaceTrack(null);
+          is(sender.track, null, "Sender track removed");
+          let stats = await sender.getStats();
+          ok(stats instanceof window.RTCStatsReport, "Stats is instance of RTCStatsReport");
+          // Number of stats in the report. This should be 0.
+          is(stats.size, 0, "Trackless sender stats report is empty");
+        }
+      }
+    );
+    test.setMediaConstraints([{audio: true}, {video: true}], []);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioStreams.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioStreams.html
new file mode 100644
index 0000000000..7ea18ab3dd
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioStreams.html
@@ -0,0 +1,23 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1091242",
+    title: "Multistream: Two audio streams"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}, {audio: true}],
+                             [{audio: true}, {audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioTracksInOneStream.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioTracksInOneStream.html
new file mode 100644
index 0000000000..99d4ad625a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioTracksInOneStream.html
@@ -0,0 +1,37 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1145407",
+    title: "Multistream: Two audio tracks in one stream"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    test.chain.insertAfter("PC_REMOTE_GET_OFFER", [
+        function PC_REMOTE_OVERRIDE_STREAM_IDS_IN_OFFER(test) {
+          test._local_offer.sdp = test._local_offer.sdp.replace(
+              /a=msid:[^\s]*/g,
+              "a=msid:foo");
+        }
+    ]);
+    test.chain.insertAfter("PC_LOCAL_GET_ANSWER", [
+        function PC_LOCAL_OVERRIDE_STREAM_IDS_IN_ANSWER(test) {
+          test._remote_answer.sdp = test._remote_answer.sdp.replace(
+              /a=msid:[^\s]*/g,
+              "a=msid:foo");
+        }
+    ]);
+    test.setMediaConstraints([{audio: true}, {audio: true}],
+                             [{audio: true}, {audio: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioVideoStreams.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioVideoStreams.html
new file mode 100644
index 0000000000..5f4bd463d4
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioVideoStreams.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+
+  createHTML({
+    bug: "1091242",
+    title: "Multistream: Two audio streams, two video streams"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true}, {video: true}, {audio: true},
+                                {video: true}],
+                             [{audio: true}, {video: true}, {audio: true},
+                                {video: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioVideoStreamsCombined.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioVideoStreamsCombined.html
new file mode 100644
index 0000000000..fcc9c6c8fa
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioVideoStreamsCombined.html
@@ -0,0 +1,70 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+
+  createHTML({
+    bug: "1091242",
+    title: "Multistream: Two audio/video streams"
+  });
+
+  runNetworkTest(async (options) => {
+    // Disable platform encodre for SW MFT encoder causes some stats
+    // exceeding the test thresholds.
+    // E.g. inbound-rtp.packetsDiscarded value=118 >= 100.
+    await matchPlatformH264CodecPrefs();
+
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{audio: true, video: true},
+                                {audio: true, video: true}],
+                             [{audio: true, video: true},
+                                {audio: true, video: true}]);
+
+    // Test stats, including coalescing of codec stats.
+    test.chain.insertAfter("PC_LOCAL_WAIT_FOR_MEDIA_FLOW",
+      [PC_LOCAL_TEST_LOCAL_STATS]);
+
+    test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW",
+      [PC_REMOTE_TEST_REMOTE_STATS]);
+
+    const testCoalescedCodecStats = stats => {
+      is([...stats.values()].filter(({type}) => type.endsWith("rtp")).length,
+        16,
+        "Expected: 4 outbound, 4 remote-inbound, 4 inbound, 4 remote-inbound");
+      const codecs = [...stats.values()]
+          .filter(({type}) => type == "codec")
+          .sort((a, b) => a.mimeType > b.mimeType);
+      is(codecs.length, 2, "Should have registered two codecs (coalesced)");
+      is(new Set(codecs.map(({transportId}) => transportId)).size, 1,
+        "Should have registered only one transport with BUNDLE");
+      const codecTypes = new Set(codecs.map(({codecType}) => codecType));
+      is(codecTypes.size, 1,
+        "Should have identical encode and decode configurations (and stats)");
+      is(codecTypes[0], undefined,
+        "Should have identical encode and decode configurations (and stats)");
+      is(codecs[0].mimeType.slice(0, 5), "audio",
+        "Should have registered an audio codec");
+      is(codecs[1].mimeType.slice(0, 5), "video",
+        "Should have registered a video codec");
+    };
+
+    test.chain.append([
+      async function PC_LOCAL_TEST_COALESCED_CODEC_STATS() {
+        testCoalescedCodecStats(await test.pcLocal._pc.getStats());
+      },
+      async function PC_REMOTE_TEST_COALESCED_CODEC_STATS() {
+        testCoalescedCodecStats(await test.pcRemote._pc.getStats());
+      },
+    ]);
+
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioVideoStreamsCombinedNoBundle.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioVideoStreamsCombinedNoBundle.html
new file mode 100644
index 0000000000..8b825db617
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoAudioVideoStreamsCombinedNoBundle.html
@@ -0,0 +1,107 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1225722",
+  title: "Multistream: Two audio/video streams without BUNDLE"
+});
+
+runNetworkTest(async (options = {}) => {
+  // Disable platform encodre for SW MFT encoder causes some stats
+  // exceeding the test thresholds.
+  // E.g. inbound-rtp.packetsDiscarded value=118 >= 100.
+  await matchPlatformH264CodecPrefs();
+
+  options.bundle = false;
+  const test = new PeerConnectionTest(options);
+  test.setMediaConstraints(
+    [{audio: true, video: true}, {audio: true, video: true}],
+    [{audio: true, video: true}, {audio: true, video: true}]
+  );
+
+  // Test stats, including that codec stats do not coalesce without BUNDLE.
+  const testNonBundledStats = async pc => {
+    // This is basically PC_*_TEST_*_STATS fleshed out, but uses
+    // sender/receiver.getStats instead of pc.getStats, since the codec stats
+    // code assumes at most one sender and at most one receiver.
+    await waitForSyncedRtcp(pc);
+    const senderPromises = pc.getSenders().map(obj => obj.getStats());
+    const receiverPromises = pc.getReceivers().map(obj => obj.getStats());
+    const senderStats = await Promise.all(senderPromises);
+    const receiverStats = await Promise.all(receiverPromises);
+    for (const stats of [...senderStats, ...receiverStats]) {
+      checkExpectedFields(stats);
+      pedanticChecks(stats);
+    }
+    for (const stats of senderStats) {
+      checkSenderStats(stats, 1);
+    }
+  };
+
+  test.chain.insertAfter("PC_LOCAL_WAIT_FOR_MEDIA_FLOW", [
+    async function PC_LOCAL_TEST_LOCAL_NONBUNDLED_STATS(test) {
+      await testNonBundledStats(test.pcLocal._pc);
+    },
+  ]);
+
+  test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW", [
+    async function PC_REMOTE_TEST_LOCAL_NONBUNDLED_STATS(test) {
+      await testNonBundledStats(test.pcRemote._pc);
+    },
+  ]);
+
+  const testNonCoalescedCodecStats = stats => {
+    const codecs = [...stats.values()]
+          .filter(({type}) => type == "codec");
+    is([...stats.values()].filter(({type}) => type.endsWith("rtp")).length, 16,
+      "Expected: 4 outbound, 4 remote-inbound, 4 inbound, 4 remote-inbound");
+    const codecTypes = new Set(codecs.map(({codecType}) => codecType));
+    is(codecTypes.size, 1,
+      "Should have identical encode and decode configurations (and stats)");
+    is(codecTypes[0], undefined,
+      "Should have identical encode and decode configurations (and stats)");
+    const transportIds = new Set(codecs.map(({transportId}) => transportId));
+    is(transportIds.size, 4,
+      "Should have registered four transports for two sendrecv streams");
+    for (const transportId of transportIds) {
+      is(codecs.filter(c => c.transportId == transportId).length, 1,
+        "Should have registered one codec per transport without BUNDLE");
+    }
+    for (const prefix of ["audio", "video"]) {
+      const prefixed = codecs.filter(c => c.mimeType.startsWith(prefix));
+      is(prefixed.length, 2, `Should have registered two ${prefix} codecs`);
+      if (prefixed.length == 2) {
+        is(prefixed[0].payloadType, prefixed[1].payloadType,
+          "same payloadType");
+        isnot(prefixed[0].transportId, prefixed[1].transportId,
+          "different transportIds");
+        is(prefixed[0].mimeType, prefixed[1].mimeType, "same mimeType");
+        is(prefixed[0].clockRate, prefixed[1].clockRate, "same clockRate");
+        is(prefixed[0].channels, prefixed[1].channels, "same channels");
+        is(prefixed[0].sdpFmtpLine, prefixed[1].sdpFmtpLine,
+          "same sdpFmtpLine");
+      }
+    }
+  };
+
+  test.chain.append([
+    async function PC_LOCAL_TEST_NON_COALESCED_CODEC_STATS() {
+      testNonCoalescedCodecStats(await test.pcLocal._pc.getStats());
+    },
+    async function PC_REMOTE_TEST_NON_COALESCED_CODEC_STATS() {
+      testNonCoalescedCodecStats(await test.pcRemote._pc.getStats());
+    },
+  ]);
+
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_twoVideoStreams.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoVideoStreams.html
new file mode 100644
index 0000000000..0ab180cc55
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoVideoStreams.html
@@ -0,0 +1,23 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1091242",
+    title: "Multistream: Two video streams"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{video: true}, {video: true}],
+                             [{video: true}, {video: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_twoVideoTracksInOneStream.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoVideoTracksInOneStream.html
new file mode 100644
index 0000000000..4eaf8b3f48
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_twoVideoTracksInOneStream.html
@@ -0,0 +1,37 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1145407",
+    title: "Multistream: Two video tracks in offerer stream"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    test.chain.insertAfter("PC_REMOTE_GET_OFFER", [
+        function PC_REMOTE_OVERRIDE_STREAM_IDS_IN_OFFER(test) {
+          test._local_offer.sdp = test._local_offer.sdp.replace(
+              /a=msid:[^\s]*/g,
+              "a=msid:foo");
+        }
+    ]);
+    test.chain.insertAfter("PC_LOCAL_GET_ANSWER", [
+        function PC_LOCAL_OVERRIDE_STREAM_IDS_IN_ANSWER(test) {
+          test._remote_answer.sdp = test._remote_answer.sdp.replace(
+              /a=msid:[^\s]*/g,
+              "a=msid:foo");
+        }
+    ]);
+    test.setMediaConstraints([{video: true}, {video: true}],
+                             [{video: true}, {video: true}]);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_verifyAudioAfterRenegotiation.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_verifyAudioAfterRenegotiation.html
new file mode 100644
index 0000000000..86ef6d4678
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_verifyAudioAfterRenegotiation.html
@@ -0,0 +1,99 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1166832",
+    title: "Renegotiation: verify audio after renegotiation"
+  });
+
+  runNetworkTest(function (options) {
+    const test = new PeerConnectionTest(options);
+    const helper = new AudioStreamHelper();
+
+    test.chain.append([
+      function CHECK_ASSUMPTIONS() {
+        is(test.pcLocal.localMediaElements.length, 1,
+           "pcLocal should have one media element");
+        is(test.pcRemote.remoteMediaElements.length, 1,
+           "pcRemote should have one media element");
+        is(test.pcLocal._pc.getLocalStreams().length, 1,
+           "pcLocal should have one stream");
+        is(test.pcRemote._pc.getRemoteStreams().length, 1,
+           "pcRemote should have one stream");
+      },
+      function CHECK_AUDIO() {
+        return Promise.resolve()
+          .then(() => info("Checking local audio enabled"))
+          .then(() => helper.checkAudioFlowing(test.pcLocal._pc.getLocalStreams()[0]))
+          .then(() => info("Checking remote audio enabled"))
+          .then(() => helper.checkAudioFlowing(test.pcRemote._pc.getRemoteStreams()[0]))
+
+          .then(() => test.pcLocal._pc.getLocalStreams()[0].getAudioTracks()[0].enabled = false)
+
+          .then(() => info("Checking local audio disabled"))
+          .then(() => helper.checkAudioNotFlowing(test.pcLocal._pc.getLocalStreams()[0]))
+          .then(() => info("Checking remote audio disabled"))
+          .then(() => helper.checkAudioNotFlowing(test.pcRemote._pc.getRemoteStreams()[0]))
+      }
+    ]);
+
+    addRenegotiation(test.chain,
+      [
+        function PC_LOCAL_ADD_SECOND_STREAM(test) {
+          test.setMediaConstraints([{audio: true}],
+                                   []);
+          return test.pcLocal.getAllUserMediaAndAddStreams([{audio: true}]);
+        },
+      ]
+    );
+
+    test.chain.append([
+      function CHECK_ASSUMPTIONS2() {
+        is(test.pcLocal.localMediaElements.length, 2,
+           "pcLocal should have two media elements");
+        is(test.pcRemote.remoteMediaElements.length, 2,
+           "pcRemote should have two media elements");
+        is(test.pcLocal._pc.getLocalStreams().length, 2,
+           "pcLocal should have two streams");
+        is(test.pcRemote._pc.getRemoteStreams().length, 2,
+           "pcRemote should have two streams");
+      },
+      function RE_CHECK_AUDIO() {
+        return Promise.resolve()
+          .then(() => info("Checking local audio enabled"))
+          .then(() => helper.checkAudioNotFlowing(test.pcLocal._pc.getLocalStreams()[0]))
+          .then(() => info("Checking remote audio enabled"))
+          .then(() => helper.checkAudioNotFlowing(test.pcRemote._pc.getRemoteStreams()[0]))
+
+          .then(() => info("Checking local2 audio enabled"))
+          .then(() => helper.checkAudioFlowing(test.pcLocal._pc.getLocalStreams()[1]))
+          .then(() => info("Checking remote2 audio enabled"))
+          .then(() => helper.checkAudioFlowing(test.pcRemote._pc.getRemoteStreams()[1]))
+
+          .then(() => test.pcLocal._pc.getLocalStreams()[1].getAudioTracks()[0].enabled = false)
+          .then(() => test.pcLocal._pc.getLocalStreams()[0].getAudioTracks()[0].enabled = true)
+
+          .then(() => info("Checking local2 audio disabled"))
+          .then(() => helper.checkAudioNotFlowing(test.pcLocal._pc.getLocalStreams()[1]))
+          .then(() => info("Checking remote2 audio disabled"))
+          .then(() => helper.checkAudioNotFlowing(test.pcRemote._pc.getRemoteStreams()[1]))
+
+          .then(() => info("Checking local audio enabled"))
+          .then(() => helper.checkAudioFlowing(test.pcLocal._pc.getLocalStreams()[0]))
+          .then(() => info("Checking remote audio enabled"))
+          .then(() => helper.checkAudioFlowing(test.pcRemote._pc.getRemoteStreams()[0]))
+      }
+    ]);
+
+    test.setMediaConstraints([{audio: true}], []);
+    return test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_verifyDescriptions.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_verifyDescriptions.html
new file mode 100644
index 0000000000..f685f7c99a
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_verifyDescriptions.html
@@ -0,0 +1,58 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1264479",
+    title: "PeerConnection verify current and pending descriptions"
+  });
+
+  const pc1 = new RTCPeerConnection();
+  const pc2 = new RTCPeerConnection();
+
+  var add = (pc, can, failed) => can && pc.addIceCandidate(can).catch(failed);
+  pc1.onicecandidate = e => add(pc2, e.candidate, generateErrorCallback());
+  pc2.onicecandidate = e => add(pc1, e.candidate, generateErrorCallback());
+
+
+  runNetworkTest(function() {
+    const v1 = createMediaElement('video', 'v1');
+    const v2 = createMediaElement('video', 'v2');
+
+    return navigator.mediaDevices.getUserMedia({ video: true, audio: true })
+    .then(stream => (v1.srcObject = stream).getTracks().forEach(t => pc1.addTrack(t, stream)))
+    .then(() => pc1.createOffer({})) // check that createOffer accepts arg.
+    .then(offer => pc1.setLocalDescription(offer))
+    .then(() => {
+      ok(!pc1.currentLocalDescription, "pc1 currentLocalDescription is empty");
+      ok(pc1.pendingLocalDescription, "pc1 pendingLocalDescription is set");
+      ok(pc1.localDescription, "pc1 localDescription is set");
+    })
+    .then(() => pc2.setRemoteDescription(pc1.localDescription))
+    .then(() => {
+      ok(!pc2.currentRemoteDescription, "pc2 currentRemoteDescription is empty");
+      ok(pc2.pendingRemoteDescription, "pc2 pendingRemoteDescription is set");
+      ok(pc2.remoteDescription, "pc2 remoteDescription is set");
+    })
+    .then(() => pc2.createAnswer({}))  // check that createAnswer accepts arg.
+    .then(answer => pc2.setLocalDescription(answer))
+    .then(() => {
+      ok(pc2.currentLocalDescription, "pc2 currentLocalDescription is set");
+      ok(!pc2.pendingLocalDescription, "pc2 pendingLocalDescription is empty");
+      ok(pc2.localDescription, "pc2 localDescription is set");
+    })
+    .then(() => pc1.setRemoteDescription(pc2.localDescription))
+    .then(() => {
+      ok(pc1.currentRemoteDescription, "pc1 currentRemoteDescription is set");
+      ok(!pc1.pendingRemoteDescription, "pc1 pendingRemoteDescription is empty");
+      ok(pc1.remoteDescription, "pc1 remoteDescription is set");
+    });
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_verifyVideoAfterRenegotiation.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_verifyVideoAfterRenegotiation.html
new file mode 100644
index 0000000000..8d4155ddff
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_verifyVideoAfterRenegotiation.html
@@ -0,0 +1,123 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1166832",
+    title: "Renegotiation: verify video after renegotiation"
+  });
+
+runNetworkTest(async () => {
+  // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+  if (navigator.userAgent.includes("Android")) {
+    await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                    ["media.webrtc.hw.h264.enabled", false]);
+  }
+
+  const test = new PeerConnectionTest();
+
+  const h1 = new CaptureStreamTestHelper2D(50, 50);
+  const canvas1 = h1.createAndAppendElement('canvas', 'source_canvas1');
+  let stream1;
+  let vremote1;
+
+  const h2 = new CaptureStreamTestHelper2D(50, 50);
+  let canvas2;
+  let stream2;
+  let vremote2;
+
+  test.setMediaConstraints([{video: true}], []);
+  test.chain.replace("PC_LOCAL_GUM", [
+    function DRAW_INITIAL_LOCAL_GREEN(test) {
+      h1.drawColor(canvas1, h1.green);
+    },
+    function PC_LOCAL_CANVAS_CAPTURESTREAM(test) {
+      stream1 = canvas1.captureStream(0);
+      test.pcLocal.attachLocalStream(stream1);
+      let i = 0;
+      return setInterval(function() {
+        try {
+          info("draw " + i ? "green" : "red");
+          h1.drawColor(canvas1, i ? h1.green : h1.red);
+          i = 1 - i;
+          stream1.requestFrame();
+          if (stream2 != null) {
+            h2.drawColor(canvas2, i ? h2.green : h2.blue);
+            stream2.requestFrame();
+          }
+        } catch (e) {
+          // ignore; stream might have shut down, and we don't bother clearing
+          // the setInterval.
+        }
+      }, 500);
+    }
+  ]);
+
+  test.chain.append([
+    function FIND_REMOTE_VIDEO() {
+      vremote1 = test.pcRemote.remoteMediaElements[0];
+      ok(!!vremote1, "Should have remote video element for pcRemote");
+    },
+    function WAIT_FOR_REMOTE_GREEN() {
+      return h1.pixelMustBecome(vremote1, h1.green, {
+        threshold: 128,
+        infoString: "pcRemote's remote should become green",
+      });
+    },
+    function WAIT_FOR_REMOTE_RED() {
+      return h1.pixelMustBecome(vremote1, h1.red, {
+        threshold: 128,
+        infoString: "pcRemote's remote should become red",
+      });
+    }
+  ]);
+
+  addRenegotiation(test.chain,
+    [
+      function PC_LOCAL_ADD_SECOND_STREAM(test) {
+        canvas2 = h2.createAndAppendElement('canvas', 'source_canvas2');
+        h2.drawColor(canvas2, h2.blue);
+        stream2 = canvas2.captureStream(0);
+
+        // can't use test.pcLocal.getAllUserMediaAndAddStreams([{video: true}]);
+        // because it doesn't let us substitute the capture stream
+        test.pcLocal.attachLocalStream(stream2);
+      }
+    ]
+  );
+
+  test.chain.append([
+    function FIND_REMOTE2_VIDEO() {
+      vremote2 = test.pcRemote.remoteMediaElements[1];
+      ok(!!vremote2, "Should have remote2 video element for pcRemote");
+    },
+    function WAIT_FOR_REMOTE2_BLUE() {
+      return h2.pixelMustBecome(vremote2, h2.blue, {
+        threshold: 128,
+        infoString: "pcRemote's remote2 should become blue",
+      });
+    },
+    function DRAW_NEW_LOCAL_GREEN(test) {
+      stream1.requestFrame();
+      h1.drawColor(canvas1, h1.green);
+    },
+    function WAIT_FOR_REMOTE1_GREEN() {
+      return h1.pixelMustBecome(vremote1, h1.green, {
+        threshold: 128,
+        infoString: "pcRemote's remote1 should become green",
+      });
+    }
+  ]);
+
+  await test.run();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_videoCodecs.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_videoCodecs.html
new file mode 100644
index 0000000000..7a245b5d8c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_videoCodecs.html
@@ -0,0 +1,142 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="stats.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1395853",
+    title: "Verify video content over WebRTC for every video codec",
+  });
+
+  async function testVideoCodec(options = {}, codec) {
+    const test = new PeerConnectionTest(options);
+    test.setMediaConstraints([{video: true}], []);
+
+    let payloadType;
+    test.chain.insertBefore("PC_LOCAL_SET_LOCAL_DESCRIPTION", [
+      function PC_LOCAL_FILTER_OUT_CODECS() {
+        const otherCodec = codecs.find(c => c != codec);
+        const otherId = sdputils.findCodecId(test.originalOffer.sdp, otherCodec.name, otherCodec.offset);
+        const otherRtpmapMatcher = new RegExp(`a=rtpmap:${otherId}.*\\r\\n`, "gi");
+
+        const id = sdputils.findCodecId(test.originalOffer.sdp, codec.name, codec.offset);
+        payloadType = Number(id);
+        if (codec.offset) {
+          isnot(id, sdputils.findCodecId(test.originalOffer.sdp, codec.name, 0),
+            "Different offsets should return different payload types");
+        }
+        test.originalOffer.sdp =
+          sdputils.removeAllButPayloadType(test.originalOffer.sdp, id);
+
+        ok(!test.originalOffer.sdp.match(new RegExp(`m=.*UDP/TLS/RTP/SAVPF.* ${otherId}[^0-9]`, "gi")),
+          `Other codec ${otherId} should be removed after filtering`);
+        ok(test.originalOffer.sdp.match(new RegExp(`m=.*UDP/TLS/RTP/SAVPF.* ${id}[^0-9]`, "gi")),
+          `Tested codec ${id} should remain after filtering`);
+
+        // We only set it now, or the framework would remove non-H264 codecs
+        // for us.
+        options.h264 = codec.name == "H264";
+      },
+    ]);
+
+    test.chain.insertAfter("PC_LOCAL_WAIT_FOR_MEDIA_FLOW",
+      [PC_LOCAL_TEST_LOCAL_STATS]);
+
+    test.chain.insertAfter("PC_REMOTE_WAIT_FOR_MEDIA_FLOW",
+      [PC_REMOTE_TEST_REMOTE_STATS]);
+
+    test.chain.append([
+      async function PC_LOCAL_TEST_CODEC() {
+        const stats = await test.pcLocal._pc.getStats();
+        let codecCount = 0;
+        stats.forEach(stat => {
+          if (stat.type == "codec") {
+            is(codecCount++, 0, "expected only one encode codec stat");
+            is(stat.payloadType, payloadType, "payloadType as expected");
+            is(stat.mimeType, `video/${codec.name}`, "mimeType as expected");
+            is(stat.codecType, "encode", "codecType as expected");
+          }
+        });
+      },
+      async function PC_REMOTE_TEST_CODEC() {
+        const stats = await test.pcRemote._pc.getStats();
+        let codecCount = 0;
+        stats.forEach(stat => {
+          if (stat.type == "codec") {
+            is(codecCount++, 0, "expected only one decode codec stat");
+            is(stat.payloadType, payloadType, "payloadType as expected");
+            is(stat.mimeType, `video/${codec.name}`, "mimeType as expected");
+            is(stat.codecType, "decode", "codecType as expected");
+          }
+        });
+      },
+      async function CHECK_VIDEO_FLOW() {
+        try {
+          const h = new VideoStreamHelper();
+          await h.checkVideoPlaying(
+              test.pcRemote.remoteMediaElements[0],
+              10, 10, 128);
+          ok(true, `Got video flow for codec ${codec.name}, offset ${codec.offset}`);
+        } catch(e) {
+          ok(false, `No video flow for codec ${codec.name}, offset ${codec.offset}: ${e}`);
+        }
+      },
+    ]);
+
+    await test.run();
+  }
+
+  // We match the name against the sdp to figure out the payload type,
+  // so all other present codecs can be removed.
+  // Use `offset` when there are multiple instances of a codec expected in an sdp.
+  const codecs = [
+    { name: "VP8" },
+    { name: "VP9" },
+    { name: "H264" },
+    { name: "H264", offset: 1 },
+  ];
+
+  runNetworkTest(async (options) => {
+    // This test expects the video being captured will change color. Use fake
+    // video device as loopback does not currently change.
+    await pushPrefs(
+      ['media.video_loopback_dev', ''],
+      ['media.navigator.streams.fake', true]);
+    for (let codec of codecs) {
+      info(`Testing video for codec ${codec.name} offset ${codec.offset}`);
+      try {
+        let enc = SpecialPowers.getBoolPref('media.webrtc.platformencoder');
+        let dec = SpecialPowers.getBoolPref('media.navigator.mediadatadecoder_h264_enabled');
+        if (codec.name == "H264") {
+          await matchPlatformH264CodecPrefs();
+          if (codec.offset == 1) {
+            // Force fake GMP codec for H.264 mode 0 because not all platforms
+            // support slice size control. Re-enable it after
+            // a. SW encoder fallback support (bug 1726617), and
+            // b. returning valid bitstream from fake GMP encoder (bug 1509012).
+            await pushPrefs(
+              ['media.webrtc.platformencoder', false],
+              ['media.navigator.mediadatadecoder_h264_enabled', false],
+            );
+          }
+        }
+        await testVideoCodec(options, codec);
+        await pushPrefs(
+          ['media.webrtc.platformencoder', enc],
+          ['media.navigator.mediadatadecoder_h264_enabled', dec],
+        );
+      } catch(e) {
+        ok(false, `Error in test for codec ${codec.name}: ${e}\n${e.stack}`);
+      }
+      info(`Tested video for codec ${codec.name}`);
+    }
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_videoRenegotiationInactiveAnswer.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_videoRenegotiationInactiveAnswer.html
new file mode 100644
index 0000000000..b77633493d
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_videoRenegotiationInactiveAnswer.html
@@ -0,0 +1,95 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+  <script type="application/javascript" src="sdpUtils.js"></script>
+  <script type="application/javascript" src="/tests/dom/canvas/test/captureStream_common.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    bug: "1213773",
+    title: "Renegotiation: answerer uses a=inactive for video"
+  });
+
+  runNetworkTest(async (options) => {
+    // [TODO] re-enable HW decoder after bug 1526207 is fixed.
+    if (navigator.userAgent.includes("Android")) {
+      await pushPrefs(["media.navigator.mediadatadecoder_vpx_enabled", false],
+                      ["media.webrtc.hw.h264.enabled", false]);
+    }
+
+    const emitter = new VideoFrameEmitter();
+    const helper = new VideoStreamHelper();
+
+    const test = new PeerConnectionTest(options);
+
+    test.chain.replace("PC_LOCAL_GUM", [
+      function PC_LOCAL_CANVAS_CAPTURESTREAM(test) {
+        test.pcLocal.attachLocalStream(emitter.stream());
+        emitter.start();
+      }
+    ]);
+
+    var haveFirstUnmuteEvent;
+
+    test.chain.insertBefore("PC_REMOTE_SET_LOCAL_DESCRIPTION", [
+      function PC_REMOTE_SETUP_ONUNMUTE_1() {
+        haveFirstUnmuteEvent = haveEvent(test.pcRemote._pc.getReceivers()[0].track, "unmute");
+      }
+    ]);
+
+    test.chain.append([
+      function PC_REMOTE_CHECK_VIDEO_UNMUTED() {
+        return haveFirstUnmuteEvent;
+      },
+      function PC_REMOTE_WAIT_FOR_FRAMES() {
+        var vremote = test.pcRemote.remoteMediaElements[0];
+        ok(vremote, "Should have remote video element for pcRemote");
+        return addFinallyToPromise(helper.checkVideoPlaying(vremote))
+            .finally(() => emitter.stop());
+      }
+    ]);
+
+    addRenegotiation(test.chain, []);
+
+    test.chain.insertAfter("PC_LOCAL_GET_ANSWER", [
+        function PC_LOCAL_REWRITE_REMOTE_SDP_INACTIVE(test) {
+          test._remote_answer.sdp =
+            sdputils.setAllMsectionsInactive(test._remote_answer.sdp);
+        }
+    ], false, 1);
+
+    test.chain.append([
+      function PC_REMOTE_ENSURE_NO_FRAMES() {
+        var vremote = test.pcRemote.remoteMediaElements[0];
+        ok(vremote, "Should have remote video element for pcRemote");
+        emitter.start();
+        return addFinallyToPromise(helper.checkVideoPaused(vremote))
+            .finally(() => emitter.stop());
+      },
+    ]);
+
+    test.chain.remove("PC_REMOTE_CHECK_STATS", 1);
+    test.chain.remove("PC_LOCAL_CHECK_STATS", 1);
+
+    addRenegotiation(test.chain, []);
+
+    test.chain.append([
+      function PC_REMOTE_WAIT_FOR_FRAMES_2() {
+        var vremote = test.pcRemote.remoteMediaElements[0];
+        ok(vremote, "Should have remote video element for pcRemote");
+        emitter.start();
+        return addFinallyToPromise(helper.checkVideoPlaying(vremote))
+            .finally(() => emitter.stop());
+      }
+    ]);
+
+    test.setMediaConstraints([{video: true}], []);
+    await test.run();
+  });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_peerConnection_webAudio.html b/dom/media/webrtc/tests/mochitests/test_peerConnection_webAudio.html
new file mode 100644
index 0000000000..1d695ecbfa
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_peerConnection_webAudio.html
@@ -0,0 +1,43 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  bug: "1081819",
+  title: "WebAudio on both input and output side of peerconnection"
+});
+
+// This tests WebAudio (a 700Hz OscillatorNode) as input to a PeerConnection.
+// It also tests that a PeerConnection works as input to WebAudio as the remote
+// stream is connected to an AnalyserNode and compared to the source node.
+
+runNetworkTest(function() {
+  const test = new PeerConnectionTest();
+  test.audioContext = new AudioContext();
+  test.setMediaConstraints([{audio: true}], []);
+  test.chain.replace("PC_LOCAL_GUM", [
+    function PC_LOCAL_WEBAUDIO_SOURCE(test) {
+      const oscillator = test.audioContext.createOscillator();
+      oscillator.type = 'sine';
+      oscillator.frequency.value = 700;
+      oscillator.start();
+      const dest = test.audioContext.createMediaStreamDestination();
+      oscillator.connect(dest);
+      test.pcLocal.attachLocalStream(dest.stream);
+    }
+  ]);
+  test.chain.append([
+    function CHECK_AUDIO_FLOW(test) {
+      return test.pcRemote.checkReceivingToneFrom(test.audioContext, test.pcLocal);
+    }
+  ]);
+  return test.run();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_selftest.html b/dom/media/webrtc/tests/mochitests/test_selftest.html
new file mode 100644
index 0000000000..3f1ce1402d
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_selftest.html
@@ -0,0 +1,37 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="pc.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+  createHTML({
+    title: "Self-test of harness functions",
+    visible: true
+  });
+
+function TEST(test) {}
+
+var catcher = func => {
+  try {
+    func();
+    return null;
+  } catch (e) {
+    return e.message;
+  }
+};
+
+runNetworkTest(() => {
+  var test = new PeerConnectionTest();
+  test.setMediaConstraints([{video: true}], [{video: true}]);
+  is(catcher(() => test.chain.replace("PC_LOCAL_SET_LOCAL_DESCRIPTION", TEST)),
+     null, "test.chain.replace works");
+  is(catcher(() => test.chain.replace("FOO", TEST)),
+     "Unknown test: FOO", "test.chain.replace catches typos");
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_setSinkId.html b/dom/media/webrtc/tests/mochitests/test_setSinkId.html
new file mode 100644
index 0000000000..0d85114a0e
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_setSinkId.html
@@ -0,0 +1,83 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+
+<script>
+  createHTML({
+    title: "SetSinkId in HTMLMediaElement",
+    bug: "934425",
+  });
+
+  const memoryReportPath = 'explicit/media/media-manager-aggregates';
+
+  /**
+   * Run a test to verify set sink id in audio element.
+   */
+  runTest(async () => {
+    await pushPrefs(["media.setsinkid.enabled", true]);
+
+    if (!SpecialPowers.getCharPref("media.audio_loopback_dev", "")) {
+      ok(false, "No loopback device set by framework. Try --use-test-media-devices");
+      return;
+    }
+
+    // Expose an audio output device.
+    SpecialPowers.wrap(document).notifyUserGestureActivation();
+    await navigator.mediaDevices.selectAudioOutput();
+
+    const allDevices = await navigator.mediaDevices.enumerateDevices();
+    const audioDevices = allDevices.filter(({kind}) => kind == 'audiooutput');
+    is(audioDevices.length, 1, "Number of output devices found");
+
+    const audio = createMediaElement("audio", "audio");
+    document.body.appendChild(audio);
+
+    is(audio.sinkId, "", "Initial value is empty string");
+
+    const p = audio.setSinkId(audioDevices[0].deviceId);
+    is(audio.sinkId, "", "Value is unchanged upon function return");
+    is(await p, undefined, "promise resolves with undefined");
+    is(audio.sinkId, audioDevices[0].deviceId, `Sink device is set, id: ${audio.sinkId}`);
+
+    await audio.setSinkId(audioDevices[0].deviceId);
+    ok(true, `Sink device is set for 2nd time for the same id: ${audio.sinkId}`);
+
+    try {
+      await audio.setSinkId("dummy sink id");
+      ok(false, "Never enter here, this must fail");
+    } catch (error) {
+      ok(true, `Set sink id expected to fail: ${error}`);
+      is(error.name, "NotFoundError", "Verify correct error");
+    }
+
+    const {usage: usage1} =
+          await collectMemoryUsage(memoryReportPath); // Provided by head.js
+
+    ok(usage1 > 0, "MediaManager memory usage should be non-zero to store \
+device ids after enumerateDevices");
+
+    const p2 = audio.setSinkId("");
+    is(audio.sinkId, audioDevices[0].deviceId,
+       'sinkId after setSinkId("") return');
+    is(await p2, undefined,
+       "promise resolution value when sinkId parameter is empty");
+    is(audio.sinkId, "", 'sinkId after setSinkId("") resolution');
+
+    await audio.setSinkId(audioDevices[0].deviceId);
+
+    const {usage: usage2, reportCount} =
+          await collectMemoryUsage(memoryReportPath);
+    is(reportCount, 1,
+       'Expect only one MediaManager to report in content processes.');
+    is(usage2, usage1, "MediaManager memory usage should return to previous \
+value after promise resolution");
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_setSinkId_default_addTrack.html b/dom/media/webrtc/tests/mochitests/test_setSinkId_default_addTrack.html
new file mode 100644
index 0000000000..64db4cad7c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_setSinkId_default_addTrack.html
@@ -0,0 +1,52 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+
+<script>
+  createHTML({
+    title: "HTMLMediaElement.setSinkId with default device and adding a track",
+    bug: "1661649",
+  });
+
+  /**
+   * Run a test to verify set sink id in audio element.
+   */
+  runTest(async () => {
+    await pushPrefs(["media.setsinkid.enabled", true]);
+
+    // Expose an audio output device.
+    SpecialPowers.wrap(document).notifyUserGestureActivation();
+    await navigator.mediaDevices.selectAudioOutput();
+
+    const allDevices = await navigator.mediaDevices.enumerateDevices();
+    const audioDevices = allDevices.filter(({kind}) => kind == 'audiooutput');
+    info(`Found  ${audioDevices.length} output devices`);
+    isnot(audioDevices.length, 0, "Found output devices");
+
+    const audio = createMediaElement("audio", "audio");
+    document.body.appendChild(audio);
+
+    audio.srcObject = await navigator.mediaDevices.getUserMedia({audio: true});
+    audio.play();
+
+    await audio.setSinkId(audioDevices[0].deviceId);
+    await audio.setSinkId("");
+    is(audio.sinkId, "", "sinkId restored to default");
+
+    audio.srcObject.addTrack((await navigator.mediaDevices.getUserMedia({audio: true})).getTracks()[0]);
+
+    await wait(0);
+
+    for (let t of audio.srcObject.getTracks()) {
+      t.stop();
+    }
+  });
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_setSinkId_preMutedElement.html b/dom/media/webrtc/tests/mochitests/test_setSinkId_preMutedElement.html
new file mode 100644
index 0000000000..fb65c3312f
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_setSinkId_preMutedElement.html
@@ -0,0 +1,100 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+<script type="application/javascript">
+createHTML({
+  title: "Test changing sink and muting before the MediaStream is set",
+  bug: "1651049",
+  visible: true
+});
+
+let getOutputDeviceId = async () => {
+  let devices = await navigator.mediaDevices.enumerateDevices();
+  let audios = devices.filter(d => d.kind == "audiooutput");
+  ok(audios.length, "One or more output devices found.");
+  return audios[0].deviceId;
+}
+
+let verifyAudioTone = async (ac, stream, freq) => {
+  const toneAnalyser = new AudioStreamAnalyser(ac, stream);
+  return toneAnalyser.waitForAnalysisSuccess(array => {
+    const lowerFreq = freq / 2;
+    const upperFreq = freq + 1000;
+    const lowerMag = array[toneAnalyser.binIndexForFrequency(lowerFreq)];
+    const freqMag = array[toneAnalyser.binIndexForFrequency(freq)];
+    const upperMag = array[toneAnalyser.binIndexForFrequency(upperFreq)];
+    info("Audio tone expected. "
+         + lowerFreq + ": " + lowerMag + ", "
+         + freq + ": " + freqMag + ", "
+         + upperFreq + ": " + upperMag);
+    return lowerMag < 50 && freqMag > 200 && upperMag < 50;
+  });
+}
+
+let verifyNoAudioTone = async (ac, stream, freq) => {
+  const toneAnalyser = new AudioStreamAnalyser(ac, stream);
+  // repeat check 100 times to make sure that it is muted.
+  let retryCnt = 0;
+  return toneAnalyser.waitForAnalysisSuccess(array => {
+    const lowerFreq = freq / 2;
+    const upperFreq = freq + 1000;
+    const lowerMag = array[toneAnalyser.binIndexForFrequency(lowerFreq)];
+    const freqMag = array[toneAnalyser.binIndexForFrequency(freq)];
+    const upperMag = array[toneAnalyser.binIndexForFrequency(upperFreq)];
+    info("No audio tone expected. "
+         + lowerFreq + ": " + lowerMag + ", "
+         + freq + ": " + freqMag + ", "
+         + upperFreq + ": " + upperMag);
+    return lowerMag == 0 && freqMag == 0 && upperMag == 0 && ++retryCnt == 100;
+  });
+}
+
+runTest(async () => {
+  let audioDevice = SpecialPowers.getCharPref("media.audio_loopback_dev", "");
+  if (!audioDevice) {
+    todo(false, "No loopback device set by framework. Try --use-test-media-devices");
+    return;
+  }
+
+  await pushPrefs(["media.setsinkid.enabled", true]);
+
+  // Implicitly expose the loopback device sink by opening the source in the
+  // same group.
+  const verifyStream = await getUserMedia({audio: true});
+  // We gonna test our tone, stop the auto created one.
+  DefaultLoopbackTone.stop();
+
+  let sinkId = await getOutputDeviceId();
+  isnot(sinkId, "", "SinkId is not null");
+
+  let audioElement = createMediaElement('audio', 'audioElement');
+  audioElement.muted = true;
+  await audioElement.setSinkId(sinkId);
+  isnot(audioElement.sinkId, "", "sinkId property of the element is not null");
+
+  // The test stream is a sine tone of 1000 Hz
+  let ac = new AudioContext();
+  const frequency = 2000;
+  let stream = createOscillatorStream(ac, frequency);
+  await verifyAudioTone(ac, stream, frequency);
+
+  audioElement.srcObject = stream;
+  audioElement.play();
+
+  // Verify the silent output using the loopback device.
+  await verifyNoAudioTone(ac, verifyStream, frequency);
+  info("output is muted");
+
+  // Clean up
+  audioElement.pause();
+  audioElement.srcObject = null;
+  verifyStream.getTracks()[0].stop();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/test_unfocused_pref.html b/dom/media/webrtc/tests/mochitests/test_unfocused_pref.html
new file mode 100644
index 0000000000..22df020f7c
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/test_unfocused_pref.html
@@ -0,0 +1,49 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <meta charset="utf-8">
+  <script src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<script>
+"use strict";
+
+createHTML({
+  // This pref exists only for a partner testing framework without WebDriver
+  // switch-to-window nor SpecialPowers to set the active window.
+  // Prefer "focusmanager.testmode".
+  title: "Test media.devices.unfocused.enabled",
+  bug: "1740824"
+});
+
+const blank_url = "/tests/docshell/test/navigation/blank.html";
+
+async function resolveOnEvent(target, name) {
+  return new Promise(r => target.addEventListener(name, r, {once: true}));
+}
+
+runTest(async () => {
+  ok(document.hasFocus(), "This test expects initial focus on the document.");
+  // 'resizable' is requested for a separate OS window on relevant platforms
+  // so that this test tests OS focus changes rather than document visibility.
+  const other = window.open(blank_url, "", "resizable");
+  SimpleTest.registerCleanupFunction(() => {
+    other.close();
+    return SimpleTest.promiseFocus(window);
+  });
+  await Promise.all([
+    resolveOnEvent(window, 'blur'),
+    SimpleTest.promiseFocus(other),
+    pushPrefs(["media.devices.unfocused.enabled", true]),
+  ]);
+  ok(!document.hasFocus(), "!document.hasFocus()");
+  await navigator.mediaDevices.enumerateDevices();
+  ok(true, "enumerateDevices() completes without focus.");
+  // The focus requirement with media.devices.unfocused.enabled false
+  // (default) is tested in
+  // testing/web-platform/mozilla/tests/mediacapture-streams/enumerateDevices-without-focus.https.html
+});
+
+</script>
+</body>
+</html>
diff --git a/dom/media/webrtc/tests/mochitests/turnConfig.js b/dom/media/webrtc/tests/mochitests/turnConfig.js
new file mode 100644
index 0000000000..1267de4ec5
--- /dev/null
+++ b/dom/media/webrtc/tests/mochitests/turnConfig.js
@@ -0,0 +1,16 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+/* An example of how to specify two TURN server configs:
+ *
+ * Note: If turn URL uses FQDN rather then an IP address the TURN relay
+ *       verification step in checkStatsIceConnectionType might fail.
+ *
+ * var turnServers = {
+ *   local: { iceServers: [{"username":"mozilla","credential":"mozilla","url":"turn:10.0.0.1"}] },
+ *   remote: { iceServers: [{"username":"firefox","credential":"firefox","url":"turn:10.0.0.2"}] }
+ * };
+ */
+
+var turnServers = {};
diff --git a/dom/media/webrtc/third_party_build/README.md b/dom/media/webrtc/third_party_build/README.md
new file mode 100644
index 0000000000..9151445c87
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/README.md
@@ -0,0 +1,17 @@
+# Vendoring libwebrtc and the fast-forward process
+
+Most of the important information about this process is contained on the fast-forward
+automation wiki page
+[here](https://wiki.mozilla.org/Media/WebRTC/libwebrtc_Update_Process/automation_plan).
+
+To skip the history and details and go directly to starting the libwebrtc fast-foward
+process, go to the
+[Operation Checklist](https://wiki.mozilla.org/Media/WebRTC/libwebrtc_Update_Process/automation_plan#Operation_Checklist).
+
+# Fixing errors reported in scripts
+
+In most cases, the scripts report errors including suggestions on how to resolve the
+issue.  If you're seeing an error message referring you to this README.md file, the
+likely issue is that you're missing environment variables that should be set in a
+config_env file in .moz-fast-forward.  The default for that file can be found at
+dom/media/webrtc/third_party_build/default_config_env.
diff --git a/dom/media/webrtc/third_party_build/build_no_op_commits.sh b/dom/media/webrtc/third_party_build/build_no_op_commits.sh
new file mode 100644
index 0000000000..eff61a5eb5
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/build_no_op_commits.sh
@@ -0,0 +1,126 @@
+#!/bin/bash
+
+function show_error_msg()
+{
+  echo "*** ERROR *** $? line $1 $0 did not complete successfully!"
+  echo "$ERROR_HELP"
+}
+ERROR_HELP=""
+
+# Print an Error message if `set -eE` causes the script to exit due to a failed command
+trap 'show_error_msg $LINENO' ERR
+
+source dom/media/webrtc/third_party_build/use_config_env.sh
+
+echo "MOZ_LIBWEBRTC_SRC: $MOZ_LIBWEBRTC_SRC"
+
+# After this point:
+# * eE: All commands should succeed.
+# * u: All variables should be defined before use.
+# * o pipefail: All stages of all pipes should succeed.
+set -eEuo pipefail
+
+CURRENT_DIR=`pwd`
+cd $MOZ_LIBWEBRTC_SRC
+
+MANUAL_INTERVENTION_COMMIT_FILE="$TMP_DIR/manual_commits.txt"
+rm -f $MANUAL_INTERVENTION_COMMIT_FILE
+
+# Find the common commit between our previous work branch and trunk
+CURRENT_RELEASE_BASE=`git merge-base branch-heads/$MOZ_PRIOR_UPSTREAM_BRANCH_HEAD_NUM master`
+
+# Write no-op files for the cherry-picked release branch commits.  For more
+# details on what this is doing, see make_upstream_revert_noop.sh.
+COMMITS=`git log -r $CURRENT_RELEASE_BASE..branch-heads/$MOZ_PRIOR_UPSTREAM_BRANCH_HEAD_NUM --format='%h'`
+for commit in $COMMITS; do
+
+  echo "Processing release branch commit $commit for no-op handling"
+
+  # Don't process the commit if the commit message is missing the customary
+  # line that shows which upstream commit is being cherry-picked.
+  CNT=`git show $commit | grep "cherry picked from commit" | wc -l | tr -d " " || true`
+  if [ $CNT != 1 ]; then
+    # record the commit to list at the end of this script as
+    # 'needing intervention'
+    echo "    no cherry-pick info found, skipping commit $commit"
+    echo "$commit" >> $MANUAL_INTERVENTION_COMMIT_FILE
+    continue
+  fi
+
+  CHERRY_PICK_COMMIT=`git show $commit | grep "cherry picked from commit" | tr -d "()" | awk '{ print $5; }'`
+  SHORT_SHA=`git show --name-only $CHERRY_PICK_COMMIT --format='%h' | head -1`
+  echo "    commit $commit cherry-picks $SHORT_SHA"
+
+  echo "We already cherry-picked this when we vendored $commit." \
+  > $STATE_DIR/$SHORT_SHA.no-op-cherry-pick-msg
+
+done
+
+# This section checks for commits that may have been cherry-picked in more
+# than one release branch.
+TARGET_RELEASE_BASE=`git merge-base $MOZ_TARGET_UPSTREAM_BRANCH_HEAD master`
+NEW_COMMITS=`git log -r $TARGET_RELEASE_BASE..$MOZ_TARGET_UPSTREAM_BRANCH_HEAD --format='%h'`
+
+# Convert the files that we've already generated for no-op detection into
+# something that we can use as a regular expression for searching.
+KNOWN_NO_OP_COMMITS=`cd $STATE_DIR ; \
+  ls *.no-op-cherry-pick-msg \
+  | sed 's/\.no-op-cherry-pick-msg//' \
+  | paste -sd '|' /dev/stdin`
+
+for commit in $NEW_COMMITS; do
+
+  echo "Processing next release branch commit $commit for no-op handling"
+
+  # Don't process the commit if the commit message is missing the customary
+  # line that shows which upstream commit is being cherry-picked.
+  CNT=`git show $commit | grep "cherry picked from commit" | wc -l | tr -d " " || true`
+  if [ $CNT != 1 ]; then
+    # record the commit to list at the end of this script as
+    # 'needing intervention'
+    echo "    no cherry-pick info found, skipping commit $commit"
+    echo "$commit" >> $MANUAL_INTERVENTION_COMMIT_FILE
+    continue
+  fi
+
+  CHERRY_PICK_COMMIT=`git show $commit | grep "cherry picked from commit" | tr -d "()" | awk '{ print $5; }'`
+  SHORT_SHA=`git show --name-only $CHERRY_PICK_COMMIT --format='%h' | head -1`
+
+  # The trick here is that we only want to include no-op processing for the
+  # commits that appear both here _and_ in the previous release's cherry-pick
+  # commits.  We check the known list of no-op commits to see if it was
+  # cherry picked in the previous release branch and then create another
+  # file for the new release branch commit that will ultimately be a no-op.
+  if [[ "$SHORT_SHA" =~ ^($KNOWN_NO_OP_COMMITS)$ ]]; then
+    echo "    commit $commit cherry-picks $SHORT_SHA"
+    cp $STATE_DIR/$SHORT_SHA.no-op-cherry-pick-msg $STATE_DIR/$commit.no-op-cherry-pick-msg
+  fi
+
+done
+
+if [ ! -f $MANUAL_INTERVENTION_COMMIT_FILE ]; then
+  echo "No commits require manual intervention"
+  exit
+fi
+
+echo $"
+Each of the following commits requires manual intervention to
+verify the source of the cherry-pick or there may be errors
+reported during the fast-forward processing.  Without this
+intervention, the common symptom is that the vendored commit
+file count (0) will not match the upstream commit file count.
+"
+
+for commit in `cat $MANUAL_INTERVENTION_COMMIT_FILE`; do
+  SUMMARY=`git show --oneline --name-only $commit | head -1`
+  echo "  '$SUMMARY'"
+done
+
+echo $"
+To manually create the no-op tracking files needed,
+run the following command for each commit in question:
+  ( export UPSTREAM_COMMIT=\"{sha-of-upstream-commit}\" ; \\
+    export PICKED_COMMIT=\"{sha-of-already-used-commit}\" ; \\
+    echo \"We already cherry-picked this when we vendored \$PICKED_COMMIT.\" \\
+    > $STATE_DIR/\$UPSTREAM_COMMIT.no-op-cherry-pick-msg )
+"
diff --git a/dom/media/webrtc/third_party_build/commit-build-file-changes.sh b/dom/media/webrtc/third_party_build/commit-build-file-changes.sh
new file mode 100644
index 0000000000..2b4c791fca
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/commit-build-file-changes.sh
@@ -0,0 +1,47 @@
+#!/bin/bash
+
+function show_error_msg()
+{
+  echo "*** ERROR *** $? line $1 $0 did not complete successfully!"
+  echo "$ERROR_HELP"
+}
+ERROR_HELP=""
+
+# Print an Error message if `set -eE` causes the script to exit due to a failed command
+trap 'show_error_msg $LINENO' ERR
+
+# All commands should be printed as they are executed
+# set -x
+
+# After this point:
+# * eE: All commands should succede.
+# * u: All variables should be defined before use.
+# * o pipefail: All stages of all pipes should succede.
+set -eEuo pipefail
+
+MOZ_BUILD_CHANGE_CNT=`hg status third_party/libwebrtc | wc -l | tr -d " "`
+echo "MOZ_BUILD_CHANGE_CNT: $MOZ_BUILD_CHANGE_CNT"
+if [ "x$MOZ_BUILD_CHANGE_CNT" != "x0" ]; then
+  CURRENT_COMMIT_SHA=`hg id -i | sed 's/+//'`
+  COMMIT_DESC=`hg --config alias.log=log log -T '{desc|firstline}' -r $CURRENT_COMMIT_SHA`
+
+  # since we have build file changes, touch the CLOBBER file
+  cat CLOBBER | egrep "^#|^$" > CLOBBER.new
+  mv CLOBBER.new CLOBBER
+  echo "Modified build files in third_party/libwebrtc - $COMMIT_DESC" >> CLOBBER
+
+  ADD_CNT=`hg status -nu third_party/libwebrtc | wc -l | tr -d " "`
+  DEL_CNT=`hg status -nd third_party/libwebrtc | wc -l | tr -d " "`
+  if [ "x$ADD_CNT" != "x0" ]; then
+    hg status -nu third_party/libwebrtc | xargs hg add
+  fi
+  if [ "x$DEL_CNT" != "x0" ]; then
+    hg status -nd third_party/libwebrtc | xargs hg rm
+  fi
+
+  hg commit -m \
+    "$COMMIT_DESC - moz.build file updates" \
+    third_party/libwebrtc CLOBBER
+fi
+
+echo "Done in $0"
diff --git a/dom/media/webrtc/third_party_build/default_config_env b/dom/media/webrtc/third_party_build/default_config_env
new file mode 100644
index 0000000000..a50694f0e3
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/default_config_env
@@ -0,0 +1,42 @@
+#!/bin/bash
+
+# Edit {path-to} to match the location of your copy of Mozilla's
+# fork of libwebrtc (at https://github.com/mozilla/libwebrtc).
+export MOZ_LIBWEBRTC_SRC=$STATE_DIR/moz-libwebrtc
+
+# Fast-forwarding each Chromium version of libwebrtc should be done
+# under a separate bugzilla bug.  This bug number is used when crafting
+# the commit summary as each upstream commit is vendored into the
+# mercurial repository.  The bug used for the v106 fast-forward was
+# 1800920.
+export MOZ_FASTFORWARD_BUG="1833237"
+
+# MOZ_NEXT_LIBWEBRTC_MILESTONE and MOZ_NEXT_FIREFOX_REL_TARGET are
+# not used during fast-forward processing, but facilitate generating this
+# default config.  To generate an default config for the next update, run
+# bash dom/media/webrtc/third_party_build/update_default_config_env.sh
+export MOZ_NEXT_LIBWEBRTC_MILESTONE=112
+export MOZ_NEXT_FIREFOX_REL_TARGET=116
+
+# For Chromium release branches, see:
+# https://chromiumdash.appspot.com/branches
+
+# Chromium's v111 release branch was 5563.  This is used to pre-stack
+# the previous release branch's commits onto the appropriate base commit
+# (the first common commit between trunk and the release branch).
+export MOZ_PRIOR_UPSTREAM_BRANCH_HEAD_NUM="5563"
+
+# New target release branch for v112 is branch-heads/5615.  This is used
+# to calculate the next upstream commit.
+export MOZ_TARGET_UPSTREAM_BRANCH_HEAD="branch-heads/5615"
+
+# For local development 'mozpatches' is fine for a branch name, but when
+# pushing the patch stack to github, it should be named something like
+# 'moz-mods-chr112-for-rel116'.
+export MOZ_LIBWEBRTC_BRANCH="mozpatches"
+
+# After elm has been merged to mozilla-central, the patch stack in
+# moz-libwebrtc should be pushed to github.  The script
+# push_official_branch.sh uses this branch name when pushing to the
+# public repo.
+export MOZ_LIBWEBRTC_OFFICIAL_BRANCH="moz-mods-chr112-for-rel116"
diff --git a/dom/media/webrtc/third_party_build/detect_upstream_revert.sh b/dom/media/webrtc/third_party_build/detect_upstream_revert.sh
new file mode 100644
index 0000000000..2dc868952d
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/detect_upstream_revert.sh
@@ -0,0 +1,93 @@
+#!/bin/bash
+
+function show_error_msg()
+{
+  echo "*** ERROR *** $? line $1 $0 did not complete successfully!"
+  echo "$ERROR_HELP"
+}
+ERROR_HELP=""
+
+# Print an Error message if `set -eE` causes the script to exit due to a failed command
+trap 'show_error_msg $LINENO' ERR
+
+source dom/media/webrtc/third_party_build/use_config_env.sh
+
+# If DEBUG_GEN is set all commands should be printed as they are executed
+if [ ! "x$DEBUG_GEN" = "x" ]; then
+  set -x
+fi
+
+if [ "x$MOZ_LIBWEBRTC_SRC" = "x" ]; then
+  echo "MOZ_LIBWEBRTC_SRC is not defined, see README.md"
+  exit
+fi
+
+if [ -d $MOZ_LIBWEBRTC_SRC ]; then
+  echo "MOZ_LIBWEBRTC_SRC is $MOZ_LIBWEBRTC_SRC"
+else
+  echo "Path $MOZ_LIBWEBRTC_SRC is not found, see README.md"
+  exit
+fi
+
+if [ "x$MOZ_LIBWEBRTC_BRANCH" = "x" ]; then
+  echo "MOZ_LIBWEBRTC_BRANCH is not defined, see README.md"
+  exit
+fi
+
+if [ "x$AUTO_FIX_REVERT_AS_NOOP" = "x" ]; then
+  AUTO_FIX_REVERT_AS_NOOP="0"
+fi
+
+find_base_commit
+find_next_commit
+
+MOZ_LIBWEBRTC_COMMIT_MSG=`cd $MOZ_LIBWEBRTC_SRC ; \
+git show --name-only --oneline $MOZ_LIBWEBRTC_NEXT_BASE \
+ | head -1 | sed 's/[^ ]* //'`
+
+echo "MOZ_LIBWEBRTC_BASE: $MOZ_LIBWEBRTC_BASE"
+echo "MOZ_LIBWEBRTC_NEXT_BASE: $MOZ_LIBWEBRTC_NEXT_BASE"
+echo "MOZ_LIBWEBRTC_COMMIT_MSG: $MOZ_LIBWEBRTC_COMMIT_MSG"
+export MOZ_LIBWEBRTC_REVERT_SHA=`cd $MOZ_LIBWEBRTC_SRC ; \
+git log --oneline -r $MOZ_LIBWEBRTC_BASE..$MOZ_TARGET_UPSTREAM_BRANCH_HEAD \
+ | grep -F "Revert \"$MOZ_LIBWEBRTC_COMMIT_MSG" \
+ | tail -1 | awk '{print $1;}' || true`
+
+echo "MOZ_LIBWEBRTC_REVERT_SHA: $MOZ_LIBWEBRTC_REVERT_SHA"
+
+if [ "x$MOZ_LIBWEBRTC_REVERT_SHA" == "x" ]; then
+  echo "no revert commit detected"
+  exit
+fi
+
+if [ "x$AUTO_FIX_REVERT_AS_NOOP" = "x1" ]; then
+  echo "AUTO_FIX_REVERT_AS_NOOP detected, fixing land/revert pair automatically"
+  bash $SCRIPT_DIR/make_upstream_revert_noop.sh
+  exit
+fi
+
+echo $"
+The next upstream commit has a corresponding future \"Revert\" commit.
+
+There are 2 common ways forward in this situation:
+1. If you're relatively certain there will not be rebase conflicts in the
+   github repo ($MOZ_LIBWEBRTC_SRC), simply run:
+       SKIP_NEXT_REVERT_CHK=1 bash $SCRIPT_DIR/loop-ff.sh
+
+2. The surer method for no rebase conflicts is to cherry-pick both the
+   next commit, and the commit that reverts the next commit onto the
+   bottom of our patch stack in github.  This pushes the likely rebase
+   conflict into the future when the upstream fix is relanded, but
+   ensures we only have to deal with the conflict once.  The following
+   commands will add the necessary commits to the bottom of our patch
+   stack in github, and leave indicator files in the home directory that
+   help loop-ff know when to invoke special no-op commit handling:
+
+       MOZ_LIBWEBRTC_BASE=$MOZ_LIBWEBRTC_BASE \\
+       MOZ_LIBWEBRTC_NEXT_BASE=$MOZ_LIBWEBRTC_NEXT_BASE \\
+       MOZ_LIBWEBRTC_REVERT_SHA=$MOZ_LIBWEBRTC_REVERT_SHA \\
+       bash $SCRIPT_DIR/make_upstream_revert_noop.sh
+
+       SKIP_NEXT_REVERT_CHK=1 bash $SCRIPT_DIR/loop-ff.sh
+"
+exit 1
diff --git a/dom/media/webrtc/third_party_build/elm_arcconfig.patch b/dom/media/webrtc/third_party_build/elm_arcconfig.patch
new file mode 100644
index 0000000000..46adb9c6f1
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/elm_arcconfig.patch
@@ -0,0 +1,10 @@
+diff --git a/.arcconfig b/.arcconfig
+--- a/.arcconfig
++++ b/.arcconfig
+@@ -1,5 +1,5 @@
+ {
+   "phabricator.uri" : "https://phabricator.services.mozilla.com/",
+-  "repository.callsign": "MOZILLACENTRAL",
++  "repository.callsign": "ELM",
+   "history.immutable": false
+ }
diff --git a/dom/media/webrtc/third_party_build/elm_rebase.sh b/dom/media/webrtc/third_party_build/elm_rebase.sh
new file mode 100644
index 0000000000..4d53373d35
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/elm_rebase.sh
@@ -0,0 +1,247 @@
+#!/bin/bash
+
+# This script exists to help with the rebase process on elm.  It rebases
+# each patch individually to make it easier to fix rebase conflicts
+# without jeopardizing earlier, sucessfully rebased commits. In order to
+# limit rebase conflicts around generated moz.build files, it regenerates
+# moz.build file commits.  It also ensures any commits with 'FLOAT' in the
+# commit summary are pushed to the top of the fast-forward stack to help
+# the sheriffs more easily merge our commit stack from elm to moz-central.
+#
+# Occasionally, there will be upstream vendored commits that break the
+# build file generation with follow on commits that fix that error. In
+# order to allow the rebase process to work more smoothly, it is possible
+# to annotate a commit with the string '(skip-generation)' and normal
+# build file generation (detected with changes to BUILD.gn files) is
+# disabled for that commit.  The script outputs instructions for handling
+# this situation.
+#
+# Note: the very first rebase operation will require some manual
+# intervention. The user will need to provide, at minimum, the commit that
+# corresponds to moz-central upon which the fast-forward stack is based.
+# It may also be necessary to provide the first commit of the
+# fast-forward stack.  Example:
+#   MOZ_BOTTOM_FF=30f0afb7e4c5 \
+#   MOZ_CURRENT_CENTRAL=cad1bd47c273 \
+#   bash dom/media/webrtc/third_party_build/elm_rebase.sh
+#
+# Assumes the top of the fast-forward stack to rebase is the current revision,
+# ".".
+
+function show_error_msg()
+{
+  echo "*** ERROR *** $? line $1 $0 did not complete successfully!"
+  echo "$ERROR_HELP"
+}
+ERROR_HELP=""
+
+# Print an Error message if `set -eE` causes the script to exit due to a failed command
+trap 'show_error_msg $LINENO' ERR
+
+source dom/media/webrtc/third_party_build/use_config_env.sh
+
+GENERATION_ERROR=$"
+Generating build files has failed.  The most common reason for this
+failure is that the current commit has an upcoming '(fix-xxxxxx)' commit
+that will then allow the build file generation to complete.  If the
+current situation seems to fit that pattern, adding a line with
+'(skip-generation)' to the commit message will ensure that future rebase
+operations do not attempt to generate build files for this commit.  It may
+be as simple as running the following commands:
+  HGPLAIN=1 hg log -T '{desc}' -r tip > $TMP_DIR/commit_message.txt
+  ed -s $TMP_DIR/commit_message.txt <<< $'3i\n(skip-generation)\n\n.\nw\nq'
+  hg commit --amend -l $TMP_DIR/commit_message.txt
+  bash $0
+"
+COMMIT_LIST_FILE=$TMP_DIR/rebase-commit-list.txt
+export HGPLAIN=1
+
+# After this point:
+# * eE: All commands should succeed.
+# * o pipefail: All stages of all pipes should succeed.
+set -eEo pipefail
+
+if [ -f $STATE_DIR/rebase_resume_state ]; then
+  source $STATE_DIR/rebase_resume_state
+else
+
+  if [ "x" == "x$MOZ_TOP_FF" ]; then
+    MOZ_TOP_FF=`hg log -r . -T"{node|short}"`
+
+    ERROR_HELP=$"
+The topmost commit to be rebased is not in the public phase. Should it be
+pushed to elm first? If this is intentional, please rerun the command and pass
+it in explicitly:
+  MOZ_TOP_FF=$MOZ_TOP_FF bash $0
+"
+    if [[ $(hg phase -r .) != *public ]]; then
+      echo "$ERROR_HELP"
+      exit 1
+    fi
+    ERROR_HELP=""
+
+    ERROR_HELP=$"
+The topmost commit to be rebased is public but has descendants. If those
+descendants should not be rebased, please rerun the command and pass the commit
+in explicitly:
+  MOZ_TOP_FF=$MOZ_TOP_FF bash $0
+"
+    if [ "x" != "x$(hg log -r 'descendants(.) and !.' -T'{node|short}')" ]; then
+      echo "$ERROR_HELP"
+      exit 1
+    fi
+    ERROR_HELP=""
+  fi
+
+  ERROR_HELP=$"
+An error here is likely because no revision for central is found.
+One possible reason for this is this is your first rebase operation.
+To 'bootstrap' the first rebase operation, please find the
+moz-central commit that the vendoring commits is based on, and
+rerun the command:
+  MOZ_CURRENT_CENTRAL={central-sha} bash $0
+
+You may also need to provide the bottom commit of the fast-forward
+stack.  The bottom commit means the commit following central.  This
+could be the sha of the .arcconfig commit if it is the bottom commit.
+That command looks like:
+  MOZ_BOTTOM_FF={base-sha} MOZ_CURRENT_CENTRAL={central-sha} bash $0
+"
+  if [ "x" == "x$MOZ_CURRENT_CENTRAL" ]; then
+    MOZ_CURRENT_CENTRAL=`hg log -r central -T"{node|short}"`
+  fi
+  if [ "x" == "x$MOZ_BOTTOM_FF" ]; then
+    MOZ_BOTTOM_FF=`hg log -r $MOZ_CURRENT_CENTRAL~-1 -T"{node|short}"`
+  fi
+  ERROR_HELP=""
+
+  if [ "x" == "x$MOZ_BOTTOM_FF" ]; then
+    echo "No value found for the bottom commit of the fast-forward commit stack."
+    exit 1
+  fi
+
+  # After this point:
+  # * eE: All commands should succeed.
+  # * u: All variables should be defined before use.
+  # * o pipefail: All stages of all pipes should succeed.
+  set -eEuo pipefail
+
+  hg pull central
+  MOZ_NEW_CENTRAL=`hg log -r central -T"{node|short}"`
+
+  echo "moz-central in elm is currently $MOZ_CURRENT_CENTRAL"
+  echo "bottom of fast-foward tree is $MOZ_BOTTOM_FF"
+  echo "top of fast-forward tree (webrtc-fast-forward) is $MOZ_TOP_FF"
+  echo "new target for elm rebase $MOZ_NEW_CENTRAL (tip of moz-central)"
+
+  hg log -T '{rev}:{node|short} {desc|firstline}\n' \
+      -r $MOZ_BOTTOM_FF::$MOZ_TOP_FF > $COMMIT_LIST_FILE
+
+  # move all FLOAT lines to end of file, and delete the "empty" tilde line
+  # line at the beginning
+  ed -s $COMMIT_LIST_FILE <<< $'g/- FLOAT -/m$\ng/^~$/d\nw\nq'
+
+  MOZ_BOOKMARK=`date "+webrtc-fast-forward-%Y-%m-%d--%H-%M"`
+  hg bookmark -r elm $MOZ_BOOKMARK
+
+  hg update $MOZ_NEW_CENTRAL
+
+  # pre-work is complete, let's write out a temporary config file that allows
+  # us to resume
+  echo $"export MOZ_CURRENT_CENTRAL=$MOZ_CURRENT_CENTRAL
+export MOZ_BOTTOM_FF=$MOZ_BOTTOM_FF
+export MOZ_TOP_FF=$MOZ_TOP_FF
+export MOZ_NEW_CENTRAL=$MOZ_NEW_CENTRAL
+export MOZ_BOOKMARK=$MOZ_BOOKMARK
+" > $STATE_DIR/rebase_resume_state
+fi # if [ -f $STATE_DIR/rebase_resume_state ]; then ; else
+
+# grab all commits
+COMMITS=`cat $COMMIT_LIST_FILE | awk '{print $1;}'`
+
+echo -n "Commits: "
+for commit in $COMMITS; do
+echo -n "$commit "
+done
+echo ""
+
+for commit in $COMMITS; do
+  echo "Processing $commit"
+  FULL_COMMIT_LINE=`head -1 $COMMIT_LIST_FILE`
+
+  function remove_commit () {
+    echo "Removing from list '$FULL_COMMIT_LINE'"
+    ed -s $COMMIT_LIST_FILE <<< $'1d\nw\nq'
+  }
+
+  IS_BUILD_COMMIT=`hg log -T '{desc|firstline}' -r $commit \
+                   | grep "file updates" | wc -l | tr -d " " || true`
+  echo "IS_BUILD_COMMIT: $IS_BUILD_COMMIT"
+  if [ "x$IS_BUILD_COMMIT" != "x0" ]; then
+    echo "Skipping $commit:"
+    hg log -T '{desc|firstline}' -r $commit
+    remove_commit
+    continue
+  fi
+
+  IS_SKIP_GEN_COMMIT=`hg log --verbose \
+                         -r $commit \
+                      | grep "skip-generation" | wc -l | tr -d " " || true`
+  echo "IS_SKIP_GEN_COMMIT: $IS_SKIP_GEN_COMMIT"
+
+  echo "Generate patch for: $commit"
+  hg export -r $commit > $TMP_DIR/rebase.patch
+
+  echo "Import patch for $commit"
+  hg import $TMP_DIR/rebase.patch || \
+  ( hg log -T '{desc}' -r $commit > $TMP_DIR/rebase_commit_message.txt ; \
+    remove_commit ; \
+    echo "Error importing: '$FULL_COMMIT_LINE'" ; \
+    echo "Please fix import errors, then:" ; \
+    echo "  hg commit -l $TMP_DIR/rebase_commit_message.txt" ; \
+    echo "  bash $0" ; \
+    exit 1 )
+
+  remove_commit
+
+  if [ "x$IS_SKIP_GEN_COMMIT" != "x0" ]; then
+    echo "Skipping build generation for $commit"
+    continue
+  fi
+
+  MODIFIED_BUILD_RELATED_FILE_CNT=`hg diff -c tip --stat \
+      --include 'third_party/libwebrtc/**BUILD.gn' \
+      --include 'third_party/libwebrtc/webrtc.gni' \
+      --include 'dom/media/webrtc/third_party_build/gn-configs/webrtc.json' \
+      | wc -l | tr -d " "`
+  echo "MODIFIED_BUILD_RELATED_FILE_CNT: $MODIFIED_BUILD_RELATED_FILE_CNT"
+  if [ "x$MODIFIED_BUILD_RELATED_FILE_CNT" != "x0" ]; then
+    echo "Regenerate build files"
+    ./mach python python/mozbuild/mozbuild/gn_processor.py \
+        dom/media/webrtc/third_party_build/gn-configs/webrtc.json || \
+    ( echo "$GENERATION_ERROR" ; exit 1 )
+
+    MOZ_BUILD_CHANGE_CNT=`hg status third_party/libwebrtc \
+        --include 'third_party/libwebrtc/**moz.build' | wc -l | tr -d " "`
+    if [ "x$MOZ_BUILD_CHANGE_CNT" != "x0" ]; then
+      bash dom/media/webrtc/third_party_build/commit-build-file-changes.sh
+      NEWEST_COMMIT=`hg log -T '{desc|firstline}' -r tip`
+      echo "NEWEST_COMMIT: $NEWEST_COMMIT"
+      echo "NEWEST_COMMIT: $NEWEST_COMMIT" >> $LOG_DIR/rebase-build-changes-commits.log
+    fi
+    echo "Done generating build files"
+  fi
+
+  echo "Done processing $commit"
+done
+
+rm $STATE_DIR/rebase_resume_state
+
+REMAINING_STEPS=$"
+The rebase process is complete.  The following steps must be completed manually:
+  ./mach bootstrap --application=browser --no-system-changes
+  ./mach build
+  hg push -r tip --force
+  hg push -B $MOZ_BOOKMARK
+"
+echo "$REMAINING_STEPS"
diff --git a/dom/media/webrtc/third_party_build/extract-for-git.py b/dom/media/webrtc/third_party_build/extract-for-git.py
new file mode 100644
index 0000000000..d2701d7dff
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/extract-for-git.py
@@ -0,0 +1,145 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+import argparse
+import os
+import re
+import subprocess
+
+# This script extracts commits that touch third party webrtc code so they can
+# be imported into Git. It filters out commits that are not part of upstream
+# code and rewrites the paths to match upstream. Finally, the commits are
+# combined into a mailbox file that can be applied with `git am`.
+LIBWEBRTC_DIR = "third_party/libwebrtc"
+
+
+def build_commit_list(revset, env):
+    """Build commit list from the specified revset.
+
+    The revset can be a single revision, e.g. 52bb9bb94661, or a range,
+    e.g. 8c08a5bb8a99::52bb9bb94661, or any other valid revset
+    (check hg help revset). Only commits that touch libwebrtc are included.
+    """
+    res = subprocess.run(
+        ["hg", "log", "-r", revset, "-M", "--template", "{node}\n", LIBWEBRTC_DIR],
+        capture_output=True,
+        text=True,
+        env=env,
+    )
+    return [line.strip() for line in res.stdout.strip().split("\n")]
+
+
+def extract_author_date(sha1, env):
+    res = subprocess.run(
+        ["hg", "log", "-r", sha1, "--template", "{author}|{date|isodate}"],
+        capture_output=True,
+        text=True,
+        env=env,
+    )
+    return res.stdout.split("|")
+
+
+def extract_description(sha1, env):
+    res = subprocess.run(
+        ["hg", "log", "-r", sha1, "--template", "{desc}"],
+        capture_output=True,
+        text=True,
+        env=env,
+    )
+    return res.stdout
+
+
+def extract_commit(sha1, env):
+    res = subprocess.run(
+        ["hg", "log", "-r", sha1, "-pg", "--template", "\n"],
+        capture_output=True,
+        text=True,
+        env=env,
+    )
+    return res.stdout
+
+
+def filter_nonwebrtc(commit):
+    filtered = []
+    skipping = False
+    for line in commit.split("\n"):
+        # Extract only patches affecting libwebrtc, but avoid commits that
+        # touch build, which is tracked by a separate repo, or that affect
+        # moz.build files which are code generated.
+        if (
+            line.startswith("diff --git a/" + LIBWEBRTC_DIR)
+            and not line.startswith("diff --git a/" + LIBWEBRTC_DIR + "/build")
+            and not line.startswith("diff --git a/" + LIBWEBRTC_DIR + "/third_party")
+            and not line.startswith(
+                "diff --git a/" + LIBWEBRTC_DIR + "/moz-patch-stack"
+            )
+            and not line.endswith("moz.build")
+        ):
+            skipping = False
+        elif line.startswith("diff --git"):
+            skipping = True
+
+        if not skipping:
+            filtered.append(line)
+    return "\n".join(filtered)
+
+
+def fixup_paths(commit):
+    # make sure we only rewrite paths in the diff-related or rename lines
+    commit = re.sub(
+        f"^rename (from|to) {LIBWEBRTC_DIR}/", "rename \\1 ", commit, flags=re.MULTILINE
+    )
+    return re.sub(f"( [ab])/{LIBWEBRTC_DIR}/", "\\1/", commit)
+
+
+def write_as_mbox(sha1, author, date, description, commit, ofile):
+    # Use same magic date as git format-patch
+    ofile.write("From {} Mon Sep 17 00:00:00 2001\n".format(sha1))
+    ofile.write("From: {}\n".format(author))
+    ofile.write("Date: {}\n".format(date))
+    description = description.split("\n")
+    ofile.write("Subject: {}\n".format(description[0]))
+    ofile.write("\n".join(description[1:]))
+    ofile.write(
+        "\nMercurial Revision: https://hg.mozilla.org/mozilla-central/rev/{}\n".format(
+            sha1
+        )
+    )
+    ofile.write(commit)
+    ofile.write("\n")
+    ofile.write("\n")
+
+
+if __name__ == "__main__":
+    commits = []
+    parser = argparse.ArgumentParser(
+        description="Format commits for upstream libwebrtc"
+    )
+    parser.add_argument(
+        "revsets", metavar="revset", type=str, nargs="+", help="A revset to process"
+    )
+    parser.add_argument(
+        "--target", choices=("libwebrtc", "build", "third_party"), default="libwebrtc"
+    )
+    args = parser.parse_args()
+
+    if args.target != "libwebrtc":
+        LIBWEBRTC_DIR = os.path.join(LIBWEBRTC_DIR, args.target)
+
+    # must run 'hg' with HGPLAIN=1 to ensure aliases don't interfere with
+    # command output.
+    env = os.environ.copy()
+    env["HGPLAIN"] = "1"
+
+    for revset in args.revsets:
+        commits.extend(build_commit_list(revset, env))
+
+    with open("mailbox.patch", "w") as ofile:
+        for sha1 in commits:
+            author, date = extract_author_date(sha1, env)
+            description = extract_description(sha1, env)
+            filtered_commit = filter_nonwebrtc(extract_commit(sha1, env))
+            if len(filtered_commit) == 0:
+                continue
+            fixedup_commit = fixup_paths(filtered_commit)
+            write_as_mbox(sha1, author, date, description, fixedup_commit, ofile)
diff --git a/dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh b/dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
new file mode 100644
index 0000000000..9ef40a41ac
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
@@ -0,0 +1,256 @@
+#!/bin/bash
+
+function show_error_msg()
+{
+  echo "*** ERROR *** $? line $1 $0 did not complete successfully!"
+  echo "$ERROR_HELP"
+}
+ERROR_HELP=""
+
+# Print an Error message if `set -eE` causes the script to exit due to a failed command
+trap 'show_error_msg $LINENO' ERR
+
+source dom/media/webrtc/third_party_build/use_config_env.sh
+
+# If DEBUG_GEN is set all commands should be printed as they are executed
+if [ ! "x$DEBUG_GEN" = "x" ]; then
+  set -x
+fi
+
+if [ "x$MOZ_LIBWEBRTC_SRC" = "x" ]; then
+  echo "MOZ_LIBWEBRTC_SRC is not defined, see README.md"
+  exit
+fi
+
+if [ -d $MOZ_LIBWEBRTC_SRC ]; then
+  echo "MOZ_LIBWEBRTC_SRC is $MOZ_LIBWEBRTC_SRC"
+else
+  echo "Path $MOZ_LIBWEBRTC_SRC is not found, see README.md"
+  exit
+fi
+
+if [ "x$MOZ_LIBWEBRTC_BRANCH" = "x" ]; then
+  echo "MOZ_LIBWEBRTC_BRANCH is not defined, see README.md"
+  exit
+fi
+
+if [ "x$HANDLE_NOOP_COMMIT" = "x" ]; then
+  HANDLE_NOOP_COMMIT=""
+fi
+
+RESUME=""
+if [ -f $STATE_DIR/resume_state ]; then
+  RESUME=`tail -1 $STATE_DIR/resume_state`
+fi
+
+GIT_IS_REBASING=`cd $MOZ_LIBWEBRTC_SRC && git status | grep "interactive rebase in progress" | wc -l | tr -d " " || true`
+if [ "x$GIT_IS_REBASING" != "x0" ]; then
+  echo "There is currently a git rebase operation in progress at $MOZ_LIBWEBRTC_SRC."
+  echo "Please resolve the rebase before attempting to continue the fast-forward"
+  echo "operation."
+  exit 1
+fi
+
+if [ "x$RESUME" = "x" ]; then
+  SKIP_TO="run"
+  # Check for modified files and abort if present.
+  MODIFIED_FILES=`hg status --exclude "third_party/libwebrtc/**.orig" third_party/libwebrtc`
+  if [ "x$MODIFIED_FILES" = "x" ]; then
+    # Completely clean the mercurial checkout before proceeding
+    hg update -C -r .
+    hg purge
+  else
+    echo "There are modified files in the checkout. Cowardly aborting!"
+    echo "$MODIFIED_FILES"
+    exit 1
+  fi
+else
+  SKIP_TO=$RESUME
+  hg revert -C third_party/libwebrtc/README.moz-ff-commit &> /dev/null
+fi
+
+find_base_commit
+find_next_commit
+
+echo "looking for $STATE_DIR/$MOZ_LIBWEBRTC_NEXT_BASE.no-op-cherry-pick-msg"
+if [ -f $STATE_DIR/$MOZ_LIBWEBRTC_NEXT_BASE.no-op-cherry-pick-msg ]; then
+  echo "***"
+  echo "*** detected special commit msg, setting HANDLE_NOOP_COMMIT"
+  echo "***"
+  HANDLE_NOOP_COMMIT="1"
+fi
+
+UPSTREAM_ADDED_FILES=""
+
+# Grab the filtered changes from git based on what we vendor.
+FILTERED_GIT_CHANGES=`./mach python $SCRIPT_DIR/filter_git_changes.py \
+   --repo-path $MOZ_LIBWEBRTC_SRC --commit-sha $MOZ_LIBWEBRTC_NEXT_BASE`
+
+# After this point:
+# * eE: All commands should succeed.
+# * u: All variables should be defined before use.
+# * o pipefail: All stages of all pipes should succeed.
+set -eEuo pipefail
+
+echo "     MOZ_LIBWEBRTC_BASE: $MOZ_LIBWEBRTC_BASE"
+echo "MOZ_LIBWEBRTC_NEXT_BASE: $MOZ_LIBWEBRTC_NEXT_BASE"
+echo "HANDLE_NOOP_COMMIT: $HANDLE_NOOP_COMMIT"
+echo " RESUME: $RESUME"
+echo "SKIP_TO: $SKIP_TO"
+
+echo "-------"
+echo "------- Write cmd-line to third_party/libwebrtc/README.moz-ff-commit"
+echo "-------"
+echo "# MOZ_LIBWEBRTC_SRC=$MOZ_LIBWEBRTC_SRC MOZ_LIBWEBRTC_BRANCH=$MOZ_LIBWEBRTC_BRANCH bash $0" \
+    >> third_party/libwebrtc/README.moz-ff-commit
+
+echo "-------"
+echo "------- Write new-base to last line of third_party/libwebrtc/README.moz-ff-commit"
+echo "-------"
+echo "# base of lastest vendoring" >> third_party/libwebrtc/README.moz-ff-commit
+echo "$MOZ_LIBWEBRTC_NEXT_BASE" >> third_party/libwebrtc/README.moz-ff-commit
+
+REBASE_HELP=$"
+The rebase operation onto $MOZ_LIBWEBRTC_NEXT_BASE has failed.  Please
+resolve all the rebase conflicts.  To fix this issue, you will need to
+jump to the github repo at $MOZ_LIBWEBRTC_SRC .
+When the github rebase is complete, re-run the script to resume the
+fast-forward process.
+"
+#"rebase_mozlibwebrtc_stack help for rebase failure"
+function rebase_mozlibwebrtc_stack {
+  echo "-------"
+  echo "------- Rebase $MOZ_LIBWEBRTC_BRANCH to $MOZ_LIBWEBRTC_NEXT_BASE"
+  echo "-------"
+  ERROR_HELP=$REBASE_HELP
+  ( cd $MOZ_LIBWEBRTC_SRC && \
+    git checkout -q $MOZ_LIBWEBRTC_BRANCH && \
+    git rebase $MOZ_LIBWEBRTC_NEXT_BASE \
+    &> $LOG_DIR/log-rebase-moz-libwebrtc.txt \
+  )
+  ERROR_HELP=""
+}
+
+function vendor_off_next_commit {
+  echo "-------"
+  echo "------- Vendor $MOZ_LIBWEBRTC_BRANCH from $MOZ_LIBWEBRTC_SRC"
+  echo "-------"
+  ./mach python $SCRIPT_DIR/vendor-libwebrtc.py \
+            --from-local $MOZ_LIBWEBRTC_SRC \
+            --commit $MOZ_LIBWEBRTC_BRANCH \
+            libwebrtc
+}
+
+# The vendoring script (called above in vendor_off_next_commit) replaces
+# the entire third_party/libwebrtc directory, which effectively removes
+# all the generated moz.build files.  It is easier (less error prone),
+# to revert only those missing moz.build files rather than attempt to
+# rebuild them because the rebuild may need json updates to work properly.
+function regen_mozbuild_files {
+  echo "-------"
+  echo "------- Restore moz.build files from repo"
+  echo "-------"
+  hg revert --include "third_party/libwebrtc/**moz.build" \
+    third_party/libwebrtc &> $LOG_DIR/log-regen-mozbuild-files.txt
+}
+
+function add_new_upstream_files {
+  if [ "x$HANDLE_NOOP_COMMIT" == "x1" ]; then
+    return
+  fi
+  UPSTREAM_ADDED_FILES=`echo "$FILTERED_GIT_CHANGES" | grep "^A" \
+      | awk '{print $2;}' || true`
+  if [ "x$UPSTREAM_ADDED_FILES" != "x" ]; then
+    echo "-------"
+    echo "------- Add new upstream files"
+    echo "-------"
+    (cd third_party/libwebrtc && hg add $UPSTREAM_ADDED_FILES)
+    echo "$UPSTREAM_ADDED_FILES" &> $LOG_DIR/log-new-upstream-files.txt
+  fi
+}
+
+function remove_deleted_upstream_files {
+  if [ "x$HANDLE_NOOP_COMMIT" == "x1" ]; then
+    return
+  fi
+  UPSTREAM_DELETED_FILES=`echo "$FILTERED_GIT_CHANGES" | grep "^D" \
+      | awk '{print $2;}' || true`
+  if [ "x$UPSTREAM_DELETED_FILES" != "x" ]; then
+    echo "-------"
+    echo "------- Remove deleted upstream files"
+    echo "-------"
+    (cd third_party/libwebrtc && hg rm $UPSTREAM_DELETED_FILES)
+    echo "$UPSTREAM_DELETED_FILES" &> $LOG_DIR/log-deleted-upstream-files.txt
+  fi
+}
+
+function handle_renamed_upstream_files {
+  if [ "x$HANDLE_NOOP_COMMIT" == "x1" ]; then
+    return
+  fi
+  UPSTREAM_RENAMED_FILES=`echo "$FILTERED_GIT_CHANGES" | grep "^R" \
+      | awk '{print $2 " " $3;}' || true`
+  if [ "x$UPSTREAM_RENAMED_FILES" != "x" ]; then
+    echo "-------"
+    echo "------- Handle renamed upstream files"
+    echo "-------"
+    (cd third_party/libwebrtc && echo "$UPSTREAM_RENAMED_FILES" | while read line; do hg rename --after $line; done)
+    echo "$UPSTREAM_RENAMED_FILES" &> $LOG_DIR/log-renamed-upstream-files.txt
+  fi
+}
+
+if [ $SKIP_TO = "run" ]; then
+  echo "resume2" > $STATE_DIR/resume_state
+  rebase_mozlibwebrtc_stack;
+fi
+
+if [ $SKIP_TO = "resume2" ]; then SKIP_TO="run"; fi
+if [ $SKIP_TO = "run" ]; then
+  echo "resume3" > $STATE_DIR/resume_state
+  vendor_off_next_commit;
+fi
+
+if [ $SKIP_TO = "resume3" ]; then SKIP_TO="run"; fi
+if [ $SKIP_TO = "run" ]; then
+  echo "resume4" > $STATE_DIR/resume_state
+  regen_mozbuild_files;
+fi
+
+if [ $SKIP_TO = "resume4" ]; then SKIP_TO="run"; fi
+if [ $SKIP_TO = "run" ]; then
+  echo "resume5" > $STATE_DIR/resume_state
+  remove_deleted_upstream_files;
+fi
+
+if [ $SKIP_TO = "resume5" ]; then SKIP_TO="run"; fi
+if [ $SKIP_TO = "run" ]; then
+  echo "resume6" > $STATE_DIR/resume_state
+  add_new_upstream_files;
+fi
+
+if [ $SKIP_TO = "resume6" ]; then SKIP_TO="run"; fi
+if [ $SKIP_TO = "run" ]; then
+  echo "resume7" > $STATE_DIR/resume_state
+  handle_renamed_upstream_files;
+fi
+
+echo "" > $STATE_DIR/resume_state
+echo "-------"
+echo "------- Commit vendored changes from $MOZ_LIBWEBRTC_NEXT_BASE"
+echo "-------"
+UPSTREAM_SHA=`cd $MOZ_LIBWEBRTC_SRC && \
+    git show --name-only $MOZ_LIBWEBRTC_NEXT_BASE \
+    | grep "^commit " | awk '{ print $NF }'`
+echo "Bug $MOZ_FASTFORWARD_BUG - Vendor libwebrtc from $MOZ_LIBWEBRTC_NEXT_BASE" \
+    > $TMP_DIR/commit_msg.txt
+echo "" >> $TMP_DIR/commit_msg.txt
+if [ -f $STATE_DIR/$MOZ_LIBWEBRTC_NEXT_BASE.no-op-cherry-pick-msg ]; then
+  cat $STATE_DIR/$MOZ_LIBWEBRTC_NEXT_BASE.no-op-cherry-pick-msg >> $TMP_DIR/commit_msg.txt
+  echo "" >> $TMP_DIR/commit_msg.txt
+  rm $STATE_DIR/$MOZ_LIBWEBRTC_NEXT_BASE.no-op-cherry-pick-msg
+fi
+echo "Upstream commit: https://webrtc.googlesource.com/src/+/$UPSTREAM_SHA" >> $TMP_DIR/commit_msg.txt
+(cd $MOZ_LIBWEBRTC_SRC && \
+git show --name-only $MOZ_LIBWEBRTC_NEXT_BASE | grep "^ ") >> $TMP_DIR/commit_msg.txt
+
+hg commit -l $TMP_DIR/commit_msg.txt third_party/libwebrtc
diff --git a/dom/media/webrtc/third_party_build/fetch_github_repo.py b/dom/media/webrtc/third_party_build/fetch_github_repo.py
new file mode 100644
index 0000000000..bfa40f5c7b
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/fetch_github_repo.py
@@ -0,0 +1,122 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+import argparse
+import os
+import re
+import shutil
+
+from run_operations import run_git, run_shell
+
+# This script fetches the moz-libwebrtc github repro with the expected
+# upstream remote and branch-heads setup.  This is used by both the
+# prep_repo.sh script as well as the restore_patch_stack.py script.
+#
+# For speed and conservation of network resources, after fetching all
+# the data, a tar of the repo is made and used if available.
+
+
+def fetch_repo(github_path, force_fetch, tar_path):
+    capture_output = False
+
+    # check for pre-existing repo - make sure we force the removal
+    if force_fetch and os.path.exists(github_path):
+        print("Removing existing repo: {}".format(github_path))
+        shutil.rmtree(github_path)
+
+    # clone https://github.com/mozilla/libwebrtc
+    if not os.path.exists(github_path):
+        # check for pre-existing tar, use it if we have it
+        if os.path.exists(tar_path):
+            print("Using tar file to reconstitute repo")
+            cmd = "cd {} ; tar --extract --gunzip --file={}".format(
+                os.path.dirname(github_path), os.path.basename(tar_path)
+            )
+            run_shell(cmd, capture_output)
+        else:
+            print("Cloning github repo")
+            run_shell(
+                "git clone https://github.com/mozilla/libwebrtc {}".format(github_path),
+                capture_output,
+            )
+
+    # setup upstream (https://webrtc.googlesource.com/src)
+    stdout_lines = run_git("git config --local --list", github_path)
+    stdout_lines = [
+        path for path in stdout_lines if re.findall("^remote.upstream.url.*", path)
+    ]
+    if len(stdout_lines) == 0:
+        print("Fetching upstream")
+        run_git("git checkout master", github_path)
+        run_git(
+            "git remote add upstream https://webrtc.googlesource.com/src", github_path
+        )
+        run_git("git fetch upstream", github_path)
+        run_git("git merge upstream/master", github_path)
+    else:
+        print(
+            "Upstream remote (https://webrtc.googlesource.com/src) already configured"
+        )
+
+    # setup upstream branch-heads
+    stdout_lines = run_git(
+        "git config --local --get-all remote.upstream.fetch", github_path
+    )
+    if len(stdout_lines) == 1:
+        print("Fetching upstream branch-heads")
+        run_git(
+            "git config --local --add remote.upstream.fetch +refs/branch-heads/*:refs/remotes/branch-heads/*",
+            github_path,
+        )
+        run_git("git fetch upstream", github_path)
+    else:
+        print("Upstream remote branch-heads already configured")
+
+    # do a sanity fetch in case this was not a freshly cloned copy of the
+    # repo, meaning it may not have all the mozilla branches present.
+    run_git("git fetch --all", github_path)
+
+    # create tar to avoid time refetching
+    if not os.path.exists(tar_path):
+        print("Creating tar file for quicker restore")
+        cmd = "cd {} ; tar --create --gzip --file={} {}".format(
+            os.path.dirname(github_path),
+            os.path.basename(tar_path),
+            os.path.basename(github_path),
+        )
+        run_shell(cmd, capture_output)
+
+
+if __name__ == "__main__":
+    default_state_dir = ".moz-fast-forward"
+    default_tar_name = "moz-libwebrtc.tar.gz"
+
+    parser = argparse.ArgumentParser(
+        description="Restore moz-libwebrtc github patch stack"
+    )
+    parser.add_argument(
+        "--repo-path",
+        required=True,
+        help="path to libwebrtc repo",
+    )
+    parser.add_argument(
+        "--force-fetch",
+        action="store_true",
+        default=False,
+        help="force rebuild an existing repo directory",
+    )
+    parser.add_argument(
+        "--tar-name",
+        default=default_tar_name,
+        help="name of tar file (defaults to {})".format(default_tar_name),
+    )
+    parser.add_argument(
+        "--state-path",
+        default=default_state_dir,
+        help="path to state directory (defaults to {})".format(default_state_dir),
+    )
+    args = parser.parse_args()
+
+    fetch_repo(
+        args.repo_path, args.force_fetch, os.path.join(args.state_path, args.tar_name)
+    )
diff --git a/dom/media/webrtc/third_party_build/filter_git_changes.py b/dom/media/webrtc/third_party_build/filter_git_changes.py
new file mode 100644
index 0000000000..f8964e145b
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/filter_git_changes.py
@@ -0,0 +1,72 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+import argparse
+import importlib
+import re
+import subprocess
+import sys
+
+sys.path.insert(0, "./dom/media/webrtc/third_party_build")
+vendor_libwebrtc = importlib.import_module("vendor-libwebrtc")
+
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser(
+        description="Get relevant change count from an upstream git commit"
+    )
+    parser.add_argument(
+        "--repo-path",
+        required=True,
+        help="path to libwebrtc repo",
+    )
+    parser.add_argument("--commit-sha", required=True, help="sha of commit to examine")
+    parser.add_argument("--diff-filter", choices=("A", "D", "R"))
+    args = parser.parse_args()
+
+    command = [
+        "git",
+        "show",
+        "--oneline",
+        "--name-status",
+        "--pretty=format:",
+        None if not args.diff_filter else "--diff-filter={}".format(args.diff_filter),
+        args.commit_sha,
+    ]
+    # strip possible empty elements from command list
+    command = [x for x in command if x is not None]
+
+    # Get the list of changes in the upstream commit.
+    res = subprocess.run(
+        command,
+        capture_output=True,
+        text=True,
+        cwd=args.repo_path,
+    )
+    if res.returncode != 0:
+        sys.exit("error: {}".format(res.stderr.strip()))
+
+    changed_files = [line.strip() for line in res.stdout.strip().split("\n")]
+    changed_files = [line for line in changed_files if line != ""]
+
+    # Fetch the list of excludes and includes used in the vendoring script.
+    exclude_list = vendor_libwebrtc.get_excluded_paths()
+    include_list = vendor_libwebrtc.get_included_path_overrides()
+
+    # First, search for changes in files that are specifically included.
+    # Do this first, because some of these files might be filtered out
+    # by the exclude list.
+    regex_includes = "|".join(["^.\t{}".format(i) for i in include_list])
+    included_files = [
+        path for path in changed_files if re.findall(regex_includes, path)
+    ]
+
+    # Convert the exclude list to a regex string.
+    regex_excludes = "|".join(["^.\t{}".format(i) for i in exclude_list])
+
+    # Filter out the excluded files/paths.
+    files_not_excluded = [
+        path for path in changed_files if not re.findall(regex_excludes, path)
+    ]
+
+    for path in included_files + files_not_excluded:
+        print(path)
diff --git a/dom/media/webrtc/third_party_build/gn-configs/README.md b/dom/media/webrtc/third_party_build/gn-configs/README.md
new file mode 100644
index 0000000000..111d2d022e
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/gn-configs/README.md
@@ -0,0 +1,16 @@
+# Generate new gn json files and moz.build files for building libwebrtc in our tree
+
+/!\ This is only supported on Linux and macOS. If you are on Windows, you can run
+the script under [WSL](https://docs.microsoft.com/en-us/windows/wsl/install).
+
+1. The script should be run from the top directory of our firefox tree.
+
+   ```
+   ./mach python python/mozbuild/mozbuild/gn_processor.py dom/media/webrtc/third_party_build/gn-configs/webrtc.json
+   ```
+
+2. Checkin all the generated/modified files and try your build!
+
+# Adding new configurations to the build
+
+Edit the `main` function in the `python/mozbuild/mozbuild/gn_processor.py` file.
diff --git a/dom/media/webrtc/third_party_build/gn-configs/webrtc.json b/dom/media/webrtc/third_party_build/gn-configs/webrtc.json
new file mode 100644
index 0000000000..9744558025
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/gn-configs/webrtc.json
@@ -0,0 +1,83 @@
+{
+  "target_dir": "third_party/libwebrtc",
+  "gn_target": "//:webrtc",
+  "gn_sandbox_variables": {
+    "COMPILE_FLAGS": {
+      "WARNINGS_AS_ERRORS": []
+    },
+    "FINAL_LIBRARY": "webrtc"
+  },
+  "mozilla_flags": ["-fobjc-arc", "-mavx2", "-mfma", "-mfpu=neon", "-msse2"],
+  "write_mozbuild_variables": {
+    "INCLUDE_TK_CFLAGS_DIRS": [
+      "third_party/libwebrtc/modules/desktop_capture/desktop_capture_gn",
+      "third_party/libwebrtc/modules/portal/portal_gn"
+    ]
+  },
+  "non_unified_sources": [
+    "third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc",
+    "third_party/libwebrtc/api/video/i210_buffer.cc",
+    "third_party/libwebrtc/api/video/i410_buffer.cc",
+    "third_party/libwebrtc/api/video/i422_buffer.cc",
+    "third_party/libwebrtc/api/video/i444_buffer.cc",
+    "third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc",
+    "third_party/libwebrtc/common_audio/vad/vad_core.c",
+    "third_party/libwebrtc/common_audio/vad/webrtc_vad.c",
+    "third_party/libwebrtc/common_audio/signal_processing/resample_by_2_mips.c",
+    "third_party/libwebrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c",
+    "third_party/libwebrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c",
+    "third_party/libwebrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c",
+    "third_party/libwebrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c",
+    "third_party/libwebrtc/modules/audio_coding/neteq/audio_vector.cc",
+    "third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer.cc",
+    "third_party/libwebrtc/modules/audio_device/android/audio_manager.cc",
+    "third_party/libwebrtc/modules/audio_device/android/audio_record_jni.cc",
+    "third_party/libwebrtc/modules/audio_device/android/audio_track_jni.cc",
+    "third_party/libwebrtc/modules/audio_device/android/opensles_player.cc",
+    "third_party/libwebrtc/modules/audio_device/linux/audio_device_pulse_linux.cc",
+    "third_party/libwebrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc",
+    "third_party/libwebrtc/modules/audio_device/win/audio_device_core_win.cc",
+    "third_party/libwebrtc/modules/audio_processing/aecm/aecm_core.cc",
+    "third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_c.cc",
+    "third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_mips.cc",
+    "third_party/libwebrtc/modules/audio_processing/aecm/aecm_core_neon.cc",
+    "third_party/libwebrtc/modules/audio_processing/aecm/echo_control_mobile.cc",
+    "third_party/libwebrtc/modules/audio_processing/echo_control_mobile_impl.cc",
+    "third_party/libwebrtc/modules/audio_processing/echo_detector/normalized_covariance_estimator.cc",
+    "third_party/libwebrtc/modules/audio_processing/gain_control_impl.cc",
+    "third_party/libwebrtc/modules/audio_processing/rms_level.cc",
+    "third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.cc",
+    "third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc",
+    "third_party/libwebrtc/modules/desktop_capture/fallback_desktop_capturer_wrapper.cc",
+    "third_party/libwebrtc/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc",
+    "third_party/libwebrtc/modules/desktop_capture/linux/wayland/moz_base_capturer_pipewire.cc",
+    "third_party/libwebrtc/modules/desktop_capture/mouse_cursor_monitor_linux.cc",
+    "third_party/libwebrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc",
+    "third_party/libwebrtc/modules/pacing/prioritized_packet_queue.cc",
+    "third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc",
+    "third_party/libwebrtc/modules/rtp_rtcp/source/flexfec_receiver.cc",
+    "third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc",
+    "third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc",
+    "third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_sender.cc",
+    "third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc",
+    "third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc",
+    "third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_egress.cc",
+    "third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_video.cc",
+    "third_party/libwebrtc/modules/rtp_rtcp/source/rtp_util.cc",
+    "third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_generator.cc",
+    "third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.cc",
+    "third_party/libwebrtc/modules/third_party/g722/g722_encode.c",
+    "third_party/libwebrtc/modules/video_capture/windows/device_info_ds.cc",
+    "third_party/libwebrtc/modules/video_capture/windows/help_functions_ds.cc",
+    "third_party/libwebrtc/modules/video_capture/windows/sink_filter_ds.cc",
+    "third_party/libwebrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc",
+    "third_party/libwebrtc/modules/video_coding/svc/scalability_structure_key_svc.cc",
+    "third_party/libwebrtc/modules/video_coding/svc/scalability_structure_simulcast.cc",
+    "third_party/libwebrtc/rtc_base/win/hstring.cc",
+    "third_party/libwebrtc/third_party/abseil-cpp/absl/strings/numbers.cc",
+    "third_party/libwebrtc/third_party/abseil-cpp/absl/synchronization/blocking_counter.cc",
+    "third_party/libwebrtc/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_posix.cc",
+    "third_party/libwebrtc/third_party/abseil-cpp/absl/time/time.cc",
+    "third_party/libwebrtc/video/rtp_video_stream_receiver2.cc"
+  ]
+}
diff --git a/dom/media/webrtc/third_party_build/lookup_branch_head.py b/dom/media/webrtc/third_party_build/lookup_branch_head.py
new file mode 100644
index 0000000000..afd0e6c791
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/lookup_branch_head.py
@@ -0,0 +1,98 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+import argparse
+import json
+import os
+import pathlib
+import sys
+import urllib.request
+
+# default cache file location in STATE_DIR location
+default_cache_path = ".moz-fast-forward/milestone.cache"
+
+
+def fetch_branch_head_dict():
+    milestone_url = (
+        "https://chromiumdash.appspot.com/fetch_milestones?only_branched=true"
+    )
+    uf = urllib.request.urlopen(milestone_url)
+    html = uf.read()
+    milestone_dict = json.loads(html)
+
+    # There is more information in the json dictionary, but we only care
+    # about the milestone (version) to branch "name" (webrtc_branch)
+    # info.  For example:
+    #  v106 -> 5249 (which translates to branch-heads/5249)
+    #  v107 -> 5304 (which translates to branch-heads/5304)
+    #
+    # As returned from web query, milestones are integers and branch
+    # "names" are strings.
+    new_dict = {}
+    for row in milestone_dict:
+        new_dict[row["milestone"]] = row["webrtc_branch"]
+
+    return new_dict
+
+
+def read_dict_from_cache(cache_path):
+    if cache_path is not None and os.path.exists(cache_path):
+        with open(cache_path, "r") as ifile:
+            return json.loads(ifile.read(), object_hook=jsonKeys2int)
+    return {}
+
+
+def write_dict_to_cache(cache_path, milestones):
+    with open(cache_path, "w") as ofile:
+        ofile.write(json.dumps(milestones))
+
+
+def get_branch_head(milestone, cache_path=default_cache_path):
+    milestones = read_dict_from_cache(cache_path)
+
+    # if the cache didn't exist or is stale, try to fetch using a web query
+    if milestone not in milestones:
+        try:
+            milestones = fetch_branch_head_dict()
+            write_dict_to_cache(cache_path, milestones)
+        except Exception:
+            pass
+
+    if milestone in milestones:
+        return milestones[milestone]
+    return None
+
+
+# From https://stackoverflow.com/questions/1450957/pythons-json-module-converts-int-dictionary-keys-to-strings
+def jsonKeys2int(x):
+    if isinstance(x, dict):
+        return {int(k): v for k, v in x.items()}
+    return x
+
+
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser(
+        description="Get libwebrtc branch-head for given chromium milestone"
+    )
+    parser.add_argument(
+        "milestone", type=int, help="integer chromium milestone (example: 106)"
+    )
+    parser.add_argument("-v", "--verbose", action="store_true")
+    parser.add_argument("-c", "--cache", type=pathlib.Path, help="path to cache file")
+    args = parser.parse_args()
+
+    # if the user provided a cache path use it, otherwise use the default
+    local_cache_path = args.cache or default_cache_path
+
+    branch_head = get_branch_head(args.milestone, local_cache_path)
+    if branch_head is None:
+        sys.exit("error: chromium milestone '{}' is not found.".format(args.milestone))
+
+    if args.verbose:
+        print(
+            "chromium milestone {} uses branch-heads/{}".format(
+                args.milestone, branch_head
+            )
+        )
+    else:
+        print(branch_head)
diff --git a/dom/media/webrtc/third_party_build/loop-ff.sh b/dom/media/webrtc/third_party_build/loop-ff.sh
new file mode 100644
index 0000000000..72a8d88091
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/loop-ff.sh
@@ -0,0 +1,236 @@
+#!/bin/bash
+
+source dom/media/webrtc/third_party_build/use_config_env.sh
+
+# file for logging loop script output
+LOOP_OUTPUT_LOG=$LOG_DIR/log-loop-ff.txt
+
+function echo_log()
+{
+  echo "===loop-ff=== $@" 2>&1| tee -a $LOOP_OUTPUT_LOG
+}
+
+function show_error_msg()
+{
+  echo_log "*** ERROR *** $? line $1 $0 did not complete successfully!"
+  echo_log "$ERROR_HELP"
+}
+ERROR_HELP=""
+
+# Print an Error message if `set -eE` causes the script to exit due to a failed command
+trap 'show_error_msg $LINENO' ERR
+
+# If DEBUG_LOOP_FF is set all commands should be printed as they are executed
+if [ ! "x$DEBUG_LOOP_FF" = "x" ]; then
+  set -x
+fi
+
+if [ "x$MOZ_LIBWEBRTC_SRC" = "x" ]; then
+  echo "MOZ_LIBWEBRTC_SRC is not defined, see README.md"
+  exit
+fi
+
+if [ ! -d $MOZ_LIBWEBRTC_SRC ]; then
+  echo "Path $MOZ_LIBWEBRTC_SRC is not found, see README.md"
+  exit
+fi
+
+if [ "x$MOZ_LIBWEBRTC_BRANCH" = "x" ]; then
+  echo "MOZ_LIBWEBRTC_BRANCH is not defined, see README.md"
+  exit
+fi
+
+if [ "x$MOZ_STOP_AFTER_COMMIT" = "x" ]; then
+  MOZ_STOP_AFTER_COMMIT=`cd $MOZ_LIBWEBRTC_SRC ; git show $MOZ_TARGET_UPSTREAM_BRANCH_HEAD --format='%h' --name-only | head -1`
+  echo "No MOZ_STOP_AFTER_COMMIT variable defined - stopping at $MOZ_TARGET_UPSTREAM_BRANCH_HEAD"
+fi
+
+if [ "x$MOZ_ADVANCE_ONE_COMMIT" = "x" ]; then
+  MOZ_ADVANCE_ONE_COMMIT=""
+fi
+
+if [ "x$SKIP_NEXT_REVERT_CHK" = "x" ]; then
+  SKIP_NEXT_REVERT_CHK="0"
+fi
+
+MOZ_CHANGED=0
+GIT_CHANGED=0
+HANDLE_NOOP_COMMIT=""
+
+# After this point:
+# * eE: All commands should succeed.
+# * u: All variables should be defined before use.
+# * o pipefail: All stages of all pipes should succeed.
+set -eEuo pipefail
+
+# start a new log with every run of this script
+rm -f $LOOP_OUTPUT_LOG
+# make sure third_party/libwebrtc/README.moz-ff-commit is the committed version
+# so we properly determine MOZ_LIBWEBRTC_BASE and MOZ_LIBWEBRTC_NEXT_BASE
+# in the loop below
+hg revert -C third_party/libwebrtc/README.moz-ff-commit &> /dev/null
+
+# check for a resume situation from fast-forward-libwebrtc.sh
+RESUME=""
+if [ -f $STATE_DIR/resume_state ]; then
+  RESUME=`tail -1 $STATE_DIR/resume_state`
+fi
+
+ERROR_HELP=$"
+It appears that initial vendoring verification has failed.
+- If you have never run the fast-forward process before, you may need to
+  prepare the github repository by running prep_repo.sh.
+- If you have previously run loop-ff.sh successfully, there may be a new
+  change to third_party/libwebrtc that should be extracted and added to
+  the patch stack in github.  It may be as easy as running:
+      ./mach python $SCRIPT_DIR/extract-for-git.py tip::tip
+      mv mailbox.patch $MOZ_LIBWEBRTC_SRC
+      (cd $MOZ_LIBWEBRTC_SRC && \\
+       git am mailbox.patch)
+"
+# if we're not in the resume situation from fast-forward-libwebrtc.sh
+if [ "x$RESUME" = "x" ]; then
+  # start off by verifying the vendoring process to make sure no changes have
+  # been added to elm to fix bugs.
+  echo_log "Verifying vendoring..."
+  bash $SCRIPT_DIR/verify_vendoring.sh
+  echo_log "Done verifying vendoring."
+fi
+ERROR_HELP=""
+
+for (( ; ; )); do
+
+find_base_commit
+find_next_commit
+
+if [ $MOZ_LIBWEBRTC_BASE == $MOZ_LIBWEBRTC_NEXT_BASE ]; then
+  echo_log "Processing complete, already at upstream $MOZ_LIBWEBRTC_BASE"
+  exit
+fi
+
+echo_log "==================="
+
+COMMITS_REMAINING=`cd $MOZ_LIBWEBRTC_SRC ; \
+   git log --oneline $MOZ_LIBWEBRTC_BASE..$MOZ_TARGET_UPSTREAM_BRANCH_HEAD \
+   | wc -l | tr -d " "`
+echo_log "Commits remaining: $COMMITS_REMAINING"
+
+ERROR_HELP=$"Some portion of the detection and/or fixing of upstream revert commits
+has failed.  Please fix the state of the git hub repo at: $MOZ_LIBWEBRTC_SRC.
+When fixed, please resume this script with the following command:
+    SKIP_NEXT_REVERT_CHK=1 bash $SCRIPT_DIR/loop-ff.sh
+"
+if [ "x$SKIP_NEXT_REVERT_CHK" == "x0" ]; then
+  echo_log "Check for upcoming revert commit"
+  AUTO_FIX_REVERT_AS_NOOP=1 bash $SCRIPT_DIR/detect_upstream_revert.sh \
+      2>&1| tee -a $LOOP_OUTPUT_LOG
+fi
+SKIP_NEXT_REVERT_CHK="0"
+ERROR_HELP=""
+
+echo_log "Looking for $STATE_DIR/$MOZ_LIBWEBRTC_NEXT_BASE.no-op-cherry-pick-msg"
+if [ -f $STATE_DIR/$MOZ_LIBWEBRTC_NEXT_BASE.no-op-cherry-pick-msg ]; then
+  echo_log "Detected special commit msg, setting HANDLE_NOOP_COMMIT=1"
+  HANDLE_NOOP_COMMIT="1"
+fi
+
+echo_log "Moving from moz-libwebrtc commit $MOZ_LIBWEBRTC_BASE to $MOZ_LIBWEBRTC_NEXT_BASE"
+bash $SCRIPT_DIR/fast-forward-libwebrtc.sh 2>&1| tee -a $LOOP_OUTPUT_LOG
+
+MOZ_CHANGED=`hg diff -c tip --stat \
+   | egrep -ve "README.moz-ff-commit|README.mozilla|files changed," \
+   | wc -l | tr -d " " || true`
+GIT_CHANGED=`./mach python $SCRIPT_DIR/filter_git_changes.py \
+   --repo-path $MOZ_LIBWEBRTC_SRC --commit-sha $MOZ_LIBWEBRTC_NEXT_BASE \
+   | wc -l | tr -d " "`
+FILE_CNT_MISMATCH_MSG=$"
+The number of files changed in the upstream commit ($GIT_CHANGED) does
+not match the number of files changed in the local Mozilla repo
+commit ($MOZ_CHANGED).  This may indicate a mismatch between the vendoring
+script and this script, or it could be a true error in the import
+processing.  Once the issue has been resolved, the following steps
+remain for this commit:
+  # generate moz.build files (may not be necessary)
+  ./mach python python/mozbuild/mozbuild/gn_processor.py \\
+      $SCRIPT_DIR/gn-configs/webrtc.json
+  # commit the updated moz.build files with the appropriate commit msg
+  bash $SCRIPT_DIR/commit-build-file-changes.sh
+  # do a (hopefully) quick test build
+  ./mach build
+"
+echo_log "Verify number of files changed MOZ($MOZ_CHANGED) GIT($GIT_CHANGED)"
+if [ "x$HANDLE_NOOP_COMMIT" == "x1" ]; then
+  echo_log "NO-OP commit detected, we expect file changed counts to differ"
+elif [ $MOZ_CHANGED -ne $GIT_CHANGED ]; then
+  echo_log "MOZ_CHANGED $MOZ_CHANGED should equal GIT_CHANGED $GIT_CHANGED"
+  echo "$FILE_CNT_MISMATCH_MSG"
+  exit 1
+fi
+HANDLE_NOOP_COMMIT=""
+
+# save the current patch stack in case we need to reconstitute it later
+./mach python $SCRIPT_DIR/save_patch_stack.py \
+    --repo-path $MOZ_LIBWEBRTC_SRC \
+    --branch $MOZ_LIBWEBRTC_BRANCH \
+    --patch-path "third_party/libwebrtc/moz-patch-stack" \
+    --state-path $STATE_DIR \
+    --target-branch-head $MOZ_TARGET_UPSTREAM_BRANCH_HEAD \
+    2>&1| tee -a $LOOP_OUTPUT_LOG
+
+MODIFIED_BUILD_RELATED_FILE_CNT=`hg diff -c tip --stat \
+    --include 'third_party/libwebrtc/**BUILD.gn' \
+    --include 'third_party/libwebrtc/webrtc.gni' \
+    | grep -v "files changed" \
+    | wc -l | tr -d " " || true`
+ERROR_HELP=$"
+Generating build files has failed.  This likely means changes to one or more
+BUILD.gn files are required.  Commit those changes following the instructions
+in https://wiki.mozilla.org/Media/WebRTC/libwebrtc_Update_Process#Operational_notes
+Then complete these steps:
+  # generate moz.build files (may not be necessary)
+  ./mach python python/mozbuild/mozbuild/gn_processor.py \\
+      $SCRIPT_DIR/gn-configs/webrtc.json
+  # commit the updated moz.build files with the appropriate commit msg
+  bash $SCRIPT_DIR/commit-build-file-changes.sh
+  # do a (hopefully) quick test build
+  ./mach build
+After a successful build, you may resume this script.
+"
+echo_log "Modified BUILD.gn (or webrtc.gni) files: $MODIFIED_BUILD_RELATED_FILE_CNT"
+if [ "x$MODIFIED_BUILD_RELATED_FILE_CNT" != "x0" ]; then
+  echo_log "Regenerate build files"
+  ./mach python python/mozbuild/mozbuild/gn_processor.py \
+      $SCRIPT_DIR/gn-configs/webrtc.json 2>&1| tee -a $LOOP_OUTPUT_LOG
+
+  MOZ_BUILD_CHANGE_CNT=`hg status third_party/libwebrtc \
+      --include 'third_party/libwebrtc/**moz.build' | wc -l | tr -d " "`
+  if [ "x$MOZ_BUILD_CHANGE_CNT" != "x0" ]; then
+    echo_log "Detected modified moz.build files, commiting"
+  fi
+
+  bash $SCRIPT_DIR/commit-build-file-changes.sh 2>&1| tee -a $LOOP_OUTPUT_LOG
+fi
+ERROR_HELP=""
+
+ERROR_HELP=$"
+The test build has failed.  Most likely this is due to an upstream api change that
+must be reflected in Mozilla code outside of the third_party/libwebrtc directory.
+"
+echo_log "Test build"
+./mach build 2>&1| tee -a $LOOP_OUTPUT_LOG
+ERROR_HELP=""
+
+if [ ! "x$MOZ_STOP_AFTER_COMMIT" = "x" ]; then
+if [ $MOZ_LIBWEBRTC_NEXT_BASE = $MOZ_STOP_AFTER_COMMIT ]; then
+  break
+fi
+fi
+
+if [ ! "x$MOZ_ADVANCE_ONE_COMMIT" = "x" ]; then
+  echo_log "Done advancing one commit."
+  exit
+fi
+
+done
+
+echo_log "Completed fast-foward to $MOZ_STOP_AFTER_COMMIT"
diff --git a/dom/media/webrtc/third_party_build/make_upstream_revert_noop.sh b/dom/media/webrtc/third_party_build/make_upstream_revert_noop.sh
new file mode 100755
index 0000000000..9f5fdd7b82
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/make_upstream_revert_noop.sh
@@ -0,0 +1,101 @@
+#!/bin/bash
+
+# This script takes the current base sha, the next base sha, and the sha
+# of the commit that reverts the next base as determined by
+# detect_upstream_revert.sh and "inserts" two commits at the bottom of the
+# moz_libwebrtc GitHub patch stack.  The two commits are exact copies of
+# the upcoming commit and its corresponding revert commit, with commit
+# messages indicating they are temporary commits.  Additionally, 2 files
+# are written that act as markers for the fast-forward script and contain
+# supplemental commit message text.
+#
+# When the fast-forward script runs, it will rebase onto the next base
+# sha.  Since we have a corresponding, identical temp commit at the bottom
+# of our patch stack, the temp commit will be absorbed as unnecessary.
+# Since the patch stack now has the temp revert commit at the bottom, this
+# results in a “no-op” commit.  The marker file indicates that specially
+# handling should occur in the fast-forward-libwebrtc.sh and loop-ff.sh
+# scripts.  This special handling includes adding the supplemental commit
+# text that explains why the commit is a no-op (or empty) commit and
+# skipping the verification of the number of files changed between our
+# mercurial repository and the GitHub repository.
+
+function show_error_msg()
+{
+  echo "*** ERROR *** $? line $1 $0 did not complete successfully!"
+  echo "$ERROR_HELP"
+}
+ERROR_HELP=""
+
+# Print an Error message if `set -eE` causes the script to exit due to a failed command
+trap 'show_error_msg $LINENO' ERR
+
+source dom/media/webrtc/third_party_build/use_config_env.sh
+
+# If DEBUG_GEN is set all commands should be printed as they are executed
+if [ ! "x$DEBUG_GEN" = "x" ]; then
+  set -x
+fi
+
+if [ "x$MOZ_LIBWEBRTC_SRC" = "x" ]; then
+  echo "MOZ_LIBWEBRTC_SRC is not defined, see README.md"
+  exit
+fi
+
+if [ -d $MOZ_LIBWEBRTC_SRC ]; then
+  echo "MOZ_LIBWEBRTC_SRC is $MOZ_LIBWEBRTC_SRC"
+else
+  echo "Path $MOZ_LIBWEBRTC_SRC is not found, see README.md"
+  exit
+fi
+
+if [ "x$MOZ_LIBWEBRTC_BRANCH" = "x" ]; then
+  echo "MOZ_LIBWEBRTC_BRANCH is not defined, see README.md"
+  exit
+fi
+
+# After this point:
+# * eE: All commands should succeed.
+# * u: All variables should be defined before use.
+# * o pipefail: All stages of all pipes should succeed.
+set -eEuo pipefail
+
+find_base_commit
+find_next_commit
+echo "MOZ_LIBWEBRTC_BASE: $MOZ_LIBWEBRTC_BASE"
+echo "MOZ_LIBWEBRTC_NEXT_BASE: $MOZ_LIBWEBRTC_NEXT_BASE"
+echo "MOZ_LIBWEBRTC_REVERT_SHA: $MOZ_LIBWEBRTC_REVERT_SHA"
+
+# These files serve dual purposes:
+# 1) They serve as marker/indicator files to loop-ff.sh to
+#    know to process the commit differently, accounting for
+#    the no-op nature of the commit and it's corresponding
+#    revert commit.
+# 2) The contain supplemental commit message text to explain
+#    why the commits are essentially empty.
+# They are written first on the off chance that the rebase
+# operation below fails and requires manual intervention,
+# thus avoiding the operator of these scripts to remember to
+# generate these two files.
+echo $"Essentially a no-op since we're going to see this change
+reverted when we vendor in $MOZ_LIBWEBRTC_REVERT_SHA." \
+> $STATE_DIR/$MOZ_LIBWEBRTC_NEXT_BASE.no-op-cherry-pick-msg
+
+echo "We already cherry-picked this when we vendored $MOZ_LIBWEBRTC_NEXT_BASE." \
+> $STATE_DIR/$MOZ_LIBWEBRTC_REVERT_SHA.no-op-cherry-pick-msg
+
+cd $MOZ_LIBWEBRTC_SRC
+rm -f $TMP_DIR/*.patch $TMP_DIR/*.patch.bak
+git checkout -b moz-cherry-pick $MOZ_LIBWEBRTC_BASE
+git format-patch -o $TMP_DIR -k --start-number 1 \
+    $MOZ_LIBWEBRTC_NEXT_BASE^..$MOZ_LIBWEBRTC_NEXT_BASE
+git format-patch -o $TMP_DIR -k --start-number 2 \
+    $MOZ_LIBWEBRTC_REVERT_SHA^..$MOZ_LIBWEBRTC_REVERT_SHA
+sed -i.bak -e "/^Subject: / s/$/ ($MOZ_LIBWEBRTC_NEXT_BASE)/" $TMP_DIR/0001*.patch
+sed -i.bak -e "/^Subject: / s/$/ ($MOZ_LIBWEBRTC_REVERT_SHA)/" $TMP_DIR/0002*.patch
+sed -i.bak -e 's/^Subject: /Subject: (tmp-cherry-pick) /' $TMP_DIR/*.patch
+git am $TMP_DIR/*.patch
+git checkout $MOZ_LIBWEBRTC_BRANCH
+git rebase moz-cherry-pick
+git branch -d moz-cherry-pick
+
diff --git a/dom/media/webrtc/third_party_build/pre-warmed-milestone.cache b/dom/media/webrtc/third_party_build/pre-warmed-milestone.cache
new file mode 100644
index 0000000000..826268db47
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/pre-warmed-milestone.cache
@@ -0,0 +1 @@
+{"109": "5414", "108": "5359", "107": "5304", "106": "5249", "105": "5195"}
diff --git a/dom/media/webrtc/third_party_build/prep_repo.sh b/dom/media/webrtc/third_party_build/prep_repo.sh
new file mode 100644
index 0000000000..9f22f63253
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/prep_repo.sh
@@ -0,0 +1,93 @@
+#!/bin/bash
+
+function show_error_msg()
+{
+  echo "*** ERROR *** $? line $1 $0 did not complete successfully!"
+  echo "$ERROR_HELP"
+}
+ERROR_HELP=""
+
+# Print an Error message if `set -eE` causes the script to exit due to a failed command
+trap 'show_error_msg $LINENO' ERR
+
+source dom/media/webrtc/third_party_build/use_config_env.sh
+
+echo "MOZ_LIBWEBRTC_SRC: $MOZ_LIBWEBRTC_SRC"
+echo "MOZ_LIBWEBRTC_BRANCH: $MOZ_LIBWEBRTC_BRANCH"
+echo "MOZ_FASTFORWARD_BUG: $MOZ_FASTFORWARD_BUG"
+
+# After this point:
+# * eE: All commands should succeed.
+# * u: All variables should be defined before use.
+# * o pipefail: All stages of all pipes should succeed.
+set -eEuo pipefail
+
+# wipe resume_state for new run
+rm -f $STATE_DIR/resume_state
+
+# If there is no cache file for the branch-head lookups done in
+# update_default_config.sh, go ahead and copy our small pre-warmed
+# version.
+if [ ! -f $STATE_DIR/milestone.cache ]; then
+  cp $SCRIPT_DIR/pre-warmed-milestone.cache $STATE_DIR/milestone.cache
+fi
+
+# fetch the github repro
+./mach python $SCRIPT_DIR/fetch_github_repo.py \
+    --repo-path $MOZ_LIBWEBRTC_SRC \
+    --state-path $STATE_DIR
+
+CURRENT_DIR=`pwd`
+cd $MOZ_LIBWEBRTC_SRC
+
+# clear any possible previous patches
+rm -f *.patch
+
+# create a new work branch and "export" a new patch stack to rebase
+# find the common commit between our upstream release branch and trunk
+CHERRY_PICK_BASE=`git merge-base branch-heads/$MOZ_PRIOR_UPSTREAM_BRANCH_HEAD_NUM master`
+echo "common commit: $CHERRY_PICK_BASE"
+
+# create a new branch at the common commit and checkout the new branch
+ERROR_HELP=$"
+Unable to create branch '$MOZ_LIBWEBRTC_BRANCH'.  This probably means
+that prep_repo.sh is being called on a repo that already has a patch
+stack in progress.  If you're sure you want to do this, the following
+commands will allow the process to continue:
+  ( cd $MOZ_LIBWEBRTC_SRC && \\
+    git checkout $MOZ_LIBWEBRTC_BRANCH && \\
+    git checkout -b $MOZ_LIBWEBRTC_BRANCH-old && \\
+    git branch -D $MOZ_LIBWEBRTC_BRANCH ) && \\
+  bash $0
+"
+git branch $MOZ_LIBWEBRTC_BRANCH $CHERRY_PICK_BASE
+ERROR_HELP=""
+git checkout $MOZ_LIBWEBRTC_BRANCH
+
+# make sure we're starting with a clean tmp directory
+rm -f $TMP_DIR/*.patch $TMP_DIR/*.patch.bak
+
+# grab the patches for all the commits in chrome's release branch for libwebrtc
+git format-patch -o $TMP_DIR -k $CHERRY_PICK_BASE..branch-heads/$MOZ_PRIOR_UPSTREAM_BRANCH_HEAD_NUM
+# tweak the release branch commit summaries to show they were cherry picked
+sed -i.bak -e "/^Subject: / s/^Subject: /Subject: (cherry-pick-branch-heads\/$MOZ_PRIOR_UPSTREAM_BRANCH_HEAD_NUM) /" $TMP_DIR/*.patch
+git am $TMP_DIR/*.patch # applies to branch mozpatches
+rm $TMP_DIR/*.patch $TMP_DIR/*.patch.bak
+
+# we don't use restore_patch_stack.py here because it would overwrite the patches
+# from the previous release branch we just added in the above step.
+
+# grab all the moz patches and apply
+git am $CURRENT_DIR/third_party/libwebrtc/moz-patch-stack/*.patch
+
+cd $CURRENT_DIR
+
+# cp all the no-op files to STATE_DIR
+NO_OP_FILE_COUNT=`ls third_party/libwebrtc/moz-patch-stack \
+    | grep "no-op-cherry-pick-msg" | wc -l | tr -d " " || true`
+if [ "x$NO_OP_FILE_COUNT" != "x0" ]; then
+  cp $CURRENT_DIR/third_party/libwebrtc/moz-patch-stack/*.no-op-cherry-pick-msg \
+     $STATE_DIR
+fi
+
+bash $SCRIPT_DIR/verify_vendoring.sh || true
diff --git a/dom/media/webrtc/third_party_build/push_official_branch.sh b/dom/media/webrtc/third_party_build/push_official_branch.sh
new file mode 100644
index 0000000000..e7ed17df2b
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/push_official_branch.sh
@@ -0,0 +1,50 @@
+#!/bin/bash
+
+# This script is a simple helper script that creates a branch name
+# in moz-libwebrtc and pushes it to origin.  You will need permissions
+# to the mozilla fork of libwebrtc in order to complete this operation.
+# The repo is: https://github.com/mozilla/libwebrtc/
+#
+# Note: this should only be run after elm has been merged to mozilla-central
+
+function show_error_msg()
+{
+  echo "*** ERROR *** $? line $1 $0 did not complete successfully!"
+  echo "$ERROR_HELP"
+}
+ERROR_HELP=""
+
+# Print an Error message if `set -eE` causes the script to exit due to a failed command
+trap 'show_error_msg $LINENO' ERR
+
+source dom/media/webrtc/third_party_build/use_config_env.sh
+
+# After this point:
+# * eE: All commands should succeed.
+# * o pipefail: All stages of all pipes should succeed.
+set -eEo pipefail
+
+cd $MOZ_LIBWEBRTC_SRC
+
+git fetch
+
+if git show-ref --quiet refs/remotes/origin/$MOZ_LIBWEBRTC_OFFICIAL_BRANCH; then
+  echo "Branch '$MOZ_LIBWEBRTC_OFFICIAL_BRANCH' already exists remotely. Nothing to do."
+
+  echo "Please ensure this branch information can be found on Bug $MOZ_FASTFORWARD_BUG"
+  echo "https://github.com/mozilla/libwebrtc/tree/$MOZ_LIBWEBRTC_OFFICIAL_BRANCH"
+  exit 0
+fi
+
+if git show-ref --quiet refs/heads/$MOZ_LIBWEBRTC_OFFICIAL_BRANCH; then
+  echo "Branch '$MOZ_LIBWEBRTC_OFFICIAL_BRANCH' already exists locally. No need to create it."
+else
+  echo "Creating branch '$MOZ_LIBWEBRTC_OFFICIAL_BRANCH'"
+  git branch $MOZ_LIBWEBRTC_OFFICIAL_BRANCH
+fi
+
+echo "Pushing the branch to https://github.com/mozilla/libwebrtc"
+git push origin $MOZ_LIBWEBRTC_OFFICIAL_BRANCH
+
+echo "Please add this new branch information to Bug $MOZ_FASTFORWARD_BUG"
+echo "https://github.com/mozilla/libwebrtc/tree/$MOZ_LIBWEBRTC_OFFICIAL_BRANCH"
diff --git a/dom/media/webrtc/third_party_build/restore_elm_arcconfig.py b/dom/media/webrtc/third_party_build/restore_elm_arcconfig.py
new file mode 100644
index 0000000000..00c0bd7309
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/restore_elm_arcconfig.py
@@ -0,0 +1,27 @@
+#!/bin/env python
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+from subprocess import run
+
+# This script sets the Arcanist configuration for the elm repo. This script
+# should be run after each repository reset.
+#
+# Usage: from the root of the repo `./mach python dom/media/webrtc/third_party_build/restore_elm_arcconfig.py`
+#
+
+ret = run(
+    [
+        "hg",
+        "import",
+        "-m",
+        "Bug 1729988 - FLOAT - REPO-elm - update .arcconfig repo callsign r=bgrins",
+        "dom/media/webrtc/third_party_build/elm_arcconfig.patch",
+    ]
+).returncode
+if ret != 0:
+    raise Exception(f"Failed to add FLOATing arcconfig patch for ELM: { ret }")
+else:
+    print("ELM .arcconfig restored. Please push this change to ELM")
diff --git a/dom/media/webrtc/third_party_build/restore_patch_stack.py b/dom/media/webrtc/third_party_build/restore_patch_stack.py
new file mode 100644
index 0000000000..6db3a64192
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/restore_patch_stack.py
@@ -0,0 +1,107 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+import argparse
+import os
+import re
+import shutil
+
+from fetch_github_repo import fetch_repo
+from run_operations import run_git, run_shell
+
+# This script restores the mozilla patch stack and no-op commit tracking
+# files.  In the case of repo corruption or a mistake made during
+# various rebase conflict resolution operations, the patch-stack can be
+# restored rather than requiring the user to restart the fast-forward
+# process from the beginning.
+
+
+def get_last_line(file_path):
+    # technique from https://stackoverflow.com/questions/46258499/how-to-read-the-last-line-of-a-file-in-python
+    with open(file_path, "rb") as f:
+        try:  # catch OSError in case of a one line file
+            f.seek(-2, os.SEEK_END)
+            while f.read(1) != b"\n":
+                f.seek(-2, os.SEEK_CUR)
+        except OSError:
+            f.seek(0)
+        return f.readline().decode().strip()
+
+
+def restore_patch_stack(
+    github_path, github_branch, patch_directory, state_directory, tar_name
+):
+    # first, refetch the repo (hopefully utilizing the tarfile for speed) so
+    # the patches apply cleanly
+    fetch_repo(github_path, True, os.path.join(state_directory, tar_name))
+
+    # remove any stale no-op-cherry-pick-msg files in state_directory
+    run_shell("rm {}/*.no-op-cherry-pick-msg || true".format(state_directory))
+
+    # lookup latest vendored commit from third_party/libwebrtc/README.moz-ff-commit
+    file = os.path.abspath("third_party/libwebrtc/README.moz-ff-commit")
+    last_vendored_commit = get_last_line(file)
+
+    # checkout the previous vendored commit with proper branch name
+    cmd = "git checkout -b {} {}".format(github_branch, last_vendored_commit)
+    run_git(cmd, github_path)
+
+    # restore the patches to moz-libwebrtc repo, use run_shell instead of
+    # run_hg to allow filepath wildcard
+    print("Restoring patch stack")
+    run_shell("cd {} && git am {}/*.patch".format(github_path, patch_directory))
+
+    # it is also helpful to restore the no-op-cherry-pick-msg files to
+    # the state directory so that if we're restoring a patch-stack we
+    # also restore the possibly consumed no-op tracking files.
+    no_op_files = [
+        path
+        for path in os.listdir(patch_directory)
+        if re.findall(".*no-op-cherry-pick-msg$", path)
+    ]
+    for file in no_op_files:
+        shutil.copy(os.path.join(patch_directory, file), state_directory)
+
+
+if __name__ == "__main__":
+    default_patch_dir = "third_party/libwebrtc/moz-patch-stack"
+    default_state_dir = ".moz-fast-forward"
+    default_tar_name = "moz-libwebrtc.tar.gz"
+
+    parser = argparse.ArgumentParser(
+        description="Restore moz-libwebrtc github patch stack"
+    )
+    parser.add_argument(
+        "--repo-path",
+        required=True,
+        help="path to libwebrtc repo",
+    )
+    parser.add_argument(
+        "--branch",
+        default="mozpatches",
+        help="moz-libwebrtc branch (defaults to mozpatches)",
+    )
+    parser.add_argument(
+        "--patch-path",
+        default=default_patch_dir,
+        help="path to save patches (defaults to {})".format(default_patch_dir),
+    )
+    parser.add_argument(
+        "--tar-name",
+        default=default_tar_name,
+        help="name of tar file (defaults to {})".format(default_tar_name),
+    )
+    parser.add_argument(
+        "--state-path",
+        default=default_state_dir,
+        help="path to state directory (defaults to {})".format(default_state_dir),
+    )
+    args = parser.parse_args()
+
+    restore_patch_stack(
+        args.repo_path,
+        args.branch,
+        os.path.abspath(args.patch_path),
+        args.state_path,
+        args.tar_name,
+    )
diff --git a/dom/media/webrtc/third_party_build/run_operations.py b/dom/media/webrtc/third_party_build/run_operations.py
new file mode 100644
index 0000000000..794ea268d8
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/run_operations.py
@@ -0,0 +1,78 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+import os
+import subprocess
+import sys
+
+# This is a collection of helper functions that use the subprocess.run
+# command to execute commands.  In the future, if we have cases where
+# we find common python functionality in utilizing the data returned
+# from these functions, that functionality can live here for easy reuse
+# between other python scripts in dom/media/webrtc/third_party_build.
+
+
+# must run 'hg' with HGPLAIN=1 to ensure aliases don't interfere with
+# command output.
+env = os.environ.copy()
+env["HGPLAIN"] = "1"
+
+
+def run_hg(cmd):
+    cmd_list = cmd.split(" ")
+    res = subprocess.run(
+        cmd_list,
+        capture_output=True,
+        text=True,
+        env=env,
+    )
+    if res.returncode != 0:
+        print(
+            "Hit return code {} running '{}'. Aborting.".format(res.returncode, cmd),
+            file=sys.stderr,
+        )
+        print(res.stderr)
+        sys.exit(1)
+    stdout = res.stdout.strip()
+    return [] if len(stdout) == 0 else stdout.split("\n")
+
+
+def run_git(cmd, working_dir):
+    cmd_list = cmd.split(" ")
+    res = subprocess.run(
+        cmd_list,
+        capture_output=True,
+        text=True,
+        cwd=working_dir,
+    )
+    if res.returncode != 0:
+        print(
+            "Hit return code {} running '{}'. Aborting.".format(res.returncode, cmd),
+            file=sys.stderr,
+        )
+        print(res.stderr)
+        sys.exit(1)
+    stdout = res.stdout.strip()
+    return [] if len(stdout) == 0 else stdout.split("\n")
+
+
+def run_shell(cmd, capture_output=True):
+    res = subprocess.run(
+        cmd,
+        shell=True,
+        capture_output=capture_output,
+        text=True,
+    )
+    if res.returncode != 0:
+        print(
+            "Hit return code {} running '{}'. Aborting.".format(res.returncode, cmd),
+            file=sys.stderr,
+        )
+        print(res.stderr)
+        sys.exit(1)
+    output_lines = []
+    if capture_output:
+        stdout = res.stdout.strip()
+        output_lines = [] if len(stdout) == 0 else stdout.split("\n")
+
+    return output_lines
diff --git a/dom/media/webrtc/third_party_build/save_patch_stack.py b/dom/media/webrtc/third_party_build/save_patch_stack.py
new file mode 100644
index 0000000000..11c007109c
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/save_patch_stack.py
@@ -0,0 +1,142 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+import argparse
+import os
+import re
+import shutil
+
+from run_operations import run_git, run_hg, run_shell
+
+# This script saves the mozilla patch stack and no-op commit tracking
+# files.  This makes our fast-forward process much more resilient by
+# saving the intermediate state after each upstream commit is processed.
+
+
+def save_patch_stack(
+    github_path,
+    github_branch,
+    patch_directory,
+    state_directory,
+    target_branch_head,
+    bug_number,
+):
+    # remove the current patch files
+    files_to_remove = os.listdir(patch_directory)
+    for file in files_to_remove:
+        os.remove(os.path.join(patch_directory, file))
+
+    # find the base of the patch stack
+    cmd = "git merge-base {} {}".format(github_branch, target_branch_head)
+    stdout_lines = run_git(cmd, github_path)
+    merge_base = stdout_lines[0]
+
+    # grab patch stack
+    cmd = "git format-patch --keep-subject --output-directory {} {}..{}".format(
+        patch_directory, merge_base, github_branch
+    )
+    run_git(cmd, github_path)
+
+    # remove the commit summary from the file name
+    patches_to_rename = os.listdir(patch_directory)
+    for file in patches_to_rename:
+        shortened_name = re.sub("^(\d\d\d\d)-.*\.patch", "\\1.patch", file)
+        os.rename(
+            os.path.join(patch_directory, file),
+            os.path.join(patch_directory, shortened_name),
+        )
+
+    # remove the unhelpful first line of the patch files that only
+    # causes diff churn.  For reasons why we can't skip creating backup
+    # files during the in-place editing, see:
+    # https://stackoverflow.com/questions/5694228/sed-in-place-flag-that-works-both-on-mac-bsd-and-linux
+    run_shell("sed -i'.bak' -e '1d' {}/*.patch".format(patch_directory))
+    run_shell("rm {}/*.patch.bak".format(patch_directory))
+
+    # it is also helpful to save the no-op-cherry-pick-msg files from
+    # the state directory so that if we're restoring a patch-stack we
+    # also restore the possibly consumed no-op tracking files.
+    no_op_files = [
+        path
+        for path in os.listdir(state_directory)
+        if re.findall(".*no-op-cherry-pick-msg$", path)
+    ]
+    for file in no_op_files:
+        shutil.copy(os.path.join(state_directory, file), patch_directory)
+
+    # get missing files (that should be marked removed)
+    cmd = "hg status --no-status --deleted {}".format(patch_directory)
+    stdout_lines = run_hg(cmd)
+    if len(stdout_lines) != 0:
+        cmd = "hg rm {}".format(" ".join(stdout_lines))
+        run_hg(cmd)
+
+    # get unknown files (that should be marked added)
+    cmd = "hg status --no-status --unknown {}".format(patch_directory)
+    stdout_lines = run_hg(cmd)
+    if len(stdout_lines) != 0:
+        cmd = "hg add {}".format(" ".join(stdout_lines))
+        run_hg(cmd)
+
+    # if any files are marked for add/remove/modify, commit them
+    cmd = "hg status --added --removed --modified {}".format(patch_directory)
+    stdout_lines = run_hg(cmd)
+    if (len(stdout_lines)) != 0:
+        print("Updating {} files in {}".format(len(stdout_lines), patch_directory))
+        if bug_number is None:
+            run_hg("hg amend")
+        else:
+            run_shell(
+                "hg commit --message 'Bug {} - updated libwebrtc patch stack'".format(
+                    bug_number
+                )
+            )
+
+
+if __name__ == "__main__":
+    default_patch_dir = "third_party/libwebrtc/moz-patch-stack"
+    default_state_dir = ".moz-fast-forward"
+
+    parser = argparse.ArgumentParser(
+        description="Save moz-libwebrtc github patch stack"
+    )
+    parser.add_argument(
+        "--repo-path",
+        required=True,
+        help="path to libwebrtc repo",
+    )
+    parser.add_argument(
+        "--branch",
+        default="mozpatches",
+        help="moz-libwebrtc branch (defaults to mozpatches)",
+    )
+    parser.add_argument(
+        "--patch-path",
+        default=default_patch_dir,
+        help="path to save patches (defaults to {})".format(default_patch_dir),
+    )
+    parser.add_argument(
+        "--state-path",
+        default=default_state_dir,
+        help="path to state directory (defaults to {})".format(default_state_dir),
+    )
+    parser.add_argument(
+        "--target-branch-head",
+        required=True,
+        help="target branch head for fast-forward, should match MOZ_TARGET_UPSTREAM_BRANCH_HEAD in config_env",
+    )
+    parser.add_argument(
+        "--separate-commit-bug-number",
+        type=int,
+        help="integer Bugzilla number (example: 1800920), if provided will write patch stack as separate commit",
+    )
+    args = parser.parse_args()
+
+    save_patch_stack(
+        args.repo_path,
+        args.branch,
+        os.path.abspath(args.patch_path),
+        args.state_path,
+        args.target_branch_head,
+        args.separate_commit_bug_number,
+    )
diff --git a/dom/media/webrtc/third_party_build/update_default_config.sh b/dom/media/webrtc/third_party_build/update_default_config.sh
new file mode 100644
index 0000000000..f4020f9b45
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/update_default_config.sh
@@ -0,0 +1,46 @@
+#!/bin/bash
+
+function show_error_msg()
+{
+  echo "*** ERROR *** $? line $1 $0 did not complete successfully!"
+  echo "$ERROR_HELP"
+}
+ERROR_HELP=""
+
+# Print an Error message if `set -eE` causes the script to exit due to a failed command
+trap 'show_error_msg $LINENO' ERR
+
+if [ "x" = "x$NEW_BUG_NUMBER" ]; then
+  echo "NEW_BUG_NUMBER is not defined.  You should probably have a new bug"
+  echo "number defined for the next fast-forward update.  Then do:"
+  echo "  NEW_BUG_NUMBER={new-bug-number} bash $0"
+  exit
+fi
+
+source dom/media/webrtc/third_party_build/use_config_env.sh
+
+if [ "x$MOZ_NEXT_LIBWEBRTC_MILESTONE" = "x" ]; then
+  echo "MOZ_NEXT_LIBWEBRTC_MILESTONE is not defined, see README.md"
+  exit
+fi
+
+if [ "x$MOZ_NEXT_FIREFOX_REL_TARGET" = "x" ]; then
+  echo "MOZ_NEXT_FIREFOX_REL_TARGET is not defined, see README.md"
+  exit
+fi
+
+
+# After this point:
+# * eE: All commands should succeed.
+# * u: All variables should be defined before use.
+# * o pipefail: All stages of all pipes should succeed.
+set -eEuo pipefail
+
+ERROR_HELP=$"
+An error has occurred running $SCRIPT_DIR/write_default_config.py
+"
+./mach python $SCRIPT_DIR/write_default_config.py \
+  --bug-number $NEW_BUG_NUMBER \
+  --milestone $MOZ_NEXT_LIBWEBRTC_MILESTONE \
+  --release-target $MOZ_NEXT_FIREFOX_REL_TARGET \
+  > $SCRIPT_DIR/default_config_env
diff --git a/dom/media/webrtc/third_party_build/use_config_env.sh b/dom/media/webrtc/third_party_build/use_config_env.sh
new file mode 100644
index 0000000000..bd78dc7e0e
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/use_config_env.sh
@@ -0,0 +1,89 @@
+#!/bin/bash
+
+# Assume that if STATE_DIR is already defined, we do not need to
+# execute this file again.
+if [ "x$STATE_DIR" != "x" ]; then
+  # no need to run script since we've already run
+  return
+fi
+
+export SCRIPT_DIR="dom/media/webrtc/third_party_build"
+# first, make sure we're running from the top of moz-central repo
+if [ ! -d $SCRIPT_DIR ]; then
+  echo "Error: unable to find directory $SCRIPT_DIR"
+  exit 1
+fi
+
+# Should we tie the location of the STATE_DIR to the path
+# in MOZ_CONFIG_PATH?  Probably.
+export STATE_DIR=`pwd`/.moz-fast-forward
+export LOG_DIR=$STATE_DIR/logs
+export TMP_DIR=$STATE_DIR/tmp
+
+if [ ! -d $STATE_DIR ]; then
+  echo "Creating missing $STATE_DIR"
+  mkdir -p $STATE_DIR
+  if [ ! -d $STATE_DIR ]; then
+    echo "error: unable to find (or create) $STATE_DIR"
+    exit 1
+  fi
+fi
+echo "Using STATE_DIR=$STATE_DIR"
+
+if [ ! -d $LOG_DIR ]; then
+  echo "Creating missing $LOG_DIR"
+  mkdir -p $LOG_DIR
+  if [ ! -d $LOG_DIR ]; then
+    echo "error: unable to find (or create) $LOG_DIR"
+    exit 1
+  fi
+fi
+echo "Using LOG_DIR=$LOG_DIR"
+
+if [ ! -d $TMP_DIR ]; then
+  echo "Creating missing $TMP_DIR"
+  mkdir -p $TMP_DIR
+  if [ ! -d $TMP_DIR ]; then
+    echo "error: unable to find (or create) $TMP_DIR"
+    exit 1
+  fi
+fi
+echo "Using TMP_DIR=$TMP_DIR"
+
+# Allow user to override default path to config_env
+if [ "x$MOZ_CONFIG_PATH" = "x" ]; then
+  MOZ_CONFIG_PATH=$STATE_DIR/config_env
+  echo "Using default MOZ_CONFIG_PATH=$MOZ_CONFIG_PATH"
+fi
+
+if [ ! -f $MOZ_CONFIG_PATH ]; then
+  echo "Creating default config file at $MOZ_CONFIG_PATH"
+  cp $SCRIPT_DIR/default_config_env $MOZ_CONFIG_PATH
+fi
+source $MOZ_CONFIG_PATH
+
+
+function find_base_commit()
+{
+  # read the last line of README.moz-ff-commit to retrieve our current base
+  # commit in moz-libwebrtc
+  MOZ_LIBWEBRTC_BASE=`tail -1 third_party/libwebrtc/README.moz-ff-commit`
+  echo "prelim MOZ_LIBWEBRTC_BASE: $MOZ_LIBWEBRTC_BASE"
+  # if we've advanced into a chrome release branch, we need to adjust the
+  # MOZ_LIBWEBRTC_BASE to the last common commit so we can now advance up
+  # the trunk commits.
+  MOZ_LIBWEBRTC_BASE=`cd $MOZ_LIBWEBRTC_SRC ; git merge-base $MOZ_LIBWEBRTC_BASE $MOZ_TARGET_UPSTREAM_BRANCH_HEAD`
+  # now make it a short hash
+  MOZ_LIBWEBRTC_BASE=`cd $MOZ_LIBWEBRTC_SRC ; git rev-parse --short $MOZ_LIBWEBRTC_BASE`
+  echo "adjusted MOZ_LIBWEBRTC_BASE: $MOZ_LIBWEBRTC_BASE"
+}
+export -f find_base_commit
+
+function find_next_commit()
+{
+  # identify the next commit above our current base commit
+  MOZ_LIBWEBRTC_NEXT_BASE=`cd $MOZ_LIBWEBRTC_SRC ; \
+  git log --oneline --ancestry-path $MOZ_LIBWEBRTC_BASE^..$MOZ_TARGET_UPSTREAM_BRANCH_HEAD \
+   | tail -2 | head -1 | awk '{print $1;}'`
+}
+export -f find_next_commit
diff --git a/dom/media/webrtc/third_party_build/vendor-libwebrtc.py b/dom/media/webrtc/third_party_build/vendor-libwebrtc.py
new file mode 100644
index 0000000000..99723849c7
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/vendor-libwebrtc.py
@@ -0,0 +1,419 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+import argparse
+import datetime
+import os
+import shutil
+import stat
+import subprocess
+import sys
+import tarfile
+
+import requests
+
+THIRDPARTY_USED_IN_FIREFOX = [
+    "abseil-cpp",
+    "google_benchmark",
+    "pffft",
+    "rnnoise",
+]
+
+LIBWEBRTC_DIR = os.path.normpath("third_party/libwebrtc")
+
+
+def get_excluded_paths():
+    return [
+        ".clang-format",
+        ".git-blame-ignore-revs",
+        ".gitignore",
+        ".vpython",
+        "CODE_OF_CONDUCT.md",
+        "ENG_REVIEW_OWNERS",
+        "PRESUBMIT.py",
+        "README.chromium",
+        "WATCHLISTS",
+        "codereview.settings",
+        "license_template.txt",
+        "native-api.md",
+        "presubmit_test.py",
+        "presubmit_test_mocks.py",
+        "pylintrc",
+        # Only the camera code under sdk/android/api/org/webrtc is used, so
+        # we remove sdk/android and add back the specific files we want.
+        "sdk/android",
+    ]
+
+
+# Paths in this list are included even if their parent directory is
+# excluded in get_excluded_paths()
+def get_included_path_overrides():
+    return [
+        "sdk/android/src/java/org/webrtc/NativeLibrary.java",
+        "sdk/android/src/java/org/webrtc/FramerateBitrateAdjuster.java",
+        "sdk/android/src/java/org/webrtc/MediaCodecVideoDecoderFactory.java",
+        "sdk/android/src/java/org/webrtc/BitrateAdjuster.java",
+        "sdk/android/src/java/org/webrtc/MediaCodecWrapperFactory.java",
+        "sdk/android/src/java/org/webrtc/WebRtcClassLoader.java",
+        "sdk/android/src/java/org/webrtc/audio/WebRtcAudioRecord.java",
+        "sdk/android/src/java/org/webrtc/audio/WebRtcAudioTrack.java",
+        "sdk/android/src/java/org/webrtc/audio/WebRtcAudioManager.java",
+        "sdk/android/src/java/org/webrtc/audio/LowLatencyAudioBufferManager.java",
+        "sdk/android/src/java/org/webrtc/audio/WebRtcAudioUtils.java",
+        "sdk/android/src/java/org/webrtc/audio/WebRtcAudioEffects.java",
+        "sdk/android/src/java/org/webrtc/audio/VolumeLogger.java",
+        "sdk/android/src/java/org/webrtc/NativeCapturerObserver.java",
+        "sdk/android/src/java/org/webrtc/MediaCodecWrapper.java",
+        "sdk/android/src/java/org/webrtc/CalledByNative.java",
+        "sdk/android/src/java/org/webrtc/Histogram.java",
+        "sdk/android/src/java/org/webrtc/EglBase10Impl.java",
+        "sdk/android/src/java/org/webrtc/EglBase14Impl.java",
+        "sdk/android/src/java/org/webrtc/MediaCodecWrapperFactoryImpl.java",
+        "sdk/android/src/java/org/webrtc/AndroidVideoDecoder.java",
+        "sdk/android/src/java/org/webrtc/BaseBitrateAdjuster.java",
+        "sdk/android/src/java/org/webrtc/HardwareVideoEncoder.java",
+        "sdk/android/src/java/org/webrtc/VideoCodecMimeType.java",
+        "sdk/android/src/java/org/webrtc/NativeAndroidVideoTrackSource.java",
+        "sdk/android/src/java/org/webrtc/VideoDecoderWrapper.java",
+        "sdk/android/src/java/org/webrtc/JNILogging.java",
+        "sdk/android/src/java/org/webrtc/CameraCapturer.java",
+        "sdk/android/src/java/org/webrtc/CameraSession.java",
+        "sdk/android/src/java/org/webrtc/H264Utils.java",
+        "sdk/android/src/java/org/webrtc/Empty.java",
+        "sdk/android/src/java/org/webrtc/DynamicBitrateAdjuster.java",
+        "sdk/android/src/java/org/webrtc/Camera1Session.java",
+        "sdk/android/src/java/org/webrtc/JniCommon.java",
+        "sdk/android/src/java/org/webrtc/NV12Buffer.java",
+        "sdk/android/src/java/org/webrtc/WrappedNativeI420Buffer.java",
+        "sdk/android/src/java/org/webrtc/GlGenericDrawer.java",
+        "sdk/android/src/java/org/webrtc/RefCountDelegate.java",
+        "sdk/android/src/java/org/webrtc/Camera2Session.java",
+        "sdk/android/src/java/org/webrtc/MediaCodecUtils.java",
+        "sdk/android/src/java/org/webrtc/CalledByNativeUnchecked.java",
+        "sdk/android/src/java/org/webrtc/VideoEncoderWrapper.java",
+        "sdk/android/src/java/org/webrtc/NV21Buffer.java",
+        "sdk/android/api/org/webrtc/RendererCommon.java",
+        "sdk/android/api/org/webrtc/YuvHelper.java",
+        "sdk/android/api/org/webrtc/LibvpxVp9Encoder.java",
+        "sdk/android/api/org/webrtc/Metrics.java",
+        "sdk/android/api/org/webrtc/CryptoOptions.java",
+        "sdk/android/api/org/webrtc/MediaConstraints.java",
+        "sdk/android/api/org/webrtc/YuvConverter.java",
+        "sdk/android/api/org/webrtc/JavaI420Buffer.java",
+        "sdk/android/api/org/webrtc/VideoDecoder.java",
+        "sdk/android/api/org/webrtc/WrappedNativeVideoDecoder.java",
+        "sdk/android/api/org/webrtc/Camera2Enumerator.java",
+        "sdk/android/api/org/webrtc/SurfaceTextureHelper.java",
+        "sdk/android/api/org/webrtc/EglBase10.java",
+        "sdk/android/api/org/webrtc/DataChannel.java",
+        "sdk/android/api/org/webrtc/audio/JavaAudioDeviceModule.java",
+        "sdk/android/api/org/webrtc/audio/AudioDeviceModule.java",
+        "sdk/android/api/org/webrtc/audio/LegacyAudioDeviceModule.java",
+        "sdk/android/api/org/webrtc/SessionDescription.java",
+        "sdk/android/api/org/webrtc/GlUtil.java",
+        "sdk/android/api/org/webrtc/VideoSource.java",
+        "sdk/android/api/org/webrtc/AudioTrack.java",
+        "sdk/android/api/org/webrtc/EglRenderer.java",
+        "sdk/android/api/org/webrtc/VideoEncoder.java",
+        "sdk/android/api/org/webrtc/VideoCapturer.java",
+        "sdk/android/api/org/webrtc/SoftwareVideoDecoderFactory.java",
+        "sdk/android/api/org/webrtc/AudioSource.java",
+        "sdk/android/api/org/webrtc/GlRectDrawer.java",
+        "sdk/android/api/org/webrtc/StatsReport.java",
+        "sdk/android/api/org/webrtc/CameraVideoCapturer.java",
+        "sdk/android/api/org/webrtc/NetEqFactoryFactory.java",
+        "sdk/android/api/org/webrtc/AudioProcessingFactory.java",
+        "sdk/android/api/org/webrtc/Camera2Capturer.java",
+        "sdk/android/api/org/webrtc/ScreenCapturerAndroid.java",
+        "sdk/android/api/org/webrtc/RefCounted.java",
+        "sdk/android/api/org/webrtc/VideoEncoderFallback.java",
+        "sdk/android/api/org/webrtc/AudioEncoderFactoryFactory.java",
+        "sdk/android/api/org/webrtc/EglBase14.java",
+        "sdk/android/api/org/webrtc/SoftwareVideoEncoderFactory.java",
+        "sdk/android/api/org/webrtc/VideoEncoderFactory.java",
+        "sdk/android/api/org/webrtc/StatsObserver.java",
+        "sdk/android/api/org/webrtc/PlatformSoftwareVideoDecoderFactory.java",
+        "sdk/android/api/org/webrtc/Camera1Capturer.java",
+        "sdk/android/api/org/webrtc/AddIceObserver.java",
+        "sdk/android/api/org/webrtc/SurfaceViewRenderer.java",
+        "sdk/android/api/org/webrtc/CameraEnumerator.java",
+        "sdk/android/api/org/webrtc/CameraEnumerationAndroid.java",
+        "sdk/android/api/org/webrtc/VideoDecoderFallback.java",
+        "sdk/android/api/org/webrtc/FileVideoCapturer.java",
+        "sdk/android/api/org/webrtc/NativeLibraryLoader.java",
+        "sdk/android/api/org/webrtc/Camera1Enumerator.java",
+        "sdk/android/api/org/webrtc/NativePeerConnectionFactory.java",
+        "sdk/android/api/org/webrtc/LibaomAv1Encoder.java",
+        "sdk/android/api/org/webrtc/BuiltinAudioEncoderFactoryFactory.java",
+        "sdk/android/api/org/webrtc/AudioDecoderFactoryFactory.java",
+        "sdk/android/api/org/webrtc/FecControllerFactoryFactoryInterface.java",
+        "sdk/android/api/org/webrtc/VideoFrameBufferType.java",
+        "sdk/android/api/org/webrtc/SdpObserver.java",
+        "sdk/android/api/org/webrtc/Predicate.java",
+        "sdk/android/api/org/webrtc/VideoFileRenderer.java",
+        "sdk/android/api/org/webrtc/WrappedNativeVideoEncoder.java",
+        "sdk/android/api/org/webrtc/LibvpxVp8Encoder.java",
+        "sdk/android/api/org/webrtc/DtmfSender.java",
+        "sdk/android/api/org/webrtc/VideoTrack.java",
+        "sdk/android/api/org/webrtc/LibvpxVp8Decoder.java",
+        "sdk/android/api/org/webrtc/GlShader.java",
+        "sdk/android/api/org/webrtc/FrameEncryptor.java",
+        "sdk/android/api/org/webrtc/EglBase.java",
+        "sdk/android/api/org/webrtc/VideoProcessor.java",
+        "sdk/android/api/org/webrtc/SSLCertificateVerifier.java",
+        "sdk/android/api/org/webrtc/VideoSink.java",
+        "sdk/android/api/org/webrtc/MediaSource.java",
+        "sdk/android/api/org/webrtc/DefaultVideoDecoderFactory.java",
+        "sdk/android/api/org/webrtc/VideoCodecInfo.java",
+        "sdk/android/api/org/webrtc/FrameDecryptor.java",
+        "sdk/android/api/org/webrtc/VideoDecoderFactory.java",
+        "sdk/android/api/org/webrtc/TextureBufferImpl.java",
+        "sdk/android/api/org/webrtc/VideoFrame.java",
+        "sdk/android/api/org/webrtc/IceCandidateErrorEvent.java",
+        "sdk/android/api/org/webrtc/CapturerObserver.java",
+        "sdk/android/api/org/webrtc/MediaStreamTrack.java",
+        "sdk/android/api/org/webrtc/GlTextureFrameBuffer.java",
+        "sdk/android/api/org/webrtc/TurnCustomizer.java",
+        "sdk/android/api/org/webrtc/TimestampAligner.java",
+        "sdk/android/api/org/webrtc/BuiltinAudioDecoderFactoryFactory.java",
+        "sdk/android/api/org/webrtc/LibvpxVp9Decoder.java",
+        "sdk/android/api/org/webrtc/SurfaceEglRenderer.java",
+        "sdk/android/api/org/webrtc/HardwareVideoDecoderFactory.java",
+        "sdk/android/api/org/webrtc/VideoCodecStatus.java",
+        "sdk/android/api/org/webrtc/Dav1dDecoder.java",
+        "sdk/android/api/org/webrtc/VideoFrameDrawer.java",
+        "sdk/android/api/org/webrtc/CallSessionFileRotatingLogSink.java",
+        "sdk/android/api/org/webrtc/EncodedImage.java",
+    ]
+
+
+def make_github_url(repo, commit):
+    if not repo.endswith("/"):
+        repo += "/"
+    return repo + "archive/" + commit + ".tar.gz"
+
+
+def make_googlesource_url(target, commit):
+    if target == "libwebrtc":
+        return "https://webrtc.googlesource.com/src.git/+archive/" + commit + ".tar.gz"
+    elif target == "build":
+        return (
+            "https://chromium.googlesource.com/chromium/src/build/+archive/"
+            + commit
+            + ".tar.gz"
+        )
+    elif target == "third_party":
+        return (
+            "https://chromium.googlesource.com/chromium/src/third_party/+archive/"
+            + commit
+            + ".tar.gz"
+        )
+
+
+def fetch(target, url):
+    print("Fetching commit from {}".format(url))
+    req = requests.get(url)
+    if req.status_code == 200:
+        with open(target + ".tar.gz", "wb") as f:
+            f.write(req.content)
+    else:
+        print(
+            "Hit status code {} fetching commit. Aborting.".format(req.status_code),
+            file=sys.stderr,
+        )
+        sys.exit(1)
+    with open(os.path.join(LIBWEBRTC_DIR, "README.mozilla"), "a") as f:
+        # write the the command line used
+        f.write("# ./mach python {}\n".format(" ".join(sys.argv[0:])))
+        f.write(
+            "{} updated from commit {} on {}.\n".format(
+                target, url, datetime.datetime.utcnow().isoformat()
+            )
+        )
+
+
+def fetch_local(target, path, commit):
+    target_archive = target + ".tar.gz"
+    cp = subprocess.run(["git", "archive", "-o", target_archive, commit], cwd=path)
+    if cp.returncode != 0:
+        print(
+            "Hit return code {} fetching commit. Aborting.".format(cp.returncode),
+            file=sys.stderr,
+        )
+        sys.exit(1)
+
+    with open(os.path.join(LIBWEBRTC_DIR, "README.mozilla"), "a") as f:
+        # write the the command line used
+        f.write("# ./mach python {}\n".format(" ".join(sys.argv[0:])))
+        f.write(
+            "{} updated from {} commit {} on {}.\n".format(
+                target, path, commit, datetime.datetime.utcnow().isoformat()
+            )
+        )
+    shutil.move(os.path.join(path, target_archive), target_archive)
+
+
+def validate_tar_member(member, path):
+    def _is_within_directory(directory, target):
+        real_directory = os.path.realpath(directory)
+        real_target = os.path.realpath(target)
+        prefix = os.path.commonprefix([real_directory, real_target])
+        return prefix == real_directory
+
+    member_path = os.path.join(path, member.name)
+    if not _is_within_directory(path, member_path):
+        raise Exception("Attempted path traversal in tar file: " + member.name)
+    if member.issym():
+        link_path = os.path.join(os.path.dirname(member_path), member.linkname)
+        if not _is_within_directory(path, link_path):
+            raise Exception("Attempted link path traversal in tar file: " + member.name)
+    if member.mode & (stat.S_ISUID | stat.S_ISGID):
+        raise Exception("Attempted setuid or setgid in tar file: " + member.name)
+
+
+def safe_extract(tar, path=".", *, numeric_owner=False):
+    def _files(tar, path):
+        for member in tar:
+            validate_tar_member(member, path)
+            yield member
+
+    tar.extractall(path, members=_files(tar, path), numeric_owner=numeric_owner)
+
+
+def unpack(target):
+    target_archive = target + ".tar.gz"
+    target_path = "tmp-" + target
+    try:
+        shutil.rmtree(target_path)
+    except FileNotFoundError:
+        pass
+    with tarfile.open(target_archive) as t:
+        safe_extract(t, path=target_path)
+
+    if target == "libwebrtc":
+        # use the top level directories from the tarfile and
+        # delete those directories in LIBWEBRTC_DIR
+        libwebrtc_used_in_firefox = os.listdir(target_path)
+        for path in libwebrtc_used_in_firefox:
+            try:
+                shutil.rmtree(os.path.join(LIBWEBRTC_DIR, path))
+            except FileNotFoundError:
+                pass
+            except NotADirectoryError:
+                pass
+
+        unused_libwebrtc_in_firefox = get_excluded_paths()
+        forced_used_in_firefox = get_included_path_overrides()
+
+        # adjust target_path if GitHub packaging is involved
+        if not os.path.exists(os.path.join(target_path, libwebrtc_used_in_firefox[0])):
+            # GitHub packs everything inside a separate directory
+            target_path = os.path.join(target_path, os.listdir(target_path)[0])
+
+        # remove any entries found in unused_libwebrtc_in_firefox from the
+        # tarfile
+        for path in unused_libwebrtc_in_firefox:
+            if os.path.isdir(os.path.join(target_path, path)):
+                shutil.rmtree(os.path.join(target_path, path))
+            else:
+                os.remove(os.path.join(target_path, path))
+
+        # move remaining top level entries from the tarfile to LIBWEBRTC_DIR
+        for path in os.listdir(target_path):
+            shutil.move(
+                os.path.join(target_path, path), os.path.join(LIBWEBRTC_DIR, path)
+            )
+
+        # An easy, but inefficient way to accomplish including specific
+        # files from directories otherwise removed.  Re-extract the tar
+        # file, and only copy over the exact files requested.
+        shutil.rmtree(target_path)
+        with tarfile.open(target_archive) as t:
+            safe_extract(t, path=target_path)
+
+        # Copy the force included files.  Note: the instinctual action
+        # is to do this prior to removing the excluded paths to avoid
+        # reextracting the tar file.  However, this causes errors due to
+        # pre-existing paths when other directories are moved out of the
+        # tar file in the "move all the top level entries from the
+        # tarfile" phase above.
+        for path in forced_used_in_firefox:
+            dest_path = os.path.join(LIBWEBRTC_DIR, path)
+            dir_path = os.path.dirname(dest_path)
+            if not os.path.exists(dir_path):
+                os.makedirs(dir_path)
+            shutil.move(os.path.join(target_path, path), dest_path)
+    elif target == "build":
+        try:
+            shutil.rmtree(os.path.join(LIBWEBRTC_DIR, "build"))
+        except FileNotFoundError:
+            pass
+        os.makedirs(os.path.join(LIBWEBRTC_DIR, "build"))
+
+        if os.path.exists(os.path.join(target_path, "linux")):
+            for path in os.listdir(target_path):
+                shutil.move(
+                    os.path.join(target_path, path),
+                    os.path.join(LIBWEBRTC_DIR, "build", path),
+                )
+        else:
+            # GitHub packs everything inside a separate directory
+            target_path = os.path.join(target_path, os.listdir(target_path)[0])
+            for path in os.listdir(target_path):
+                shutil.move(
+                    os.path.join(target_path, path),
+                    os.path.join(LIBWEBRTC_DIR, "build", path),
+                )
+    elif target == "third_party":
+        try:
+            shutil.rmtree(os.path.join(LIBWEBRTC_DIR, "third_party"))
+        except FileNotFoundError:
+            pass
+        except NotADirectoryError:
+            pass
+
+        if os.path.exists(os.path.join(target_path, THIRDPARTY_USED_IN_FIREFOX[0])):
+            for path in THIRDPARTY_USED_IN_FIREFOX:
+                shutil.move(
+                    os.path.join(target_path, path),
+                    os.path.join(LIBWEBRTC_DIR, "third_party", path),
+                )
+        else:
+            # GitHub packs everything inside a separate directory
+            target_path = os.path.join(target_path, os.listdir(target_path)[0])
+            for path in THIRDPARTY_USED_IN_FIREFOX:
+                shutil.move(
+                    os.path.join(target_path, path),
+                    os.path.join(LIBWEBRTC_DIR, "third_party", path),
+                )
+
+
+def cleanup(target):
+    os.remove(target + ".tar.gz")
+    shutil.rmtree("tmp-" + target)
+
+
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser(description="Update libwebrtc")
+    parser.add_argument("target", choices=("libwebrtc", "build", "third_party"))
+    group = parser.add_mutually_exclusive_group(required=True)
+    group.add_argument("--from-github", type=str)
+    group.add_argument("--from-googlesource", action="store_true", default=False)
+    group.add_argument("--from-local", type=str)
+    parser.add_argument("--commit", type=str, default="master")
+    parser.add_argument("--skip-fetch", action="store_true", default=False)
+    parser.add_argument("--skip-cleanup", action="store_true", default=False)
+    args = parser.parse_args()
+
+    os.makedirs(LIBWEBRTC_DIR, exist_ok=True)
+
+    if not args.skip_fetch:
+        if args.from_github:
+            fetch(args.target, make_github_url(args.from_github, args.commit))
+        elif args.from_googlesource:
+            fetch(args.target, make_googlesource_url(args.target, args.commit))
+        elif args.from_local:
+            fetch_local(args.target, args.from_local, args.commit)
+    unpack(args.target)
+    if not args.skip_cleanup:
+        cleanup(args.target)
diff --git a/dom/media/webrtc/third_party_build/verify_vendoring.sh b/dom/media/webrtc/third_party_build/verify_vendoring.sh
new file mode 100644
index 0000000000..3d2c161075
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/verify_vendoring.sh
@@ -0,0 +1,55 @@
+#!/bin/bash
+
+function show_error_msg()
+{
+  echo "*** ERROR *** $? line $1 $0 did not complete successfully!"
+  echo "$ERROR_HELP"
+}
+ERROR_HELP=""
+
+# Print an Error message if `set -eE` causes the script to exit due to a failed command
+trap 'show_error_msg $LINENO' ERR
+
+source dom/media/webrtc/third_party_build/use_config_env.sh
+
+echo "MOZ_LIBWEBRTC_SRC: $MOZ_LIBWEBRTC_SRC"
+echo "MOZ_LIBWEBRTC_BRANCH: $MOZ_LIBWEBRTC_BRANCH"
+echo "MOZ_FASTFORWARD_BUG: $MOZ_FASTFORWARD_BUG"
+
+# After this point:
+# * eE: All commands should succeed.
+# * u: All variables should be defined before use.
+# * o pipefail: All stages of all pipes should succeed.
+set -eEuo pipefail
+
+./mach python $SCRIPT_DIR/vendor-libwebrtc.py \
+        --from-local $MOZ_LIBWEBRTC_SRC \
+        --commit $MOZ_LIBWEBRTC_BRANCH \
+        libwebrtc
+
+hg revert -q \
+   --include "third_party/libwebrtc/**moz.build" \
+   --include "third_party/libwebrtc/README.mozilla" \
+   third_party/libwebrtc
+
+FILE_CHANGE_CNT=`hg status third_party/libwebrtc | wc -l | tr -d " "`
+if [ "x$FILE_CHANGE_CNT" != "x0" ]; then
+  echo "***"
+  echo "There are changes after vendoring - running extract-for-git.py"
+  echo "is recommended.  First, find the mercurial commit after the"
+  echo "previous fast-forward landing.  The commands you want will look"
+  echo "something like:"
+  echo "  ./mach python $SCRIPT_DIR/extract-for-git.py {after-ff-commit}::{tip-of-central}"
+  echo "  mv mailbox.patch $MOZ_LIBWEBRTC_SRC"
+  echo "  (cd $MOZ_LIBWEBRTC_SRC && \\"
+  echo "   git am mailbox.patch)"
+  echo ""
+  echo "After adding the new changes from moz-central to the moz-libwebrtc"
+  echo "patch stack, you may re-run this command to verify vendoring:"
+  echo "  bash $0"
+
+  exit 1
+fi
+
+
+echo "Done - vendoring has been verified."
diff --git a/dom/media/webrtc/third_party_build/webrtc.mozbuild b/dom/media/webrtc/third_party_build/webrtc.mozbuild
new file mode 100644
index 0000000000..30169c36c2
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/webrtc.mozbuild
@@ -0,0 +1,40 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+# This file mimics the defines used by any headers in libwebrtc that get
+# included in Mozilla code.
+# We currently audit these by hand. Searching for the defines upstream is a good
+# start:
+# https://source.chromium.org/search?q=usage:%23if.*def.*%5B%5E_%5D$%20AND%20path:%5C.h%20AND%20path:%5Ethird_party%2Fwebrtc&ss=chromium
+
+if CONFIG['MOZ_WEBRTC']:
+    DEFINES['HAVE_UINT64_T'] = True
+    DEFINES['WEBRTC_MOZILLA_BUILD'] = True
+    DEFINES['RTC_ENABLE_VP9'] = True
+
+    if CONFIG['OS_TARGET'] != 'WINNT':
+        DEFINES['WEBRTC_POSIX'] = True
+        DEFINES['WEBRTC_BUILD_LIBEVENT'] = True
+
+    if CONFIG['OS_TARGET'] == 'Linux':
+        DEFINES['WEBRTC_LINUX'] = True
+    elif CONFIG['OS_TARGET'] == 'Darwin':
+        DEFINES['WEBRTC_MAC'] = True
+    elif CONFIG['OS_TARGET'] == 'WINNT':
+        DEFINES['WEBRTC_WIN'] = True
+        DEFINES['RTC_ENABLE_WIN_WGC'] = False
+        DEFINES['HAVE_WINSOCK2_H'] = True
+    elif CONFIG['OS_TARGET'] in ('DragonFly', 'FreeBSD', 'NetBSD', 'OpenBSD'):
+        DEFINES['WEBRTC_BSD'] = True
+    elif CONFIG['OS_TARGET'] == 'Android':
+        DEFINES['WEBRTC_LINUX'] = True
+        DEFINES['WEBRTC_ANDROID'] = True
+
+    if CONFIG['MOZ_X11']:
+        DEFINES['WEBRTC_USE_X11'] = True
+
+    if CONFIG["MOZ_WIDGET_TOOLKIT"] == "gtk":
+        DEFINES['WEBRTC_USE_PIPEWIRE'] = True
diff --git a/dom/media/webrtc/third_party_build/write_default_config.py b/dom/media/webrtc/third_party_build/write_default_config.py
new file mode 100644
index 0000000000..204789e052
--- /dev/null
+++ b/dom/media/webrtc/third_party_build/write_default_config.py
@@ -0,0 +1,104 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+import argparse
+import sys
+from string import Template
+
+sys.path.insert(0, "./dom/media/webrtc/third_party_build")
+import lookup_branch_head
+
+text = """#!/bin/bash
+
+# Edit {path-to} to match the location of your copy of Mozilla's
+# fork of libwebrtc (at https://github.com/mozilla/libwebrtc).
+export MOZ_LIBWEBRTC_SRC=$$STATE_DIR/moz-libwebrtc
+
+# Fast-forwarding each Chromium version of libwebrtc should be done
+# under a separate bugzilla bug.  This bug number is used when crafting
+# the commit summary as each upstream commit is vendored into the
+# mercurial repository.  The bug used for the v106 fast-forward was
+# 1800920.
+export MOZ_FASTFORWARD_BUG="$bugnum"
+
+# MOZ_NEXT_LIBWEBRTC_MILESTONE and MOZ_NEXT_FIREFOX_REL_TARGET are
+# not used during fast-forward processing, but facilitate generating this
+# default config.  To generate an default config for the next update, run
+# bash dom/media/webrtc/third_party_build/update_default_config_env.sh
+export MOZ_NEXT_LIBWEBRTC_MILESTONE=$m2
+export MOZ_NEXT_FIREFOX_REL_TARGET=$t2
+
+# For Chromium release branches, see:
+# https://chromiumdash.appspot.com/branches
+
+# Chromium's v$m1 release branch was $bh1.  This is used to pre-stack
+# the previous release branch's commits onto the appropriate base commit
+# (the first common commit between trunk and the release branch).
+export MOZ_PRIOR_UPSTREAM_BRANCH_HEAD_NUM="$bh1"
+
+# New target release branch for v$m2 is branch-heads/$bh2.  This is used
+# to calculate the next upstream commit.
+export MOZ_TARGET_UPSTREAM_BRANCH_HEAD="branch-heads/$bh2"
+
+# For local development 'mozpatches' is fine for a branch name, but when
+# pushing the patch stack to github, it should be named something like
+# 'moz-mods-chr$m2-for-rel$t2'.
+export MOZ_LIBWEBRTC_BRANCH="mozpatches"
+
+# After elm has been merged to mozilla-central, the patch stack in
+# moz-libwebrtc should be pushed to github.  The script
+# push_official_branch.sh uses this branch name when pushing to the
+# public repo.
+export MOZ_LIBWEBRTC_OFFICIAL_BRANCH="moz-mods-chr$m2-for-rel$t2"
+"""
+
+
+def build_default_config_env(bug_number, milestone, target):
+    prior_branch_head = lookup_branch_head.get_branch_head(milestone)
+    if prior_branch_head is None:
+        sys.exit("error: chromium milestone '{}' is not found.".format(milestone))
+    new_branch_head = lookup_branch_head.get_branch_head(milestone + 1)
+    if new_branch_head is None:
+        sys.exit(
+            "error: next chromium milestone '{}' is not found.".format(milestone + 1)
+        )
+
+    s = Template(text)
+    return s.substitute(
+        bugnum=bug_number,
+        m1=milestone,
+        m2=milestone + 1,
+        t2=target + 1,
+        bh1=prior_branch_head,
+        bh2=new_branch_head,
+    )
+
+
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser(
+        description="Updates the default_config_env file for new release/milestone"
+    )
+    parser.add_argument(
+        "--bug-number",
+        required=True,
+        type=int,
+        help="integer Bugzilla number (example: 1800920)",
+    )
+    parser.add_argument(
+        "--milestone",
+        required=True,
+        type=int,
+        help="integer chromium milestone (example: 106)",
+    )
+    parser.add_argument(
+        "--release-target",
+        required=True,
+        type=int,
+        help="integer firefox release (example: 110)",
+    )
+    args = parser.parse_args()
+
+    print(
+        build_default_config_env(args.bug_number, args.milestone, args.release_target),
+        end="",
+    )
diff --git a/dom/media/webrtc/transport/README b/dom/media/webrtc/transport/README
new file mode 100644
index 0000000000..3630b2aa1f
--- /dev/null
+++ b/dom/media/webrtc/transport/README
@@ -0,0 +1,45 @@
+This is a generic media transport system for WebRTC.
+
+The basic model is that you have a TransportFlow which contains a
+series of TransportLayers, each of which gets an opportunity to
+manipulate data up and down the stack (think SysV STREAMS or a
+standard networking stack). You can also address individual
+sublayers to manipulate them or to bypass reading and writing
+at an upper layer; WebRTC uses this to implement DTLS-SRTP.
+
+
+DATAFLOW MODEL
+Unlike the existing nsSocket I/O system, this is a push rather
+than a pull system. Clients of the interface do writes downward
+with SendPacket() and receive notification of incoming packets
+via callbacks registed via sigslot.h. It is the responsibility
+of the bottom layer (or any other layer which needs to reference
+external events) to arrange for that somehow; typically by
+using nsITimer or the SocketTansportService.
+
+This sort of push model is a much better fit for the demands
+of WebRTC, expecially because ICE contexts span multiple
+network transports.
+
+
+THREADING MODEL
+There are no thread locks. It is the responsibility of the caller to
+arrange that any given TransportLayer/TransportFlow is only
+manipulated in one thread at once. One good way to do this is to run
+everything on the STS thread. Many of the existing layer implementations
+(TransportLayerIce, TransportLayerLoopback) already run on STS so in those
+cases you must run on STS, though you can do setup on the main thread and
+then activate them on the STS.
+
+
+EXISTING TRANSPORT LAYERS
+The following transport layers are currently implemented:
+
+* DTLS -- a wrapper around NSS's DTLS [RFC 6347] stack
+* ICE  -- a wrapper around the nICEr ICE [RFC 5245] stack.
+* Loopback -- a loopback IO mechanism
+* Logging -- a passthrough that just logs its data
+
+The last two are primarily for debugging.
+
+
diff --git a/dom/media/webrtc/transport/SrtpFlow.cpp b/dom/media/webrtc/transport/SrtpFlow.cpp
new file mode 100644
index 0000000000..827d1a0f6d
--- /dev/null
+++ b/dom/media/webrtc/transport/SrtpFlow.cpp
@@ -0,0 +1,259 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+#include "logging.h"
+#include "SrtpFlow.h"
+
+#include "srtp.h"
+
+#include "transportlayerdtls.h"
+
+#include "mozilla/RefPtr.h"
+
+using namespace mozilla;
+
+namespace mozilla {
+
+MOZ_MTLOG_MODULE("mtransport")
+bool SrtpFlow::initialized;  // Static
+
+SrtpFlow::~SrtpFlow() {
+  if (session_) {
+    srtp_dealloc(session_);
+  }
+}
+
+unsigned int SrtpFlow::KeySize(int cipher_suite) {
+  srtp_profile_t profile = static_cast<srtp_profile_t>(cipher_suite);
+  return srtp_profile_get_master_key_length(profile);
+}
+
+unsigned int SrtpFlow::SaltSize(int cipher_suite) {
+  srtp_profile_t profile = static_cast<srtp_profile_t>(cipher_suite);
+  return srtp_profile_get_master_salt_length(profile);
+}
+
+RefPtr<SrtpFlow> SrtpFlow::Create(int cipher_suite, bool inbound,
+                                  const void* key, size_t key_len) {
+  nsresult res = Init();
+  if (!NS_SUCCEEDED(res)) return nullptr;
+
+  RefPtr<SrtpFlow> flow = new SrtpFlow();
+
+  if (!key) {
+    MOZ_MTLOG(ML_ERROR, "Null SRTP key specified");
+    return nullptr;
+  }
+
+  if ((key_len > SRTP_MAX_KEY_LENGTH) || (key_len < SRTP_MIN_KEY_LENGTH)) {
+    MOZ_ASSERT(false, "Invalid SRTP key length");
+    return nullptr;
+  }
+
+  srtp_policy_t policy;
+  memset(&policy, 0, sizeof(srtp_policy_t));
+
+  // Note that we set the same cipher suite for RTP and RTCP
+  // since any flow can only have one cipher suite with DTLS-SRTP
+  switch (cipher_suite) {
+    case kDtlsSrtpAeadAes256Gcm:
+      MOZ_MTLOG(ML_DEBUG, "Setting SRTP cipher suite SRTP_AEAD_AES_256_GCM");
+      srtp_crypto_policy_set_aes_gcm_256_16_auth(&policy.rtp);
+      srtp_crypto_policy_set_aes_gcm_256_16_auth(&policy.rtcp);
+      break;
+    case kDtlsSrtpAeadAes128Gcm:
+      MOZ_MTLOG(ML_DEBUG, "Setting SRTP cipher suite SRTP_AEAD_AES_128_GCM");
+      srtp_crypto_policy_set_aes_gcm_128_16_auth(&policy.rtp);
+      srtp_crypto_policy_set_aes_gcm_128_16_auth(&policy.rtcp);
+      break;
+    case kDtlsSrtpAes128CmHmacSha1_80:
+      MOZ_MTLOG(ML_DEBUG,
+                "Setting SRTP cipher suite SRTP_AES128_CM_HMAC_SHA1_80");
+      srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtp);
+      srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp);
+      break;
+    case kDtlsSrtpAes128CmHmacSha1_32:
+      MOZ_MTLOG(ML_DEBUG,
+                "Setting SRTP cipher suite SRTP_AES128_CM_HMAC_SHA1_32");
+      srtp_crypto_policy_set_aes_cm_128_hmac_sha1_32(&policy.rtp);
+      srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(
+          &policy.rtcp);  // 80-bit per RFC 5764
+      break;              // S 4.1.2.
+    default:
+      MOZ_MTLOG(ML_ERROR, "Request to set unknown SRTP cipher suite");
+      return nullptr;
+  }
+  // This key is copied into the srtp_t object, so we don't
+  // need to keep it.
+  policy.key =
+      const_cast<unsigned char*>(static_cast<const unsigned char*>(key));
+  policy.ssrc.type = inbound ? ssrc_any_inbound : ssrc_any_outbound;
+  policy.ssrc.value = 0;
+  policy.window_size =
+      1024;  // Use the Chrome value.  Needs to be revisited.  Default is 128
+  policy.allow_repeat_tx = 1;  // Use Chrome value; needed for NACK mode to work
+  policy.next = nullptr;
+
+  // Now make the session
+  srtp_err_status_t r = srtp_create(&flow->session_, &policy);
+  if (r != srtp_err_status_ok) {
+    MOZ_MTLOG(ML_ERROR, "Error creating srtp session");
+    return nullptr;
+  }
+
+  return flow;
+}
+
+nsresult SrtpFlow::CheckInputs(bool protect, void* in, int in_len, int max_len,
+                               int* out_len) {
+  MOZ_ASSERT(in);
+  if (!in) {
+    MOZ_MTLOG(ML_ERROR, "NULL input value");
+    return NS_ERROR_NULL_POINTER;
+  }
+
+  if (in_len < 0) {
+    MOZ_MTLOG(ML_ERROR, "Input length is negative");
+    return NS_ERROR_ILLEGAL_VALUE;
+  }
+
+  if (max_len < 0) {
+    MOZ_MTLOG(ML_ERROR, "Max output length is negative");
+    return NS_ERROR_ILLEGAL_VALUE;
+  }
+
+  if (protect) {
+    if ((max_len < SRTP_MAX_EXPANSION) ||
+        ((max_len - SRTP_MAX_EXPANSION) < in_len)) {
+      MOZ_MTLOG(ML_ERROR, "Output too short");
+      return NS_ERROR_ILLEGAL_VALUE;
+    }
+  } else {
+    if (in_len > max_len) {
+      MOZ_MTLOG(ML_ERROR, "Output too short");
+      return NS_ERROR_ILLEGAL_VALUE;
+    }
+  }
+
+  return NS_OK;
+}
+
+nsresult SrtpFlow::ProtectRtp(void* in, int in_len, int max_len, int* out_len) {
+  nsresult res = CheckInputs(true, in, in_len, max_len, out_len);
+  if (NS_FAILED(res)) return res;
+
+  int len = in_len;
+  srtp_err_status_t r = srtp_protect(session_, in, &len);
+
+  if (r != srtp_err_status_ok) {
+    MOZ_MTLOG(ML_ERROR, "Error protecting SRTP packet");
+    return NS_ERROR_FAILURE;
+  }
+
+  MOZ_ASSERT(len <= max_len);
+  *out_len = len;
+
+  MOZ_MTLOG(ML_DEBUG,
+            "Successfully protected an SRTP packet of len " << *out_len);
+
+  return NS_OK;
+}
+
+nsresult SrtpFlow::UnprotectRtp(void* in, int in_len, int max_len,
+                                int* out_len) {
+  nsresult res = CheckInputs(false, in, in_len, max_len, out_len);
+  if (NS_FAILED(res)) return res;
+
+  int len = in_len;
+  srtp_err_status_t r = srtp_unprotect(session_, in, &len);
+
+  if (r != srtp_err_status_ok) {
+    MOZ_MTLOG(ML_ERROR, "Error unprotecting SRTP packet error=" << (int)r);
+    return NS_ERROR_FAILURE;
+  }
+
+  MOZ_ASSERT(len <= max_len);
+  *out_len = len;
+
+  MOZ_MTLOG(ML_DEBUG,
+            "Successfully unprotected an SRTP packet of len " << *out_len);
+
+  return NS_OK;
+}
+
+nsresult SrtpFlow::ProtectRtcp(void* in, int in_len, int max_len,
+                               int* out_len) {
+  nsresult res = CheckInputs(true, in, in_len, max_len, out_len);
+  if (NS_FAILED(res)) return res;
+
+  int len = in_len;
+  srtp_err_status_t r = srtp_protect_rtcp(session_, in, &len);
+
+  if (r != srtp_err_status_ok) {
+    MOZ_MTLOG(ML_ERROR, "Error protecting SRTCP packet");
+    return NS_ERROR_FAILURE;
+  }
+
+  MOZ_ASSERT(len <= max_len);
+  *out_len = len;
+
+  MOZ_MTLOG(ML_DEBUG,
+            "Successfully protected an SRTCP packet of len " << *out_len);
+
+  return NS_OK;
+}
+
+nsresult SrtpFlow::UnprotectRtcp(void* in, int in_len, int max_len,
+                                 int* out_len) {
+  nsresult res = CheckInputs(false, in, in_len, max_len, out_len);
+  if (NS_FAILED(res)) return res;
+
+  int len = in_len;
+  srtp_err_status_t r = srtp_unprotect_rtcp(session_, in, &len);
+
+  if (r != srtp_err_status_ok) {
+    MOZ_MTLOG(ML_ERROR, "Error unprotecting SRTCP packet error=" << (int)r);
+    return NS_ERROR_FAILURE;
+  }
+
+  MOZ_ASSERT(len <= max_len);
+  *out_len = len;
+
+  MOZ_MTLOG(ML_DEBUG,
+            "Successfully unprotected an SRTCP packet of len " << *out_len);
+
+  return NS_OK;
+}
+
+// Statics
+void SrtpFlow::srtp_event_handler(srtp_event_data_t* data) {
+  // TODO(ekr@rtfm.com): Implement this
+  MOZ_CRASH();
+}
+
+nsresult SrtpFlow::Init() {
+  if (!initialized) {
+    srtp_err_status_t r = srtp_init();
+    if (r != srtp_err_status_ok) {
+      MOZ_MTLOG(ML_ERROR, "Could not initialize SRTP");
+      MOZ_ASSERT(PR_FALSE);
+      return NS_ERROR_FAILURE;
+    }
+
+    r = srtp_install_event_handler(&SrtpFlow::srtp_event_handler);
+    if (r != srtp_err_status_ok) {
+      MOZ_MTLOG(ML_ERROR, "Could not install SRTP event handler");
+      MOZ_ASSERT(PR_FALSE);
+      return NS_ERROR_FAILURE;
+    }
+
+    initialized = true;
+  }
+
+  return NS_OK;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/transport/SrtpFlow.h b/dom/media/webrtc/transport/SrtpFlow.h
new file mode 100644
index 0000000000..92fbfcf1a5
--- /dev/null
+++ b/dom/media/webrtc/transport/SrtpFlow.h
@@ -0,0 +1,69 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+#ifndef srtpflow_h__
+#define srtpflow_h__
+
+#include "mozilla/RefPtr.h"
+#include "nsISupportsImpl.h"
+#include "srtp.h"
+
+namespace mozilla {
+
+#define SRTP_ICM_MASTER_KEY_LENGTH 16
+#define SRTP_ICM_MASTER_SALT_LENGTH 14
+#define SRTP_ICM_MAX_MASTER_LENGTH \
+  (SRTP_ICM_MASTER_KEY_LENGTH + SRTP_ICM_MASTER_SALT_LENGTH)
+
+#define SRTP_GCM_MASTER_KEY_MIN_LENGTH 16
+#define SRTP_GCM_MASTER_KEY_MAX_LENGTH 32
+#define SRTP_GCM_MASTER_SALT_LENGTH 12
+
+#define SRTP_GCM_MIN_MASTER_LENGTH \
+  (SRTP_GCM_MASTER_KEY_MIN_LENGTH + SRTP_GCM_MASTER_SALT_LENGTH)
+#define SRTP_GCM_MAX_MASTER_LENGTH \
+  (SRTP_GCM_MASTER_KEY_MAX_LENGTH + SRTP_GCM_MASTER_SALT_LENGTH)
+
+#define SRTP_MIN_KEY_LENGTH SRTP_GCM_MIN_MASTER_LENGTH
+#define SRTP_MAX_KEY_LENGTH SRTP_GCM_MAX_MASTER_LENGTH
+
+// SRTCP requires an auth tag *plus* a 4-byte index-plus-'E'-bit value (see
+// RFC 3711)
+#define SRTP_MAX_EXPANSION (SRTP_MAX_TRAILER_LEN + 4)
+
+class SrtpFlow {
+  ~SrtpFlow();
+
+ public:
+  static unsigned int KeySize(int cipher_suite);
+  static unsigned int SaltSize(int cipher_suite);
+
+  static RefPtr<SrtpFlow> Create(int cipher_suite, bool inbound,
+                                 const void* key, size_t key_len);
+
+  nsresult ProtectRtp(void* in, int in_len, int max_len, int* out_len);
+  nsresult UnprotectRtp(void* in, int in_len, int max_len, int* out_len);
+  nsresult ProtectRtcp(void* in, int in_len, int max_len, int* out_len);
+  nsresult UnprotectRtcp(void* in, int in_len, int max_len, int* out_len);
+
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(SrtpFlow)
+
+  static void srtp_event_handler(srtp_event_data_t* data);
+
+ private:
+  SrtpFlow() : session_(nullptr) {}
+
+  nsresult CheckInputs(bool protect, void* in, int in_len, int max_len,
+                       int* out_len);
+
+  static nsresult Init();
+  static bool initialized;  // Was libsrtp initialized? Only happens once.
+
+  srtp_t session_;
+};
+
+}  // namespace mozilla
+#endif
diff --git a/dom/media/webrtc/transport/WebrtcTCPSocketWrapper.cpp b/dom/media/webrtc/transport/WebrtcTCPSocketWrapper.cpp
new file mode 100644
index 0000000000..2bff019878
--- /dev/null
+++ b/dom/media/webrtc/transport/WebrtcTCPSocketWrapper.cpp
@@ -0,0 +1,123 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set sw=2 ts=8 et tw=80 ft=cpp : */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "WebrtcTCPSocketWrapper.h"
+
+#include "mozilla/net/WebrtcTCPSocketChild.h"
+
+#include "nsNetCID.h"
+#include "nsProxyRelease.h"
+#include "nsServiceManagerUtils.h"
+
+#include "nr_socket_proxy_config.h"
+
+namespace mozilla::net {
+
+using std::shared_ptr;
+
+WebrtcTCPSocketWrapper::WebrtcTCPSocketWrapper(
+    WebrtcTCPSocketCallback* aCallbacks)
+    : mProxyCallbacks(aCallbacks),
+      mWebrtcTCPSocket(nullptr),
+      mMainThread(nullptr),
+      mSocketThread(nullptr) {
+  mMainThread = GetMainThreadSerialEventTarget();
+  mSocketThread = do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID);
+  MOZ_RELEASE_ASSERT(mMainThread, "no main thread");
+  MOZ_RELEASE_ASSERT(mSocketThread, "no socket thread");
+}
+
+WebrtcTCPSocketWrapper::~WebrtcTCPSocketWrapper() {
+  MOZ_ASSERT(!mWebrtcTCPSocket, "webrtc TCP socket non-null");
+
+  // If we're never opened then we never get an OnClose from our parent process.
+  // We need to release our callbacks here safely.
+  NS_ProxyRelease("WebrtcTCPSocketWrapper::CleanUpCallbacks", mSocketThread,
+                  mProxyCallbacks.forget());
+}
+
+void WebrtcTCPSocketWrapper::AsyncOpen(
+    const nsCString& aHost, const int& aPort, const nsCString& aLocalAddress,
+    const int& aLocalPort, bool aUseTls,
+    const shared_ptr<NrSocketProxyConfig>& aConfig) {
+  if (!NS_IsMainThread()) {
+    MOZ_ALWAYS_SUCCEEDS(mMainThread->Dispatch(
+        NewRunnableMethod<const nsCString, const int, const nsCString,
+                          const int, bool,
+                          const shared_ptr<NrSocketProxyConfig>>(
+            "WebrtcTCPSocketWrapper::AsyncOpen", this,
+            &WebrtcTCPSocketWrapper::AsyncOpen, aHost, aPort, aLocalAddress,
+            aLocalPort, aUseTls, aConfig)));
+    return;
+  }
+
+  MOZ_ASSERT(!mWebrtcTCPSocket, "wrapper already open");
+  mWebrtcTCPSocket = new WebrtcTCPSocketChild(this);
+  mWebrtcTCPSocket->AsyncOpen(aHost, aPort, aLocalAddress, aLocalPort, aUseTls,
+                              aConfig);
+}
+
+void WebrtcTCPSocketWrapper::SendWrite(nsTArray<uint8_t>&& aReadData) {
+  if (!NS_IsMainThread()) {
+    MOZ_ALWAYS_SUCCEEDS(
+        mMainThread->Dispatch(NewRunnableMethod<nsTArray<uint8_t>&&>(
+            "WebrtcTCPSocketWrapper::SendWrite", this,
+            &WebrtcTCPSocketWrapper::SendWrite, std::move(aReadData))));
+    return;
+  }
+
+  MOZ_ASSERT(mWebrtcTCPSocket, "webrtc TCP socket should be non-null");
+  mWebrtcTCPSocket->SendWrite(aReadData);
+}
+
+void WebrtcTCPSocketWrapper::Close() {
+  if (!NS_IsMainThread()) {
+    MOZ_ALWAYS_SUCCEEDS(mMainThread->Dispatch(
+        NewRunnableMethod("WebrtcTCPSocketWrapper::Close", this,
+                          &WebrtcTCPSocketWrapper::Close)));
+    return;
+  }
+
+  // We're only open if we have a channel. Also Close() should be idempotent.
+  if (mWebrtcTCPSocket) {
+    RefPtr<WebrtcTCPSocketChild> child = mWebrtcTCPSocket;
+    mWebrtcTCPSocket = nullptr;
+    child->SendClose();
+  }
+}
+
+void WebrtcTCPSocketWrapper::OnClose(nsresult aReason) {
+  MOZ_ASSERT(NS_IsMainThread(), "not on main thread");
+  MOZ_ASSERT(mProxyCallbacks, "webrtc TCP callbacks should be non-null");
+
+  MOZ_ALWAYS_SUCCEEDS(mSocketThread->Dispatch(NewRunnableMethod<nsresult>(
+      "WebrtcTCPSocketWrapper::OnClose", mProxyCallbacks,
+      &WebrtcTCPSocketCallback::OnClose, aReason)));
+
+  NS_ProxyRelease("WebrtcTCPSocketWrapper::CleanUpCallbacks", mSocketThread,
+                  mProxyCallbacks.forget());
+}
+
+void WebrtcTCPSocketWrapper::OnRead(nsTArray<uint8_t>&& aReadData) {
+  MOZ_ASSERT(NS_IsMainThread(), "not on main thread");
+  MOZ_ASSERT(mProxyCallbacks, "webrtc TCP callbacks should be non-null");
+
+  MOZ_ALWAYS_SUCCEEDS(
+      mSocketThread->Dispatch(NewRunnableMethod<nsTArray<uint8_t>&&>(
+          "WebrtcTCPSocketWrapper::OnRead", mProxyCallbacks,
+          &WebrtcTCPSocketCallback::OnRead, std::move(aReadData))));
+}
+
+void WebrtcTCPSocketWrapper::OnConnected(const nsACString& aProxyType) {
+  MOZ_ASSERT(NS_IsMainThread(), "not on main thread");
+  MOZ_ASSERT(mProxyCallbacks, "webrtc TCP callbacks should be non-null");
+
+  MOZ_ALWAYS_SUCCEEDS(mSocketThread->Dispatch(NewRunnableMethod<nsCString>(
+      "WebrtcTCPSocketWrapper::OnConnected", mProxyCallbacks,
+      &WebrtcTCPSocketCallback::OnConnected, aProxyType)));
+}
+
+}  // namespace mozilla::net
diff --git a/dom/media/webrtc/transport/WebrtcTCPSocketWrapper.h b/dom/media/webrtc/transport/WebrtcTCPSocketWrapper.h
new file mode 100644
index 0000000000..c0775ee9aa
--- /dev/null
+++ b/dom/media/webrtc/transport/WebrtcTCPSocketWrapper.h
@@ -0,0 +1,69 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set sw=2 ts=8 et tw=80 ft=cpp : */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef webrtc_tcp_socket_wrapper__
+#define webrtc_tcp_socket_wrapper__
+
+#include <memory>
+
+#include "nsCOMPtr.h"
+#include "nsTArray.h"
+
+#include "mozilla/net/WebrtcTCPSocketCallback.h"
+
+class nsIEventTarget;
+
+namespace mozilla {
+
+class NrSocketProxyConfig;
+
+namespace net {
+
+class WebrtcTCPSocketChild;
+
+/**
+ * WebrtcTCPSocketWrapper is a protector class for mtransport and IPDL.
+ * mtransport and IPDL cannot include headers from each other due to conflicting
+ * typedefs. Also it helps users by dispatching calls to the appropriate thread
+ * based on mtransport's and IPDL's threading requirements.
+ *
+ * WebrtcTCPSocketWrapper is only used in the child process.
+ * WebrtcTCPSocketWrapper does not dispatch for the parent process.
+ * WebrtcTCPSocketCallback calls are dispatched to the STS thread.
+ * IPDL calls are dispatched to the main thread.
+ */
+class WebrtcTCPSocketWrapper : public WebrtcTCPSocketCallback {
+ public:
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(WebrtcTCPSocketWrapper, override)
+
+  explicit WebrtcTCPSocketWrapper(WebrtcTCPSocketCallback* aCallbacks);
+
+  virtual void AsyncOpen(const nsCString& aHost, const int& aPort,
+                         const nsCString& aLocalAddress, const int& aLocalPort,
+                         bool aUseTls,
+                         const std::shared_ptr<NrSocketProxyConfig>& aConfig);
+  virtual void SendWrite(nsTArray<uint8_t>&& aReadData);
+  virtual void Close();
+
+  // WebrtcTCPSocketCallback
+  virtual void OnClose(nsresult aReason) override;
+  virtual void OnConnected(const nsACString& aProxyType) override;
+  virtual void OnRead(nsTArray<uint8_t>&& aReadData) override;
+
+ protected:
+  RefPtr<WebrtcTCPSocketCallback> mProxyCallbacks;
+  RefPtr<WebrtcTCPSocketChild> mWebrtcTCPSocket;
+
+  nsCOMPtr<nsIEventTarget> mMainThread;
+  nsCOMPtr<nsIEventTarget> mSocketThread;
+
+  virtual ~WebrtcTCPSocketWrapper();
+};
+
+}  // namespace net
+}  // namespace mozilla
+
+#endif  // webrtc_tcp_socket_wrapper__
diff --git a/dom/media/webrtc/transport/build/moz.build b/dom/media/webrtc/transport/build/moz.build
new file mode 100644
index 0000000000..7349ee5d71
--- /dev/null
+++ b/dom/media/webrtc/transport/build/moz.build
@@ -0,0 +1,44 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+include("/ipc/chromium/chromium-config.mozbuild")
+
+EXPORTS.transport += [
+    "../dtlsidentity.h",
+    "../m_cpp_utils.h",
+    "../mediapacket.h",
+    "../nr_socket_proxy_config.h",
+    "../nricectx.h",
+    "../nricemediastream.h",
+    "../nriceresolverfake.h",
+    "../nricestunaddr.h",
+    "../rlogconnector.h",
+    "../runnable_utils.h",
+    "../sigslot.h",
+    "../simpletokenbucket.h",
+    "../SrtpFlow.h",
+    "../stun_socket_filter.h",
+    "../transportflow.h",
+    "../transportlayer.h",
+    "../transportlayerdtls.h",
+    "../transportlayerice.h",
+    "../transportlayerlog.h",
+    "../transportlayerloopback.h",
+    "../transportlayersrtp.h",
+]
+
+include("../common.build")
+
+# Add libFuzzer configuration directives
+include("/tools/fuzzing/libfuzzer-config.mozbuild")
+
+# These files cannot be built in unified mode because of the redefinition of
+# getLogModule, UNIMPLEMENTED, nr_socket_long_term_violation_time,
+# nr_socket_short_term_violation_time, nrappkit/IPDL typedef conflicts in
+# PBrowserOrId and WebrtcTCPSocketChild.
+SOURCES += transport_cppsrcs
+
+FINAL_LIBRARY = "xul"
diff --git a/dom/media/webrtc/transport/common.build b/dom/media/webrtc/transport/common.build
new file mode 100644
index 0000000000..b907b57ee8
--- /dev/null
+++ b/dom/media/webrtc/transport/common.build
@@ -0,0 +1,94 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+transport_lcppsrcs = [
+    "dtlsidentity.cpp",
+    "mediapacket.cpp",
+    "nr_socket_proxy_config.cpp",
+    "nr_socket_prsock.cpp",
+    "nr_socket_tcp.cpp",
+    "nr_timer.cpp",
+    "nricectx.cpp",
+    "nricemediastream.cpp",
+    "nriceresolver.cpp",
+    "nriceresolverfake.cpp",
+    "nricestunaddr.cpp",
+    "nrinterfaceprioritizer.cpp",
+    "rlogconnector.cpp",
+    "simpletokenbucket.cpp",
+    "SrtpFlow.cpp",
+    "stun_socket_filter.cpp",
+    "test_nr_socket.cpp",
+    "transportflow.cpp",
+    "transportlayer.cpp",
+    "transportlayerdtls.cpp",
+    "transportlayerice.cpp",
+    "transportlayerlog.cpp",
+    "transportlayerloopback.cpp",
+    "transportlayersrtp.cpp",
+    "WebrtcTCPSocketWrapper.cpp",
+]
+
+transport_cppsrcs = [
+    "/dom/media/webrtc/transport/%s" % s for s in sorted(transport_lcppsrcs)
+]
+
+LOCAL_INCLUDES += [
+    "/dom/media/webrtc/transport/",
+    "/dom/media/webrtc/transport/third_party/",
+    "/dom/media/webrtc/transport/third_party/nICEr/src/crypto",
+    "/dom/media/webrtc/transport/third_party/nICEr/src/ice",
+    "/dom/media/webrtc/transport/third_party/nICEr/src/net",
+    "/dom/media/webrtc/transport/third_party/nICEr/src/stun",
+    "/dom/media/webrtc/transport/third_party/nICEr/src/util",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/event",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/log",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/plugin",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/port/generic/include",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/registry",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/share",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/stats",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr",
+    "/third_party/libsrtp/src/crypto/include",
+    "/third_party/libsrtp/src/include",
+]
+
+if CONFIG["OS_TARGET"] in ["Darwin", "DragonFly", "FreeBSD", "NetBSD", "OpenBSD"]:
+    if CONFIG["OS_TARGET"] == "Darwin":
+        DEFINES["DARWIN"] = True
+    else:
+        DEFINES["BSD"] = True
+    LOCAL_INCLUDES += [
+        "/dom/media/webrtc/transport/third_party/nrappkit/src/port/darwin/include",
+    ]
+elif CONFIG["OS_TARGET"] == "Linux":
+    DEFINES["LINUX"] = True
+    LOCAL_INCLUDES += [
+        "/dom/media/webrtc/transport/third_party/nrappkit/src/port/linux/include",
+    ]
+elif CONFIG["OS_TARGET"] == "Android":
+    DEFINES["LINUX"] = True
+    DEFINES["ANDROID"] = True
+    LOCAL_INCLUDES += [
+        "/dom/media/webrtc/transport/third_party/nrappkit/src/port/android/include",
+    ]
+elif CONFIG["OS_TARGET"] == "WINNT":
+    DEFINES["WIN"] = True
+    # for stun.h
+    DEFINES["WIN32"] = True
+    LOCAL_INCLUDES += [
+        "/dom/media/webrtc/transport/third_party/nrappkit/src/port/win32/include",
+    ]
+
+for var in ("HAVE_STRDUP", "NR_SOCKET_IS_VOID_PTR"):
+    DEFINES[var] = True
+
+DEFINES["R_DEFINED_INT2"] = "int16_t"
+DEFINES["R_DEFINED_UINT2"] = "uint16_t"
+DEFINES["R_DEFINED_INT4"] = "int32_t"
+DEFINES["R_DEFINED_UINT4"] = "uint32_t"
+DEFINES["R_DEFINED_INT8"] = "int64_t"
+DEFINES["R_DEFINED_UINT8"] = "uint64_t"
diff --git a/dom/media/webrtc/transport/dtlsidentity.cpp b/dom/media/webrtc/transport/dtlsidentity.cpp
new file mode 100644
index 0000000000..c466ce7d88
--- /dev/null
+++ b/dom/media/webrtc/transport/dtlsidentity.cpp
@@ -0,0 +1,331 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "dtlsidentity.h"
+
+#include "cert.h"
+#include "cryptohi.h"
+#include "keyhi.h"
+#include "nsError.h"
+#include "pk11pub.h"
+#include "sechash.h"
+#include "mozpkix/nss_scoped_ptrs.h"
+#include "secerr.h"
+#include "sslerr.h"
+
+#include "mozilla/Sprintf.h"
+
+namespace mozilla {
+
+SECItem* WrapPrivateKeyInfoWithEmptyPassword(
+    SECKEYPrivateKey* pk) /* encrypt this private key */
+{
+  if (!pk) {
+    PR_SetError(SEC_ERROR_INVALID_ARGS, 0);
+    return nullptr;
+  }
+
+  UniquePK11SlotInfo slot(PK11_GetInternalSlot());
+  if (!slot) {
+    return nullptr;
+  }
+
+  // For private keys, NSS cannot export anything other than RSA, but we need EC
+  // also. So, we use the private key encryption function to serialize instead,
+  // using a hard-coded dummy password; this is not intended to provide any
+  // additional security, it just works around a limitation in NSS.
+  SECItem dummyPassword = {siBuffer, nullptr, 0};
+  UniqueSECKEYEncryptedPrivateKeyInfo epki(PK11_ExportEncryptedPrivKeyInfo(
+      slot.get(), SEC_OID_AES_128_CBC, &dummyPassword, pk, 1, nullptr));
+
+  if (!epki) {
+    return nullptr;
+  }
+
+  return SEC_ASN1EncodeItem(
+      nullptr, nullptr, epki.get(),
+      NSS_Get_SECKEY_EncryptedPrivateKeyInfoTemplate(nullptr, false));
+}
+
+SECStatus UnwrapPrivateKeyInfoWithEmptyPassword(
+    SECItem* derPKI, const UniqueCERTCertificate& aCert,
+    SECKEYPrivateKey** privk) {
+  if (!derPKI || !aCert || !privk) {
+    PR_SetError(SEC_ERROR_INVALID_ARGS, 0);
+    return SECFailure;
+  }
+
+  UniqueSECKEYPublicKey publicKey(CERT_ExtractPublicKey(aCert.get()));
+  // This is a pointer to data inside publicKey
+  SECItem* publicValue = nullptr;
+  switch (publicKey->keyType) {
+    case dsaKey:
+      publicValue = &publicKey->u.dsa.publicValue;
+      break;
+    case dhKey:
+      publicValue = &publicKey->u.dh.publicValue;
+      break;
+    case rsaKey:
+      publicValue = &publicKey->u.rsa.modulus;
+      break;
+    case ecKey:
+      publicValue = &publicKey->u.ec.publicValue;
+      break;
+    default:
+      MOZ_ASSERT(false);
+      PR_SetError(SSL_ERROR_BAD_CERTIFICATE, 0);
+      return SECFailure;
+  }
+
+  UniquePK11SlotInfo slot(PK11_GetInternalSlot());
+  if (!slot) {
+    return SECFailure;
+  }
+
+  UniquePLArenaPool temparena(PORT_NewArena(DER_DEFAULT_CHUNKSIZE));
+  if (!temparena) {
+    return SECFailure;
+  }
+
+  SECKEYEncryptedPrivateKeyInfo* epki =
+      PORT_ArenaZNew(temparena.get(), SECKEYEncryptedPrivateKeyInfo);
+  if (!epki) {
+    return SECFailure;
+  }
+
+  SECStatus rv = SEC_ASN1DecodeItem(
+      temparena.get(), epki,
+      NSS_Get_SECKEY_EncryptedPrivateKeyInfoTemplate(nullptr, false), derPKI);
+  if (rv != SECSuccess) {
+    // If SEC_ASN1DecodeItem fails, we cannot assume anything about the
+    // validity of the data in epki. The best we can do is free the arena
+    // and return.
+    return rv;
+  }
+
+  // See comment in WrapPrivateKeyInfoWithEmptyPassword about this
+  // dummy password stuff.
+  SECItem dummyPassword = {siBuffer, nullptr, 0};
+  return PK11_ImportEncryptedPrivateKeyInfoAndReturnKey(
+      slot.get(), epki, &dummyPassword, nullptr, publicValue, false, false,
+      publicKey->keyType, KU_ALL, privk, nullptr);
+}
+
+nsresult DtlsIdentity::Serialize(nsTArray<uint8_t>* aKeyDer,
+                                 nsTArray<uint8_t>* aCertDer) {
+  ScopedSECItem derPki(WrapPrivateKeyInfoWithEmptyPassword(private_key_.get()));
+  if (!derPki) {
+    return NS_ERROR_FAILURE;
+  }
+
+  aKeyDer->AppendElements(derPki->data, derPki->len);
+  aCertDer->AppendElements(cert_->derCert.data, cert_->derCert.len);
+  return NS_OK;
+}
+
+/* static */
+RefPtr<DtlsIdentity> DtlsIdentity::Deserialize(
+    const nsTArray<uint8_t>& aKeyDer, const nsTArray<uint8_t>& aCertDer,
+    SSLKEAType authType) {
+  SECItem certDer = {siBuffer, const_cast<uint8_t*>(aCertDer.Elements()),
+                     static_cast<unsigned int>(aCertDer.Length())};
+  UniqueCERTCertificate cert(CERT_NewTempCertificate(
+      CERT_GetDefaultCertDB(), &certDer, nullptr, true, true));
+
+  SECItem derPKI = {siBuffer, const_cast<uint8_t*>(aKeyDer.Elements()),
+                    static_cast<unsigned int>(aKeyDer.Length())};
+
+  SECKEYPrivateKey* privateKey;
+  if (UnwrapPrivateKeyInfoWithEmptyPassword(&derPKI, cert, &privateKey) !=
+      SECSuccess) {
+    MOZ_ASSERT(false);
+    return nullptr;
+  }
+
+  return new DtlsIdentity(UniqueSECKEYPrivateKey(privateKey), std::move(cert),
+                          authType);
+}
+
+RefPtr<DtlsIdentity> DtlsIdentity::Generate() {
+  UniquePK11SlotInfo slot(PK11_GetInternalSlot());
+  if (!slot) {
+    return nullptr;
+  }
+
+  uint8_t random_name[16];
+
+  SECStatus rv =
+      PK11_GenerateRandomOnSlot(slot.get(), random_name, sizeof(random_name));
+  if (rv != SECSuccess) return nullptr;
+
+  std::string name;
+  char chunk[3];
+  for (unsigned char r_name : random_name) {
+    SprintfLiteral(chunk, "%.2x", r_name);
+    name += chunk;
+  }
+
+  std::string subject_name_string = "CN=" + name;
+  UniqueCERTName subject_name(CERT_AsciiToName(subject_name_string.c_str()));
+  if (!subject_name) {
+    return nullptr;
+  }
+
+  unsigned char paramBuf[12];  // OIDs are small
+  SECItem ecdsaParams = {siBuffer, paramBuf, sizeof(paramBuf)};
+  SECOidData* oidData = SECOID_FindOIDByTag(SEC_OID_SECG_EC_SECP256R1);
+  if (!oidData || (oidData->oid.len > (sizeof(paramBuf) - 2))) {
+    return nullptr;
+  }
+  ecdsaParams.data[0] = SEC_ASN1_OBJECT_ID;
+  ecdsaParams.data[1] = oidData->oid.len;
+  memcpy(ecdsaParams.data + 2, oidData->oid.data, oidData->oid.len);
+  ecdsaParams.len = oidData->oid.len + 2;
+
+  SECKEYPublicKey* pubkey;
+  UniqueSECKEYPrivateKey private_key(
+      PK11_GenerateKeyPair(slot.get(), CKM_EC_KEY_PAIR_GEN, &ecdsaParams,
+                           &pubkey, PR_FALSE, PR_TRUE, nullptr));
+  if (private_key == nullptr) return nullptr;
+  UniqueSECKEYPublicKey public_key(pubkey);
+  pubkey = nullptr;
+
+  UniqueCERTSubjectPublicKeyInfo spki(
+      SECKEY_CreateSubjectPublicKeyInfo(public_key.get()));
+  if (!spki) {
+    return nullptr;
+  }
+
+  UniqueCERTCertificateRequest certreq(
+      CERT_CreateCertificateRequest(subject_name.get(), spki.get(), nullptr));
+  if (!certreq) {
+    return nullptr;
+  }
+
+  // From 1 day before todayto 30 days after.
+  // This is a sort of arbitrary range designed to be valid
+  // now with some slack in case the other side expects
+  // some before expiry.
+  //
+  // Note: explicit casts necessary to avoid
+  //       warning C4307: '*' : integral constant overflow
+  static const PRTime oneDay = PRTime(PR_USEC_PER_SEC) * PRTime(60)  // sec
+                               * PRTime(60)                          // min
+                               * PRTime(24);                         // hours
+  PRTime now = PR_Now();
+  PRTime notBefore = now - oneDay;
+  PRTime notAfter = now + (PRTime(30) * oneDay);
+
+  UniqueCERTValidity validity(CERT_CreateValidity(notBefore, notAfter));
+  if (!validity) {
+    return nullptr;
+  }
+
+  unsigned long serial;
+  // Note: This serial in principle could collide, but it's unlikely
+  rv = PK11_GenerateRandomOnSlot(
+      slot.get(), reinterpret_cast<unsigned char*>(&serial), sizeof(serial));
+  if (rv != SECSuccess) {
+    return nullptr;
+  }
+
+  // NB: CERTCertificates created with CERT_CreateCertificate are not safe to
+  // use with other NSS functions like CERT_DupCertificate.
+  // The strategy here is to create a tbsCertificate ("to-be-signed
+  // certificate"), encode it, and sign it, resulting in a signed DER
+  // certificate that can be decoded into a CERTCertificate.
+  UniqueCERTCertificate tbsCertificate(CERT_CreateCertificate(
+      serial, subject_name.get(), validity.get(), certreq.get()));
+  if (!tbsCertificate) {
+    return nullptr;
+  }
+
+  PLArenaPool* arena = tbsCertificate->arena;
+
+  rv = SECOID_SetAlgorithmID(arena, &tbsCertificate->signature,
+                             SEC_OID_ANSIX962_ECDSA_SHA256_SIGNATURE, nullptr);
+  if (rv != SECSuccess) return nullptr;
+
+  // Set version to X509v3.
+  *(tbsCertificate->version.data) = SEC_CERTIFICATE_VERSION_3;
+  tbsCertificate->version.len = 1;
+
+  SECItem innerDER;
+  innerDER.len = 0;
+  innerDER.data = nullptr;
+
+  if (!SEC_ASN1EncodeItem(arena, &innerDER, tbsCertificate.get(),
+                          SEC_ASN1_GET(CERT_CertificateTemplate))) {
+    return nullptr;
+  }
+
+  SECItem* certDer = PORT_ArenaZNew(arena, SECItem);
+  if (!certDer) {
+    return nullptr;
+  }
+
+  rv = SEC_DerSignData(arena, certDer, innerDER.data, innerDER.len,
+                       private_key.get(),
+                       SEC_OID_ANSIX962_ECDSA_SHA256_SIGNATURE);
+  if (rv != SECSuccess) {
+    return nullptr;
+  }
+
+  UniqueCERTCertificate certificate(CERT_NewTempCertificate(
+      CERT_GetDefaultCertDB(), certDer, nullptr, false, true));
+
+  return new DtlsIdentity(std::move(private_key), std::move(certificate),
+                          ssl_kea_ecdh);
+}
+
+const std::string DtlsIdentity::DEFAULT_HASH_ALGORITHM = "sha-256";
+
+nsresult DtlsIdentity::ComputeFingerprint(DtlsDigest* digest) const {
+  const UniqueCERTCertificate& c = cert();
+  MOZ_ASSERT(c);
+
+  return ComputeFingerprint(c, digest);
+}
+
+nsresult DtlsIdentity::ComputeFingerprint(const UniqueCERTCertificate& cert,
+                                          DtlsDigest* digest) {
+  MOZ_ASSERT(cert);
+
+  HASH_HashType ht;
+
+  if (digest->algorithm_ == "sha-1") {
+    ht = HASH_AlgSHA1;
+  } else if (digest->algorithm_ == "sha-224") {
+    ht = HASH_AlgSHA224;
+  } else if (digest->algorithm_ == "sha-256") {
+    ht = HASH_AlgSHA256;
+  } else if (digest->algorithm_ == "sha-384") {
+    ht = HASH_AlgSHA384;
+  } else if (digest->algorithm_ == "sha-512") {
+    ht = HASH_AlgSHA512;
+  } else {
+    return NS_ERROR_INVALID_ARG;
+  }
+
+  const SECHashObject* ho = HASH_GetHashObject(ht);
+  MOZ_ASSERT(ho);
+  if (!ho) {
+    return NS_ERROR_INVALID_ARG;
+  }
+
+  MOZ_ASSERT(ho->length >= 20);  // Double check
+  digest->value_.resize(ho->length);
+
+  SECStatus rv = HASH_HashBuf(ho->type, digest->value_.data(),
+                              cert->derCert.data, cert->derCert.len);
+  if (rv != SECSuccess) {
+    return NS_ERROR_FAILURE;
+  }
+
+  return NS_OK;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/transport/dtlsidentity.h b/dom/media/webrtc/transport/dtlsidentity.h
new file mode 100644
index 0000000000..b4f7686618
--- /dev/null
+++ b/dom/media/webrtc/transport/dtlsidentity.h
@@ -0,0 +1,101 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+#ifndef dtls_identity_h__
+#define dtls_identity_h__
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "ScopedNSSTypes.h"
+#include "m_cpp_utils.h"
+#include "mozilla/RefPtr.h"
+#include "nsISupportsImpl.h"
+#include "nsTArray.h"
+#include "sslt.h"
+
+// All code in this module requires NSS to be live.
+// Callers must initialize NSS and implement the nsNSSShutdownObject
+// protocol.
+namespace mozilla {
+
+class DtlsDigest {
+ public:
+  const static size_t kMaxDtlsDigestLength = HASH_LENGTH_MAX;
+  DtlsDigest() = default;
+  explicit DtlsDigest(const std::string& algorithm) : algorithm_(algorithm) {}
+  DtlsDigest(const std::string& algorithm, const std::vector<uint8_t>& value)
+      : algorithm_(algorithm), value_(value) {
+    MOZ_ASSERT(value.size() <= kMaxDtlsDigestLength);
+  }
+  ~DtlsDigest() = default;
+
+  bool operator!=(const DtlsDigest& rhs) const { return !operator==(rhs); }
+
+  bool operator==(const DtlsDigest& rhs) const {
+    if (algorithm_ != rhs.algorithm_) {
+      return false;
+    }
+
+    return value_ == rhs.value_;
+  }
+
+  std::string algorithm_;
+  std::vector<uint8_t> value_;
+};
+
+typedef std::vector<DtlsDigest> DtlsDigestList;
+
+class DtlsIdentity final {
+ public:
+  // This constructor takes ownership of privkey and cert.
+  DtlsIdentity(UniqueSECKEYPrivateKey privkey, UniqueCERTCertificate cert,
+               SSLKEAType authType)
+      : private_key_(std::move(privkey)),
+        cert_(std::move(cert)),
+        auth_type_(authType) {}
+
+  // Allows serialization/deserialization; cannot write IPC serialization code
+  // directly for DtlsIdentity, since IPC-able types need to be constructable
+  // on the stack.
+  nsresult Serialize(nsTArray<uint8_t>* aKeyDer, nsTArray<uint8_t>* aCertDer);
+  static RefPtr<DtlsIdentity> Deserialize(const nsTArray<uint8_t>& aKeyDer,
+                                          const nsTArray<uint8_t>& aCertDer,
+                                          SSLKEAType authType);
+
+  // This is only for use in tests, or for external linkage.  It makes a (bad)
+  // instance of this class.
+  static RefPtr<DtlsIdentity> Generate();
+
+  // These don't create copies or transfer ownership. If you want these to live
+  // on, make a copy.
+  const UniqueCERTCertificate& cert() const { return cert_; }
+  const UniqueSECKEYPrivateKey& privkey() const { return private_key_; }
+  // Note: this uses SSLKEAType because that is what the libssl API requires.
+  // This is a giant confusing mess, but libssl indexes certificates based on a
+  // key exchange type, not authentication type (as you might have reasonably
+  // expected).
+  SSLKEAType auth_type() const { return auth_type_; }
+
+  nsresult ComputeFingerprint(DtlsDigest* digest) const;
+  static nsresult ComputeFingerprint(const UniqueCERTCertificate& cert,
+                                     DtlsDigest* digest);
+
+  static const std::string DEFAULT_HASH_ALGORITHM;
+  enum { HASH_ALGORITHM_MAX_LENGTH = 64 };
+
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(DtlsIdentity)
+
+ private:
+  ~DtlsIdentity() = default;
+  DISALLOW_COPY_ASSIGN(DtlsIdentity);
+
+  UniqueSECKEYPrivateKey private_key_;
+  UniqueCERTCertificate cert_;
+  SSLKEAType auth_type_;
+};
+}  // namespace mozilla
+#endif
diff --git a/dom/media/webrtc/transport/fuzztest/moz.build b/dom/media/webrtc/transport/fuzztest/moz.build
new file mode 100644
index 0000000000..f22a5a702b
--- /dev/null
+++ b/dom/media/webrtc/transport/fuzztest/moz.build
@@ -0,0 +1,31 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+Library("FuzzingStun")
+
+DEFINES["HAVE_STRDUP"] = True
+
+LOCAL_INCLUDES += [
+    "/dom/media/webrtc/transport/third_party/nICEr/src/net",
+    "/dom/media/webrtc/transport/third_party/nICEr/src/stun",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/event",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/log",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/plugin",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/port/darwin/include",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/share",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/stats",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr",
+    "/ipc/chromium/src",
+]
+
+# Add libFuzzer configuration directives
+include("/tools/fuzzing/libfuzzer-config.mozbuild")
+
+SOURCES += [
+    "stun_parser_libfuzz.cpp",
+]
+
+FINAL_LIBRARY = "xul-gtest"
diff --git a/dom/media/webrtc/transport/fuzztest/stun_parser_libfuzz.cpp b/dom/media/webrtc/transport/fuzztest/stun_parser_libfuzz.cpp
new file mode 100644
index 0000000000..73bae5024c
--- /dev/null
+++ b/dom/media/webrtc/transport/fuzztest/stun_parser_libfuzz.cpp
@@ -0,0 +1,35 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include <string>
+
+#include "gtest/gtest.h"
+
+#include "FuzzingInterface.h"
+
+extern "C" {
+#include <csi_platform.h>
+#include "stun_msg.h"
+#include "stun_codec.h"
+}
+
+int FuzzingInitStunParser(int* argc, char*** argv) { return 0; }
+
+static int RunStunParserFuzzing(const uint8_t* data, size_t size) {
+  nr_stun_message* req = 0;
+
+  UCHAR* mes = (UCHAR*)data;
+
+  if (!nr_stun_message_create2(&req, mes, size)) {
+    nr_stun_decode_message(req, nullptr, nullptr);
+    nr_stun_message_destroy(&req);
+  }
+
+  return 0;
+}
+
+MOZ_FUZZING_INTERFACE_RAW(FuzzingInitStunParser, RunStunParserFuzzing,
+                          StunParser);
diff --git a/dom/media/webrtc/transport/ipc/NrIceStunAddrMessageUtils.h b/dom/media/webrtc/transport/ipc/NrIceStunAddrMessageUtils.h
new file mode 100644
index 0000000000..34ec46726e
--- /dev/null
+++ b/dom/media/webrtc/transport/ipc/NrIceStunAddrMessageUtils.h
@@ -0,0 +1,54 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef mozilla_net_NrIceStunAddrMessageUtils_h
+#define mozilla_net_NrIceStunAddrMessageUtils_h
+
+// forward declare NrIceStunAddr for --disable-webrtc builds where
+// the header will not be available.
+namespace mozilla {
+class NrIceStunAddr;
+}  // namespace mozilla
+
+#include "ipc/IPCMessageUtils.h"
+#ifdef MOZ_WEBRTC
+#  include "transport/nricestunaddr.h"
+#endif
+
+namespace IPC {
+
+template <>
+struct ParamTraits<mozilla::NrIceStunAddr> {
+  static void Write(MessageWriter* aWriter,
+                    const mozilla::NrIceStunAddr& aParam) {
+#ifdef MOZ_WEBRTC
+    const size_t bufSize = aParam.SerializationBufferSize();
+    char* buffer = new char[bufSize];
+    aParam.Serialize(buffer, bufSize);
+    aWriter->WriteBytes((void*)buffer, bufSize);
+    delete[] buffer;
+#endif
+  }
+
+  static bool Read(MessageReader* aReader, mozilla::NrIceStunAddr* aResult) {
+#ifdef MOZ_WEBRTC
+    const size_t bufSize = aResult->SerializationBufferSize();
+    char* buffer = new char[bufSize];
+    bool result = aReader->ReadBytesInto((void*)buffer, bufSize);
+
+    if (result) {
+      result = result && (NS_OK == aResult->Deserialize(buffer, bufSize));
+    }
+    delete[] buffer;
+
+    return result;
+#else
+    return false;
+#endif
+  }
+};
+
+}  // namespace IPC
+
+#endif  // mozilla_net_NrIceStunAddrMessageUtils_h
diff --git a/dom/media/webrtc/transport/ipc/PStunAddrsParams.h b/dom/media/webrtc/transport/ipc/PStunAddrsParams.h
new file mode 100644
index 0000000000..315925609d
--- /dev/null
+++ b/dom/media/webrtc/transport/ipc/PStunAddrsParams.h
@@ -0,0 +1,33 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef PStunAddrsParams_h
+#define PStunAddrsParams_h
+
+#include "mozilla/Maybe.h"
+#include "nsTArray.h"
+
+#ifdef MOZ_WEBRTC
+#  include "transport/nricestunaddr.h"
+#endif
+
+namespace mozilla {
+namespace net {
+
+// Need to define typedef in .h file--can't seem to in ipdl.h file?
+#ifdef MOZ_WEBRTC
+typedef nsTArray<NrIceStunAddr> NrIceStunAddrArray;
+#else
+// a "dummy" typedef for --disabled-webrtc builds when the definition
+// for NrIceStunAddr is not available (otherwise we get complaints
+// about missing definitions for contructor and destructor)
+typedef nsTArray<int> NrIceStunAddrArray;
+#endif
+
+typedef Maybe<nsCString> MaybeNsCString;
+
+}  // namespace net
+}  // namespace mozilla
+
+#endif  // PStunAddrsParams_h
diff --git a/dom/media/webrtc/transport/ipc/PStunAddrsRequest.ipdl b/dom/media/webrtc/transport/ipc/PStunAddrsRequest.ipdl
new file mode 100644
index 0000000000..c775f73f99
--- /dev/null
+++ b/dom/media/webrtc/transport/ipc/PStunAddrsRequest.ipdl
@@ -0,0 +1,35 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+include protocol PNecko;
+
+using mozilla::net::NrIceStunAddrArray from "mozilla/net/PStunAddrsParams.h";
+using mozilla::net::MaybeNsCString from "mozilla/net/PStunAddrsParams.h";
+
+include "mozilla/net/NrIceStunAddrMessageUtils.h";
+
+namespace mozilla {
+namespace net {
+
+[ManualDealloc, ChildImpl=virtual, ParentImpl=virtual]
+async protocol PStunAddrsRequest
+{
+  manager PNecko;
+
+parent:
+  async GetStunAddrs();
+
+  async RegisterMDNSHostname(nsCString hostname, nsCString address);
+  async QueryMDNSHostname(nsCString hostname);
+  async UnregisterMDNSHostname(nsCString hostname);
+
+  async __delete__();
+
+child:
+  async OnMDNSQueryComplete(nsCString hostname, MaybeNsCString address);
+  async OnStunAddrsAvailable(NrIceStunAddrArray iceStunAddrs);
+};
+
+} // namespace net
+} // namespace mozilla
diff --git a/dom/media/webrtc/transport/ipc/PWebrtcTCPSocket.ipdl b/dom/media/webrtc/transport/ipc/PWebrtcTCPSocket.ipdl
new file mode 100644
index 0000000000..3bc2a89828
--- /dev/null
+++ b/dom/media/webrtc/transport/ipc/PWebrtcTCPSocket.ipdl
@@ -0,0 +1,42 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set sw=2 ts=8 et tw=80 ft=cpp : */
+
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+include protocol PNecko;
+include protocol PSocketProcess;
+
+include WebrtcProxyConfig;
+
+using mozilla::dom::TabId from "mozilla/dom/ipc/IdType.h";
+
+namespace mozilla {
+namespace net {
+
+[ManualDealloc, ChildImpl=virtual, ParentImpl=virtual]
+async protocol PWebrtcTCPSocket
+{
+  manager PNecko or PSocketProcess;
+
+parent:
+  async AsyncOpen(nsCString aHost,
+                  int32_t aPort,
+                  nsCString aLocalAddress,
+                  int32_t aLocalPort,
+                  bool aUseTls,
+                  WebrtcProxyConfig? aProxyConfig);
+  async Write(uint8_t[] aWriteData);
+  async Close();
+
+child:
+  async OnClose(nsresult aReason);
+  async OnConnected(nsCString aProxyType);
+  async OnRead(uint8_t[] aReadData);
+
+  async __delete__();
+};
+
+} // namespace net
+} // namespace mozilla
diff --git a/dom/media/webrtc/transport/ipc/StunAddrsRequestChild.cpp b/dom/media/webrtc/transport/ipc/StunAddrsRequestChild.cpp
new file mode 100644
index 0000000000..057fcbd330
--- /dev/null
+++ b/dom/media/webrtc/transport/ipc/StunAddrsRequestChild.cpp
@@ -0,0 +1,45 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "StunAddrsRequestChild.h"
+
+#include "mozilla/net/NeckoChild.h"
+#include "nsISerialEventTarget.h"
+
+using namespace mozilla::ipc;
+
+namespace mozilla::net {
+
+StunAddrsRequestChild::StunAddrsRequestChild(StunAddrsListener* listener)
+    : mListener(listener) {
+  gNeckoChild->SendPStunAddrsRequestConstructor(this);
+  // IPDL holds a reference until IPDL channel gets destroyed
+  AddIPDLReference();
+}
+
+mozilla::ipc::IPCResult StunAddrsRequestChild::RecvOnMDNSQueryComplete(
+    const nsACString& hostname, const Maybe<nsCString>& address) {
+  if (mListener) {
+    mListener->OnMDNSQueryComplete(PromiseFlatCString(hostname), address);
+  }
+  return IPC_OK();
+}
+
+mozilla::ipc::IPCResult StunAddrsRequestChild::RecvOnStunAddrsAvailable(
+    const NrIceStunAddrArray& addrs) {
+  if (mListener) {
+    mListener->OnStunAddrsAvailable(addrs);
+  }
+  return IPC_OK();
+}
+
+void StunAddrsRequestChild::Cancel() { mListener = nullptr; }
+
+NS_IMPL_ADDREF(StunAddrsRequestChild)
+NS_IMPL_RELEASE(StunAddrsRequestChild)
+
+NS_IMPL_ADDREF(StunAddrsListener)
+NS_IMPL_RELEASE(StunAddrsListener)
+
+}  // namespace mozilla::net
diff --git a/dom/media/webrtc/transport/ipc/StunAddrsRequestChild.h b/dom/media/webrtc/transport/ipc/StunAddrsRequestChild.h
new file mode 100644
index 0000000000..f487f52baf
--- /dev/null
+++ b/dom/media/webrtc/transport/ipc/StunAddrsRequestChild.h
@@ -0,0 +1,64 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef mozilla_net_StunAddrsRequestChild_h
+#define mozilla_net_StunAddrsRequestChild_h
+
+#include "mozilla/net/PStunAddrsRequestChild.h"
+
+class nsISerialEventTarget;
+
+namespace mozilla::net {
+
+class StunAddrsListener {
+ public:
+  virtual void OnMDNSQueryComplete(const nsCString& hostname,
+                                   const Maybe<nsCString>& address) = 0;
+  virtual void OnStunAddrsAvailable(const NrIceStunAddrArray& addrs) = 0;
+
+  NS_IMETHOD_(MozExternalRefCountType) AddRef();
+  NS_IMETHOD_(MozExternalRefCountType) Release();
+
+ protected:
+  virtual ~StunAddrsListener() = default;
+
+  ThreadSafeAutoRefCnt mRefCnt;
+  NS_DECL_OWNINGTHREAD
+};
+
+class StunAddrsRequestChild final : public PStunAddrsRequestChild {
+  friend class PStunAddrsRequestChild;
+
+ public:
+  explicit StunAddrsRequestChild(StunAddrsListener* listener);
+
+  NS_IMETHOD_(MozExternalRefCountType) AddRef();
+  NS_IMETHOD_(MozExternalRefCountType) Release();
+
+  // Not sure why AddIPDLReference & ReleaseIPDLReference don't come
+  // from PStunAddrsRequestChild since the IPC plumbing seem to
+  // expect this.
+  void AddIPDLReference() { AddRef(); }
+  void ReleaseIPDLReference() { Release(); }
+
+  void Cancel();
+
+ protected:
+  virtual ~StunAddrsRequestChild() = default;
+
+  virtual mozilla::ipc::IPCResult RecvOnMDNSQueryComplete(
+      const nsACString& aHostname, const Maybe<nsCString>& aAddress) override;
+
+  virtual mozilla::ipc::IPCResult RecvOnStunAddrsAvailable(
+      const NrIceStunAddrArray& addrs) override;
+
+  RefPtr<StunAddrsListener> mListener;
+
+  ThreadSafeAutoRefCnt mRefCnt;
+  NS_DECL_OWNINGTHREAD
+};
+
+}  // namespace mozilla::net
+
+#endif  // mozilla_net_StunAddrsRequestChild_h
diff --git a/dom/media/webrtc/transport/ipc/StunAddrsRequestParent.cpp b/dom/media/webrtc/transport/ipc/StunAddrsRequestParent.cpp
new file mode 100644
index 0000000000..23ef6dea73
--- /dev/null
+++ b/dom/media/webrtc/transport/ipc/StunAddrsRequestParent.cpp
@@ -0,0 +1,262 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "StunAddrsRequestParent.h"
+
+#include "../runnable_utils.h"
+#include "mozilla/StaticPtr.h"
+#include "nsIThread.h"
+#include "nsNetUtil.h"
+
+#include "transport/nricectx.h"
+#include "transport/nricemediastream.h"  // needed only for including nricectx.h
+#include "transport/nricestunaddr.h"
+
+#include "../mdns_service/mdns_service.h"
+
+extern "C" {
+#include "local_addr.h"
+}
+
+using namespace mozilla::ipc;
+
+namespace mozilla::net {
+
+static void mdns_service_resolved(void* cb, const char* hostname,
+                                  const char* addr) {
+  StunAddrsRequestParent* self = static_cast<StunAddrsRequestParent*>(cb);
+  self->OnQueryComplete(nsCString(hostname), Some(nsCString(addr)));
+}
+
+void mdns_service_timedout(void* cb, const char* hostname) {
+  StunAddrsRequestParent* self = static_cast<StunAddrsRequestParent*>(cb);
+  self->OnQueryComplete(nsCString(hostname), Nothing());
+}
+
+StunAddrsRequestParent::StunAddrsRequestParent() : mIPCClosed(false) {
+  NS_GetMainThread(getter_AddRefs(mMainThread));
+
+  nsresult res;
+  mSTSThread = do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &res);
+  MOZ_ASSERT(mSTSThread);
+}
+
+StunAddrsRequestParent::~StunAddrsRequestParent() {
+  ASSERT_ON_THREAD(mMainThread);
+}
+
+mozilla::ipc::IPCResult StunAddrsRequestParent::RecvGetStunAddrs() {
+  ASSERT_ON_THREAD(mMainThread);
+
+  if (mIPCClosed) {
+    return IPC_OK();
+  }
+
+  RUN_ON_THREAD(mSTSThread,
+                WrapRunnable(RefPtr<StunAddrsRequestParent>(this),
+                             &StunAddrsRequestParent::GetStunAddrs_s),
+                NS_DISPATCH_NORMAL);
+
+  return IPC_OK();
+}
+
+mozilla::ipc::IPCResult StunAddrsRequestParent::RecvRegisterMDNSHostname(
+    const nsACString& aHostname, const nsACString& aAddress) {
+  ASSERT_ON_THREAD(mMainThread);
+
+  if (mIPCClosed) {
+    return IPC_OK();
+  }
+
+  if (mSharedMDNSService) {
+    mSharedMDNSService->RegisterHostname(aHostname.BeginReading(),
+                                         aAddress.BeginReading());
+  }
+
+  return IPC_OK();
+}
+
+mozilla::ipc::IPCResult StunAddrsRequestParent::RecvQueryMDNSHostname(
+    const nsACString& aHostname) {
+  ASSERT_ON_THREAD(mMainThread);
+
+  if (mIPCClosed) {
+    return IPC_OK();
+  }
+
+  if (mSharedMDNSService) {
+    mSharedMDNSService->QueryHostname(this, aHostname.BeginReading());
+  }
+
+  return IPC_OK();
+}
+
+mozilla::ipc::IPCResult StunAddrsRequestParent::RecvUnregisterMDNSHostname(
+    const nsACString& aHostname) {
+  ASSERT_ON_THREAD(mMainThread);
+
+  if (mIPCClosed) {
+    return IPC_OK();
+  }
+
+  if (mSharedMDNSService) {
+    mSharedMDNSService->UnregisterHostname(aHostname.BeginReading());
+  }
+
+  return IPC_OK();
+}
+
+mozilla::ipc::IPCResult StunAddrsRequestParent::Recv__delete__() {
+  // see note below in ActorDestroy
+  mIPCClosed = true;
+  return IPC_OK();
+}
+
+void StunAddrsRequestParent::OnQueryComplete(const nsACString& hostname,
+                                             const Maybe<nsCString>& address) {
+  RUN_ON_THREAD(mMainThread,
+                WrapRunnable(RefPtr<StunAddrsRequestParent>(this),
+                             &StunAddrsRequestParent::OnQueryComplete_m,
+                             nsCString(hostname), address),
+                NS_DISPATCH_NORMAL);
+}
+
+void StunAddrsRequestParent::ActorDestroy(ActorDestroyReason why) {
+  // We may still have refcount>0 if we haven't made it through
+  // GetStunAddrs_s and SendStunAddrs_m yet, but child process
+  // has crashed.  We must not send any more msgs to child, or
+  // IPDL will kill chrome process, too.
+  mIPCClosed = true;
+
+  // We need to stop the mDNS service here to ensure that we don't
+  // end up with any messages queued for the main thread after the
+  // destructors run. Because of Bug 1569311, all of the
+  // StunAddrsRequestParent instances end up being destroyed one
+  // after the other, so it is ok to free the shared service when
+  // the first one is destroyed rather than waiting for the last one.
+  // If this behaviour changes, we would potentially end up starting
+  // and stopping instances repeatedly and should add a refcount and
+  // a way of cancelling pending queries to avoid churn in that case.
+  if (mSharedMDNSService) {
+    mSharedMDNSService = nullptr;
+  }
+}
+
+void StunAddrsRequestParent::GetStunAddrs_s() {
+  ASSERT_ON_THREAD(mSTSThread);
+
+  // get the stun addresses while on STS thread
+  NrIceStunAddrArray addrs = NrIceCtx::GetStunAddrs();
+
+  if (mIPCClosed) {
+    return;
+  }
+
+  // in order to return the result over IPC, we need to be on main thread
+  RUN_ON_THREAD(
+      mMainThread,
+      WrapRunnable(RefPtr<StunAddrsRequestParent>(this),
+                   &StunAddrsRequestParent::SendStunAddrs_m, std::move(addrs)),
+      NS_DISPATCH_NORMAL);
+}
+
+void StunAddrsRequestParent::SendStunAddrs_m(const NrIceStunAddrArray& addrs) {
+  ASSERT_ON_THREAD(mMainThread);
+
+  if (mIPCClosed) {
+    // nothing to do: child probably crashed
+    return;
+  }
+
+  // This means that the mDNS service will continue running until shutdown
+  // once started. The StunAddrsRequestParent destructor does not run until
+  // shutdown anyway (see Bug 1569311), so there is not much we can do about
+  // this here. One option would be to add a check if there are no hostnames
+  // registered after UnregisterHostname is called, and if so, stop the mDNS
+  // service at that time (see Bug 1569955.)
+  if (!mSharedMDNSService) {
+    std::ostringstream o;
+    char buffer[16];
+    for (auto& addr : addrs) {
+      if (addr.localAddr().addr.ip_version == NR_IPV4 &&
+          !nr_transport_addr_is_loopback(&addr.localAddr().addr)) {
+        nr_transport_addr_get_addrstring(&addr.localAddr().addr, buffer, 16);
+        o << buffer << ";";
+      }
+    }
+    std::string addrstring = o.str();
+    if (!addrstring.empty()) {
+      mSharedMDNSService = new MDNSServiceWrapper(addrstring);
+    }
+  }
+
+  // send the new addresses back to the child
+  Unused << SendOnStunAddrsAvailable(addrs);
+}
+
+void StunAddrsRequestParent::OnQueryComplete_m(
+    const nsACString& hostname, const Maybe<nsCString>& address) {
+  ASSERT_ON_THREAD(mMainThread);
+
+  if (mIPCClosed) {
+    // nothing to do: child probably crashed
+    return;
+  }
+
+  // send the hostname and address back to the child
+  Unused << SendOnMDNSQueryComplete(hostname, address);
+}
+
+StaticRefPtr<StunAddrsRequestParent::MDNSServiceWrapper>
+    StunAddrsRequestParent::mSharedMDNSService;
+
+NS_IMPL_ADDREF(StunAddrsRequestParent)
+NS_IMPL_RELEASE(StunAddrsRequestParent)
+
+StunAddrsRequestParent::MDNSServiceWrapper::MDNSServiceWrapper(
+    const std::string& ifaddr)
+    : ifaddr(ifaddr) {}
+
+void StunAddrsRequestParent::MDNSServiceWrapper::RegisterHostname(
+    const char* hostname, const char* address) {
+  StartIfRequired();
+  if (mMDNSService) {
+    mdns_service_register_hostname(mMDNSService, hostname, address);
+  }
+}
+
+void StunAddrsRequestParent::MDNSServiceWrapper::QueryHostname(
+    void* data, const char* hostname) {
+  StartIfRequired();
+  if (mMDNSService) {
+    mdns_service_query_hostname(mMDNSService, data, mdns_service_resolved,
+                                mdns_service_timedout, hostname);
+  }
+}
+
+void StunAddrsRequestParent::MDNSServiceWrapper::UnregisterHostname(
+    const char* hostname) {
+  StartIfRequired();
+  if (mMDNSService) {
+    mdns_service_unregister_hostname(mMDNSService, hostname);
+  }
+}
+
+StunAddrsRequestParent::MDNSServiceWrapper::~MDNSServiceWrapper() {
+  if (mMDNSService) {
+    mdns_service_stop(mMDNSService);
+    mMDNSService = nullptr;
+  }
+}
+
+void StunAddrsRequestParent::MDNSServiceWrapper::StartIfRequired() {
+  if (!mMDNSService) {
+    mMDNSService = mdns_service_start(ifaddr.c_str());
+  }
+}
+
+NS_IMPL_ADDREF(StunAddrsRequestParent::MDNSServiceWrapper)
+NS_IMPL_RELEASE(StunAddrsRequestParent::MDNSServiceWrapper)
+
+}  // namespace mozilla::net
diff --git a/dom/media/webrtc/transport/ipc/StunAddrsRequestParent.h b/dom/media/webrtc/transport/ipc/StunAddrsRequestParent.h
new file mode 100644
index 0000000000..33e71abbc7
--- /dev/null
+++ b/dom/media/webrtc/transport/ipc/StunAddrsRequestParent.h
@@ -0,0 +1,82 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef mozilla_net_StunAddrsRequestParent_h
+#define mozilla_net_StunAddrsRequestParent_h
+
+#include "mozilla/net/PStunAddrsRequestParent.h"
+
+struct MDNSService;
+
+namespace mozilla::net {
+
+class StunAddrsRequestParent : public PStunAddrsRequestParent {
+  friend class PStunAddrsRequestParent;
+
+ public:
+  StunAddrsRequestParent();
+
+  NS_IMETHOD_(MozExternalRefCountType) AddRef();
+  NS_IMETHOD_(MozExternalRefCountType) Release();
+
+  mozilla::ipc::IPCResult Recv__delete__() override;
+
+  void OnQueryComplete(const nsACString& hostname,
+                       const Maybe<nsCString>& address);
+
+ protected:
+  virtual ~StunAddrsRequestParent();
+
+  virtual mozilla::ipc::IPCResult RecvGetStunAddrs() override;
+  virtual mozilla::ipc::IPCResult RecvRegisterMDNSHostname(
+      const nsACString& hostname, const nsACString& address) override;
+  virtual mozilla::ipc::IPCResult RecvQueryMDNSHostname(
+      const nsACString& hostname) override;
+  virtual mozilla::ipc::IPCResult RecvUnregisterMDNSHostname(
+      const nsACString& hostname) override;
+  virtual void ActorDestroy(ActorDestroyReason why) override;
+
+  nsCOMPtr<nsIThread> mMainThread;
+  nsCOMPtr<nsISerialEventTarget> mSTSThread;
+
+  void GetStunAddrs_s();
+  void SendStunAddrs_m(const NrIceStunAddrArray& addrs);
+
+  void OnQueryComplete_m(const nsACString& hostname,
+                         const Maybe<nsCString>& address);
+
+  ThreadSafeAutoRefCnt mRefCnt;
+  NS_DECL_OWNINGTHREAD
+
+ private:
+  bool mIPCClosed;  // true if IPDL channel has been closed (child crash)
+
+  class MDNSServiceWrapper {
+   public:
+    explicit MDNSServiceWrapper(const std::string& ifaddr);
+    void RegisterHostname(const char* hostname, const char* address);
+    void QueryHostname(void* data, const char* hostname);
+    void UnregisterHostname(const char* hostname);
+
+    NS_IMETHOD_(MozExternalRefCountType) AddRef();
+    NS_IMETHOD_(MozExternalRefCountType) Release();
+
+   protected:
+    ThreadSafeAutoRefCnt mRefCnt;
+    NS_DECL_OWNINGTHREAD
+
+   private:
+    virtual ~MDNSServiceWrapper();
+    void StartIfRequired();
+
+    std::string ifaddr;
+    MDNSService* mMDNSService = nullptr;
+  };
+
+  static StaticRefPtr<MDNSServiceWrapper> mSharedMDNSService;
+};
+
+}  // namespace mozilla::net
+
+#endif  // mozilla_net_StunAddrsRequestParent_h
diff --git a/dom/media/webrtc/transport/ipc/WebrtcProxyConfig.ipdlh b/dom/media/webrtc/transport/ipc/WebrtcProxyConfig.ipdlh
new file mode 100644
index 0000000000..e93e82a6a3
--- /dev/null
+++ b/dom/media/webrtc/transport/ipc/WebrtcProxyConfig.ipdlh
@@ -0,0 +1,23 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set sw=2 ts=8 et tw=80 ft=cpp : */
+
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+using mozilla::dom::TabId from "mozilla/dom/ipc/IdType.h";
+include NeckoChannelParams;
+
+namespace mozilla {
+namespace net {
+
+struct WebrtcProxyConfig {
+  TabId tabId;
+  nsCString alpn;
+  LoadInfoArgs loadInfoArgs;
+  bool forceProxy;
+};
+
+} // namespace net
+} // namespace mozilla
+
diff --git a/dom/media/webrtc/transport/ipc/WebrtcTCPSocket.cpp b/dom/media/webrtc/transport/ipc/WebrtcTCPSocket.cpp
new file mode 100644
index 0000000000..447b4cc741
--- /dev/null
+++ b/dom/media/webrtc/transport/ipc/WebrtcTCPSocket.cpp
@@ -0,0 +1,785 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set sw=2 ts=8 et tw=80 ft=cpp : */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "WebrtcTCPSocket.h"
+
+#include "nsHttpChannel.h"
+#include "nsIChannel.h"
+#include "nsIClassOfService.h"
+#include "nsIContentPolicy.h"
+#include "nsIIOService.h"
+#include "nsILoadInfo.h"
+#include "nsIProtocolProxyService.h"
+#include "nsIURIMutator.h"
+#include "nsICookieJarSettings.h"
+#include "nsProxyRelease.h"
+#include "nsString.h"
+#include "mozilla/dom/ContentProcessManager.h"
+#include "mozilla/dom/BrowserParent.h"
+#include "nsISocketTransportService.h"
+#include "nsICancelable.h"
+#include "nsSocketTransportService2.h"
+
+#include "WebrtcTCPSocketCallback.h"
+#include "WebrtcTCPSocketLog.h"
+
+namespace mozilla::net {
+
+class WebrtcTCPData {
+ public:
+  explicit WebrtcTCPData(nsTArray<uint8_t>&& aData) : mData(std::move(aData)) {
+    MOZ_COUNT_CTOR(WebrtcTCPData);
+  }
+
+  MOZ_COUNTED_DTOR(WebrtcTCPData)
+
+  const nsTArray<uint8_t>& GetData() const { return mData; }
+
+ private:
+  nsTArray<uint8_t> mData;
+};
+
+NS_IMPL_ISUPPORTS(WebrtcTCPSocket, nsIAuthPromptProvider,
+                  nsIHttpUpgradeListener, nsIInputStreamCallback,
+                  nsIInterfaceRequestor, nsIOutputStreamCallback,
+                  nsIRequestObserver, nsIStreamListener,
+                  nsIProtocolProxyCallback)
+
+WebrtcTCPSocket::WebrtcTCPSocket(WebrtcTCPSocketCallback* aCallbacks)
+    : mProxyCallbacks(aCallbacks),
+      mClosed(false),
+      mOpened(false),
+      mWriteOffset(0),
+      mAuthProvider(nullptr),
+      mTransport(nullptr),
+      mSocketIn(nullptr),
+      mSocketOut(nullptr) {
+  LOG(("WebrtcTCPSocket::WebrtcTCPSocket %p\n", this));
+  mMainThread = GetMainThreadSerialEventTarget();
+  mSocketThread = do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID);
+  MOZ_RELEASE_ASSERT(mMainThread, "no main thread");
+  MOZ_RELEASE_ASSERT(mSocketThread, "no socket thread");
+}
+
+WebrtcTCPSocket::~WebrtcTCPSocket() {
+  LOG(("WebrtcTCPSocket::~WebrtcTCPSocket %p\n", this));
+
+  NS_ProxyRelease("WebrtcTCPSocket::CleanUpAuthProvider", mMainThread,
+                  mAuthProvider.forget());
+}
+
+void WebrtcTCPSocket::SetTabId(dom::TabId aTabId) {
+  MOZ_ASSERT(NS_IsMainThread());
+  dom::ContentProcessManager* cpm = dom::ContentProcessManager::GetSingleton();
+  if (cpm) {
+    dom::ContentParentId cpId = cpm->GetTabProcessId(aTabId);
+    mAuthProvider = cpm->GetBrowserParentByProcessAndTabId(cpId, aTabId);
+  }
+}
+
+nsresult WebrtcTCPSocket::Write(nsTArray<uint8_t>&& aWriteData) {
+  LOG(("WebrtcTCPSocket::Write %p\n", this));
+  MOZ_ASSERT(NS_IsMainThread());
+  nsresult rv = mSocketThread->Dispatch(NewRunnableMethod<nsTArray<uint8_t>&&>(
+      "WebrtcTCPSocket::Write", this, &WebrtcTCPSocket::EnqueueWrite_s,
+      std::move(aWriteData)));
+  NS_WARNING_ASSERTION(NS_SUCCEEDED(rv), "Failed to dispatch to STS");
+
+  return rv;
+}
+
+nsresult WebrtcTCPSocket::Close() {
+  LOG(("WebrtcTCPSocket::Close %p\n", this));
+
+  CloseWithReason(NS_OK);
+
+  return NS_OK;
+}
+
+void WebrtcTCPSocket::CloseWithReason(nsresult aReason) {
+  LOG(("WebrtcTCPSocket::CloseWithReason %p reason=%u\n", this,
+       static_cast<uint32_t>(aReason)));
+
+  if (!OnSocketThread()) {
+    MOZ_ASSERT(NS_IsMainThread(), "not on main thread");
+
+    // Let's pretend we got an open even if we didn't to prevent an Open later.
+    mOpened = true;
+
+    DebugOnly<nsresult> rv =
+        mSocketThread->Dispatch(NewRunnableMethod<nsresult>(
+            "WebrtcTCPSocket::CloseWithReason", this,
+            &WebrtcTCPSocket::CloseWithReason, aReason));
+
+    // This was MOZ_ALWAYS_SUCCEEDS, but that now uses NS_WARNING_ASSERTION.
+    // In order to convert this back to MOZ_ALWAYS_SUCCEEDS we would need
+    // OnSocketThread to return true if we're shutting down and doing the
+    // "running all of STS's queued events on main" thing.
+    NS_WARNING_ASSERTION(NS_SUCCEEDED(rv), "Failed to dispatch to STS");
+
+    return;
+  }
+
+  if (mClosed) {
+    return;
+  }
+
+  mClosed = true;
+
+  if (mSocketIn) {
+    mSocketIn->AsyncWait(nullptr, 0, 0, nullptr);
+    mSocketIn = nullptr;
+  }
+
+  if (mSocketOut) {
+    mSocketOut->AsyncWait(nullptr, 0, 0, nullptr);
+    mSocketOut = nullptr;
+  }
+
+  if (mTransport) {
+    mTransport->Close(NS_BASE_STREAM_CLOSED);
+    mTransport = nullptr;
+  }
+
+  NS_ProxyRelease("WebrtcTCPSocket::CleanUpAuthProvider", mMainThread,
+                  mAuthProvider.forget());
+  InvokeOnClose(aReason);
+}
+
+nsresult WebrtcTCPSocket::Open(
+    const nsACString& aHost, const int& aPort, const nsACString& aLocalAddress,
+    const int& aLocalPort, bool aUseTls,
+    const Maybe<net::WebrtcProxyConfig>& aProxyConfig) {
+  LOG(("WebrtcTCPSocket::Open %p remote-host=%s local-addr=%s local-port=%d",
+       this, PromiseFlatCString(aHost).get(),
+       PromiseFlatCString(aLocalAddress).get(), aLocalPort));
+  MOZ_ASSERT(NS_IsMainThread());
+
+  if (NS_WARN_IF(mOpened)) {
+    LOG(("WebrtcTCPSocket %p: TCP socket already open\n", this));
+    CloseWithReason(NS_ERROR_FAILURE);
+    return NS_ERROR_FAILURE;
+  }
+
+  mOpened = true;
+  const nsLiteralCString schemePrefix = aUseTls ? "https://"_ns : "http://"_ns;
+  nsAutoCString spec(schemePrefix);
+
+  bool ipv6Literal = aHost.Find(":") != kNotFound;
+  if (ipv6Literal) {
+    spec += "[";
+    spec += aHost;
+    spec += "]";
+  } else {
+    spec += aHost;
+  }
+
+  nsresult rv = NS_MutateURI(NS_STANDARDURLMUTATOR_CONTRACTID)
+                    .SetSpec(spec)
+                    .SetPort(aPort)
+                    .Finalize(mURI);
+
+  if (NS_WARN_IF(NS_FAILED(rv))) {
+    CloseWithReason(NS_ERROR_FAILURE);
+    return NS_ERROR_FAILURE;
+  }
+
+  mTls = aUseTls;
+  mLocalAddress = aLocalAddress;
+  mLocalPort = aLocalPort;
+  mProxyConfig = aProxyConfig;
+
+  if (!mProxyConfig.isSome()) {
+    OpenWithoutHttpProxy(nullptr);
+    return NS_OK;
+  }
+
+  // We need to figure out whether a proxy needs to be used for mURI before
+  // we can start on establishing a connection.
+  rv = DoProxyConfigLookup();
+
+  if (NS_WARN_IF(NS_FAILED(rv))) {
+    CloseWithReason(rv);
+  }
+
+  return rv;
+}
+
+nsresult WebrtcTCPSocket::DoProxyConfigLookup() {
+  MOZ_ASSERT(NS_IsMainThread());
+  nsresult rv;
+  nsCOMPtr<nsIProtocolProxyService> pps =
+      do_GetService(NS_PROTOCOLPROXYSERVICE_CONTRACTID, &rv);
+  if (NS_WARN_IF(NS_FAILED(rv))) {
+    return rv;
+  }
+
+  nsCOMPtr<nsIChannel> channel;
+  rv = NS_NewChannel(getter_AddRefs(channel), mURI,
+                     nsContentUtils::GetSystemPrincipal(),
+                     nsILoadInfo::SEC_ALLOW_CROSS_ORIGIN_SEC_CONTEXT_IS_NULL,
+                     nsIContentPolicy::TYPE_OTHER);
+  if (NS_WARN_IF(NS_FAILED(rv))) {
+    return rv;
+  }
+
+  rv = pps->AsyncResolve(channel,
+                         nsIProtocolProxyService::RESOLVE_PREFER_HTTPS_PROXY |
+                             nsIProtocolProxyService::RESOLVE_ALWAYS_TUNNEL,
+                         this, nullptr, getter_AddRefs(mProxyRequest));
+  if (NS_WARN_IF(NS_FAILED(rv))) {
+    return rv;
+  }
+
+  // We pick back up in OnProxyAvailable
+
+  return NS_OK;
+}
+
+NS_IMETHODIMP WebrtcTCPSocket::OnProxyAvailable(nsICancelable* aRequest,
+                                                nsIChannel* aChannel,
+                                                nsIProxyInfo* aProxyinfo,
+                                                nsresult aResult) {
+  MOZ_ASSERT(NS_IsMainThread());
+  mProxyRequest = nullptr;
+
+  if (NS_SUCCEEDED(aResult) && aProxyinfo) {
+    nsresult rv = aProxyinfo->GetType(mProxyType);
+    if (NS_WARN_IF(NS_FAILED(rv))) {
+      CloseWithReason(rv);
+      return rv;
+    }
+
+    if (mProxyType == "http" || mProxyType == "https") {
+      rv = OpenWithHttpProxy();
+      if (NS_WARN_IF(NS_FAILED(rv))) {
+        CloseWithReason(rv);
+      }
+      return rv;
+    }
+
+    if (mProxyType == "socks" || mProxyType == "socks4" ||
+        mProxyType == "direct") {
+      OpenWithoutHttpProxy(aProxyinfo);
+      return NS_OK;
+    }
+  }
+
+  OpenWithoutHttpProxy(nullptr);
+
+  return NS_OK;
+}
+
+void WebrtcTCPSocket::OpenWithoutHttpProxy(nsIProxyInfo* aSocksProxyInfo) {
+  if (!OnSocketThread()) {
+    DebugOnly<nsresult> rv =
+        mSocketThread->Dispatch(NewRunnableMethod<nsCOMPtr<nsIProxyInfo>>(
+            "WebrtcTCPSocket::OpenWithoutHttpProxy", this,
+            &WebrtcTCPSocket::OpenWithoutHttpProxy, aSocksProxyInfo));
+    NS_WARNING_ASSERTION(NS_SUCCEEDED(rv), "Failed to dispatch to STS");
+    return;
+  }
+
+  LOG(("WebrtcTCPSocket::OpenWithoutHttpProxy %p\n", this));
+
+  if (mClosed) {
+    return;
+  }
+
+  if (NS_WARN_IF(mProxyConfig.isSome() && mProxyConfig->forceProxy() &&
+                 !aSocksProxyInfo)) {
+    CloseWithReason(NS_ERROR_FAILURE);
+    return;
+  }
+
+  nsCString host;
+  int32_t port;
+
+  nsresult rv = mURI->GetHost(host);
+  if (NS_WARN_IF(NS_FAILED(rv))) {
+    CloseWithReason(rv);
+    return;
+  }
+
+  rv = mURI->GetPort(&port);
+  if (NS_WARN_IF(NS_FAILED(rv))) {
+    CloseWithReason(rv);
+    return;
+  }
+
+  AutoTArray<nsCString, 1> socketTypes;
+  if (mTls) {
+    socketTypes.AppendElement("ssl"_ns);
+  }
+
+  nsCOMPtr<nsISocketTransportService> sts =
+      do_GetService("@mozilla.org/network/socket-transport-service;1");
+  rv = sts->CreateTransport(socketTypes, host, port, aSocksProxyInfo, nullptr,
+                            getter_AddRefs(mTransport));
+  if (NS_WARN_IF(NS_FAILED(rv))) {
+    CloseWithReason(rv);
+    return;
+  }
+
+  mTransport->SetReuseAddrPort(true);
+
+  PRNetAddr prAddr;
+  if (NS_WARN_IF(PR_SUCCESS !=
+                 PR_InitializeNetAddr(PR_IpAddrAny, mLocalPort, &prAddr))) {
+    CloseWithReason(NS_ERROR_FAILURE);
+    return;
+  }
+
+  if (NS_WARN_IF(PR_SUCCESS !=
+                 PR_StringToNetAddr(mLocalAddress.BeginReading(), &prAddr))) {
+    CloseWithReason(NS_ERROR_FAILURE);
+    return;
+  }
+
+  mozilla::net::NetAddr addr(&prAddr);
+  rv = mTransport->Bind(&addr);
+  if (NS_WARN_IF(NS_FAILED(rv))) {
+    CloseWithReason(rv);
+    return;
+  }
+
+  // Binding to a V4 address is not sufficient to cause this socket to use
+  // V4, and the same goes for V6. So, we disable as needed here.
+  uint32_t flags = 0;
+  if (addr.raw.family == AF_INET) {
+    flags |= nsISocketTransport::DISABLE_IPV6;
+  } else if (addr.raw.family == AF_INET6) {
+    flags |= nsISocketTransport::DISABLE_IPV4;
+  } else {
+    MOZ_CRASH();
+  }
+
+  mTransport->SetConnectionFlags(flags);
+
+  nsCOMPtr<nsIInputStream> socketIn;
+  rv = mTransport->OpenInputStream(0, 0, 0, getter_AddRefs(socketIn));
+  if (NS_WARN_IF(NS_FAILED(rv))) {
+    CloseWithReason(rv);
+    return;
+  }
+  mSocketIn = do_QueryInterface(socketIn);
+  if (NS_WARN_IF(!mSocketIn)) {
+    CloseWithReason(NS_ERROR_NULL_POINTER);
+    return;
+  }
+
+  nsCOMPtr<nsIOutputStream> socketOut;
+  rv = mTransport->OpenOutputStream(nsITransport::OPEN_UNBUFFERED, 0, 0,
+                                    getter_AddRefs(socketOut));
+  if (NS_WARN_IF(NS_FAILED(rv))) {
+    CloseWithReason(rv);
+    return;
+  }
+  mSocketOut = do_QueryInterface(socketOut);
+  if (NS_WARN_IF(!mSocketOut)) {
+    CloseWithReason(NS_ERROR_NULL_POINTER);
+    return;
+  }
+
+  FinishOpen();
+}
+
+nsresult WebrtcTCPSocket::OpenWithHttpProxy() {
+  MOZ_ASSERT(NS_IsMainThread(), "not on main thread");
+  LOG(("WebrtcTCPSocket::OpenWithHttpProxy %p\n", this));
+  nsresult rv;
+  nsCOMPtr<nsIIOService> ioService;
+  ioService = do_GetService(NS_IOSERVICE_CONTRACTID, &rv);
+  if (NS_FAILED(rv)) {
+    LOG(("WebrtcTCPSocket %p: io service missing\n", this));
+    return rv;
+  }
+
+  nsCOMPtr<nsILoadInfo> loadInfo;
+  Maybe<net::LoadInfoArgs> loadInfoArgs = Some(mProxyConfig->loadInfoArgs());
+
+  // FIXME: We don't know the remote type of the process which provided these
+  // LoadInfoArgs. Pass in `NOT_REMOTE_TYPE` as the origin process to blindly
+  // accept whatever value was passed by the other side for now, as we aren't
+  // using it for security checks here.
+  // If this code ever starts checking the triggering remote type, this needs to
+  // be changed.
+  rv = ipc::LoadInfoArgsToLoadInfo(loadInfoArgs, NOT_REMOTE_TYPE,
+                                   getter_AddRefs(loadInfo));
+  if (NS_FAILED(rv)) {
+    LOG(("WebrtcTCPSocket %p: could not init load info\n", this));
+    return rv;
+  }
+
+  // -need to always tunnel since we're using a proxy
+  // -there shouldn't be an opportunity to send cookies, but explicitly disallow
+  // them anyway.
+  // -the previous proxy tunnel didn't support redirects e.g. 307. don't need to
+  // introduce new behavior. can't follow redirects on connect anyway.
+  nsCOMPtr<nsIChannel> localChannel;
+  rv = ioService->NewChannelFromURIWithProxyFlags(
+      mURI, nullptr,
+      // Proxy flags are overridden by SetConnectOnly()
+      0, loadInfo->LoadingNode(), loadInfo->GetLoadingPrincipal(),
+      loadInfo->TriggeringPrincipal(),
+      nsILoadInfo::SEC_COOKIES_OMIT |
+          // We need this flag to allow loads from any origin since this channel
+          // is being used to CONNECT to an HTTP proxy.
+          nsILoadInfo::SEC_ALLOW_CROSS_ORIGIN_SEC_CONTEXT_IS_NULL,
+      nsIContentPolicy::TYPE_PROXIED_WEBRTC_MEDIA,
+      getter_AddRefs(localChannel));
+  if (NS_FAILED(rv)) {
+    LOG(("WebrtcTCPSocket %p: bad open channel\n", this));
+    return rv;
+  }
+
+  nsCOMPtr<nsILoadInfo> channelLoadInfo = localChannel->LoadInfo();
+  nsCOMPtr<nsICookieJarSettings> cookieJarSettings;
+  loadInfo->GetCookieJarSettings(getter_AddRefs(cookieJarSettings));
+  channelLoadInfo->SetCookieJarSettings(cookieJarSettings);
+
+  RefPtr<nsHttpChannel> httpChannel;
+  CallQueryInterface(localChannel, httpChannel.StartAssignment());
+
+  if (!httpChannel) {
+    LOG(("WebrtcTCPSocket %p: not an http channel\n", this));
+    return NS_ERROR_FAILURE;
+  }
+
+  httpChannel->SetNotificationCallbacks(this);
+
+  // don't block webrtc proxy setup with other requests
+  // often more than one of these channels will be created all at once by ICE
+  nsCOMPtr<nsIClassOfService> cos = do_QueryInterface(localChannel);
+  if (cos) {
+    cos->AddClassFlags(nsIClassOfService::Unblocked |
+                       nsIClassOfService::DontThrottle);
+  } else {
+    LOG(("WebrtcTCPSocket %p: could not set class of service\n", this));
+    return NS_ERROR_FAILURE;
+  }
+
+  rv = httpChannel->HTTPUpgrade(mProxyConfig->alpn(), this);
+  if (NS_WARN_IF(NS_FAILED(rv))) {
+    return rv;
+  }
+  rv = httpChannel->SetConnectOnly();
+  if (NS_WARN_IF(NS_FAILED(rv))) {
+    return rv;
+  }
+
+  rv = httpChannel->AsyncOpen(this);
+
+  if (NS_FAILED(rv)) {
+    LOG(("WebrtcTCPSocket %p: cannot async open\n", this));
+    return rv;
+  }
+
+  // This picks back up in OnTransportAvailable once we have connected to the
+  // proxy, and performed the http upgrade to switch the proxy into passthrough
+  // mode.
+
+  return NS_OK;
+}
+
+void WebrtcTCPSocket::EnqueueWrite_s(nsTArray<uint8_t>&& aWriteData) {
+  LOG(("WebrtcTCPSocket::EnqueueWrite %p\n", this));
+  MOZ_ASSERT(OnSocketThread(), "not on socket thread");
+
+  if (mClosed) {
+    return;
+  }
+
+  mWriteQueue.emplace_back(std::move(aWriteData));
+
+  if (mSocketOut) {
+    mSocketOut->AsyncWait(this, 0, 0, nullptr);
+  }
+}
+
+void WebrtcTCPSocket::InvokeOnClose(nsresult aReason) {
+  LOG(("WebrtcTCPSocket::InvokeOnClose %p\n", this));
+
+  if (!NS_IsMainThread()) {
+    MOZ_ALWAYS_SUCCEEDS(mMainThread->Dispatch(
+        NewRunnableMethod<nsresult>("WebrtcTCPSocket::InvokeOnClose", this,
+                                    &WebrtcTCPSocket::InvokeOnClose, aReason)));
+    return;
+  }
+
+  MOZ_ASSERT(mProxyCallbacks, "webrtc TCP callback should be non-null");
+
+  if (mProxyRequest) {
+    mProxyRequest->Cancel(aReason);
+    mProxyRequest = nullptr;
+  }
+
+  mProxyCallbacks->OnClose(aReason);
+  mProxyCallbacks = nullptr;
+}
+
+void WebrtcTCPSocket::InvokeOnConnected() {
+  LOG(("WebrtcTCPSocket::InvokeOnConnected %p\n", this));
+
+  if (!NS_IsMainThread()) {
+    MOZ_ALWAYS_SUCCEEDS(mMainThread->Dispatch(
+        NewRunnableMethod("WebrtcTCPSocket::InvokeOnConnected", this,
+                          &WebrtcTCPSocket::InvokeOnConnected)));
+    return;
+  }
+
+  MOZ_ASSERT(mProxyCallbacks, "webrtc TCP callback should be non-null");
+
+  mProxyCallbacks->OnConnected(mProxyType);
+}
+
+void WebrtcTCPSocket::InvokeOnRead(nsTArray<uint8_t>&& aReadData) {
+  LOG(("WebrtcTCPSocket::InvokeOnRead %p count=%zu\n", this,
+       aReadData.Length()));
+
+  if (!NS_IsMainThread()) {
+    MOZ_ALWAYS_SUCCEEDS(
+        mMainThread->Dispatch(NewRunnableMethod<nsTArray<uint8_t>&&>(
+            "WebrtcTCPSocket::InvokeOnRead", this,
+            &WebrtcTCPSocket::InvokeOnRead, std::move(aReadData))));
+    return;
+  }
+
+  MOZ_ASSERT(mProxyCallbacks, "webrtc TCP callback should be non-null");
+
+  mProxyCallbacks->OnRead(std::move(aReadData));
+}
+
+// nsIHttpUpgradeListener
+NS_IMETHODIMP
+WebrtcTCPSocket::OnTransportAvailable(nsISocketTransport* aTransport,
+                                      nsIAsyncInputStream* aSocketIn,
+                                      nsIAsyncOutputStream* aSocketOut) {
+  // This is called only in the http proxy case, once we have connected to the
+  // http proxy and performed the http upgrade to switch it over to passthrough
+  // mode. That process is started async by OpenWithHttpProxy.
+  LOG(("WebrtcTCPSocket::OnTransportAvailable %p\n", this));
+  MOZ_ASSERT(OnSocketThread(), "not on socket thread");
+  MOZ_ASSERT(!mTransport,
+             "already called transport available on webrtc TCP socket");
+
+  // Cancel any pending callbacks. The caller doesn't always cancel these
+  // awaits. We need to make sure they don't get them.
+  aSocketIn->AsyncWait(nullptr, 0, 0, nullptr);
+  aSocketOut->AsyncWait(nullptr, 0, 0, nullptr);
+
+  if (mClosed) {
+    LOG(("WebrtcTCPSocket::OnTransportAvailable %p closed\n", this));
+    return NS_OK;
+  }
+
+  mTransport = aTransport;
+  mSocketIn = aSocketIn;
+  mSocketOut = aSocketOut;
+
+  // pulled from nr_socket_prsock.cpp
+  uint32_t minBufferSize = 256 * 1024;
+  nsresult rv = mTransport->SetSendBufferSize(minBufferSize);
+  if (NS_FAILED(rv)) {
+    LOG(("WebrtcProxyChannel::OnTransportAvailable %p send failed\n", this));
+    CloseWithReason(rv);
+    return rv;
+  }
+  rv = mTransport->SetRecvBufferSize(minBufferSize);
+  if (NS_FAILED(rv)) {
+    LOG(("WebrtcProxyChannel::OnTransportAvailable %p recv failed\n", this));
+    CloseWithReason(rv);
+    return rv;
+  }
+
+  FinishOpen();
+  return NS_OK;
+}
+
+void WebrtcTCPSocket::FinishOpen() {
+  MOZ_ASSERT(OnSocketThread());
+  // mTransport, mSocketIn, and mSocketOut are all set. We may have set them in
+  // OnTransportAvailable (in the http/https proxy case), or in
+  // OpenWithoutHttpProxy. From here on out, this class functions the same for
+  // these two cases.
+
+  mSocketIn->AsyncWait(this, 0, 0, nullptr);
+
+  InvokeOnConnected();
+}
+
+NS_IMETHODIMP
+WebrtcTCPSocket::OnUpgradeFailed(nsresult aErrorCode) {
+  LOG(("WebrtcTCPSocket::OnUpgradeFailed %p\n", this));
+  MOZ_ASSERT(OnSocketThread(), "not on socket thread");
+  MOZ_ASSERT(!mTransport,
+             "already called transport available on webrtc TCP socket");
+
+  if (mClosed) {
+    LOG(("WebrtcTCPSocket::OnUpgradeFailed %p closed\n", this));
+    return NS_OK;
+  }
+
+  CloseWithReason(aErrorCode);
+
+  return NS_OK;
+}
+
+NS_IMETHODIMP
+WebrtcTCPSocket::OnWebSocketConnectionAvailable(
+    WebSocketConnectionBase* aConnection) {
+  return NS_ERROR_NOT_IMPLEMENTED;
+}
+
+// nsIRequestObserver (from nsIStreamListener)
+NS_IMETHODIMP
+WebrtcTCPSocket::OnStartRequest(nsIRequest* aRequest) {
+  LOG(("WebrtcTCPSocket::OnStartRequest %p\n", this));
+
+  return NS_OK;
+}
+
+NS_IMETHODIMP
+WebrtcTCPSocket::OnStopRequest(nsIRequest* aRequest, nsresult aStatusCode) {
+  LOG(("WebrtcTCPSocket::OnStopRequest %p status=%u\n", this,
+       static_cast<uint32_t>(aStatusCode)));
+
+  // see nsHttpChannel::ProcessFailedProxyConnect for most error codes
+  if (NS_FAILED(aStatusCode)) {
+    CloseWithReason(aStatusCode);
+    return aStatusCode;
+  }
+
+  return NS_OK;
+}
+
+// nsIStreamListener
+NS_IMETHODIMP
+WebrtcTCPSocket::OnDataAvailable(nsIRequest* aRequest,
+                                 nsIInputStream* aInputStream, uint64_t aOffset,
+                                 uint32_t aCount) {
+  LOG(("WebrtcTCPSocket::OnDataAvailable %p count=%u\n", this, aCount));
+  MOZ_ASSERT(0, "unreachable data available");
+  return NS_OK;
+}
+
+// nsIInputStreamCallback
+NS_IMETHODIMP
+WebrtcTCPSocket::OnInputStreamReady(nsIAsyncInputStream* in) {
+  LOG(("WebrtcTCPSocket::OnInputStreamReady %p unwritten=%zu\n", this,
+       CountUnwrittenBytes()));
+  MOZ_ASSERT(OnSocketThread(), "not on socket thread");
+  MOZ_ASSERT(!mClosed, "webrtc TCP socket closed");
+  MOZ_ASSERT(mTransport, "webrtc TCP socket not connected");
+  MOZ_ASSERT(mSocketIn == in, "wrong input stream");
+
+  char buffer[9216];
+  uint32_t remainingCapacity = sizeof(buffer);
+  uint32_t read = 0;
+
+  while (remainingCapacity > 0) {
+    uint32_t count = 0;
+    nsresult rv = mSocketIn->Read(buffer + read, remainingCapacity, &count);
+    if (rv == NS_BASE_STREAM_WOULD_BLOCK) {
+      break;
+    }
+
+    if (NS_FAILED(rv)) {
+      LOG(("WebrtcTCPSocket::OnInputStreamReady %p failed %u\n", this,
+           static_cast<uint32_t>(rv)));
+      CloseWithReason(rv);
+      return rv;
+    }
+
+    // base stream closed
+    if (count == 0) {
+      LOG(("WebrtcTCPSocket::OnInputStreamReady %p connection closed\n", this));
+      CloseWithReason(NS_ERROR_FAILURE);
+      return NS_OK;
+    }
+
+    remainingCapacity -= count;
+    read += count;
+  }
+
+  if (read > 0) {
+    nsTArray<uint8_t> array(read);
+    array.AppendElements(buffer, read);
+
+    InvokeOnRead(std::move(array));
+  }
+
+  mSocketIn->AsyncWait(this, 0, 0, nullptr);
+
+  return NS_OK;
+}
+
+// nsIOutputStreamCallback
+NS_IMETHODIMP
+WebrtcTCPSocket::OnOutputStreamReady(nsIAsyncOutputStream* out) {
+  LOG(("WebrtcTCPSocket::OnOutputStreamReady %p unwritten=%zu\n", this,
+       CountUnwrittenBytes()));
+  MOZ_ASSERT(OnSocketThread(), "not on socket thread");
+  MOZ_ASSERT(!mClosed, "webrtc TCP socket closed");
+  MOZ_ASSERT(mTransport, "webrtc TCP socket not connected");
+  MOZ_ASSERT(mSocketOut == out, "wrong output stream");
+
+  while (!mWriteQueue.empty()) {
+    const WebrtcTCPData& data = mWriteQueue.front();
+
+    char* buffer = reinterpret_cast<char*>(
+                       const_cast<uint8_t*>(data.GetData().Elements())) +
+                   mWriteOffset;
+    uint32_t toWrite = data.GetData().Length() - mWriteOffset;
+
+    uint32_t wrote = 0;
+    nsresult rv = mSocketOut->Write(buffer, toWrite, &wrote);
+    if (rv == NS_BASE_STREAM_WOULD_BLOCK) {
+      mSocketOut->AsyncWait(this, 0, 0, nullptr);
+      return NS_OK;
+    }
+
+    if (NS_FAILED(rv)) {
+      LOG(("WebrtcTCPSocket::OnOutputStreamReady %p failed %u\n", this,
+           static_cast<uint32_t>(rv)));
+      CloseWithReason(rv);
+      return NS_OK;
+    }
+
+    mWriteOffset += wrote;
+
+    if (toWrite == wrote) {
+      mWriteOffset = 0;
+      mWriteQueue.pop_front();
+    }
+  }
+
+  return NS_OK;
+}
+
+// nsIInterfaceRequestor
+NS_IMETHODIMP
+WebrtcTCPSocket::GetInterface(const nsIID& iid, void** result) {
+  LOG(("WebrtcTCPSocket::GetInterface %p\n", this));
+
+  return QueryInterface(iid, result);
+}
+
+size_t WebrtcTCPSocket::CountUnwrittenBytes() const {
+  size_t count = 0;
+
+  for (const WebrtcTCPData& data : mWriteQueue) {
+    count += data.GetData().Length();
+  }
+
+  MOZ_ASSERT(count >= mWriteOffset, "offset exceeds write buffer length");
+
+  count -= mWriteOffset;
+
+  return count;
+}
+
+}  // namespace mozilla::net
diff --git a/dom/media/webrtc/transport/ipc/WebrtcTCPSocket.h b/dom/media/webrtc/transport/ipc/WebrtcTCPSocket.h
new file mode 100644
index 0000000000..632ba47d32
--- /dev/null
+++ b/dom/media/webrtc/transport/ipc/WebrtcTCPSocket.h
@@ -0,0 +1,104 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set sw=2 ts=8 et tw=80 ft=cpp : */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef webrtc_tcp_socket_h__
+#define webrtc_tcp_socket_h__
+
+#include <list>
+
+#include "nsCOMPtr.h"
+#include "nsIAsyncInputStream.h"
+#include "nsIAsyncOutputStream.h"
+#include "nsIAuthPromptProvider.h"
+#include "nsIHttpChannelInternal.h"
+#include "nsIInterfaceRequestor.h"
+#include "nsIStreamListener.h"
+#include "nsStringFwd.h"
+#include "nsTArray.h"
+#include "nsIProtocolProxyCallback.h"
+#include "mozilla/net/WebrtcProxyConfig.h"
+
+class nsISocketTransport;
+
+namespace mozilla::net {
+
+class WebrtcTCPSocketCallback;
+class WebrtcTCPData;
+
+class WebrtcTCPSocket : public nsIHttpUpgradeListener,
+                        public nsIStreamListener,
+                        public nsIInputStreamCallback,
+                        public nsIOutputStreamCallback,
+                        public nsIInterfaceRequestor,
+                        public nsIAuthPromptProvider,
+                        public nsIProtocolProxyCallback {
+ public:
+  NS_DECL_NSIHTTPUPGRADELISTENER
+  NS_DECL_NSIINPUTSTREAMCALLBACK
+  NS_DECL_NSIINTERFACEREQUESTOR
+  NS_DECL_NSIOUTPUTSTREAMCALLBACK
+  NS_DECL_NSIREQUESTOBSERVER
+  NS_DECL_NSISTREAMLISTENER
+  NS_DECL_THREADSAFE_ISUPPORTS
+  NS_FORWARD_SAFE_NSIAUTHPROMPTPROVIDER(mAuthProvider)
+  NS_DECL_NSIPROTOCOLPROXYCALLBACK
+
+  explicit WebrtcTCPSocket(WebrtcTCPSocketCallback* aCallbacks);
+
+  void SetTabId(dom::TabId aTabId);
+  nsresult Open(const nsACString& aHost, const int& aPort,
+                const nsACString& aLocalAddress, const int& aLocalPort,
+                bool aUseTls,
+                const Maybe<net::WebrtcProxyConfig>& aProxyConfig);
+  nsresult Write(nsTArray<uint8_t>&& aBytes);
+  nsresult Close();
+
+  size_t CountUnwrittenBytes() const;
+
+ protected:
+  virtual ~WebrtcTCPSocket();
+
+  // protected for gtests
+  virtual void InvokeOnClose(nsresult aReason);
+  virtual void InvokeOnConnected();
+  virtual void InvokeOnRead(nsTArray<uint8_t>&& aReadData);
+
+  RefPtr<WebrtcTCPSocketCallback> mProxyCallbacks;
+
+ private:
+  bool mClosed;
+  bool mOpened;
+  nsCOMPtr<nsIURI> mURI;
+  bool mTls = false;
+  Maybe<WebrtcProxyConfig> mProxyConfig;
+  nsCString mLocalAddress;
+  uint16_t mLocalPort = 0;
+  nsCString mProxyType;
+
+  nsresult DoProxyConfigLookup();
+  nsresult OpenWithHttpProxy();
+  void OpenWithoutHttpProxy(nsIProxyInfo* aSocksProxyInfo);
+  void FinishOpen();
+  void EnqueueWrite_s(nsTArray<uint8_t>&& aWriteData);
+
+  void CloseWithReason(nsresult aReason);
+
+  size_t mWriteOffset;
+  std::list<WebrtcTCPData> mWriteQueue;
+  nsCOMPtr<nsIAuthPromptProvider> mAuthProvider;
+
+  // Indicates that the channel is CONNECTed
+  nsCOMPtr<nsISocketTransport> mTransport;
+  nsCOMPtr<nsIAsyncInputStream> mSocketIn;
+  nsCOMPtr<nsIAsyncOutputStream> mSocketOut;
+  nsCOMPtr<nsIEventTarget> mMainThread;
+  nsCOMPtr<nsIEventTarget> mSocketThread;
+  nsCOMPtr<nsICancelable> mProxyRequest;
+};
+
+}  // namespace mozilla::net
+
+#endif  // webrtc_tcp_socket_h__
diff --git a/dom/media/webrtc/transport/ipc/WebrtcTCPSocketCallback.h b/dom/media/webrtc/transport/ipc/WebrtcTCPSocketCallback.h
new file mode 100644
index 0000000000..1929e55ac2
--- /dev/null
+++ b/dom/media/webrtc/transport/ipc/WebrtcTCPSocketCallback.h
@@ -0,0 +1,28 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set sw=2 ts=8 et tw=80 ft=cpp : */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef webrtc_tcp_socket_callback_h__
+#define webrtc_tcp_socket_callback_h__
+
+#include "nsTArray.h"
+
+namespace mozilla::net {
+
+class WebrtcTCPSocketCallback {
+ public:
+  NS_INLINE_DECL_PURE_VIRTUAL_REFCOUNTING
+
+  virtual void OnClose(nsresult aReason) = 0;
+  virtual void OnConnected(const nsACString& aProxyType) = 0;
+  virtual void OnRead(nsTArray<uint8_t>&& aReadData) = 0;
+
+ protected:
+  virtual ~WebrtcTCPSocketCallback() = default;
+};
+
+}  // namespace mozilla::net
+
+#endif  // webrtc_tcp_socket_callback_h__
diff --git a/dom/media/webrtc/transport/ipc/WebrtcTCPSocketChild.cpp b/dom/media/webrtc/transport/ipc/WebrtcTCPSocketChild.cpp
new file mode 100644
index 0000000000..52d1ba8ab2
--- /dev/null
+++ b/dom/media/webrtc/transport/ipc/WebrtcTCPSocketChild.cpp
@@ -0,0 +1,96 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set sw=2 ts=8 et tw=80 ft=cpp : */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "WebrtcTCPSocketChild.h"
+
+#include "mozilla/net/NeckoChild.h"
+#include "mozilla/net/SocketProcessChild.h"
+
+#include "LoadInfo.h"
+
+#include "WebrtcTCPSocketLog.h"
+#include "WebrtcTCPSocketCallback.h"
+
+using namespace mozilla::ipc;
+
+namespace mozilla::net {
+
+mozilla::ipc::IPCResult WebrtcTCPSocketChild::RecvOnClose(
+    const nsresult& aReason) {
+  LOG(("WebrtcTCPSocketChild::RecvOnClose %p\n", this));
+
+  MOZ_ASSERT(mProxyCallbacks, "webrtc TCP callbacks should be non-null");
+  mProxyCallbacks->OnClose(aReason);
+  mProxyCallbacks = nullptr;
+
+  return IPC_OK();
+}
+
+mozilla::ipc::IPCResult WebrtcTCPSocketChild::RecvOnConnected(
+    const nsACString& aProxyType) {
+  LOG(("WebrtcTCPSocketChild::RecvOnConnected %p\n", this));
+
+  MOZ_ASSERT(mProxyCallbacks, "webrtc TCP callbacks should be non-null");
+  mProxyCallbacks->OnConnected(aProxyType);
+
+  return IPC_OK();
+}
+
+mozilla::ipc::IPCResult WebrtcTCPSocketChild::RecvOnRead(
+    nsTArray<uint8_t>&& aReadData) {
+  LOG(("WebrtcTCPSocketChild::RecvOnRead %p\n", this));
+
+  MOZ_ASSERT(mProxyCallbacks, "webrtc TCP callbacks should be non-null");
+  mProxyCallbacks->OnRead(std::move(aReadData));
+
+  return IPC_OK();
+}
+
+WebrtcTCPSocketChild::WebrtcTCPSocketChild(
+    WebrtcTCPSocketCallback* aProxyCallbacks)
+    : mProxyCallbacks(aProxyCallbacks) {
+  MOZ_COUNT_CTOR(WebrtcTCPSocketChild);
+
+  LOG(("WebrtcTCPSocketChild::WebrtcTCPSocketChild %p\n", this));
+}
+
+WebrtcTCPSocketChild::~WebrtcTCPSocketChild() {
+  MOZ_COUNT_DTOR(WebrtcTCPSocketChild);
+
+  LOG(("WebrtcTCPSocketChild::~WebrtcTCPSocketChild %p\n", this));
+}
+
+void WebrtcTCPSocketChild::AsyncOpen(
+    const nsACString& aHost, const int& aPort, const nsACString& aLocalAddress,
+    const int& aLocalPort, bool aUseTls,
+    const std::shared_ptr<NrSocketProxyConfig>& aProxyConfig) {
+  LOG(("WebrtcTCPSocketChild::AsyncOpen %p %s:%d\n", this,
+       PromiseFlatCString(aHost).get(), aPort));
+
+  MOZ_ASSERT(NS_IsMainThread(), "not main thread");
+
+  AddIPDLReference();
+
+  Maybe<net::WebrtcProxyConfig> proxyConfig;
+  Maybe<dom::TabId> tabId;
+  if (aProxyConfig) {
+    proxyConfig = Some(aProxyConfig->GetConfig());
+    tabId = Some(proxyConfig->tabId());
+  }
+
+  if (IsNeckoChild()) {
+    // We're on a content process
+    gNeckoChild->SendPWebrtcTCPSocketConstructor(this, tabId);
+  } else if (IsSocketProcessChild()) {
+    // We're on a socket process
+    SocketProcessChild::GetSingleton()->SendPWebrtcTCPSocketConstructor(this,
+                                                                        tabId);
+  }
+
+  SendAsyncOpen(aHost, aPort, aLocalAddress, aLocalPort, aUseTls, proxyConfig);
+}
+
+}  // namespace mozilla::net
diff --git a/dom/media/webrtc/transport/ipc/WebrtcTCPSocketChild.h b/dom/media/webrtc/transport/ipc/WebrtcTCPSocketChild.h
new file mode 100644
index 0000000000..638ffcaac3
--- /dev/null
+++ b/dom/media/webrtc/transport/ipc/WebrtcTCPSocketChild.h
@@ -0,0 +1,47 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set sw=2 ts=8 et tw=80 ft=cpp : */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef mozilla_net_WebrtcTCPSocketChild_h
+#define mozilla_net_WebrtcTCPSocketChild_h
+
+#include "mozilla/net/PWebrtcTCPSocketChild.h"
+#include "mozilla/dom/ipc/IdType.h"
+#include "transport/nr_socket_proxy_config.h"
+
+namespace mozilla::net {
+
+class WebrtcTCPSocketCallback;
+
+class WebrtcTCPSocketChild : public PWebrtcTCPSocketChild {
+ public:
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(WebrtcTCPSocketChild)
+
+  mozilla::ipc::IPCResult RecvOnClose(const nsresult& aReason) override;
+
+  mozilla::ipc::IPCResult RecvOnConnected(
+      const nsACString& aProxyType) override;
+
+  mozilla::ipc::IPCResult RecvOnRead(nsTArray<uint8_t>&& aReadData) override;
+
+  explicit WebrtcTCPSocketChild(WebrtcTCPSocketCallback* aProxyCallbacks);
+
+  void AsyncOpen(const nsACString& aHost, const int& aPort,
+                 const nsACString& aLocalAddress, const int& aLocalPort,
+                 bool aUseTls,
+                 const std::shared_ptr<NrSocketProxyConfig>& aProxyConfig);
+
+  void AddIPDLReference() { AddRef(); }
+  void ReleaseIPDLReference() { Release(); }
+
+ protected:
+  virtual ~WebrtcTCPSocketChild();
+
+  RefPtr<WebrtcTCPSocketCallback> mProxyCallbacks;
+};
+
+}  // namespace mozilla::net
+
+#endif  // mozilla_net_WebrtcTCPSocketChild_h
diff --git a/dom/media/webrtc/transport/ipc/WebrtcTCPSocketLog.cpp b/dom/media/webrtc/transport/ipc/WebrtcTCPSocketLog.cpp
new file mode 100644
index 0000000000..a6b0dcdb75
--- /dev/null
+++ b/dom/media/webrtc/transport/ipc/WebrtcTCPSocketLog.cpp
@@ -0,0 +1,11 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set sw=2 ts=8 et tw=80 ft=cpp : */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "WebrtcTCPSocketLog.h"
+
+namespace mozilla::net {
+LazyLogModule webrtcTCPSocketLog("WebrtcTCPSocket");
+}  // namespace mozilla::net
diff --git a/dom/media/webrtc/transport/ipc/WebrtcTCPSocketLog.h b/dom/media/webrtc/transport/ipc/WebrtcTCPSocketLog.h
new file mode 100644
index 0000000000..72d3d0064b
--- /dev/null
+++ b/dom/media/webrtc/transport/ipc/WebrtcTCPSocketLog.h
@@ -0,0 +1,20 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set sw=2 ts=8 et tw=80 ft=cpp : */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef webrtc_tcp_socket_log_h__
+#define webrtc_tcp_socket_log_h__
+
+#include "mozilla/Logging.h"
+
+namespace mozilla::net {
+extern LazyLogModule webrtcTCPSocketLog;
+}  // namespace mozilla::net
+
+#undef LOG
+#define LOG(args) \
+  MOZ_LOG(mozilla::net::webrtcTCPSocketLog, mozilla::LogLevel::Debug, args)
+
+#endif  // webrtc_tcp_socket_log_h__
diff --git a/dom/media/webrtc/transport/ipc/WebrtcTCPSocketParent.cpp b/dom/media/webrtc/transport/ipc/WebrtcTCPSocketParent.cpp
new file mode 100644
index 0000000000..0df8962757
--- /dev/null
+++ b/dom/media/webrtc/transport/ipc/WebrtcTCPSocketParent.cpp
@@ -0,0 +1,122 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set sw=2 ts=8 et tw=80 ft=cpp : */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "WebrtcTCPSocketParent.h"
+
+#include "mozilla/net/NeckoParent.h"
+
+#include "WebrtcTCPSocket.h"
+#include "WebrtcTCPSocketLog.h"
+
+using namespace mozilla::dom;
+using namespace mozilla::ipc;
+
+namespace mozilla::net {
+
+mozilla::ipc::IPCResult WebrtcTCPSocketParent::RecvAsyncOpen(
+    const nsACString& aHost, const int& aPort, const nsACString& aLocalAddress,
+    const int& aLocalPort, const bool& aUseTls,
+    const Maybe<WebrtcProxyConfig>& aProxyConfig) {
+  LOG(("WebrtcTCPSocketParent::RecvAsyncOpen %p to %s:%d\n", this,
+       PromiseFlatCString(aHost).get(), aPort));
+
+  MOZ_ASSERT(mChannel, "webrtc TCP socket should be non-null");
+  if (!mChannel) {
+    return IPC_FAIL(this, "Called with null channel.");
+  }
+
+  mChannel->Open(aHost, aPort, aLocalAddress, aLocalPort, aUseTls,
+                 aProxyConfig);
+
+  return IPC_OK();
+}
+
+mozilla::ipc::IPCResult WebrtcTCPSocketParent::RecvWrite(
+    nsTArray<uint8_t>&& aWriteData) {
+  LOG(("WebrtcTCPSocketParent::RecvWrite %p for %zu\n", this,
+       aWriteData.Length()));
+
+  // Need to check this here in case there are Writes in the queue after OnClose
+  if (mChannel) {
+    mChannel->Write(std::move(aWriteData));
+  }
+
+  return IPC_OK();
+}
+
+mozilla::ipc::IPCResult WebrtcTCPSocketParent::RecvClose() {
+  LOG(("WebrtcTCPSocketParent::RecvClose %p\n", this));
+
+  CleanupChannel();
+
+  IProtocol* mgr = Manager();
+  if (!Send__delete__(this)) {
+    return IPC_FAIL_NO_REASON(mgr);
+  }
+
+  return IPC_OK();
+}
+
+void WebrtcTCPSocketParent::ActorDestroy(ActorDestroyReason aWhy) {
+  LOG(("WebrtcTCPSocketParent::ActorDestroy %p for %d\n", this, aWhy));
+
+  CleanupChannel();
+}
+
+WebrtcTCPSocketParent::WebrtcTCPSocketParent(const Maybe<dom::TabId>& aTabId) {
+  MOZ_COUNT_CTOR(WebrtcTCPSocketParent);
+
+  LOG(("WebrtcTCPSocketParent::WebrtcTCPSocketParent %p\n", this));
+
+  mChannel = new WebrtcTCPSocket(this);
+  if (aTabId.isSome()) {
+    mChannel->SetTabId(*aTabId);
+  }
+}
+
+WebrtcTCPSocketParent::~WebrtcTCPSocketParent() {
+  MOZ_COUNT_DTOR(WebrtcTCPSocketParent);
+
+  LOG(("WebrtcTCPSocketParent::~WebrtcTCPSocketParent %p\n", this));
+
+  CleanupChannel();
+}
+
+// WebrtcTCPSocketCallback
+void WebrtcTCPSocketParent::OnClose(nsresult aReason) {
+  LOG(("WebrtcTCPSocketParent::OnClose %p\n", this));
+
+  if (mChannel) {
+    Unused << SendOnClose(aReason);
+  }
+
+  CleanupChannel();
+}
+
+void WebrtcTCPSocketParent::OnRead(nsTArray<uint8_t>&& aReadData) {
+  LOG(("WebrtcTCPSocketParent::OnRead %p %zu\n", this, aReadData.Length()));
+
+  if (mChannel && !SendOnRead(std::move(aReadData))) {
+    CleanupChannel();
+  }
+}
+
+void WebrtcTCPSocketParent::OnConnected(const nsACString& aProxyType) {
+  LOG(("WebrtcTCPSocketParent::OnConnected %p\n", this));
+
+  if (mChannel && !SendOnConnected(aProxyType)) {
+    CleanupChannel();
+  }
+}
+
+void WebrtcTCPSocketParent::CleanupChannel() {
+  if (mChannel) {
+    mChannel->Close();
+    mChannel = nullptr;
+  }
+}
+
+}  // namespace mozilla::net
diff --git a/dom/media/webrtc/transport/ipc/WebrtcTCPSocketParent.h b/dom/media/webrtc/transport/ipc/WebrtcTCPSocketParent.h
new file mode 100644
index 0000000000..df4462609d
--- /dev/null
+++ b/dom/media/webrtc/transport/ipc/WebrtcTCPSocketParent.h
@@ -0,0 +1,59 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set sw=2 ts=8 et tw=80 ft=cpp : */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef mozilla_net_WebrtcTCPSocketParent_h
+#define mozilla_net_WebrtcTCPSocketParent_h
+
+#include "mozilla/net/PWebrtcTCPSocketParent.h"
+
+#include "WebrtcTCPSocketCallback.h"
+
+class nsIAuthPromptProvider;
+
+namespace mozilla::net {
+
+class WebrtcTCPSocket;
+
+class WebrtcTCPSocketParent : public PWebrtcTCPSocketParent,
+                              public WebrtcTCPSocketCallback {
+ public:
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(WebrtcTCPSocketParent, override)
+
+  mozilla::ipc::IPCResult RecvAsyncOpen(
+      const nsACString& aHost, const int& aPort,
+      const nsACString& aLocalAddress, const int& aLocalPort,
+      const bool& aUseTls,
+      const Maybe<WebrtcProxyConfig>& aProxyConfig) override;
+
+  mozilla::ipc::IPCResult RecvWrite(nsTArray<uint8_t>&& aWriteData) override;
+
+  mozilla::ipc::IPCResult RecvClose() override;
+
+  void ActorDestroy(ActorDestroyReason aWhy) override;
+
+  explicit WebrtcTCPSocketParent(const Maybe<dom::TabId>& aTabId);
+
+  // WebrtcTCPSocketCallback
+  void OnClose(nsresult aReason) override;
+  void OnConnected(const nsACString& aProxyType) override;
+  void OnRead(nsTArray<uint8_t>&& bytes) override;
+
+  void AddIPDLReference() { AddRef(); }
+  void ReleaseIPDLReference() { Release(); }
+
+ protected:
+  virtual ~WebrtcTCPSocketParent();
+
+ private:
+  void CleanupChannel();
+
+  // Indicates that IPC is open.
+  RefPtr<WebrtcTCPSocket> mChannel;
+};
+
+}  // namespace mozilla::net
+
+#endif  // mozilla_net_WebrtcTCPSocketParent_h
diff --git a/dom/media/webrtc/transport/ipc/moz.build b/dom/media/webrtc/transport/ipc/moz.build
new file mode 100644
index 0000000000..a2a72bb624
--- /dev/null
+++ b/dom/media/webrtc/transport/ipc/moz.build
@@ -0,0 +1,54 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+EXPORTS.mozilla.net += [
+    "NrIceStunAddrMessageUtils.h",
+    "PStunAddrsParams.h",
+    "StunAddrsRequestChild.h",
+    "StunAddrsRequestParent.h",
+    "WebrtcTCPSocket.h",
+    "WebrtcTCPSocketCallback.h",
+    "WebrtcTCPSocketChild.h",
+    "WebrtcTCPSocketParent.h",
+]
+
+UNIFIED_SOURCES += [
+    "StunAddrsRequestChild.cpp",
+    "StunAddrsRequestParent.cpp",
+    "WebrtcTCPSocket.cpp",
+    "WebrtcTCPSocketChild.cpp",
+    "WebrtcTCPSocketLog.cpp",
+    "WebrtcTCPSocketParent.cpp",
+]
+
+IPDL_SOURCES += [
+    "PStunAddrsRequest.ipdl",
+    "PWebrtcTCPSocket.ipdl",
+    "WebrtcProxyConfig.ipdlh",
+]
+
+include("/ipc/chromium/chromium-config.mozbuild")
+
+FINAL_LIBRARY = "xul"
+
+DEFINES["R_DEFINED_INT2"] = "int16_t"
+DEFINES["R_DEFINED_UINT2"] = "uint16_t"
+DEFINES["R_DEFINED_INT4"] = "int32_t"
+DEFINES["R_DEFINED_UINT4"] = "uint32_t"
+# These are defined to avoid a conflict between typedefs in winsock2.h and
+# r_types.h. This is safe because these types are unused by the code here,
+# but still deeply unfortunate. There is similar code in the win32 version of
+# csi_platform.h, but that trick does not work here, even if that file is
+# directly included.
+DEFINES["R_DEFINED_INT8"] = "int8_t"
+DEFINES["R_DEFINED_UINT8"] = "uint8_t"
+
+LOCAL_INCLUDES += [
+    "/dom/media/webrtc/jsapi",
+    "/dom/media/webrtc/transport/third_party/nICEr/src/net",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr",
+    "/media/webrtc",
+    "/netwerk/base",
+    "/netwerk/protocol/http",
+]
diff --git a/dom/media/webrtc/transport/logging.h b/dom/media/webrtc/transport/logging.h
new file mode 100644
index 0000000000..98cbd13819
--- /dev/null
+++ b/dom/media/webrtc/transport/logging.h
@@ -0,0 +1,65 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+#ifndef logging_h__
+#define logging_h__
+
+#include <sstream>
+#include "mozilla/Logging.h"
+
+#ifdef MOZILLA_INTERNAL_API
+
+#  define ML_ERROR mozilla::LogLevel::Error
+#  define ML_WARNING mozilla::LogLevel::Warning
+#  define ML_NOTICE mozilla::LogLevel::Info
+#  define ML_INFO mozilla::LogLevel::Debug
+#  define ML_DEBUG mozilla::LogLevel::Verbose
+
+#  define MOZ_MTLOG_MODULE(n)                       \
+    static mozilla::LogModule* getLogModule() {     \
+      static mozilla::LazyLogModule log(n);         \
+      return static_cast<mozilla::LogModule*>(log); \
+    }
+
+#  define MOZ_MTLOG(level, b)                                      \
+    do {                                                           \
+      if (MOZ_LOG_TEST(getLogModule(), level)) {                   \
+        std::stringstream str;                                     \
+        str << b;                                                  \
+        MOZ_LOG(getLogModule(), level, ("%s", str.str().c_str())); \
+      }                                                            \
+    } while (0)
+#else
+// When building mtransport outside of XUL, for example in stand-alone gtests,
+// PR_Logging needs to be used instead of mozilla logging.
+
+#  include "prlog.h"
+
+#  define ML_ERROR PR_LOG_ERROR
+#  define ML_WARNING PR_LOG_WARNING
+#  define ML_NOTICE PR_LOG_INFO
+#  define ML_INFO PR_LOG_DEBUG
+#  define ML_DEBUG PR_LOG_VERBOSE
+
+#  define MOZ_MTLOG_MODULE(n)                \
+    static PRLogModuleInfo* getLogModule() { \
+      static PRLogModuleInfo* log;           \
+      if (!log) log = PR_NewLogModule(n);    \
+      return log;                            \
+    }
+
+#  define MOZ_MTLOG(level, b)                                     \
+    do {                                                          \
+      if (PR_LOG_TEST(getLogModule(), level)) {                   \
+        std::stringstream str;                                    \
+        str << b;                                                 \
+        PR_LOG(getLogModule(), level, ("%s", str.str().c_str())); \
+      }                                                           \
+    } while (0)
+#endif  // MOZILLA_INTERNAL_API
+#endif  // logging_h__
diff --git a/dom/media/webrtc/transport/m_cpp_utils.h b/dom/media/webrtc/transport/m_cpp_utils.h
new file mode 100644
index 0000000000..a1469f798a
--- /dev/null
+++ b/dom/media/webrtc/transport/m_cpp_utils.h
@@ -0,0 +1,25 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+#ifndef m_cpp_utils_h__
+#define m_cpp_utils_h__
+
+#include "mozilla/Attributes.h"
+
+namespace mozilla {
+
+#define DISALLOW_ASSIGNMENT(T) void operator=(const T& other) = delete
+
+#define DISALLOW_COPY(T) T(const T& other) = delete
+
+#define DISALLOW_COPY_ASSIGN(T) \
+  DISALLOW_COPY(T);             \
+  DISALLOW_ASSIGNMENT(T)
+
+}  // namespace mozilla
+#endif
diff --git a/dom/media/webrtc/transport/mdns_service/Cargo.toml b/dom/media/webrtc/transport/mdns_service/Cargo.toml
new file mode 100644
index 0000000000..ec7e182d4b
--- /dev/null
+++ b/dom/media/webrtc/transport/mdns_service/Cargo.toml
@@ -0,0 +1,14 @@
+[package]
+name = "mdns_service"
+version = "0.1.1"
+authors = ["Dan Minor <dminor@mozilla.com>"]
+edition = "2018"
+license = "MPL-2.0"
+
+[dependencies]
+byteorder = "1.3.1"
+dns-parser = "0.8.0"
+gecko-profiler = { path = "../../../../../tools/profiler/rust-api" }
+log = "0.4"
+socket2 = { version = "0.4", features = ["all"] }
+uuid = { version = "1.0", features = ["v4"] }
diff --git a/dom/media/webrtc/transport/mdns_service/mdns_service.h b/dom/media/webrtc/transport/mdns_service/mdns_service.h
new file mode 100644
index 0000000000..5e8c252ebc
--- /dev/null
+++ b/dom/media/webrtc/transport/mdns_service/mdns_service.h
@@ -0,0 +1,28 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include <cstdarg>
+#include <cstdint>
+#include <cstdlib>
+#include <new>
+
+struct MDNSService;
+
+extern "C" {
+
+void mdns_service_register_hostname(MDNSService* serv, const char* hostname,
+                                    const char* addr);
+
+MDNSService* mdns_service_start(const char* ifaddr);
+
+void mdns_service_stop(MDNSService* serv);
+
+void mdns_service_query_hostname(
+    MDNSService* serv, void* data,
+    void (*resolved)(void* data, const char* hostname, const char* address),
+    void (*timedout)(void* data, const char* hostname), const char* hostname);
+
+void mdns_service_unregister_hostname(MDNSService* serv, const char* hostname);
+
+}  // extern "C"
diff --git a/dom/media/webrtc/transport/mdns_service/src/lib.rs b/dom/media/webrtc/transport/mdns_service/src/lib.rs
new file mode 100644
index 0000000000..fbd07b45f8
--- /dev/null
+++ b/dom/media/webrtc/transport/mdns_service/src/lib.rs
@@ -0,0 +1,843 @@
+/* -*- Mode: rust; rust-indent-offset: 2 -*- */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+use byteorder::{BigEndian, WriteBytesExt};
+use socket2::{Domain, Socket, Type};
+use std::collections::HashMap;
+use std::collections::LinkedList;
+use std::ffi::{c_void, CStr, CString};
+use std::io;
+use std::net;
+use std::os::raw::c_char;
+use std::sync::mpsc::channel;
+use std::thread;
+use std::time;
+use uuid::Uuid;
+
+#[macro_use]
+extern crate log;
+
+struct Callback {
+    data: *const c_void,
+    resolved: unsafe extern "C" fn(*const c_void, *const c_char, *const c_char),
+    timedout: unsafe extern "C" fn(*const c_void, *const c_char),
+}
+
+unsafe impl Send for Callback {}
+
+fn hostname_resolved(callback: &Callback, hostname: &str, addr: &str) {
+    if let Ok(hostname) = CString::new(hostname) {
+        if let Ok(addr) = CString::new(addr) {
+            unsafe {
+                (callback.resolved)(callback.data, hostname.as_ptr(), addr.as_ptr());
+            }
+        }
+    }
+}
+
+fn hostname_timedout(callback: &Callback, hostname: &str) {
+    if let Ok(hostname) = CString::new(hostname) {
+        unsafe {
+            (callback.timedout)(callback.data, hostname.as_ptr());
+        }
+    }
+}
+
+// This code is derived from code for creating questions in the dns-parser
+// crate. It would be nice to upstream this, or something similar.
+fn create_answer(id: u16, answers: &[(String, &[u8])]) -> Result<Vec<u8>, io::Error> {
+    let mut buf = Vec::with_capacity(512);
+    let head = dns_parser::Header {
+        id,
+        query: false,
+        opcode: dns_parser::Opcode::StandardQuery,
+        authoritative: true,
+        truncated: false,
+        recursion_desired: false,
+        recursion_available: false,
+        authenticated_data: false,
+        checking_disabled: false,
+        response_code: dns_parser::ResponseCode::NoError,
+        questions: 0,
+        answers: answers.len() as u16,
+        nameservers: 0,
+        additional: 0,
+    };
+
+    buf.extend([0u8; 12].iter());
+    head.write(&mut buf[..12]);
+
+    for (name, addr) in answers {
+        for part in name.split('.') {
+            if part.len() > 62 {
+                return Err(io::Error::new(
+                    io::ErrorKind::Other,
+                    "Name part length too long",
+                ));
+            }
+            let ln = part.len() as u8;
+            buf.push(ln);
+            buf.extend(part.as_bytes());
+        }
+        buf.push(0);
+
+        if addr.len() == 4 {
+            buf.write_u16::<BigEndian>(dns_parser::Type::A as u16)?;
+        } else {
+            buf.write_u16::<BigEndian>(dns_parser::Type::AAAA as u16)?;
+        }
+        // set cache flush bit
+        buf.write_u16::<BigEndian>(dns_parser::Class::IN as u16 | (0x1 << 15))?;
+        buf.write_u32::<BigEndian>(120)?;
+        buf.write_u16::<BigEndian>(addr.len() as u16)?;
+        buf.extend(*addr);
+    }
+
+    Ok(buf)
+}
+
+fn create_query(id: u16, queries: &[String]) -> Result<Vec<u8>, io::Error> {
+    let mut buf = Vec::with_capacity(512);
+    let head = dns_parser::Header {
+        id,
+        query: true,
+        opcode: dns_parser::Opcode::StandardQuery,
+        authoritative: false,
+        truncated: false,
+        recursion_desired: false,
+        recursion_available: false,
+        authenticated_data: false,
+        checking_disabled: false,
+        response_code: dns_parser::ResponseCode::NoError,
+        questions: queries.len() as u16,
+        answers: 0,
+        nameservers: 0,
+        additional: 0,
+    };
+
+    buf.extend([0u8; 12].iter());
+    head.write(&mut buf[..12]);
+
+    for name in queries {
+        for part in name.split('.') {
+            assert!(part.len() < 63);
+            let ln = part.len() as u8;
+            buf.push(ln);
+            buf.extend(part.as_bytes());
+        }
+        buf.push(0);
+
+        buf.write_u16::<BigEndian>(dns_parser::QueryType::A as u16)?;
+        buf.write_u16::<BigEndian>(dns_parser::QueryClass::IN as u16)?;
+    }
+
+    Ok(buf)
+}
+
+fn handle_queries(
+    socket: &std::net::UdpSocket,
+    mdns_addr: &std::net::SocketAddr,
+    pending_queries: &mut HashMap<String, Query>,
+    unsent_queries: &mut LinkedList<Query>,
+) {
+    if pending_queries.len() < 50 {
+        let mut queries: Vec<Query> = Vec::new();
+        while queries.len() < 5 && !unsent_queries.is_empty() {
+            if let Some(query) = unsent_queries.pop_front() {
+                if !pending_queries.contains_key(&query.hostname) {
+                    queries.push(query);
+                }
+            }
+        }
+        if !queries.is_empty() {
+            let query_hostnames: Vec<String> =
+                queries.iter().map(|q| q.hostname.to_string()).collect();
+
+            if let Ok(buf) = create_query(0, &query_hostnames) {
+                match socket.send_to(&buf, &mdns_addr) {
+                    Ok(_) => {
+                        for query in queries {
+                            pending_queries.insert(query.hostname.to_string(), query);
+                        }
+                    }
+                    Err(err) => {
+                        warn!("Sending mDNS query failed: {}", err);
+                        if err.kind() != io::ErrorKind::PermissionDenied {
+                            for query in queries {
+                                unsent_queries.push_back(query);
+                            }
+                        } else {
+                            for query in queries {
+                                hostname_timedout(&query.callback, &query.hostname);
+                            }
+                        }
+                    }
+                }
+            }
+        }
+    }
+
+    let now = time::Instant::now();
+    let expired: Vec<String> = pending_queries
+        .iter()
+        .filter(|(_, query)| now.duration_since(query.timestamp).as_secs() >= 3)
+        .map(|(hostname, _)| hostname.to_string())
+        .collect();
+    for hostname in expired {
+        if let Some(mut query) = pending_queries.remove(&hostname) {
+            query.attempts += 1;
+            if query.attempts < 3 {
+                query.timestamp = now;
+                unsent_queries.push_back(query);
+            } else {
+                hostname_timedout(&query.callback, &hostname);
+            }
+        }
+    }
+}
+
+fn handle_mdns_socket(
+    socket: &std::net::UdpSocket,
+    mdns_addr: &std::net::SocketAddr,
+    mut buffer: &mut [u8],
+    hosts: &mut HashMap<String, Vec<u8>>,
+    pending_queries: &mut HashMap<String, Query>,
+) -> bool {
+    // Record a simple marker to see how often this is called.
+    gecko_profiler::add_untyped_marker(
+        "handle_mdns_socket",
+        gecko_profiler::gecko_profiler_category!(Network),
+        Default::default(),
+    );
+
+    match socket.recv_from(&mut buffer) {
+        Ok((amt, _)) => {
+            if amt > 0 {
+                let buffer = &buffer[0..amt];
+                match dns_parser::Packet::parse(&buffer) {
+                    Ok(parsed) => {
+                        let mut answers: Vec<(String, &[u8])> = Vec::new();
+
+                        // If a packet contains both both questions and
+                        // answers, the questions should be ignored.
+                        if parsed.answers.is_empty() {
+                            parsed
+                                .questions
+                                .iter()
+                                .filter(|question| question.qtype == dns_parser::QueryType::A)
+                                .for_each(|question| {
+                                    let qname = question.qname.to_string();
+                                    trace!("mDNS question: {} {:?}", qname, question.qtype);
+                                    if let Some(octets) = hosts.get(&qname) {
+                                        trace!("Sending mDNS answer for {}: {:?}", qname, octets);
+                                        answers.push((qname, &octets));
+                                    }
+                                });
+                        }
+                        for answer in parsed.answers {
+                            let hostname = answer.name.to_string();
+                            match pending_queries.get(&hostname) {
+                                Some(query) => {
+                                    match answer.data {
+                                        dns_parser::RData::A(dns_parser::rdata::a::Record(
+                                            addr,
+                                        )) => {
+                                            let addr = addr.to_string();
+                                            trace!("mDNS response: {} {}", hostname, addr);
+                                            hostname_resolved(&query.callback, &hostname, &addr);
+                                        }
+                                        dns_parser::RData::AAAA(
+                                            dns_parser::rdata::aaaa::Record(addr),
+                                        ) => {
+                                            let addr = addr.to_string();
+                                            trace!("mDNS response: {} {}", hostname, addr);
+                                            hostname_resolved(&query.callback, &hostname, &addr);
+                                        }
+                                        _ => {}
+                                    }
+                                    pending_queries.remove(&hostname);
+                                }
+                                None => {
+                                    continue;
+                                }
+                            }
+                        }
+                        // TODO: If we did not answer every query in this
+                        // question, we should wait for a random amount of time
+                        // so as to not collide with someone else responding to
+                        // this query.
+                        if !answers.is_empty() {
+                            if let Ok(buf) = create_answer(parsed.header.id, &answers) {
+                                if let Err(err) = socket.send_to(&buf, &mdns_addr) {
+                                    warn!("Sending mDNS answer failed: {}", err);
+                                }
+                            }
+                        }
+                    }
+                    Err(err) => {
+                        warn!("Could not parse mDNS packet: {}", err);
+                    }
+                }
+            }
+        }
+        Err(err) => {
+            if err.kind() != io::ErrorKind::Interrupted
+                && err.kind() != io::ErrorKind::TimedOut
+                && err.kind() != io::ErrorKind::WouldBlock
+            {
+                error!("Socket error: {}", err);
+                return false;
+            }
+        }
+    }
+
+    true
+}
+
+fn validate_hostname(hostname: &str) -> bool {
+    match hostname.find(".local") {
+        Some(index) => match hostname.get(0..index) {
+            Some(uuid) => match uuid.get(0..36) {
+                Some(initial) => match Uuid::parse_str(initial) {
+                    Ok(_) => {
+                        // Oddly enough, Safari does not generate valid UUIDs,
+                        // the last part sometimes contains more than 12 digits.
+                        match uuid.get(36..) {
+                            Some(trailing) => {
+                                for c in trailing.chars() {
+                                    if !c.is_ascii_hexdigit() {
+                                        return false;
+                                    }
+                                }
+                                true
+                            }
+                            None => true,
+                        }
+                    }
+                    Err(_) => false,
+                },
+                None => false,
+            },
+            None => false,
+        },
+        None => false,
+    }
+}
+
+enum ServiceControl {
+    Register {
+        hostname: String,
+        address: String,
+    },
+    Query {
+        callback: Callback,
+        hostname: String,
+    },
+    Unregister {
+        hostname: String,
+    },
+    Stop,
+}
+
+struct Query {
+    hostname: String,
+    callback: Callback,
+    timestamp: time::Instant,
+    attempts: i32,
+}
+
+impl Query {
+    fn new(hostname: &str, callback: Callback) -> Query {
+        Query {
+            hostname: hostname.to_string(),
+            callback,
+            timestamp: time::Instant::now(),
+            attempts: 0,
+        }
+    }
+}
+
+pub struct MDNSService {
+    handle: Option<std::thread::JoinHandle<()>>,
+    sender: Option<std::sync::mpsc::Sender<ServiceControl>>,
+}
+
+impl MDNSService {
+    fn register_hostname(&mut self, hostname: &str, address: &str) {
+        if let Some(sender) = &self.sender {
+            if let Err(err) = sender.send(ServiceControl::Register {
+                hostname: hostname.to_string(),
+                address: address.to_string(),
+            }) {
+                warn!(
+                    "Could not send register hostname {} message: {}",
+                    hostname, err
+                );
+            }
+        }
+    }
+
+    fn query_hostname(&mut self, callback: Callback, hostname: &str) {
+        if let Some(sender) = &self.sender {
+            if let Err(err) = sender.send(ServiceControl::Query {
+                callback,
+                hostname: hostname.to_string(),
+            }) {
+                warn!(
+                    "Could not send query hostname {} message: {}",
+                    hostname, err
+                );
+            }
+        }
+    }
+
+    fn unregister_hostname(&mut self, hostname: &str) {
+        if let Some(sender) = &self.sender {
+            if let Err(err) = sender.send(ServiceControl::Unregister {
+                hostname: hostname.to_string(),
+            }) {
+                warn!(
+                    "Could not send unregister hostname {} message: {}",
+                    hostname, err
+                );
+            }
+        }
+    }
+
+    fn start(&mut self, addrs: Vec<std::net::Ipv4Addr>) -> io::Result<()> {
+        let (sender, receiver) = channel();
+        self.sender = Some(sender);
+
+        let mdns_addr = std::net::Ipv4Addr::new(224, 0, 0, 251);
+        let port = 5353;
+
+        let socket = Socket::new(Domain::IPV4, Type::DGRAM, None)?;
+        socket.set_reuse_address(true)?;
+
+        #[cfg(not(target_os = "windows"))]
+        socket.set_reuse_port(true)?;
+        socket.bind(&socket2::SockAddr::from(std::net::SocketAddr::from((
+            [0, 0, 0, 0],
+            port,
+        ))))?;
+
+        let socket = std::net::UdpSocket::from(socket);
+        socket.set_multicast_loop_v4(true)?;
+        socket.set_read_timeout(Some(time::Duration::from_millis(1)))?;
+        socket.set_write_timeout(Some(time::Duration::from_millis(1)))?;
+        for addr in addrs {
+            if let Err(err) = socket.join_multicast_v4(&mdns_addr, &addr) {
+                warn!(
+                    "Could not join multicast group on interface: {:?}: {}",
+                    addr, err
+                );
+            }
+        }
+
+        let thread_name = "mdns_service";
+        let builder = thread::Builder::new().name(thread_name.into());
+        self.handle = Some(builder.spawn(move || {
+            gecko_profiler::register_thread(thread_name);
+            let mdns_addr = std::net::SocketAddr::from(([224, 0, 0, 251], port));
+            let mut buffer: [u8; 9_000] = [0; 9_000];
+            let mut hosts = HashMap::new();
+            let mut unsent_queries = LinkedList::new();
+            let mut pending_queries = HashMap::new();
+            loop {
+                match receiver.try_recv() {
+                    Ok(msg) => match msg {
+                        ServiceControl::Register { hostname, address } => {
+                            if !validate_hostname(&hostname) {
+                                warn!("Not registering invalid hostname: {}", hostname);
+                                continue;
+                            }
+                            trace!("Registering {} for: {}", hostname, address);
+                            match address.parse().and_then(|ip| {
+                                Ok(match ip {
+                                    net::IpAddr::V4(ip) => ip.octets().to_vec(),
+                                    net::IpAddr::V6(ip) => ip.octets().to_vec(),
+                                })
+                            }) {
+                                Ok(octets) => {
+                                    let mut v = Vec::new();
+                                    v.extend(octets);
+                                    hosts.insert(hostname, v);
+                                }
+                                Err(err) => {
+                                    warn!(
+                                        "Could not parse address for {}: {}: {}",
+                                        hostname, address, err
+                                    );
+                                }
+                            }
+                        }
+                        ServiceControl::Query { callback, hostname } => {
+                            trace!("Querying {}", hostname);
+                            if !validate_hostname(&hostname) {
+                                warn!("Not sending mDNS query for invalid hostname: {}", hostname);
+                                continue;
+                            }
+                            unsent_queries.push_back(Query::new(&hostname, callback));
+                        }
+                        ServiceControl::Unregister { hostname } => {
+                            trace!("Unregistering {}", hostname);
+                            hosts.remove(&hostname);
+                        }
+                        ServiceControl::Stop => {
+                            trace!("Stopping");
+                            break;
+                        }
+                    },
+                    Err(std::sync::mpsc::TryRecvError::Disconnected) => {
+                        break;
+                    }
+                    Err(std::sync::mpsc::TryRecvError::Empty) => {}
+                }
+
+                handle_queries(
+                    &socket,
+                    &mdns_addr,
+                    &mut pending_queries,
+                    &mut unsent_queries,
+                );
+
+                if !handle_mdns_socket(
+                    &socket,
+                    &mdns_addr,
+                    &mut buffer,
+                    &mut hosts,
+                    &mut pending_queries,
+                ) {
+                    break;
+                }
+            }
+            gecko_profiler::unregister_thread();
+        })?);
+
+        Ok(())
+    }
+
+    fn stop(self) {
+        if let Some(sender) = self.sender {
+            if let Err(err) = sender.send(ServiceControl::Stop) {
+                warn!("Could not stop mDNS Service: {}", err);
+            }
+            if let Some(handle) = self.handle {
+                if handle.join().is_err() {
+                    error!("Error on thread join");
+                }
+            }
+        }
+    }
+
+    fn new() -> MDNSService {
+        MDNSService {
+            handle: None,
+            sender: None,
+        }
+    }
+}
+
+/// # Safety
+///
+/// This function must only be called with a valid MDNSService pointer.
+/// This hostname and address arguments must be zero terminated strings.
+#[no_mangle]
+pub unsafe extern "C" fn mdns_service_register_hostname(
+    serv: *mut MDNSService,
+    hostname: *const c_char,
+    address: *const c_char,
+) {
+    assert!(!serv.is_null());
+    assert!(!hostname.is_null());
+    assert!(!address.is_null());
+    let hostname = CStr::from_ptr(hostname).to_string_lossy();
+    let address = CStr::from_ptr(address).to_string_lossy();
+    (*serv).register_hostname(&hostname, &address);
+}
+
+/// # Safety
+///
+/// This ifaddrs argument must be a zero terminated string.
+#[no_mangle]
+pub unsafe extern "C" fn mdns_service_start(ifaddrs: *const c_char) -> *mut MDNSService {
+    assert!(!ifaddrs.is_null());
+    let mut r = Box::new(MDNSService::new());
+    let ifaddrs = CStr::from_ptr(ifaddrs).to_string_lossy();
+    let addrs: Vec<std::net::Ipv4Addr> =
+        ifaddrs.split(';').filter_map(|x| x.parse().ok()).collect();
+
+    if addrs.is_empty() {
+        warn!("Could not parse interface addresses from: {}", ifaddrs);
+    } else if let Err(err) = r.start(addrs) {
+        warn!("Could not start mDNS Service: {}", err);
+    }
+
+    Box::into_raw(r)
+}
+
+/// # Safety
+///
+/// This function must only be called with a valid MDNSService pointer.
+#[no_mangle]
+pub unsafe extern "C" fn mdns_service_stop(serv: *mut MDNSService) {
+    assert!(!serv.is_null());
+    let boxed = Box::from_raw(serv);
+    boxed.stop();
+}
+
+/// # Safety
+///
+/// This function must only be called with a valid MDNSService pointer.
+/// The data argument will be passed back into the resolved and timedout
+/// functions. The object it points to must not be freed until the MDNSService
+/// has stopped.
+#[no_mangle]
+pub unsafe extern "C" fn mdns_service_query_hostname(
+    serv: *mut MDNSService,
+    data: *const c_void,
+    resolved: unsafe extern "C" fn(*const c_void, *const c_char, *const c_char),
+    timedout: unsafe extern "C" fn(*const c_void, *const c_char),
+    hostname: *const c_char,
+) {
+    assert!(!serv.is_null());
+    assert!(!data.is_null());
+    assert!(!hostname.is_null());
+    let hostname = CStr::from_ptr(hostname).to_string_lossy();
+    let callback = Callback {
+        data,
+        resolved,
+        timedout,
+    };
+    (*serv).query_hostname(callback, &hostname);
+}
+
+/// # Safety
+///
+/// This function must only be called with a valid MDNSService pointer.
+/// This function should only be called once per hostname.
+#[no_mangle]
+pub unsafe extern "C" fn mdns_service_unregister_hostname(
+    serv: *mut MDNSService,
+    hostname: *const c_char,
+) {
+    assert!(!serv.is_null());
+    assert!(!hostname.is_null());
+    let hostname = CStr::from_ptr(hostname).to_string_lossy();
+    (*serv).unregister_hostname(&hostname);
+}
+
+#[cfg(test)]
+mod tests {
+    use crate::create_query;
+    use crate::validate_hostname;
+    use crate::Callback;
+    use crate::MDNSService;
+    use socket2::{Domain, Socket, Type};
+    use std::collections::HashSet;
+    use std::ffi::c_void;
+    use std::io;
+    use std::iter::FromIterator;
+    use std::os::raw::c_char;
+    use std::thread;
+    use std::time;
+    use uuid::Uuid;
+
+    #[no_mangle]
+    pub unsafe extern "C" fn mdns_service_resolved(
+        _: *const c_void,
+        _: *const c_char,
+        _: *const c_char,
+    ) -> () {
+    }
+
+    #[no_mangle]
+    pub unsafe extern "C" fn mdns_service_timedout(_: *const c_void, _: *const c_char) -> () {}
+
+    fn listen_until(addr: &std::net::Ipv4Addr, stop: u64) -> thread::JoinHandle<Vec<String>> {
+        let port = 5353;
+
+        let socket = Socket::new(Domain::IPV4, Type::DGRAM, None).unwrap();
+        socket.set_reuse_address(true).unwrap();
+
+        #[cfg(not(target_os = "windows"))]
+        socket.set_reuse_port(true).unwrap();
+        socket
+            .bind(&socket2::SockAddr::from(std::net::SocketAddr::from((
+                [0, 0, 0, 0],
+                port,
+            ))))
+            .unwrap();
+
+        let socket = std::net::UdpSocket::from(socket);
+        socket.set_multicast_loop_v4(true).unwrap();
+        socket
+            .set_read_timeout(Some(time::Duration::from_millis(10)))
+            .unwrap();
+        socket
+            .set_write_timeout(Some(time::Duration::from_millis(10)))
+            .unwrap();
+        socket
+            .join_multicast_v4(&std::net::Ipv4Addr::new(224, 0, 0, 251), &addr)
+            .unwrap();
+
+        let mut buffer: [u8; 9_000] = [0; 9_000];
+        thread::spawn(move || {
+            let start = time::Instant::now();
+            let mut questions = Vec::new();
+            while time::Instant::now().duration_since(start).as_secs() < stop {
+                match socket.recv_from(&mut buffer) {
+                    Ok((amt, _)) => {
+                        if amt > 0 {
+                            let buffer = &buffer[0..amt];
+                            match dns_parser::Packet::parse(&buffer) {
+                                Ok(parsed) => {
+                                    parsed
+                                        .questions
+                                        .iter()
+                                        .filter(|question| {
+                                            question.qtype == dns_parser::QueryType::A
+                                        })
+                                        .for_each(|question| {
+                                            let qname = question.qname.to_string();
+                                            questions.push(qname);
+                                        });
+                                }
+                                Err(err) => {
+                                    warn!("Could not parse mDNS packet: {}", err);
+                                }
+                            }
+                        }
+                    }
+                    Err(err) => {
+                        if err.kind() != io::ErrorKind::WouldBlock
+                            && err.kind() != io::ErrorKind::TimedOut
+                        {
+                            error!("Socket error: {}", err);
+                            break;
+                        }
+                    }
+                }
+            }
+            questions
+        })
+    }
+
+    #[test]
+    fn test_validate_hostname() {
+        assert_eq!(
+            validate_hostname("e17f08d4-689a-4df6-ba31-35bb9f041100.local"),
+            true
+        );
+        assert_eq!(
+            validate_hostname("62240723-ae6d-4f6a-99b8-94a233e3f84a2.local"),
+            true
+        );
+        assert_eq!(
+            validate_hostname("62240723-ae6d-4f6a-99b8.94e3f84a2.local"),
+            false
+        );
+        assert_eq!(validate_hostname("hi there"), false);
+    }
+
+    #[test]
+    fn start_stop() {
+        let mut service = MDNSService::new();
+        let addr = "127.0.0.1".parse().unwrap();
+        service.start(vec![addr]).unwrap();
+        service.stop();
+    }
+
+    #[test]
+    fn simple_query() {
+        let mut service = MDNSService::new();
+        let addr = "127.0.0.1".parse().unwrap();
+        let handle = listen_until(&addr, 1);
+
+        service.start(vec![addr]).unwrap();
+
+        let callback = Callback {
+            data: 0 as *const c_void,
+            resolved: mdns_service_resolved,
+            timedout: mdns_service_timedout,
+        };
+        let hostname = Uuid::new_v4().as_hyphenated().to_string() + ".local";
+        service.query_hostname(callback, &hostname);
+        service.stop();
+        let questions = handle.join().unwrap();
+        assert!(questions.contains(&hostname));
+    }
+
+    #[test]
+    fn rate_limited_query() {
+        let mut service = MDNSService::new();
+        let addr = "127.0.0.1".parse().unwrap();
+        let handle = listen_until(&addr, 1);
+
+        service.start(vec![addr]).unwrap();
+
+        let mut hostnames = HashSet::new();
+        for _ in 0..100 {
+            let callback = Callback {
+                data: 0 as *const c_void,
+                resolved: mdns_service_resolved,
+                timedout: mdns_service_timedout,
+            };
+            let hostname = Uuid::new_v4().as_hyphenated().to_string() + ".local";
+            service.query_hostname(callback, &hostname);
+            hostnames.insert(hostname);
+        }
+        service.stop();
+        let questions = HashSet::from_iter(handle.join().unwrap().iter().map(|x| x.to_string()));
+        let intersection: HashSet<&String> = questions.intersection(&hostnames).collect();
+        assert_eq!(intersection.len(), 50);
+    }
+
+    #[test]
+    fn repeat_failed_query() {
+        let mut service = MDNSService::new();
+        let addr = "127.0.0.1".parse().unwrap();
+        let handle = listen_until(&addr, 4);
+
+        service.start(vec![addr]).unwrap();
+
+        let hostname = Uuid::new_v4().as_hyphenated().to_string() + ".local";
+        let callback = Callback {
+            data: 0 as *const c_void,
+            resolved: mdns_service_resolved,
+            timedout: mdns_service_timedout,
+        };
+        service.query_hostname(callback, &hostname);
+        thread::sleep(time::Duration::from_secs(4));
+        service.stop();
+
+        let questions: Vec<String> = handle
+            .join()
+            .unwrap()
+            .iter()
+            .filter(|x| *x == &hostname)
+            .map(|x| x.to_string())
+            .collect();
+        assert_eq!(questions.len(), 2);
+    }
+
+    #[test]
+    fn multiple_queries_in_a_single_packet() {
+        let mut hostnames: Vec<String> = Vec::new();
+        for _ in 0..100 {
+            let hostname = Uuid::new_v4().as_hyphenated().to_string() + ".local";
+            hostnames.push(hostname);
+        }
+
+        match create_query(42, &hostnames) {
+            Ok(q) => match dns_parser::Packet::parse(&q) {
+                Ok(parsed) => {
+                    assert_eq!(parsed.questions.len(), 100);
+                }
+                Err(_) => assert!(false),
+            },
+            Err(_) => assert!(false),
+        }
+    }
+}
diff --git a/dom/media/webrtc/transport/mediapacket.cpp b/dom/media/webrtc/transport/mediapacket.cpp
new file mode 100644
index 0000000000..38ab65ed2a
--- /dev/null
+++ b/dom/media/webrtc/transport/mediapacket.cpp
@@ -0,0 +1,145 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "mediapacket.h"
+
+#include <cstring>
+#include "ipc/IPCMessageUtils.h"
+
+namespace mozilla {
+
+void MediaPacket::Copy(const uint8_t* data, size_t len, size_t capacity) {
+  if (capacity < len) {
+    capacity = len;
+  }
+  data_.reset(new uint8_t[capacity]);
+  len_ = len;
+  capacity_ = capacity;
+  memcpy(data_.get(), data, len);
+}
+
+MediaPacket MediaPacket::Clone() const {
+  MediaPacket newPacket;
+  newPacket.type_ = type_;
+  newPacket.sdp_level_ = sdp_level_;
+  newPacket.Copy(data_.get(), len_, capacity_);
+  return newPacket;
+}
+
+void MediaPacket::Serialize(IPC::MessageWriter* aWriter) const {
+  aWriter->WriteUInt32(len_);
+  aWriter->WriteUInt32(capacity_);
+  if (len_) {
+    aWriter->WriteBytes(data_.get(), len_);
+  }
+  aWriter->WriteUInt32(encrypted_len_);
+  if (encrypted_len_) {
+    aWriter->WriteBytes(encrypted_data_.get(), encrypted_len_);
+  }
+  aWriter->WriteInt32(sdp_level_.isSome() ? *sdp_level_ : -1);
+  aWriter->WriteInt32(type_);
+}
+
+bool MediaPacket::Deserialize(IPC::MessageReader* aReader) {
+  Reset();
+  uint32_t len;
+  if (!aReader->ReadUInt32(&len)) {
+    return false;
+  }
+  uint32_t capacity;
+  if (!aReader->ReadUInt32(&capacity)) {
+    return false;
+  }
+  if (len) {
+    MOZ_RELEASE_ASSERT(capacity >= len);
+    UniquePtr<uint8_t[]> data(new uint8_t[capacity]);
+    if (!aReader->ReadBytesInto(data.get(), len)) {
+      return false;
+    }
+    data_ = std::move(data);
+    len_ = len;
+    capacity_ = capacity;
+  }
+
+  if (!aReader->ReadUInt32(&len)) {
+    return false;
+  }
+  if (len) {
+    UniquePtr<uint8_t[]> data(new uint8_t[len]);
+    if (!aReader->ReadBytesInto(data.get(), len)) {
+      return false;
+    }
+    encrypted_data_ = std::move(data);
+    encrypted_len_ = len;
+  }
+
+  int32_t sdp_level;
+  if (!aReader->ReadInt32(&sdp_level)) {
+    return false;
+  }
+
+  if (sdp_level >= 0) {
+    sdp_level_ = Some(sdp_level);
+  }
+
+  int32_t type;
+  if (!aReader->ReadInt32(&type)) {
+    return false;
+  }
+  type_ = static_cast<Type>(type);
+  return true;
+}
+
+static bool IsRtp(const uint8_t* data, size_t len) {
+  if (len < 2) return false;
+
+  // Check if this is a RTCP packet. Logic based on the types listed in
+  // media/webrtc/trunk/src/modules/rtp_rtcp/source/rtp_utility.cc
+
+  // Anything outside this range is RTP.
+  if ((data[1] < 192) || (data[1] > 207)) return true;
+
+  if (data[1] == 192)  // FIR
+    return false;
+
+  if (data[1] == 193)  // NACK, but could also be RTP. This makes us sad
+    return true;       // but it's how webrtc.org behaves.
+
+  if (data[1] == 194) return true;
+
+  if (data[1] == 195)  // IJ.
+    return false;
+
+  if ((data[1] > 195) && (data[1] < 200))  // the > 195 is redundant
+    return true;
+
+  if ((data[1] >= 200) && (data[1] <= 207))  // SR, RR, SDES, BYE,
+    return false;                            // APP, RTPFB, PSFB, XR
+
+  MOZ_ASSERT(false);  // Not reached, belt and suspenders.
+  return true;
+}
+
+void MediaPacket::Categorize() {
+  SetType(MediaPacket::UNCLASSIFIED);
+
+  if (!data_ || len_ < 4) {
+    return;
+  }
+
+  if (data_[0] >= 20 && data_[0] <= 63) {
+    // DTLS per RFC 7983
+    SetType(MediaPacket::DTLS);
+  } else if (data_[0] > 127 && data_[0] < 192) {
+    // RTP/RTCP per RFC 7983
+    if (IsRtp(data_.get(), len_)) {
+      SetType(MediaPacket::SRTP);
+    } else {
+      SetType(MediaPacket::SRTCP);
+    }
+  }
+}
+}  // namespace mozilla
diff --git a/dom/media/webrtc/transport/mediapacket.h b/dom/media/webrtc/transport/mediapacket.h
new file mode 100644
index 0000000000..563056c4a4
--- /dev/null
+++ b/dom/media/webrtc/transport/mediapacket.h
@@ -0,0 +1,117 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef mediapacket_h__
+#define mediapacket_h__
+
+#include <cstddef>
+#include <cstdint>
+#include "mozilla/UniquePtr.h"
+#include "mozilla/Maybe.h"
+
+class PickleIterator;
+
+namespace IPC {
+class Message;
+class MessageReader;
+class MessageWriter;
+}  // namespace IPC
+
+namespace mozilla {
+
+// TODO: It might be worthwhile to teach this class how to "borrow" a buffer.
+// That would make it easier to misuse, however, so maybe not worth it.
+class MediaPacket {
+ public:
+  MediaPacket() = default;
+  MediaPacket(MediaPacket&& orig) = default;
+  MediaPacket& operator=(MediaPacket&& orig) = default;
+
+  MediaPacket Clone() const;
+
+  // Takes ownership of the passed-in data
+  void Take(UniquePtr<uint8_t[]>&& data, size_t len, size_t capacity = 0) {
+    data_ = std::move(data);
+    len_ = len;
+    if (capacity < len) {
+      capacity = len;
+    }
+    capacity_ = capacity;
+  }
+
+  void Reset() {
+    data_.reset();
+    len_ = 0;
+    capacity_ = 0;
+    encrypted_data_.reset();
+    encrypted_len_ = 0;
+    sdp_level_.reset();
+  }
+
+  // Copies the passed-in data
+  void Copy(const uint8_t* data, size_t len, size_t capacity = 0);
+
+  uint8_t* data() const { return data_.get(); }
+
+  size_t len() const { return len_; }
+
+  void SetLength(size_t length) { len_ = length; }
+
+  size_t capacity() const { return capacity_; }
+
+  Maybe<size_t>& sdp_level() { return sdp_level_; }
+
+  void CopyDataToEncrypted() {
+    encrypted_data_ = std::move(data_);
+    encrypted_len_ = len_;
+    Copy(encrypted_data_.get(), len_);
+  }
+
+  const uint8_t* encrypted_data() const { return encrypted_data_.get(); }
+
+  size_t encrypted_len() const { return encrypted_len_; }
+
+  enum Type { UNCLASSIFIED, SRTP, SRTCP, DTLS, RTP, RTCP, SCTP };
+
+  void Categorize();
+
+  void SetType(Type type) { type_ = type; }
+
+  Type type() const { return type_; }
+
+  void Serialize(IPC::MessageWriter* aWriter) const;
+  bool Deserialize(IPC::MessageReader* aReader);
+
+ private:
+  UniquePtr<uint8_t[]> data_;
+  size_t len_ = 0;
+  size_t capacity_ = 0;
+  // Encrypted form of the data, if there is one.
+  UniquePtr<uint8_t[]> encrypted_data_;
+  size_t encrypted_len_ = 0;
+  // SDP level that this packet belongs to, if known.
+  Maybe<size_t> sdp_level_;
+  Type type_ = UNCLASSIFIED;
+};
+}  // namespace mozilla
+
+namespace IPC {
+template <typename>
+struct ParamTraits;
+
+template <>
+struct ParamTraits<mozilla::MediaPacket> {
+  static void Write(MessageWriter* aWriter,
+                    const mozilla::MediaPacket& aParam) {
+    aParam.Serialize(aWriter);
+  }
+
+  static bool Read(MessageReader* aReader, mozilla::MediaPacket* aResult) {
+    return aResult->Deserialize(aReader);
+  }
+};
+}  // namespace IPC
+#endif  // mediapacket_h__
diff --git a/dom/media/webrtc/transport/moz.build b/dom/media/webrtc/transport/moz.build
new file mode 100644
index 0000000000..fcf265e22f
--- /dev/null
+++ b/dom/media/webrtc/transport/moz.build
@@ -0,0 +1,23 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+with Files("**"):
+    BUG_COMPONENT = ("Core", "WebRTC: Networking")
+
+include("/ipc/chromium/chromium-config.mozbuild")
+
+DIRS += [
+    "/dom/media/webrtc/transport/third_party",
+    "/dom/media/webrtc/transport/build",
+    "/dom/media/webrtc/transport/ipc",
+    "/dom/media/webrtc/transport/srtp",
+]
+
+# Add libFuzzer configuration directives
+include("/tools/fuzzing/libfuzzer-config.mozbuild")
+
+if CONFIG["FUZZING_INTERFACES"]:
+    TEST_DIRS += ["fuzztest"]
diff --git a/dom/media/webrtc/transport/nr_socket_proxy_config.cpp b/dom/media/webrtc/transport/nr_socket_proxy_config.cpp
new file mode 100644
index 0000000000..8c91a5d975
--- /dev/null
+++ b/dom/media/webrtc/transport/nr_socket_proxy_config.cpp
@@ -0,0 +1,34 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set sw=2 ts=8 et tw=80 ft=cpp : */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "nr_socket_proxy_config.h"
+
+#include "mozilla/net/WebrtcProxyConfig.h"
+
+namespace mozilla {
+
+class NrSocketProxyConfig::Private {
+ public:
+  net::WebrtcProxyConfig mProxyConfig;
+};
+
+NrSocketProxyConfig::NrSocketProxyConfig(
+    const net::WebrtcProxyConfig& aProxyConfig)
+    : mPrivate(new Private({aProxyConfig})) {}
+
+NrSocketProxyConfig::NrSocketProxyConfig(NrSocketProxyConfig&& aOrig)
+    : mPrivate(std::move(aOrig.mPrivate)) {}
+
+NrSocketProxyConfig::~NrSocketProxyConfig() = default;
+
+const net::WebrtcProxyConfig& NrSocketProxyConfig::GetConfig() const {
+  return mPrivate->mProxyConfig;
+}
+
+bool NrSocketProxyConfig::GetForceProxy() const {
+  return mPrivate->mProxyConfig.forceProxy();
+}
+}  // namespace mozilla
diff --git a/dom/media/webrtc/transport/nr_socket_proxy_config.h b/dom/media/webrtc/transport/nr_socket_proxy_config.h
new file mode 100644
index 0000000000..55ef4fdcb1
--- /dev/null
+++ b/dom/media/webrtc/transport/nr_socket_proxy_config.h
@@ -0,0 +1,41 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set sw=2 ts=8 et tw=80 ft=cpp : */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef nr_socket_proxy_config__
+#define nr_socket_proxy_config__
+
+#include <memory>
+
+namespace mozilla {
+namespace net {
+class WebrtcProxyConfig;
+}
+
+class NrSocketProxyConfig {
+ public:
+  explicit NrSocketProxyConfig(const net::WebrtcProxyConfig& aProxyConfig);
+
+  // We need to actually write the default impl ourselves, because the compiler
+  // needs to know how to destroy mPrivate in case an exception is thrown, even
+  // though we disable exceptions in our build.
+  NrSocketProxyConfig(NrSocketProxyConfig&& aOrig);
+
+  ~NrSocketProxyConfig();
+
+  const net::WebrtcProxyConfig& GetConfig() const;
+  bool GetForceProxy() const;
+
+ private:
+  // dom::ProxyConfig includes stuff that conflicts with nICEr includes.
+  // Make it possible to include this header file without tripping over this
+  // problem.
+  class Private;
+  std::unique_ptr<Private> mPrivate;
+};
+
+}  // namespace mozilla
+
+#endif  // nr_socket_proxy_config__
diff --git a/dom/media/webrtc/transport/nr_socket_prsock.cpp b/dom/media/webrtc/transport/nr_socket_prsock.cpp
new file mode 100644
index 0000000000..3c99a2d0c8
--- /dev/null
+++ b/dom/media/webrtc/transport/nr_socket_prsock.cpp
@@ -0,0 +1,1785 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+/*
+Modified version of nr_socket_local, adapted for NSPR
+*/
+
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+/*
+Original code from nICEr and nrappkit.
+
+nICEr copyright:
+
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+
+nrappkit copyright:
+
+   Copyright (C) 2001-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   ekr@rtfm.com  Thu Dec 20 20:14:49 2001
+*/
+
+#include <csi_platform.h>
+#include <stdio.h>
+#include <string.h>
+#include <sys/types.h>
+#include <assert.h>
+#include <errno.h>
+#include <string>
+
+#include "nspr.h"
+#include "prerror.h"
+#include "prio.h"
+#include "prnetdb.h"
+
+#include "mozilla/net/DNS.h"
+#include "nsCOMPtr.h"
+#include "nsASocketHandler.h"
+#include "nsISocketTransportService.h"
+#include "nsNetCID.h"
+#include "nsISupportsImpl.h"
+#include "nsServiceManagerUtils.h"
+#include "nsComponentManagerUtils.h"
+#include "nsXPCOM.h"
+#include "nsXULAppAPI.h"
+#include "runnable_utils.h"
+#include "mozilla/SyncRunnable.h"
+#include "nsTArray.h"
+#include "nsISocketFilter.h"
+#include "nsDebug.h"
+#include "nsNetUtil.h"
+
+#ifdef XP_WIN
+#  include "mozilla/WindowsVersion.h"
+#endif
+
+#if defined(MOZILLA_INTERNAL_API)
+// csi_platform.h deep in nrappkit defines LOG_INFO and LOG_WARNING
+#  ifdef LOG_INFO
+#    define LOG_TEMP_INFO LOG_INFO
+#    undef LOG_INFO
+#  endif
+#  ifdef LOG_WARNING
+#    define LOG_TEMP_WARNING LOG_WARNING
+#    undef LOG_WARNING
+#  endif
+#  if defined(LOG_DEBUG)
+#    define LOG_TEMP_DEBUG LOG_DEBUG
+#    undef LOG_DEBUG
+#  endif
+#  undef strlcpy
+
+#  include "mozilla/dom/network/UDPSocketChild.h"
+
+#  ifdef LOG_TEMP_INFO
+#    define LOG_INFO LOG_TEMP_INFO
+#  endif
+#  ifdef LOG_TEMP_WARNING
+#    define LOG_WARNING LOG_TEMP_WARNING
+#  endif
+
+#  ifdef LOG_TEMP_DEBUG
+#    define LOG_DEBUG LOG_TEMP_DEBUG
+#  endif
+#  ifdef XP_WIN
+#    ifdef LOG_DEBUG
+#      undef LOG_DEBUG
+#    endif
+// cloned from csi_platform.h.  Win32 doesn't like how we hide symbols
+#    define LOG_DEBUG 7
+#  endif
+#endif
+
+extern "C" {
+#include "nr_api.h"
+#include "async_wait.h"
+#include "nr_socket.h"
+#include "nr_socket_local.h"
+#include "stun_hint.h"
+}
+#include "nr_socket_prsock.h"
+#include "simpletokenbucket.h"
+#include "test_nr_socket.h"
+#include "nr_socket_tcp.h"
+#include "nr_socket_proxy_config.h"
+
+// Implement the nsISupports ref counting
+namespace mozilla {
+
+#if defined(MOZILLA_INTERNAL_API)
+class SingletonThreadHolder final {
+ private:
+  ~SingletonThreadHolder() {
+    r_log(LOG_GENERIC, LOG_DEBUG, "Deleting SingletonThreadHolder");
+    if (mThread) {
+      // Likely a connection is somehow being held in CC or GC
+      NS_WARNING(
+          "SingletonThreads should be Released and shut down before exit!");
+      mThread->Shutdown();
+      mThread = nullptr;
+    }
+  }
+
+  DISALLOW_COPY_ASSIGN(SingletonThreadHolder);
+
+ public:
+  // Must be threadsafe for StaticRefPtr/ClearOnShutdown
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(SingletonThreadHolder)
+
+  explicit SingletonThreadHolder(const nsACString& aName) : mName(aName) {
+    mParentThread = NS_GetCurrentThread();
+  }
+
+  nsIThread* GetThread() { return mThread; }
+
+  /*
+   * Keep track of how many instances are using a SingletonThreadHolder.
+   * When no one is using it, shut it down
+   */
+  void AddUse() {
+    MOZ_ASSERT(mParentThread == NS_GetCurrentThread());
+    MOZ_ASSERT(int32_t(mUseCount) >= 0, "illegal refcnt");
+    nsrefcnt count = ++mUseCount;
+    if (count == 1) {
+      // idle -> in-use
+      nsresult rv = NS_NewNamedThread(mName, getter_AddRefs(mThread));
+      MOZ_RELEASE_ASSERT(NS_SUCCEEDED(rv) && mThread,
+                         "Should successfully create mtransport I/O thread");
+      r_log(LOG_GENERIC, LOG_DEBUG, "Created wrapped SingletonThread %p",
+            mThread.get());
+    }
+    r_log(LOG_GENERIC, LOG_DEBUG, "AddUse_i: %lu", (unsigned long)count);
+  }
+
+  void ReleaseUse() {
+    MOZ_ASSERT(mParentThread == NS_GetCurrentThread());
+    nsrefcnt count = --mUseCount;
+    MOZ_ASSERT(int32_t(mUseCount) >= 0, "illegal refcnt");
+    if (mThread && count == 0) {
+      // in-use -> idle -- no one forcing it to remain instantiated
+      r_log(LOG_GENERIC, LOG_DEBUG, "Shutting down wrapped SingletonThread %p",
+            mThread.get());
+      mThread->AsyncShutdown();
+      mThread = nullptr;
+      // It'd be nice to use a timer instead...  But be careful of
+      // xpcom-shutdown-threads in that case
+    }
+    r_log(LOG_GENERIC, LOG_DEBUG, "ReleaseUse_i: %lu", (unsigned long)count);
+  }
+
+ private:
+  nsCString mName;
+  nsAutoRefCnt mUseCount;
+  nsCOMPtr<nsIThread> mParentThread;
+  nsCOMPtr<nsIThread> mThread;
+};
+
+static StaticRefPtr<SingletonThreadHolder> sThread;
+
+static void ClearSingletonOnShutdown() {
+  // We expect everybody to have done ReleaseUse() at the latest during
+  // xpcom-shutdown-threads. So we need to live longer than that.
+  ClearOnShutdown(&sThread, ShutdownPhase::XPCOMShutdownFinal);
+}
+#endif
+
+static nsIThread* GetIOThreadAndAddUse_s() {
+  // Always runs on STS thread!
+#if defined(MOZILLA_INTERNAL_API)
+  // We need to safely release this on shutdown to avoid leaks
+  if (!sThread) {
+    sThread = new SingletonThreadHolder("mtransport"_ns);
+    NS_DispatchToMainThread(mozilla::WrapRunnableNM(&ClearSingletonOnShutdown));
+  }
+  // Mark that we're using the shared thread and need it to stick around
+  sThread->AddUse();
+  return sThread->GetThread();
+#else
+  static nsCOMPtr<nsIThread> sThread;
+  if (!sThread) {
+    (void)NS_NewNamedThread("mtransport", getter_AddRefs(sThread));
+  }
+  return sThread;
+#endif
+}
+
+NrSocketIpc::NrSocketIpc(nsIEventTarget* aThread) : io_thread_(aThread) {}
+
+static TimeStamp nr_socket_short_term_violation_time;
+static TimeStamp nr_socket_long_term_violation_time;
+
+TimeStamp NrSocketBase::short_term_violation_time() {
+  return nr_socket_short_term_violation_time;
+}
+
+TimeStamp NrSocketBase::long_term_violation_time() {
+  return nr_socket_long_term_violation_time;
+}
+
+// NrSocketBase implementation
+// async_event APIs
+int NrSocketBase::async_wait(int how, NR_async_cb cb, void* cb_arg,
+                             char* function, int line) {
+  uint16_t flag;
+
+  switch (how) {
+    case NR_ASYNC_WAIT_READ:
+      flag = PR_POLL_READ;
+      break;
+    case NR_ASYNC_WAIT_WRITE:
+      flag = PR_POLL_WRITE;
+      break;
+    default:
+      return R_BAD_ARGS;
+  }
+
+  cbs_[how] = cb;
+  cb_args_[how] = cb_arg;
+  poll_flags_ |= flag;
+
+  return 0;
+}
+
+int NrSocketBase::cancel(int how) {
+  uint16_t flag;
+
+  switch (how) {
+    case NR_ASYNC_WAIT_READ:
+      flag = PR_POLL_READ;
+      break;
+    case NR_ASYNC_WAIT_WRITE:
+      flag = PR_POLL_WRITE;
+      break;
+    default:
+      return R_BAD_ARGS;
+  }
+
+  poll_flags_ &= ~flag;
+
+  return 0;
+}
+
+void NrSocketBase::fire_callback(int how) {
+  // This can't happen unless we are armed because we only set
+  // the flags if we are armed
+  MOZ_ASSERT(cbs_[how]);
+
+  // Now cancel so that we need to be re-armed. Note that
+  // the re-arming probably happens in the callback we are
+  // about to fire.
+  cancel(how);
+
+  cbs_[how](this, how, cb_args_[how]);
+}
+
+// NrSocket implementation
+NS_IMPL_QUERY_INTERFACE0(NrSocket)
+
+// The nsASocket callbacks
+void NrSocket::OnSocketReady(PRFileDesc* fd, int16_t outflags) {
+  if (outflags & PR_POLL_READ & poll_flags()) fire_callback(NR_ASYNC_WAIT_READ);
+  if (outflags & PR_POLL_WRITE & poll_flags())
+    fire_callback(NR_ASYNC_WAIT_WRITE);
+  if (outflags & (PR_POLL_ERR | PR_POLL_NVAL | PR_POLL_HUP))
+    // TODO: Bug 946423: how do we notify the upper layers about this?
+    close();
+}
+
+void NrSocket::OnSocketDetached(PRFileDesc* fd) {
+  r_log(LOG_GENERIC, LOG_DEBUG, "Socket %p detached", fd);
+}
+
+void NrSocket::IsLocal(bool* aIsLocal) {
+  // TODO(jesup): better check? Does it matter? (likely no)
+  *aIsLocal = false;
+}
+
+// async_event APIs
+int NrSocket::async_wait(int how, NR_async_cb cb, void* cb_arg, char* function,
+                         int line) {
+  int r = NrSocketBase::async_wait(how, cb, cb_arg, function, line);
+
+  if (!r) {
+    mPollFlags = poll_flags();
+  }
+
+  return r;
+}
+
+int NrSocket::cancel(int how) {
+  int r = NrSocketBase::cancel(how);
+
+  if (!r) {
+    mPollFlags = poll_flags();
+  }
+
+  return r;
+}
+
+// Helper functions for addresses
+static int nr_transport_addr_to_praddr(const nr_transport_addr* addr,
+                                       PRNetAddr* naddr) {
+  int _status;
+
+  memset(naddr, 0, sizeof(*naddr));
+
+  switch (addr->protocol) {
+    case IPPROTO_TCP:
+      break;
+    case IPPROTO_UDP:
+      break;
+    default:
+      ABORT(R_BAD_ARGS);
+  }
+
+  switch (addr->ip_version) {
+    case NR_IPV4:
+      naddr->inet.family = PR_AF_INET;
+      naddr->inet.port = addr->u.addr4.sin_port;
+      naddr->inet.ip = addr->u.addr4.sin_addr.s_addr;
+      break;
+    case NR_IPV6:
+      naddr->ipv6.family = PR_AF_INET6;
+      naddr->ipv6.port = addr->u.addr6.sin6_port;
+      naddr->ipv6.flowinfo = addr->u.addr6.sin6_flowinfo;
+      memcpy(&naddr->ipv6.ip, &addr->u.addr6.sin6_addr, sizeof(in6_addr));
+      naddr->ipv6.scope_id = addr->u.addr6.sin6_scope_id;
+      break;
+    default:
+      ABORT(R_BAD_ARGS);
+  }
+
+  _status = 0;
+abort:
+  return (_status);
+}
+
+// XXX schien@mozilla.com: copy from PRNetAddrToNetAddr,
+// should be removed after fix the link error in signaling_unittests
+static int praddr_to_netaddr(const PRNetAddr* prAddr, net::NetAddr* addr) {
+  int _status;
+
+  switch (prAddr->raw.family) {
+    case PR_AF_INET:
+      addr->inet.family = AF_INET;
+      addr->inet.port = prAddr->inet.port;
+      addr->inet.ip = prAddr->inet.ip;
+      break;
+    case PR_AF_INET6:
+      addr->inet6.family = AF_INET6;
+      addr->inet6.port = prAddr->ipv6.port;
+      addr->inet6.flowinfo = prAddr->ipv6.flowinfo;
+      memcpy(&addr->inet6.ip, &prAddr->ipv6.ip, sizeof(addr->inet6.ip.u8));
+      addr->inet6.scope_id = prAddr->ipv6.scope_id;
+      break;
+    default:
+      MOZ_ASSERT(false);
+      ABORT(R_BAD_ARGS);
+  }
+
+  _status = 0;
+abort:
+  return (_status);
+}
+
+static int nr_transport_addr_to_netaddr(const nr_transport_addr* addr,
+                                        net::NetAddr* naddr) {
+  int r, _status;
+  PRNetAddr praddr;
+
+  if ((r = nr_transport_addr_to_praddr(addr, &praddr))) {
+    ABORT(r);
+  }
+
+  if ((r = praddr_to_netaddr(&praddr, naddr))) {
+    ABORT(r);
+  }
+
+  _status = 0;
+abort:
+  return (_status);
+}
+
+int nr_netaddr_to_transport_addr(const net::NetAddr* netaddr,
+                                 nr_transport_addr* addr, int protocol) {
+  int _status;
+  int r;
+
+  switch (netaddr->raw.family) {
+    case AF_INET:
+      if ((r = nr_ip4_port_to_transport_addr(ntohl(netaddr->inet.ip),
+                                             ntohs(netaddr->inet.port),
+                                             protocol, addr)))
+        ABORT(r);
+      break;
+    case AF_INET6:
+      if ((r = nr_ip6_port_to_transport_addr((in6_addr*)&netaddr->inet6.ip.u8,
+                                             ntohs(netaddr->inet6.port),
+                                             protocol, addr)))
+        ABORT(r);
+      break;
+    default:
+      MOZ_ASSERT(false);
+      ABORT(R_BAD_ARGS);
+  }
+  _status = 0;
+abort:
+  return (_status);
+}
+
+int nr_praddr_to_transport_addr(const PRNetAddr* praddr,
+                                nr_transport_addr* addr, int protocol,
+                                int keep) {
+  int _status;
+  int r;
+  struct sockaddr_in ip4;
+  struct sockaddr_in6 ip6;
+
+  switch (praddr->raw.family) {
+    case PR_AF_INET:
+      ip4.sin_family = PF_INET;
+      ip4.sin_addr.s_addr = praddr->inet.ip;
+      ip4.sin_port = praddr->inet.port;
+      if ((r = nr_sockaddr_to_transport_addr((sockaddr*)&ip4, protocol, keep,
+                                             addr)))
+        ABORT(r);
+      break;
+    case PR_AF_INET6:
+      ip6.sin6_family = PF_INET6;
+      ip6.sin6_port = praddr->ipv6.port;
+      ip6.sin6_flowinfo = praddr->ipv6.flowinfo;
+      memcpy(&ip6.sin6_addr, &praddr->ipv6.ip, sizeof(in6_addr));
+      ip6.sin6_scope_id = praddr->ipv6.scope_id;
+      if ((r = nr_sockaddr_to_transport_addr((sockaddr*)&ip6, protocol, keep,
+                                             addr)))
+        ABORT(r);
+      break;
+    default:
+      MOZ_ASSERT(false);
+      ABORT(R_BAD_ARGS);
+  }
+
+  _status = 0;
+abort:
+  return (_status);
+}
+
+/*
+ * nr_transport_addr_get_addrstring_and_port
+ * convert nr_transport_addr to IP address string and port number
+ */
+int nr_transport_addr_get_addrstring_and_port(const nr_transport_addr* addr,
+                                              nsACString* host, int32_t* port) {
+  int r, _status;
+  char addr_string[64];
+
+  // We cannot directly use |nr_transport_addr.as_string| because it contains
+  // more than ip address, therefore, we need to explicity convert it
+  // from |nr_transport_addr_get_addrstring|.
+  if ((r = nr_transport_addr_get_addrstring(addr, addr_string,
+                                            sizeof(addr_string)))) {
+    ABORT(r);
+  }
+
+  if ((r = nr_transport_addr_get_port(addr, port))) {
+    ABORT(r);
+  }
+
+  *host = addr_string;
+
+  _status = 0;
+abort:
+  return (_status);
+}
+
+// nr_socket APIs (as member functions)
+int NrSocket::create(nr_transport_addr* addr) {
+  int r, _status;
+
+  PRStatus status;
+  PRNetAddr naddr;
+
+  nsresult rv;
+  nsCOMPtr<nsISocketTransportService> stservice =
+      do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &rv);
+
+  if (!NS_SUCCEEDED(rv)) {
+    ABORT(R_INTERNAL);
+  }
+
+  if ((r = nr_transport_addr_to_praddr(addr, &naddr))) ABORT(r);
+
+  switch (addr->protocol) {
+    case IPPROTO_UDP:
+      if (!(fd_ = PR_OpenUDPSocket(naddr.raw.family))) {
+        r_log(LOG_GENERIC, LOG_CRIT,
+              "Couldn't create UDP socket, "
+              "family=%d, err=%d",
+              naddr.raw.family, PR_GetError());
+        ABORT(R_INTERNAL);
+      }
+#ifdef XP_WIN
+      if (!mozilla::IsWin8OrLater()) {
+        // Increase default send and receive buffer sizes on <= Win7 to be able
+        // to receive and send an unpaced HD (>= 720p = 1280x720 - I Frame ~ 21K
+        // size) stream without losing packets. Manual testing showed that 100K
+        // buffer size was not enough and the packet loss dis-appeared with 256K
+        // buffer size. See bug 1252769 for future improvements of this.
+        PRSize min_buffer_size = 256 * 1024;
+        PRSocketOptionData opt_rcvbuf;
+        opt_rcvbuf.option = PR_SockOpt_RecvBufferSize;
+        if ((status = PR_GetSocketOption(fd_, &opt_rcvbuf)) == PR_SUCCESS) {
+          if (opt_rcvbuf.value.recv_buffer_size < min_buffer_size) {
+            opt_rcvbuf.value.recv_buffer_size = min_buffer_size;
+            if ((status = PR_SetSocketOption(fd_, &opt_rcvbuf)) != PR_SUCCESS) {
+              r_log(LOG_GENERIC, LOG_CRIT,
+                    "Couldn't set socket receive buffer size: %d", status);
+            }
+          } else {
+            r_log(LOG_GENERIC, LOG_INFO,
+                  "Socket receive buffer size is already: %d",
+                  opt_rcvbuf.value.recv_buffer_size);
+          }
+        } else {
+          r_log(LOG_GENERIC, LOG_CRIT,
+                "Couldn't get socket receive buffer size: %d", status);
+        }
+        PRSocketOptionData opt_sndbuf;
+        opt_sndbuf.option = PR_SockOpt_SendBufferSize;
+        if ((status = PR_GetSocketOption(fd_, &opt_sndbuf)) == PR_SUCCESS) {
+          if (opt_sndbuf.value.recv_buffer_size < min_buffer_size) {
+            opt_sndbuf.value.recv_buffer_size = min_buffer_size;
+            if ((status = PR_SetSocketOption(fd_, &opt_sndbuf)) != PR_SUCCESS) {
+              r_log(LOG_GENERIC, LOG_CRIT,
+                    "Couldn't set socket send buffer size: %d", status);
+            }
+          } else {
+            r_log(LOG_GENERIC, LOG_INFO,
+                  "Socket send buffer size is already: %d",
+                  opt_sndbuf.value.recv_buffer_size);
+          }
+        } else {
+          r_log(LOG_GENERIC, LOG_CRIT,
+                "Couldn't get socket send buffer size: %d", status);
+        }
+      }
+#endif
+      break;
+    case IPPROTO_TCP:
+      // TODO: Rewrite this to use WebrtcTcpSocket.
+      // Also use the same logic for TLS.
+      if (my_addr_.fqdn[0] != '\0') ABORT(R_INTERNAL);
+
+      if (!(fd_ = PR_OpenTCPSocket(naddr.raw.family))) {
+        r_log(LOG_GENERIC, LOG_CRIT,
+              "Couldn't create TCP socket, "
+              "family=%d, err=%d",
+              naddr.raw.family, PR_GetError());
+        ABORT(R_INTERNAL);
+      }
+      // Set ReuseAddr for TCP sockets to enable having several
+      // sockets bound to same local IP and port
+      PRSocketOptionData opt_reuseaddr;
+      opt_reuseaddr.option = PR_SockOpt_Reuseaddr;
+      opt_reuseaddr.value.reuse_addr = PR_TRUE;
+      status = PR_SetSocketOption(fd_, &opt_reuseaddr);
+      if (status != PR_SUCCESS) {
+        r_log(LOG_GENERIC, LOG_CRIT,
+              "Couldn't set reuse addr socket option: %d", status);
+        ABORT(R_INTERNAL);
+      }
+      // And also set ReusePort for platforms supporting this socket option
+      PRSocketOptionData opt_reuseport;
+      opt_reuseport.option = PR_SockOpt_Reuseport;
+      opt_reuseport.value.reuse_port = PR_TRUE;
+      status = PR_SetSocketOption(fd_, &opt_reuseport);
+      if (status != PR_SUCCESS) {
+        if (PR_GetError() != PR_OPERATION_NOT_SUPPORTED_ERROR) {
+          r_log(LOG_GENERIC, LOG_CRIT,
+                "Couldn't set reuse port socket option: %d", status);
+          ABORT(R_INTERNAL);
+        }
+      }
+      // Try to speedup packet delivery by disabling TCP Nagle
+      PRSocketOptionData opt_nodelay;
+      opt_nodelay.option = PR_SockOpt_NoDelay;
+      opt_nodelay.value.no_delay = PR_TRUE;
+      status = PR_SetSocketOption(fd_, &opt_nodelay);
+      if (status != PR_SUCCESS) {
+        r_log(LOG_GENERIC, LOG_WARNING,
+              "Couldn't set Nodelay socket option: %d", status);
+      }
+      break;
+    default:
+      ABORT(R_INTERNAL);
+  }
+
+  status = PR_Bind(fd_, &naddr);
+  if (status != PR_SUCCESS) {
+    r_log(LOG_GENERIC, LOG_CRIT, "Couldn't bind socket to address %s",
+          addr->as_string);
+    ABORT(R_INTERNAL);
+  }
+
+  r_log(LOG_GENERIC, LOG_DEBUG, "Creating socket %p with addr %s", fd_,
+        addr->as_string);
+  nr_transport_addr_copy(&my_addr_, addr);
+
+  /* If we have a wildcard port, patch up the addr */
+  if (nr_transport_addr_is_wildcard(addr)) {
+    status = PR_GetSockName(fd_, &naddr);
+    if (status != PR_SUCCESS) {
+      r_log(LOG_GENERIC, LOG_CRIT, "Couldn't get sock name for socket");
+      ABORT(R_INTERNAL);
+    }
+
+    if ((r = nr_praddr_to_transport_addr(&naddr, &my_addr_, addr->protocol, 1)))
+      ABORT(r);
+  }
+
+  // Set nonblocking
+  PRSocketOptionData opt_nonblock;
+  opt_nonblock.option = PR_SockOpt_Nonblocking;
+  opt_nonblock.value.non_blocking = PR_TRUE;
+  status = PR_SetSocketOption(fd_, &opt_nonblock);
+  if (status != PR_SUCCESS) {
+    r_log(LOG_GENERIC, LOG_CRIT, "Couldn't make socket nonblocking");
+    ABORT(R_INTERNAL);
+  }
+
+  // Remember our thread.
+  ststhread_ = do_QueryInterface(stservice, &rv);
+  if (!NS_SUCCEEDED(rv)) ABORT(R_INTERNAL);
+
+  // Finally, register with the STS
+  rv = stservice->AttachSocket(fd_, this);
+  if (!NS_SUCCEEDED(rv)) {
+    r_log(LOG_GENERIC, LOG_CRIT, "Couldn't attach socket to STS, rv=%u",
+          static_cast<unsigned>(rv));
+    ABORT(R_INTERNAL);
+  }
+
+  _status = 0;
+
+abort:
+  return (_status);
+}
+
+static int ShouldDrop(size_t len) {
+  // Global rate limiting for stun requests, to mitigate the ice hammer DoS
+  // (see http://tools.ietf.org/html/draft-thomson-mmusic-ice-webrtc)
+
+  // Tolerate rate of 8k/sec, for one second.
+  static SimpleTokenBucket burst(16384 * 1, 16384);
+  // Tolerate rate of 7.2k/sec over twenty seconds.
+  static SimpleTokenBucket sustained(7372 * 20, 7372);
+
+  // Check number of tokens in each bucket.
+  if (burst.getTokens(UINT32_MAX) < len) {
+    r_log(LOG_GENERIC, LOG_ERR,
+          "Short term global rate limit for STUN requests exceeded.");
+#ifdef MOZILLA_INTERNAL_API
+    nr_socket_short_term_violation_time = TimeStamp::Now();
+#endif
+
+// Bug 1013007
+#if !EARLY_BETA_OR_EARLIER
+    return R_WOULDBLOCK;
+#else
+    MOZ_ASSERT(false,
+               "Short term global rate limit for STUN requests exceeded. Go "
+               "bug bcampen@mozilla.com if you weren't intentionally "
+               "spamming ICE candidates, or don't know what that means.");
+#endif
+  }
+
+  if (sustained.getTokens(UINT32_MAX) < len) {
+    r_log(LOG_GENERIC, LOG_ERR,
+          "Long term global rate limit for STUN requests exceeded.");
+#ifdef MOZILLA_INTERNAL_API
+    nr_socket_long_term_violation_time = TimeStamp::Now();
+#endif
+// Bug 1013007
+#if !EARLY_BETA_OR_EARLIER
+    return R_WOULDBLOCK;
+#else
+    MOZ_ASSERT(false,
+               "Long term global rate limit for STUN requests exceeded. Go "
+               "bug bcampen@mozilla.com if you weren't intentionally "
+               "spamming ICE candidates, or don't know what that means.");
+#endif
+  }
+
+  // Take len tokens from both buckets.
+  // (not threadsafe, but no problem since this is only called from STS)
+  burst.getTokens(len);
+  sustained.getTokens(len);
+  return 0;
+}
+
+// This should be called on the STS thread.
+int NrSocket::sendto(const void* msg, size_t len, int flags,
+                     const nr_transport_addr* to) {
+  ASSERT_ON_THREAD(ststhread_);
+  int r, _status;
+  PRNetAddr naddr;
+  int32_t status;
+
+  if ((r = nr_transport_addr_to_praddr(to, &naddr))) ABORT(r);
+
+  if (fd_ == nullptr) ABORT(R_EOD);
+
+  if (nr_is_stun_request_message((UCHAR*)msg, len) && ShouldDrop(len)) {
+    ABORT(R_WOULDBLOCK);
+  }
+
+  // TODO: Convert flags?
+  status = PR_SendTo(fd_, msg, len, flags, &naddr, PR_INTERVAL_NO_WAIT);
+  if (status < 0 || (size_t)status != len) {
+    if (PR_GetError() == PR_WOULD_BLOCK_ERROR) ABORT(R_WOULDBLOCK);
+
+    r_log(LOG_GENERIC, LOG_INFO, "Error in sendto %s: %d", to->as_string,
+          PR_GetError());
+    ABORT(R_IO_ERROR);
+  }
+
+  _status = 0;
+abort:
+  return (_status);
+}
+
+int NrSocket::recvfrom(void* buf, size_t maxlen, size_t* len, int flags,
+                       nr_transport_addr* from) {
+  ASSERT_ON_THREAD(ststhread_);
+  int r, _status;
+  PRNetAddr nfrom;
+  int32_t status;
+
+  status = PR_RecvFrom(fd_, buf, maxlen, flags, &nfrom, PR_INTERVAL_NO_WAIT);
+  if (status <= 0) {
+    if (PR_GetError() == PR_WOULD_BLOCK_ERROR) ABORT(R_WOULDBLOCK);
+    r_log(LOG_GENERIC, LOG_INFO, "Error in recvfrom: %d", (int)PR_GetError());
+    ABORT(R_IO_ERROR);
+  }
+  *len = status;
+
+  if ((r = nr_praddr_to_transport_addr(&nfrom, from, my_addr_.protocol, 0)))
+    ABORT(r);
+
+  // r_log(LOG_GENERIC,LOG_DEBUG,"Read %d bytes from %s",*len,addr->as_string);
+
+  _status = 0;
+abort:
+  return (_status);
+}
+
+int NrSocket::getaddr(nr_transport_addr* addrp) {
+  ASSERT_ON_THREAD(ststhread_);
+  return nr_transport_addr_copy(addrp, &my_addr_);
+}
+
+// Close the socket so that the STS will detach and then kill it
+void NrSocket::close() {
+  ASSERT_ON_THREAD(ststhread_);
+  mCondition = NS_BASE_STREAM_CLOSED;
+  cancel(NR_ASYNC_WAIT_READ);
+  cancel(NR_ASYNC_WAIT_WRITE);
+}
+
+int NrSocket::connect(const nr_transport_addr* addr) {
+  ASSERT_ON_THREAD(ststhread_);
+  int r, _status;
+  PRNetAddr naddr;
+  int32_t connect_status, getsockname_status;
+
+  if ((r = nr_transport_addr_to_praddr(addr, &naddr))) ABORT(r);
+
+  if (!fd_) ABORT(R_EOD);
+
+  // Note: this just means we tried to connect, not that we
+  // are actually live.
+  connect_invoked_ = true;
+  connect_status = PR_Connect(fd_, &naddr, PR_INTERVAL_NO_WAIT);
+  if (connect_status != PR_SUCCESS) {
+    if (PR_GetError() != PR_IN_PROGRESS_ERROR) {
+      r_log(LOG_GENERIC, LOG_CRIT, "PR_Connect failed: %d", PR_GetError());
+      ABORT(R_IO_ERROR);
+    }
+  }
+
+  // If our local address is wildcard, then fill in the
+  // address now.
+  if (nr_transport_addr_is_wildcard(&my_addr_)) {
+    getsockname_status = PR_GetSockName(fd_, &naddr);
+    if (getsockname_status != PR_SUCCESS) {
+      r_log(LOG_GENERIC, LOG_CRIT, "Couldn't get sock name for socket");
+      ABORT(R_INTERNAL);
+    }
+
+    if ((r = nr_praddr_to_transport_addr(&naddr, &my_addr_, addr->protocol, 1)))
+      ABORT(r);
+  }
+
+  // Now return the WOULDBLOCK if needed.
+  if (connect_status != PR_SUCCESS) {
+    ABORT(R_WOULDBLOCK);
+  }
+
+  _status = 0;
+abort:
+  return (_status);
+}
+
+int NrSocket::write(const void* msg, size_t len, size_t* written) {
+  ASSERT_ON_THREAD(ststhread_);
+  int _status;
+  int32_t status;
+
+  if (!connect_invoked_) ABORT(R_FAILED);
+
+  status = PR_Write(fd_, msg, len);
+  if (status < 0) {
+    if (PR_GetError() == PR_WOULD_BLOCK_ERROR) ABORT(R_WOULDBLOCK);
+    r_log(LOG_GENERIC, LOG_INFO, "Error in write");
+    ABORT(R_IO_ERROR);
+  }
+
+  *written = status;
+
+  _status = 0;
+abort:
+  return _status;
+}
+
+int NrSocket::read(void* buf, size_t maxlen, size_t* len) {
+  ASSERT_ON_THREAD(ststhread_);
+  int _status;
+  int32_t status;
+
+  if (!connect_invoked_) ABORT(R_FAILED);
+
+  status = PR_Read(fd_, buf, maxlen);
+  if (status < 0) {
+    if (PR_GetError() == PR_WOULD_BLOCK_ERROR) ABORT(R_WOULDBLOCK);
+    r_log(LOG_GENERIC, LOG_INFO, "Error in read");
+    ABORT(R_IO_ERROR);
+  }
+  if (status == 0) ABORT(R_EOD);
+
+  *len = (size_t)status;  // Guaranteed to be > 0
+  _status = 0;
+abort:
+  return (_status);
+}
+
+int NrSocket::listen(int backlog) {
+  ASSERT_ON_THREAD(ststhread_);
+  int32_t status;
+  int _status;
+
+  assert(fd_);
+  status = PR_Listen(fd_, backlog);
+  if (status != PR_SUCCESS) {
+    r_log(LOG_GENERIC, LOG_CRIT, "%s: PR_GetError() == %d", __FUNCTION__,
+          PR_GetError());
+    ABORT(R_IO_ERROR);
+  }
+
+  _status = 0;
+abort:
+  return (_status);
+}
+
+int NrSocket::accept(nr_transport_addr* addrp, nr_socket** sockp) {
+  ASSERT_ON_THREAD(ststhread_);
+  int _status, r;
+  PRStatus status;
+  PRFileDesc* prfd;
+  PRNetAddr nfrom;
+  NrSocket* sock = nullptr;
+  nsresult rv;
+  PRSocketOptionData opt_nonblock, opt_nodelay;
+  nsCOMPtr<nsISocketTransportService> stservice =
+      do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &rv);
+
+  if (NS_FAILED(rv)) {
+    ABORT(R_INTERNAL);
+  }
+
+  if (!fd_) ABORT(R_EOD);
+
+  prfd = PR_Accept(fd_, &nfrom, PR_INTERVAL_NO_WAIT);
+
+  if (!prfd) {
+    if (PR_GetError() == PR_WOULD_BLOCK_ERROR) ABORT(R_WOULDBLOCK);
+
+    ABORT(R_IO_ERROR);
+  }
+
+  sock = new NrSocket();
+
+  sock->fd_ = prfd;
+  nr_transport_addr_copy(&sock->my_addr_, &my_addr_);
+
+  if ((r = nr_praddr_to_transport_addr(&nfrom, addrp, my_addr_.protocol, 0)))
+    ABORT(r);
+
+  // Set nonblocking
+  opt_nonblock.option = PR_SockOpt_Nonblocking;
+  opt_nonblock.value.non_blocking = PR_TRUE;
+  status = PR_SetSocketOption(prfd, &opt_nonblock);
+  if (status != PR_SUCCESS) {
+    r_log(LOG_GENERIC, LOG_CRIT,
+          "Failed to make accepted socket nonblocking: %d", status);
+    ABORT(R_INTERNAL);
+  }
+  // Disable TCP Nagle
+  opt_nodelay.option = PR_SockOpt_NoDelay;
+  opt_nodelay.value.no_delay = PR_TRUE;
+  status = PR_SetSocketOption(prfd, &opt_nodelay);
+  if (status != PR_SUCCESS) {
+    r_log(LOG_GENERIC, LOG_WARNING,
+          "Failed to set Nodelay on accepted socket: %d", status);
+  }
+
+  // Should fail only with OOM
+  if ((r = nr_socket_create_int(static_cast<void*>(sock), sock->vtbl(), sockp)))
+    ABORT(r);
+
+  // Remember our thread.
+  sock->ststhread_ = do_QueryInterface(stservice, &rv);
+  if (NS_FAILED(rv)) ABORT(R_INTERNAL);
+
+  // Finally, register with the STS
+  rv = stservice->AttachSocket(prfd, sock);
+  if (NS_FAILED(rv)) {
+    ABORT(R_INTERNAL);
+  }
+
+  sock->connect_invoked_ = true;
+
+  // Add a reference so that we can delete it in destroy()
+  sock->AddRef();
+  _status = 0;
+abort:
+  if (_status) {
+    delete sock;
+  }
+
+  return (_status);
+}
+
+NS_IMPL_ISUPPORTS(NrUdpSocketIpcProxy, nsIUDPSocketInternal)
+
+nsresult NrUdpSocketIpcProxy::Init(const RefPtr<NrUdpSocketIpc>& socket) {
+  nsresult rv;
+  sts_thread_ = do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &rv);
+  if (NS_FAILED(rv)) {
+    MOZ_ASSERT(false, "Failed to get STS thread");
+    return rv;
+  }
+
+  socket_ = socket;
+  return NS_OK;
+}
+
+NrUdpSocketIpcProxy::~NrUdpSocketIpcProxy() {
+  // Send our ref to STS to be released
+  RUN_ON_THREAD(sts_thread_, mozilla::WrapRelease(socket_.forget()),
+                NS_DISPATCH_NORMAL);
+}
+
+// IUDPSocketInternal interfaces
+// callback while error happened in UDP socket operation
+NS_IMETHODIMP NrUdpSocketIpcProxy::CallListenerError(const nsACString& message,
+                                                     const nsACString& filename,
+                                                     uint32_t line_number) {
+  return socket_->CallListenerError(message, filename, line_number);
+}
+
+// callback while receiving UDP packet
+NS_IMETHODIMP NrUdpSocketIpcProxy::CallListenerReceivedData(
+    const nsACString& host, uint16_t port, const nsTArray<uint8_t>& data) {
+  return socket_->CallListenerReceivedData(host, port, data);
+}
+
+// callback while UDP socket is opened
+NS_IMETHODIMP NrUdpSocketIpcProxy::CallListenerOpened() {
+  return socket_->CallListenerOpened();
+}
+
+// callback while UDP socket is connected
+NS_IMETHODIMP NrUdpSocketIpcProxy::CallListenerConnected() {
+  return socket_->CallListenerConnected();
+}
+
+// callback while UDP socket is closed
+NS_IMETHODIMP NrUdpSocketIpcProxy::CallListenerClosed() {
+  return socket_->CallListenerClosed();
+}
+
+// NrUdpSocketIpc Implementation
+NrUdpSocketIpc::NrUdpSocketIpc()
+    : NrSocketIpc(GetIOThreadAndAddUse_s()),
+      monitor_("NrUdpSocketIpc"),
+      err_(false),
+      state_(NR_INIT) {}
+
+NrUdpSocketIpc::~NrUdpSocketIpc() = default;
+
+void NrUdpSocketIpc::Destroy() {
+#if defined(MOZILLA_INTERNAL_API)
+  // destroy_i also dispatches back to STS to call ReleaseUse, to avoid shutting
+  // down the IO thread before close() runs.
+  // We use a NonOwning runnable because our refcount has already gone to 0.
+  io_thread_->Dispatch(NewNonOwningRunnableMethod(
+      "NrUdpSocketIpc::Destroy", this, &NrUdpSocketIpc::destroy_i));
+#endif
+}
+
+// IUDPSocketInternal interfaces
+// callback while error happened in UDP socket operation
+NS_IMETHODIMP NrUdpSocketIpc::CallListenerError(const nsACString& message,
+                                                const nsACString& filename,
+                                                uint32_t line_number) {
+  ASSERT_ON_THREAD(io_thread_);
+
+  r_log(LOG_GENERIC, LOG_ERR, "UDP socket error:%s at %s:%d this=%p",
+        message.BeginReading(), filename.BeginReading(), line_number,
+        (void*)this);
+
+  ReentrantMonitorAutoEnter mon(monitor_);
+  err_ = true;
+  monitor_.NotifyAll();
+
+  return NS_OK;
+}
+
+// callback while receiving UDP packet
+NS_IMETHODIMP NrUdpSocketIpc::CallListenerReceivedData(
+    const nsACString& host, uint16_t port, const nsTArray<uint8_t>& data) {
+  ASSERT_ON_THREAD(io_thread_);
+
+  PRNetAddr addr;
+  memset(&addr, 0, sizeof(addr));
+
+  {
+    ReentrantMonitorAutoEnter mon(monitor_);
+
+    if (PR_SUCCESS != PR_StringToNetAddr(host.BeginReading(), &addr)) {
+      err_ = true;
+      MOZ_ASSERT(false, "Failed to convert remote host to PRNetAddr");
+      return NS_OK;
+    }
+
+    // Use PR_IpAddrNull to avoid address being reset to 0.
+    if (PR_SUCCESS !=
+        PR_SetNetAddr(PR_IpAddrNull, addr.raw.family, port, &addr)) {
+      err_ = true;
+      MOZ_ASSERT(false, "Failed to set port in PRNetAddr");
+      return NS_OK;
+    }
+  }
+
+  auto buf = MakeUnique<MediaPacket>();
+  buf->Copy(data.Elements(), data.Length());
+  RefPtr<nr_udp_message> msg(new nr_udp_message(addr, std::move(buf)));
+
+  RUN_ON_THREAD(sts_thread_,
+                mozilla::WrapRunnable(RefPtr<NrUdpSocketIpc>(this),
+                                      &NrUdpSocketIpc::recv_callback_s, msg),
+                NS_DISPATCH_NORMAL);
+  return NS_OK;
+}
+
+nsresult NrUdpSocketIpc::SetAddress() {
+  uint16_t port = socket_child_->LocalPort();
+
+  nsAutoCString address(socket_child_->LocalAddress());
+
+  PRNetAddr praddr;
+  if (PR_SUCCESS != PR_InitializeNetAddr(PR_IpAddrAny, port, &praddr)) {
+    err_ = true;
+    MOZ_ASSERT(false, "Failed to set port in PRNetAddr");
+    return NS_OK;
+  }
+
+  if (PR_SUCCESS != PR_StringToNetAddr(address.BeginReading(), &praddr)) {
+    err_ = true;
+    MOZ_ASSERT(false, "Failed to convert local host to PRNetAddr");
+    return NS_OK;
+  }
+
+  nr_transport_addr expected_addr;
+  if (nr_transport_addr_copy(&expected_addr, &my_addr_)) {
+    err_ = true;
+    MOZ_ASSERT(false, "Failed to copy my_addr_");
+  }
+
+  if (nr_praddr_to_transport_addr(&praddr, &my_addr_, IPPROTO_UDP, 1)) {
+    err_ = true;
+    MOZ_ASSERT(false, "Failed to copy local host to my_addr_");
+  }
+
+  if (!nr_transport_addr_is_wildcard(&expected_addr) &&
+      nr_transport_addr_cmp(&expected_addr, &my_addr_,
+                            NR_TRANSPORT_ADDR_CMP_MODE_ADDR)) {
+    err_ = true;
+    MOZ_ASSERT(false, "Address of opened socket is not expected");
+  }
+
+  return NS_OK;
+}
+
+// callback while UDP socket is opened
+NS_IMETHODIMP NrUdpSocketIpc::CallListenerOpened() {
+  ASSERT_ON_THREAD(io_thread_);
+  ReentrantMonitorAutoEnter mon(monitor_);
+
+  r_log(LOG_GENERIC, LOG_DEBUG, "UDP socket opened this=%p", (void*)this);
+  nsresult rv = SetAddress();
+  if (NS_FAILED(rv)) {
+    return rv;
+  }
+
+  mon.NotifyAll();
+
+  return NS_OK;
+}
+
+// callback while UDP socket is connected
+NS_IMETHODIMP NrUdpSocketIpc::CallListenerConnected() {
+  ASSERT_ON_THREAD(io_thread_);
+
+  ReentrantMonitorAutoEnter mon(monitor_);
+
+  r_log(LOG_GENERIC, LOG_DEBUG, "UDP socket connected this=%p", (void*)this);
+  MOZ_ASSERT(state_ == NR_CONNECTED);
+
+  nsresult rv = SetAddress();
+  if (NS_FAILED(rv)) {
+    mon.NotifyAll();
+    return rv;
+  }
+
+  r_log(LOG_GENERIC, LOG_INFO, "Exit UDP socket connected");
+  mon.NotifyAll();
+
+  return NS_OK;
+}
+
+// callback while UDP socket is closed
+NS_IMETHODIMP NrUdpSocketIpc::CallListenerClosed() {
+  ASSERT_ON_THREAD(io_thread_);
+
+  ReentrantMonitorAutoEnter mon(monitor_);
+
+  r_log(LOG_GENERIC, LOG_DEBUG, "UDP socket closed this=%p", (void*)this);
+  MOZ_ASSERT(state_ == NR_CONNECTED || state_ == NR_CLOSING);
+  state_ = NR_CLOSED;
+
+  return NS_OK;
+}
+
+//
+// NrSocketBase methods.
+//
+int NrUdpSocketIpc::create(nr_transport_addr* addr) {
+  ASSERT_ON_THREAD(sts_thread_);
+
+  int r, _status;
+  nsresult rv;
+  int32_t port;
+  nsCString host;
+
+  ReentrantMonitorAutoEnter mon(monitor_);
+
+  if (state_ != NR_INIT) {
+    ABORT(R_INTERNAL);
+  }
+
+  sts_thread_ = do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &rv);
+  if (NS_FAILED(rv)) {
+    MOZ_ASSERT(false, "Failed to get STS thread");
+    ABORT(R_INTERNAL);
+  }
+
+  if ((r = nr_transport_addr_get_addrstring_and_port(addr, &host, &port))) {
+    ABORT(r);
+  }
+
+  // wildcard address will be resolved at NrUdpSocketIpc::CallListenerVoid
+  if ((r = nr_transport_addr_copy(&my_addr_, addr))) {
+    ABORT(r);
+  }
+
+  state_ = NR_CONNECTING;
+
+  MOZ_ASSERT(io_thread_);
+  RUN_ON_THREAD(io_thread_,
+                mozilla::WrapRunnable(RefPtr<NrUdpSocketIpc>(this),
+                                      &NrUdpSocketIpc::create_i, host,
+                                      static_cast<uint16_t>(port)),
+                NS_DISPATCH_NORMAL);
+
+  // Wait until socket creation complete.
+  mon.Wait();
+
+  if (err_) {
+    close();
+    ABORT(R_INTERNAL);
+  }
+
+  state_ = NR_CONNECTED;
+
+  _status = 0;
+abort:
+  return (_status);
+}
+
+int NrUdpSocketIpc::sendto(const void* msg, size_t len, int flags,
+                           const nr_transport_addr* to) {
+  ASSERT_ON_THREAD(sts_thread_);
+
+  ReentrantMonitorAutoEnter mon(monitor_);
+
+  // If send err happened before, simply return the error.
+  if (err_) {
+    return R_IO_ERROR;
+  }
+
+  if (state_ != NR_CONNECTED) {
+    return R_INTERNAL;
+  }
+
+  int r;
+  net::NetAddr addr;
+  if ((r = nr_transport_addr_to_netaddr(to, &addr))) {
+    return r;
+  }
+
+  if (nr_is_stun_request_message((UCHAR*)msg, len) && ShouldDrop(len)) {
+    return R_WOULDBLOCK;
+  }
+
+  UniquePtr<MediaPacket> buf(new MediaPacket);
+  buf->Copy(static_cast<const uint8_t*>(msg), len);
+
+  RUN_ON_THREAD(
+      io_thread_,
+      mozilla::WrapRunnable(RefPtr<NrUdpSocketIpc>(this),
+                            &NrUdpSocketIpc::sendto_i, addr, std::move(buf)),
+      NS_DISPATCH_NORMAL);
+  return 0;
+}
+
+void NrUdpSocketIpc::close() {
+  r_log(LOG_GENERIC, LOG_DEBUG, "NrUdpSocketIpc::close()");
+
+  ASSERT_ON_THREAD(sts_thread_);
+
+  ReentrantMonitorAutoEnter mon(monitor_);
+  state_ = NR_CLOSING;
+
+  RUN_ON_THREAD(io_thread_,
+                mozilla::WrapRunnable(RefPtr<NrUdpSocketIpc>(this),
+                                      &NrUdpSocketIpc::close_i),
+                NS_DISPATCH_NORMAL);
+
+  // remove all enqueued messages
+  std::queue<RefPtr<nr_udp_message>> empty;
+  std::swap(received_msgs_, empty);
+}
+
+int NrUdpSocketIpc::recvfrom(void* buf, size_t maxlen, size_t* len, int flags,
+                             nr_transport_addr* from) {
+  ASSERT_ON_THREAD(sts_thread_);
+
+  ReentrantMonitorAutoEnter mon(monitor_);
+
+  int r, _status;
+  uint32_t consumed_len;
+
+  *len = 0;
+
+  if (state_ != NR_CONNECTED) {
+    ABORT(R_INTERNAL);
+  }
+
+  if (received_msgs_.empty()) {
+    ABORT(R_WOULDBLOCK);
+  }
+
+  {
+    RefPtr<nr_udp_message> msg(received_msgs_.front());
+
+    received_msgs_.pop();
+
+    if ((r = nr_praddr_to_transport_addr(&msg->from, from, IPPROTO_UDP, 0))) {
+      err_ = true;
+      MOZ_ASSERT(false, "Get bogus address for received UDP packet");
+      ABORT(r);
+    }
+
+    consumed_len = std::min(maxlen, msg->data->len());
+    if (consumed_len < msg->data->len()) {
+      r_log(LOG_GENERIC, LOG_DEBUG,
+            "Partial received UDP packet will be discard");
+    }
+
+    memcpy(buf, msg->data->data(), consumed_len);
+    *len = consumed_len;
+  }
+
+  _status = 0;
+abort:
+  return (_status);
+}
+
+int NrUdpSocketIpc::getaddr(nr_transport_addr* addrp) {
+  ASSERT_ON_THREAD(sts_thread_);
+
+  ReentrantMonitorAutoEnter mon(monitor_);
+
+  return nr_transport_addr_copy(addrp, &my_addr_);
+}
+
+int NrUdpSocketIpc::connect(const nr_transport_addr* addr) {
+  int r, _status;
+  int32_t port;
+  nsCString host;
+
+  ReentrantMonitorAutoEnter mon(monitor_);
+  r_log(LOG_GENERIC, LOG_DEBUG, "NrUdpSocketIpc::connect(%s) this=%p",
+        addr->as_string, (void*)this);
+
+  if ((r = nr_transport_addr_get_addrstring_and_port(addr, &host, &port))) {
+    ABORT(r);
+  }
+
+  RUN_ON_THREAD(io_thread_,
+                mozilla::WrapRunnable(RefPtr<NrUdpSocketIpc>(this),
+                                      &NrUdpSocketIpc::connect_i, host,
+                                      static_cast<uint16_t>(port)),
+                NS_DISPATCH_NORMAL);
+
+  // Wait until connect() completes.
+  mon.Wait();
+
+  r_log(LOG_GENERIC, LOG_DEBUG,
+        "NrUdpSocketIpc::connect this=%p completed err_ = %s", (void*)this,
+        err_ ? "true" : "false");
+
+  if (err_) {
+    ABORT(R_INTERNAL);
+  }
+
+  _status = 0;
+abort:
+  return _status;
+}
+
+int NrUdpSocketIpc::write(const void* msg, size_t len, size_t* written) {
+  MOZ_ASSERT(false);
+  return R_INTERNAL;
+}
+
+int NrUdpSocketIpc::read(void* buf, size_t maxlen, size_t* len) {
+  MOZ_ASSERT(false);
+  return R_INTERNAL;
+}
+
+int NrUdpSocketIpc::listen(int backlog) {
+  MOZ_ASSERT(false);
+  return R_INTERNAL;
+}
+
+int NrUdpSocketIpc::accept(nr_transport_addr* addrp, nr_socket** sockp) {
+  MOZ_ASSERT(false);
+  return R_INTERNAL;
+}
+
+// IO thread executors
+void NrUdpSocketIpc::create_i(const nsACString& host, const uint16_t port) {
+  ASSERT_ON_THREAD(io_thread_);
+
+  uint32_t minBuffSize = 0;
+  RefPtr<dom::UDPSocketChild> socketChild = new dom::UDPSocketChild();
+
+  // This can spin the event loop; don't do that with the monitor held
+  socketChild->SetBackgroundSpinsEvents();
+
+  ReentrantMonitorAutoEnter mon(monitor_);
+  if (!socket_child_) {
+    socket_child_ = socketChild;
+    socket_child_->SetFilterName(
+        nsCString(NS_NETWORK_SOCKET_FILTER_HANDLER_STUN_SUFFIX));
+  } else {
+    socketChild = nullptr;
+  }
+
+  RefPtr<NrUdpSocketIpcProxy> proxy(new NrUdpSocketIpcProxy);
+  nsresult rv = proxy->Init(this);
+  if (NS_FAILED(rv)) {
+    err_ = true;
+    mon.NotifyAll();
+    return;
+  }
+
+#ifdef XP_WIN
+  if (!mozilla::IsWin8OrLater()) {
+    // Increase default receive and send buffer size on <= Win7 to be able to
+    // receive and send an unpaced HD (>= 720p = 1280x720 - I Frame ~ 21K size)
+    // stream without losing packets.
+    // Manual testing showed that 100K buffer size was not enough and the
+    // packet loss dis-appeared with 256K buffer size.
+    // See bug 1252769 for future improvements of this.
+    minBuffSize = 256 * 1024;
+  }
+#endif
+  // XXX bug 1126232 - don't use null Principal!
+  if (NS_FAILED(socket_child_->Bind(proxy, nullptr, host, port,
+                                    /* addressReuse = */ false,
+                                    /* loopback = */ false,
+                                    /* recv buffer size */ minBuffSize,
+                                    /* send buffer size */ minBuffSize))) {
+    err_ = true;
+    MOZ_ASSERT(false, "Failed to create UDP socket");
+    mon.NotifyAll();
+    return;
+  }
+}
+
+void NrUdpSocketIpc::connect_i(const nsACString& host, const uint16_t port) {
+  ASSERT_ON_THREAD(io_thread_);
+  nsresult rv;
+  ReentrantMonitorAutoEnter mon(monitor_);
+
+  RefPtr<NrUdpSocketIpcProxy> proxy(new NrUdpSocketIpcProxy);
+  rv = proxy->Init(this);
+  if (NS_FAILED(rv)) {
+    err_ = true;
+    mon.NotifyAll();
+    return;
+  }
+
+  socket_child_->Connect(proxy, host, port);
+}
+
+void NrUdpSocketIpc::sendto_i(const net::NetAddr& addr,
+                              UniquePtr<MediaPacket> buf) {
+  ASSERT_ON_THREAD(io_thread_);
+
+  ReentrantMonitorAutoEnter mon(monitor_);
+
+  if (!socket_child_) {
+    MOZ_ASSERT(false);
+    err_ = true;
+    return;
+  }
+  if (NS_FAILED(
+          socket_child_->SendWithAddress(&addr, buf->data(), buf->len()))) {
+    err_ = true;
+  }
+}
+
+void NrUdpSocketIpc::close_i() {
+  ASSERT_ON_THREAD(io_thread_);
+
+  if (socket_child_) {
+    socket_child_->Close();
+    socket_child_ = nullptr;
+  }
+}
+
+#if defined(MOZILLA_INTERNAL_API)
+
+static void ReleaseIOThread_s() { sThread->ReleaseUse(); }
+
+void NrUdpSocketIpc::destroy_i() {
+  close_i();
+
+  RUN_ON_THREAD(sts_thread_, WrapRunnableNM(&ReleaseIOThread_s),
+                NS_DISPATCH_NORMAL);
+}
+#endif
+
+void NrUdpSocketIpc::recv_callback_s(RefPtr<nr_udp_message> msg) {
+  ASSERT_ON_THREAD(sts_thread_);
+
+  {
+    ReentrantMonitorAutoEnter mon(monitor_);
+    if (state_ != NR_CONNECTED) {
+      return;
+    }
+  }
+
+  // enqueue received message
+  received_msgs_.push(msg);
+
+  if ((poll_flags() & PR_POLL_READ)) {
+    fire_callback(NR_ASYNC_WAIT_READ);
+  }
+}
+
+}  // namespace mozilla
+
+using namespace mozilla;
+
+// Bridge to the nr_socket interface
+static int nr_socket_local_destroy(void** objp);
+static int nr_socket_local_sendto(void* obj, const void* msg, size_t len,
+                                  int flags, const nr_transport_addr* to);
+static int nr_socket_local_recvfrom(void* obj, void* restrict buf,
+                                    size_t maxlen, size_t* len, int flags,
+                                    nr_transport_addr* from);
+static int nr_socket_local_getfd(void* obj, NR_SOCKET* fd);
+static int nr_socket_local_getaddr(void* obj, nr_transport_addr* addrp);
+static int nr_socket_local_close(void* obj);
+static int nr_socket_local_connect(void* obj, const nr_transport_addr* addr);
+static int nr_socket_local_write(void* obj, const void* msg, size_t len,
+                                 size_t* written);
+static int nr_socket_local_read(void* obj, void* restrict buf, size_t maxlen,
+                                size_t* len);
+static int nr_socket_local_listen(void* obj, int backlog);
+static int nr_socket_local_accept(void* obj, nr_transport_addr* addrp,
+                                  nr_socket** sockp);
+
+static nr_socket_vtbl nr_socket_local_vtbl = {2,
+                                              nr_socket_local_destroy,
+                                              nr_socket_local_sendto,
+                                              nr_socket_local_recvfrom,
+                                              nr_socket_local_getfd,
+                                              nr_socket_local_getaddr,
+                                              nr_socket_local_connect,
+                                              nr_socket_local_write,
+                                              nr_socket_local_read,
+                                              nr_socket_local_close,
+                                              nr_socket_local_listen,
+                                              nr_socket_local_accept};
+
+/* static */
+int NrSocketBase::CreateSocket(
+    nr_transport_addr* addr, RefPtr<NrSocketBase>* sock,
+    const std::shared_ptr<NrSocketProxyConfig>& config) {
+  int r, _status;
+
+  if (IsForbiddenAddress(addr)) {
+    ABORT(R_REJECTED);
+  }
+
+  if (config && config->GetForceProxy() && addr->protocol == IPPROTO_UDP) {
+    ABORT(R_REJECTED);
+  }
+
+  // create IPC bridge for content process
+  if (XRE_IsParentProcess()) {
+    // TODO: Make NrTcpSocket work on the parent process
+    *sock = new NrSocket();
+  } else if (XRE_IsSocketProcess()) {
+    if (addr->protocol == IPPROTO_TCP) {
+      *sock = new NrTcpSocket(config);
+    } else {
+      *sock = new NrSocket();
+    }
+  } else {
+    if (addr->protocol == IPPROTO_TCP) {
+      *sock = new NrTcpSocket(config);
+    } else {
+      *sock = new NrUdpSocketIpc();
+    }
+  }
+
+  r = (*sock)->create(addr);
+  if (r) ABORT(r);
+
+  _status = 0;
+abort:
+  if (_status) {
+    *sock = nullptr;
+  }
+  return _status;
+}
+
+// static
+bool NrSocketBase::IsForbiddenAddress(nr_transport_addr* addr) {
+  int r, port;
+
+  r = nr_transport_addr_get_port(addr, &port);
+  if (r) {
+    return true;
+  }
+
+  // allow auto assigned ports
+  if (port != 0) {
+    // Don't need to check an override scheme
+    nsresult rv = NS_CheckPortSafety(port, nullptr);
+    if (NS_FAILED(rv)) {
+      return true;
+    }
+  }
+
+  return false;
+}
+
+static int nr_socket_local_destroy(void** objp) {
+  if (!objp || !*objp) return 0;
+
+  NrSocketBase* sock = static_cast<NrSocketBase*>(*objp);
+  *objp = nullptr;
+
+  sock->close();    // Signal STS that we want not to listen
+  sock->Release();  // Decrement the ref count
+
+  return 0;
+}
+
+static int nr_socket_local_sendto(void* obj, const void* msg, size_t len,
+                                  int flags, const nr_transport_addr* addr) {
+  NrSocketBase* sock = static_cast<NrSocketBase*>(obj);
+
+  return sock->sendto(msg, len, flags, addr);
+}
+
+static int nr_socket_local_recvfrom(void* obj, void* restrict buf,
+                                    size_t maxlen, size_t* len, int flags,
+                                    nr_transport_addr* addr) {
+  NrSocketBase* sock = static_cast<NrSocketBase*>(obj);
+
+  return sock->recvfrom(buf, maxlen, len, flags, addr);
+}
+
+static int nr_socket_local_getfd(void* obj, NR_SOCKET* fd) {
+  NrSocketBase* sock = static_cast<NrSocketBase*>(obj);
+
+  *fd = sock;
+
+  return 0;
+}
+
+static int nr_socket_local_getaddr(void* obj, nr_transport_addr* addrp) {
+  NrSocketBase* sock = static_cast<NrSocketBase*>(obj);
+
+  return sock->getaddr(addrp);
+}
+
+static int nr_socket_local_close(void* obj) {
+  NrSocketBase* sock = static_cast<NrSocketBase*>(obj);
+
+  sock->close();
+
+  return 0;
+}
+
+static int nr_socket_local_write(void* obj, const void* msg, size_t len,
+                                 size_t* written) {
+  NrSocketBase* sock = static_cast<NrSocketBase*>(obj);
+
+  return sock->write(msg, len, written);
+}
+
+static int nr_socket_local_read(void* obj, void* restrict buf, size_t maxlen,
+                                size_t* len) {
+  NrSocketBase* sock = static_cast<NrSocketBase*>(obj);
+
+  return sock->read(buf, maxlen, len);
+}
+
+static int nr_socket_local_connect(void* obj, const nr_transport_addr* addr) {
+  NrSocketBase* sock = static_cast<NrSocketBase*>(obj);
+
+  return sock->connect(addr);
+}
+
+static int nr_socket_local_listen(void* obj, int backlog) {
+  NrSocketBase* sock = static_cast<NrSocketBase*>(obj);
+
+  return sock->listen(backlog);
+}
+
+static int nr_socket_local_accept(void* obj, nr_transport_addr* addrp,
+                                  nr_socket** sockp) {
+  NrSocketBase* sock = static_cast<NrSocketBase*>(obj);
+
+  return sock->accept(addrp, sockp);
+}
+
+// Implement async api
+int NR_async_wait(NR_SOCKET sock, int how, NR_async_cb cb, void* cb_arg,
+                  char* function, int line) {
+  NrSocketBase* s = static_cast<NrSocketBase*>(sock);
+
+  return s->async_wait(how, cb, cb_arg, function, line);
+}
+
+int NR_async_cancel(NR_SOCKET sock, int how) {
+  NrSocketBase* s = static_cast<NrSocketBase*>(sock);
+
+  return s->cancel(how);
+}
+
+nr_socket_vtbl* NrSocketBase::vtbl() { return &nr_socket_local_vtbl; }
diff --git a/dom/media/webrtc/transport/nr_socket_prsock.h b/dom/media/webrtc/transport/nr_socket_prsock.h
new file mode 100644
index 0000000000..010fcb59bc
--- /dev/null
+++ b/dom/media/webrtc/transport/nr_socket_prsock.h
@@ -0,0 +1,320 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+/* Some source code here from nICEr. Copyright is:
+
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+// Implementation of nICEr/nr_socket that is tied to the Gecko
+// SocketTransportService.
+
+#ifndef nr_socket_prsock__
+#define nr_socket_prsock__
+
+#include <memory>
+#include <queue>
+
+#include "nspr.h"
+#include "prio.h"
+
+#include "nsCOMPtr.h"
+#include "nsASocketHandler.h"
+#include "nsXPCOM.h"
+#include "nsIEventTarget.h"
+#include "nsIUDPSocketChild.h"
+#include "nsProxyRelease.h"
+#include "nsThreadUtils.h"
+
+#include "mediapacket.h"
+#include "m_cpp_utils.h"
+#include "mozilla/ReentrantMonitor.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/TimeStamp.h"
+#include "mozilla/ClearOnShutdown.h"
+
+// nICEr includes
+extern "C" {
+#include "transport_addr.h"
+#include "async_wait.h"
+}
+
+// Stub declaration for nICEr type
+typedef struct nr_socket_vtbl_ nr_socket_vtbl;
+typedef struct nr_socket_ nr_socket;
+
+#if defined(MOZILLA_INTERNAL_API)
+namespace mozilla {
+class NrSocketProxyConfig;
+}  // namespace mozilla
+#endif
+
+namespace mozilla {
+
+namespace net {
+union NetAddr;
+}
+
+namespace dom {
+class UDPSocketChild;
+}  // namespace dom
+
+class NrSocketBase {
+ public:
+  NrSocketBase() : connect_invoked_(false), poll_flags_(0) {
+    memset(cbs_, 0, sizeof(cbs_));
+    memset(cb_args_, 0, sizeof(cb_args_));
+    memset(&my_addr_, 0, sizeof(my_addr_));
+  }
+  virtual ~NrSocketBase() = default;
+
+  // Factory method; will create either an NrSocket, NrUdpSocketIpc, or
+  // NrTcpSocketIpc as appropriate.
+  static int CreateSocket(nr_transport_addr* addr, RefPtr<NrSocketBase>* sock,
+                          const std::shared_ptr<NrSocketProxyConfig>& config);
+  static bool IsForbiddenAddress(nr_transport_addr* addr);
+
+  // the nr_socket APIs
+  virtual int create(nr_transport_addr* addr) = 0;
+  virtual int sendto(const void* msg, size_t len, int flags,
+                     const nr_transport_addr* to) = 0;
+  virtual int recvfrom(void* buf, size_t maxlen, size_t* len, int flags,
+                       nr_transport_addr* from) = 0;
+  virtual int getaddr(nr_transport_addr* addrp) = 0;
+  virtual void close() = 0;
+  virtual int connect(const nr_transport_addr* addr) = 0;
+  virtual int write(const void* msg, size_t len, size_t* written) = 0;
+  virtual int read(void* buf, size_t maxlen, size_t* len) = 0;
+  virtual int listen(int backlog) = 0;
+  virtual int accept(nr_transport_addr* addrp, nr_socket** sockp) = 0;
+
+  // Implementations of the async_event APIs
+  virtual int async_wait(int how, NR_async_cb cb, void* cb_arg, char* function,
+                         int line);
+  virtual int cancel(int how);
+
+  // nsISupport reference counted interface
+  NS_INLINE_DECL_PURE_VIRTUAL_REFCOUNTING
+
+  uint32_t poll_flags() { return poll_flags_; }
+
+  virtual nr_socket_vtbl* vtbl();  // To access in test classes.
+
+  static TimeStamp short_term_violation_time();
+  static TimeStamp long_term_violation_time();
+  const nr_transport_addr& my_addr() const { return my_addr_; }
+
+  void fire_callback(int how);
+
+ protected:
+  bool connect_invoked_;
+  nr_transport_addr my_addr_;
+
+ private:
+  NR_async_cb cbs_[NR_ASYNC_WAIT_WRITE + 1];
+  void* cb_args_[NR_ASYNC_WAIT_WRITE + 1];
+  uint32_t poll_flags_;
+};
+
+class NrSocket : public NrSocketBase, public nsASocketHandler {
+ public:
+  NrSocket() : fd_(nullptr) {}
+
+  // Implement nsASocket
+  virtual void OnSocketReady(PRFileDesc* fd, int16_t outflags) override;
+  virtual void OnSocketDetached(PRFileDesc* fd) override;
+  virtual void IsLocal(bool* aIsLocal) override;
+  virtual uint64_t ByteCountSent() override { return 0; }
+  virtual uint64_t ByteCountReceived() override { return 0; }
+
+  // nsISupports methods
+  NS_IMETHOD QueryInterface(REFNSIID aIID, void** aInstancePtr) override;
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING_WITH_DESTROY(NrSocket, Destroy(),
+                                                     override);
+  virtual void Destroy() { delete this; }
+
+  // Implementations of the async_event APIs
+  virtual int async_wait(int how, NR_async_cb cb, void* cb_arg, char* function,
+                         int line) override;
+  virtual int cancel(int how) override;
+
+  // Implementations of the nr_socket APIs
+  virtual int create(nr_transport_addr* addr)
+      override;  // (really init, but it's called create)
+  virtual int sendto(const void* msg, size_t len, int flags,
+                     const nr_transport_addr* to) override;
+  virtual int recvfrom(void* buf, size_t maxlen, size_t* len, int flags,
+                       nr_transport_addr* from) override;
+  virtual int getaddr(nr_transport_addr* addrp) override;
+  virtual void close() override;
+  virtual int connect(const nr_transport_addr* addr) override;
+  virtual int write(const void* msg, size_t len, size_t* written) override;
+  virtual int read(void* buf, size_t maxlen, size_t* len) override;
+  virtual int listen(int backlog) override;
+  virtual int accept(nr_transport_addr* addrp, nr_socket** sockp) override;
+
+ protected:
+  virtual ~NrSocket() {
+    if (fd_) PR_Close(fd_);
+  }
+
+  DISALLOW_COPY_ASSIGN(NrSocket);
+
+  PRFileDesc* fd_;
+  nsCOMPtr<nsIEventTarget> ststhread_;
+};
+
+struct nr_udp_message {
+  nr_udp_message(const PRNetAddr& from, UniquePtr<MediaPacket>&& data)
+      : from(from), data(std::move(data)) {}
+
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(nr_udp_message);
+
+  PRNetAddr from;
+  UniquePtr<MediaPacket> data;
+
+ private:
+  ~nr_udp_message() = default;
+  DISALLOW_COPY_ASSIGN(nr_udp_message);
+};
+
+class NrSocketIpc : public NrSocketBase {
+ public:
+  enum NrSocketIpcState {
+    NR_INIT,
+    NR_CONNECTING,
+    NR_CONNECTED,
+    NR_CLOSING,
+    NR_CLOSED,
+  };
+
+  NrSocketIpc(nsIEventTarget* aThread);
+
+ protected:
+  nsCOMPtr<nsIEventTarget> sts_thread_;
+  // Note: for UDP PBackground, this is a thread held by SingletonThreadHolder.
+  // For TCP PNecko, this is MainThread (and TCPSocket requires MainThread
+  // currently)
+  const nsCOMPtr<nsIEventTarget> io_thread_;
+  virtual ~NrSocketIpc() = default;
+
+ private:
+  DISALLOW_COPY_ASSIGN(NrSocketIpc);
+};
+
+class NrUdpSocketIpc : public NrSocketIpc {
+ public:
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(NrUdpSocketIpc, override)
+
+  NS_IMETHODIMP CallListenerError(const nsACString& message,
+                                  const nsACString& filename,
+                                  uint32_t line_number);
+  NS_IMETHODIMP CallListenerReceivedData(const nsACString& host, uint16_t port,
+                                         const nsTArray<uint8_t>& data);
+  NS_IMETHODIMP CallListenerOpened();
+  NS_IMETHODIMP CallListenerConnected();
+  NS_IMETHODIMP CallListenerClosed();
+
+  NrUdpSocketIpc();
+
+  // Implementations of the NrSocketBase APIs
+  virtual int create(nr_transport_addr* addr) override;
+  virtual int sendto(const void* msg, size_t len, int flags,
+                     const nr_transport_addr* to) override;
+  virtual int recvfrom(void* buf, size_t maxlen, size_t* len, int flags,
+                       nr_transport_addr* from) override;
+  virtual int getaddr(nr_transport_addr* addrp) override;
+  virtual void close() override;
+  virtual int connect(const nr_transport_addr* addr) override;
+  virtual int write(const void* msg, size_t len, size_t* written) override;
+  virtual int read(void* buf, size_t maxlen, size_t* len) override;
+  virtual int listen(int backlog) override;
+  virtual int accept(nr_transport_addr* addrp, nr_socket** sockp) override;
+
+ private:
+  virtual ~NrUdpSocketIpc();
+  virtual void Destroy();
+
+  DISALLOW_COPY_ASSIGN(NrUdpSocketIpc);
+
+  nsresult SetAddress();  // Set the local address from parent info.
+
+  // Main or private thread executors of the NrSocketBase APIs
+  void create_i(const nsACString& host, const uint16_t port);
+  void connect_i(const nsACString& host, const uint16_t port);
+  void sendto_i(const net::NetAddr& addr, UniquePtr<MediaPacket> buf);
+  void close_i();
+#if defined(MOZILLA_INTERNAL_API) && !defined(MOZILLA_XPCOMRT_API)
+  void destroy_i();
+#endif
+  // STS thread executor
+  void recv_callback_s(RefPtr<nr_udp_message> msg);
+
+  ReentrantMonitor monitor_ MOZ_UNANNOTATED;  // protects err_and state_
+  bool err_;
+  NrSocketIpcState state_;
+
+  std::queue<RefPtr<nr_udp_message>> received_msgs_;
+
+  // only accessed from the io_thread
+  RefPtr<dom::UDPSocketChild> socket_child_;
+};
+
+// The socket child holds onto one of these, which just passes callbacks
+// through and makes sure the ref to the NrSocketIpc is released on STS.
+class NrUdpSocketIpcProxy : public nsIUDPSocketInternal {
+ public:
+  NS_DECL_THREADSAFE_ISUPPORTS
+  NS_DECL_NSIUDPSOCKETINTERNAL
+
+  nsresult Init(const RefPtr<NrUdpSocketIpc>& socket);
+
+ private:
+  virtual ~NrUdpSocketIpcProxy();
+
+  RefPtr<NrUdpSocketIpc> socket_;
+  nsCOMPtr<nsIEventTarget> sts_thread_;
+};
+
+int nr_netaddr_to_transport_addr(const net::NetAddr* netaddr,
+                                 nr_transport_addr* addr, int protocol);
+int nr_praddr_to_transport_addr(const PRNetAddr* praddr,
+                                nr_transport_addr* addr, int protocol,
+                                int keep);
+int nr_transport_addr_get_addrstring_and_port(const nr_transport_addr* addr,
+                                              nsACString* host, int32_t* port);
+}  // namespace mozilla
+#endif
diff --git a/dom/media/webrtc/transport/nr_socket_tcp.cpp b/dom/media/webrtc/transport/nr_socket_tcp.cpp
new file mode 100644
index 0000000000..a9cc5db312
--- /dev/null
+++ b/dom/media/webrtc/transport/nr_socket_tcp.cpp
@@ -0,0 +1,310 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set sw=2 ts=8 et tw=80 ft=cpp : */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+Copyright (c) 2013, Mozilla
+
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in the
+ documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance, Mozilla nor
+ the names of its contributors may be used to endorse or promote
+ products derived from this software without specific prior written
+ permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include "nr_socket_tcp.h"
+
+#include "mozilla/ErrorNames.h"
+
+#include "WebrtcTCPSocketWrapper.h"
+
+namespace mozilla {
+using namespace net;
+
+using std::shared_ptr;
+
+class NrTcpSocketData {
+ public:
+  explicit NrTcpSocketData(nsTArray<uint8_t>&& aData)
+      : mData(std::move(aData)) {
+    MOZ_COUNT_CTOR(NrTcpSocketData);
+  }
+
+  MOZ_COUNTED_DTOR(NrTcpSocketData)
+
+  const nsTArray<uint8_t>& GetData() const { return mData; }
+
+ private:
+  nsTArray<uint8_t> mData;
+};
+
+NrTcpSocket::NrTcpSocket(const shared_ptr<NrSocketProxyConfig>& aConfig)
+    : mClosed(false),
+      mReadOffset(0),
+      mConfig(aConfig),
+      mWebrtcTCPSocket(nullptr) {
+  r_log(LOG_GENERIC, LOG_DEBUG, "NrTcpSocket::NrTcpSocket %p\n", this);
+}
+
+NrTcpSocket::~NrTcpSocket() {
+  r_log(LOG_GENERIC, LOG_DEBUG, "NrTcpSocket::~NrTcpSocket %p\n", this);
+  MOZ_ASSERT(!mWebrtcTCPSocket, "webrtc TCP socket not null");
+}
+
+int NrTcpSocket::create(nr_transport_addr* aAddr) {
+  r_log(LOG_GENERIC, LOG_DEBUG, "NrTcpSocket::create %p\n", this);
+  int32_t port;
+  nsCString host;
+
+  // Sanity check
+  if (nr_transport_addr_get_addrstring_and_port(aAddr, &host, &port)) {
+    return R_FAILED;
+  }
+
+  if (nr_transport_addr_copy(&my_addr_, aAddr)) {
+    return R_FAILED;
+  }
+
+  return 0;
+}
+
+int NrTcpSocket::connect(const nr_transport_addr* aAddr) {
+  r_log(LOG_GENERIC, LOG_DEBUG, "NrTcpSocket::connect %p\n", this);
+
+  nsCString remote_host;
+  int remote_port;
+
+  if (NS_WARN_IF(nr_transport_addr_get_addrstring_and_port(aAddr, &remote_host,
+                                                           &remote_port))) {
+    return R_FAILED;
+  }
+
+  bool use_tls = aAddr->tls;
+
+  nsCString local_addr;
+  int local_port;
+
+  if (NS_WARN_IF(nr_transport_addr_get_addrstring_and_port(
+          &my_addr_, &local_addr, &local_port))) {
+    return R_FAILED;
+  }
+
+  mWebrtcTCPSocket = new WebrtcTCPSocketWrapper(this);
+
+  mWebrtcTCPSocket->AsyncOpen(remote_host, remote_port, local_addr, local_port,
+                              use_tls, mConfig);
+
+  // trigger nr_socket_buffered to set write/read callback
+  return R_WOULDBLOCK;
+}
+
+void NrTcpSocket::close() {
+  r_log(LOG_GENERIC, LOG_DEBUG, "NrTcpSocket::close %p\n", this);
+
+  if (mClosed) {
+    return;
+  }
+
+  mClosed = true;
+
+  // We're not always open at this point.
+  if (mWebrtcTCPSocket) {
+    mWebrtcTCPSocket->Close();
+    mWebrtcTCPSocket = nullptr;
+  }
+}
+
+int NrTcpSocket::write(const void* aBuffer, size_t aCount, size_t* aWrote) {
+  r_log(LOG_GENERIC, LOG_DEBUG, "NrTcpSocket::write %p count=%zu\n", this,
+        aCount);
+
+  if (mClosed) {
+    return R_FAILED;
+  }
+
+  if (!aWrote) {
+    return R_FAILED;
+  }
+
+  if (NS_WARN_IF(!mWebrtcTCPSocket)) {
+    return R_FAILED;
+  }
+
+  *aWrote = aCount;
+
+  if (aCount > 0) {
+    nsTArray<uint8_t> writeData;
+    writeData.SetLength(aCount);
+    memcpy(writeData.Elements(), aBuffer, aCount);
+
+    mWebrtcTCPSocket->SendWrite(std::move(writeData));
+  }
+
+  return 0;
+}
+
+int NrTcpSocket::read(void* aBuffer, size_t aCount, size_t* aRead) {
+  r_log(LOG_GENERIC, LOG_DEBUG, "NrTcpSocket::read %p\n", this);
+
+  if (mClosed) {
+    return R_FAILED;
+  }
+
+  if (!aRead) {
+    return R_FAILED;
+  }
+
+  *aRead = 0;
+
+  if (mReadQueue.empty()) {
+    return R_WOULDBLOCK;
+  }
+
+  while (aCount > 0 && !mReadQueue.empty()) {
+    const NrTcpSocketData& data = mReadQueue.front();
+
+    size_t remainingCount = data.GetData().Length() - mReadOffset;
+    size_t amountToCopy = std::min(aCount, remainingCount);
+
+    char* buffer = static_cast<char*>(aBuffer) + (*aRead);
+
+    memcpy(buffer, data.GetData().Elements() + mReadOffset, amountToCopy);
+
+    mReadOffset += amountToCopy;
+    *aRead += amountToCopy;
+    aCount -= amountToCopy;
+
+    if (remainingCount == amountToCopy) {
+      mReadOffset = 0;
+      mReadQueue.pop_front();
+    }
+  }
+
+  return 0;
+}
+
+int NrTcpSocket::getaddr(nr_transport_addr* aAddr) {
+  r_log(LOG_GENERIC, LOG_DEBUG, "NrTcpSocket::getaddr %p\n", this);
+  return nr_transport_addr_copy(aAddr, &my_addr_);
+}
+
+int NrTcpSocket::sendto(const void* aBuffer, size_t aCount, int aFlags,
+                        const nr_transport_addr* aAddr) {
+  // never call this
+  MOZ_ASSERT(0);
+  return R_FAILED;
+}
+
+int NrTcpSocket::recvfrom(void* aBuffer, size_t aCount, size_t* aRead,
+                          int aFlags, nr_transport_addr* aAddr) {
+  // never call this
+  MOZ_ASSERT(0);
+  return R_FAILED;
+}
+
+int NrTcpSocket::listen(int aBacklog) {
+  r_log(LOG_GENERIC, LOG_DEBUG, "NrTcpSocket::listen %p\n", this);
+  return R_INTERNAL;
+}
+
+int NrTcpSocket::accept(nr_transport_addr* aAddr, nr_socket** aSocket) {
+  r_log(LOG_GENERIC, LOG_DEBUG, "NrTcpSocket::accept %p\n", this);
+  return R_INTERNAL;
+}
+
+// WebrtcTCPSocketCallback
+void NrTcpSocket::OnClose(nsresult aReason) {
+  nsCString errorName;
+  GetErrorName(aReason, errorName);
+
+  r_log(LOG_GENERIC, LOG_ERR, "NrTcpSocket::OnClose %p reason=%u name=%s\n",
+        this, static_cast<uint32_t>(aReason), errorName.get());
+
+  close();
+
+  DoCallbacks();
+}
+
+void NrTcpSocket::OnConnected(const nsACString& aProxyType) {
+  r_log(LOG_GENERIC, LOG_DEBUG, "NrTcpSocket::OnConnected %p\n", this);
+  if (aProxyType != "" && aProxyType != "direct") {
+    my_addr_.is_proxied = true;
+  }
+
+  DoCallbacks();
+}
+
+void NrTcpSocket::OnRead(nsTArray<uint8_t>&& aReadData) {
+  r_log(LOG_GENERIC, LOG_DEBUG, "NrTcpSocket::OnRead %p read=%zu\n", this,
+        aReadData.Length());
+
+  mReadQueue.emplace_back(std::move(aReadData));
+
+  DoCallbacks();
+}
+
+void NrTcpSocket::DoCallbacks() {
+  size_t lastCount = -1;
+  size_t currentCount = 0;
+  while ((poll_flags() & PR_POLL_READ) != 0 &&
+         // Make sure whatever is reading knows we're closed. This doesn't need
+         // to happen for writes since ICE doesn't like a failing writable.
+         (mClosed || (currentCount = CountUnreadBytes()) > 0) &&
+         lastCount != currentCount) {
+    fire_callback(NR_ASYNC_WAIT_READ);
+    lastCount = currentCount;
+  }
+
+  // We're always ready to write after we're connected. The parent process will
+  // buffer writes for us.
+  if (!mClosed && mWebrtcTCPSocket && (poll_flags() & PR_POLL_WRITE) != 0) {
+    fire_callback(NR_ASYNC_WAIT_WRITE);
+  }
+}
+
+size_t NrTcpSocket::CountUnreadBytes() const {
+  size_t count = 0;
+
+  for (const NrTcpSocketData& data : mReadQueue) {
+    count += data.GetData().Length();
+  }
+
+  MOZ_ASSERT(count >= mReadOffset, "offset exceeds read buffer length");
+
+  count -= mReadOffset;
+
+  return count;
+}
+
+void NrTcpSocket::AssignChannel_DoNotUse(WebrtcTCPSocketWrapper* aWrapper) {
+  mWebrtcTCPSocket = aWrapper;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/transport/nr_socket_tcp.h b/dom/media/webrtc/transport/nr_socket_tcp.h
new file mode 100644
index 0000000000..ebf806a924
--- /dev/null
+++ b/dom/media/webrtc/transport/nr_socket_tcp.h
@@ -0,0 +1,117 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set sw=2 ts=8 et tw=80 ft=cpp : */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+Copyright (c) 2013, Mozilla
+
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+ notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+ notice, this list of conditions and the following disclaimer in the
+ documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance, Mozilla nor
+ the names of its contributors may be used to endorse or promote
+ products derived from this software without specific prior written
+ permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef nr_socket_tcp_h__
+#define nr_socket_tcp_h__
+
+#include <list>
+
+#include "mozilla/net/WebrtcTCPSocketCallback.h"
+
+#include "nsTArray.h"
+
+extern "C" {
+#include "nr_api.h"
+#include "nr_socket.h"
+#include "transport_addr.h"
+}
+
+#include "nr_socket_prsock.h"
+
+namespace mozilla {
+using namespace net;
+
+namespace net {
+class WebrtcTCPSocketWrapper;
+}  // namespace net
+
+class NrTcpSocketData;
+class NrSocketProxyConfig;
+
+class NrTcpSocket : public NrSocketBase, public WebrtcTCPSocketCallback {
+ public:
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(NrTcpSocket, override)
+
+  explicit NrTcpSocket(const std::shared_ptr<NrSocketProxyConfig>& aConfig);
+
+  // NrSocketBase
+  int create(nr_transport_addr* aAddr) override;
+  int connect(const nr_transport_addr* aAddr) override;
+  void close() override;
+  int write(const void* aBuffer, size_t aCount, size_t* aWrote) override;
+  int read(void* aBuffer, size_t aCount, size_t* aRead) override;
+  int getaddr(nr_transport_addr* aAddr) override;
+  int sendto(const void* aBuffer, size_t aCount, int aFlags,
+             const nr_transport_addr* aAddr) override;
+  int recvfrom(void* aBuffer, size_t aCount, size_t* aRead, int aFlags,
+               nr_transport_addr* aAddr) override;
+  int listen(int aBacklog) override;
+  int accept(nr_transport_addr* aAddr, nr_socket** aSocket) override;
+
+  // WebrtcTCPSocketCallback
+  void OnClose(nsresult aReason) override;
+  void OnConnected(const nsACString& aProxyType) override;
+  void OnRead(nsTArray<uint8_t>&& aReadData) override;
+
+  size_t CountUnreadBytes() const;
+
+  // for gtests
+  void AssignChannel_DoNotUse(WebrtcTCPSocketWrapper* aWrapper);
+
+ protected:
+  virtual ~NrTcpSocket();
+
+ private:
+  void DoCallbacks();
+
+  bool mClosed;
+
+  size_t mReadOffset;
+  std::list<NrTcpSocketData> mReadQueue;
+
+  std::shared_ptr<NrSocketProxyConfig> mConfig;
+
+  RefPtr<WebrtcTCPSocketWrapper> mWebrtcTCPSocket;
+};
+
+}  // namespace mozilla
+
+#endif  // nr_socket_tcp_h__
diff --git a/dom/media/webrtc/transport/nr_timer.cpp b/dom/media/webrtc/transport/nr_timer.cpp
new file mode 100644
index 0000000000..3d5671c7a4
--- /dev/null
+++ b/dom/media/webrtc/transport/nr_timer.cpp
@@ -0,0 +1,256 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original code by: ekr@rtfm.com
+
+// Implementation of the NR timer interface
+
+// Some code here copied from nrappkit. The license was.
+
+/**
+   Copyright (C) 2004, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   ekr@rtfm.com  Sun Feb 22 19:35:24 2004
+ */
+
+#include <string>
+
+#include "nsCOMPtr.h"
+#include "nsServiceManagerUtils.h"
+#include "nsIEventTarget.h"
+#include "nsINamed.h"
+#include "nsITimer.h"
+#include "nsNetCID.h"
+#include "runnable_utils.h"
+#include "mozilla/DebugOnly.h"
+#include "mozilla/UniquePtr.h"
+
+extern "C" {
+#include "async_wait.h"
+#include "async_timer.h"
+#include "r_errors.h"
+#include "r_log.h"
+}
+
+namespace mozilla {
+
+class nrappkitCallback {
+ public:
+  nrappkitCallback(NR_async_cb cb, void* cb_arg, const char* function, int line)
+      : cb_(cb), cb_arg_(cb_arg), function_(function), line_(line) {}
+  virtual ~nrappkitCallback() = default;
+
+  virtual void Cancel() = 0;
+
+ protected:
+  /* additional members */
+  NR_async_cb cb_;
+  void* cb_arg_;
+  std::string function_;
+  int line_;
+};
+
+class nrappkitTimerCallback : public nrappkitCallback,
+                              public nsITimerCallback,
+                              public nsINamed {
+ public:
+  // We're going to release ourself in the callback, so we need to be threadsafe
+  NS_DECL_THREADSAFE_ISUPPORTS
+  NS_DECL_NSITIMERCALLBACK
+
+  nrappkitTimerCallback(NR_async_cb cb, void* cb_arg, const char* function,
+                        int line)
+      : nrappkitCallback(cb, cb_arg, function, line), timer_(nullptr) {}
+
+  void SetTimer(already_AddRefed<nsITimer>&& timer) { timer_ = timer; }
+
+  virtual void Cancel() override {
+    AddRef();  // Cancelling the timer causes the callback it holds to
+               // be released. AddRef() keeps us alive.
+    timer_->Cancel();
+    timer_ = nullptr;
+    Release();  // Will cause deletion of this object.
+  }
+
+  NS_IMETHOD
+  GetName(nsACString& aName) override {
+    aName.AssignLiteral("nrappkitTimerCallback");
+    return NS_OK;
+  }
+
+ private:
+  nsCOMPtr<nsITimer> timer_;
+  virtual ~nrappkitTimerCallback() = default;
+};
+
+NS_IMPL_ISUPPORTS(nrappkitTimerCallback, nsITimerCallback, nsINamed)
+
+NS_IMETHODIMP nrappkitTimerCallback::Notify(nsITimer* timer) {
+  r_log(LOG_GENERIC, LOG_DEBUG, "Timer callback fired (set in %s:%d)",
+        function_.c_str(), line_);
+  MOZ_RELEASE_ASSERT(timer == timer_);
+  cb_(nullptr, 0, cb_arg_);
+
+  // Allow the timer to go away.
+  timer_ = nullptr;
+  return NS_OK;
+}
+
+class nrappkitScheduledCallback : public nrappkitCallback {
+ public:
+  nrappkitScheduledCallback(NR_async_cb cb, void* cb_arg, const char* function,
+                            int line)
+      : nrappkitCallback(cb, cb_arg, function, line) {}
+
+  void Run() {
+    if (cb_) {
+      cb_(nullptr, 0, cb_arg_);
+    }
+  }
+
+  virtual void Cancel() override { cb_ = nullptr; }
+
+  ~nrappkitScheduledCallback() = default;
+};
+
+}  // namespace mozilla
+
+using namespace mozilla;
+
+static nsCOMPtr<nsIEventTarget> GetSTSThread() {
+  nsresult rv;
+
+  nsCOMPtr<nsIEventTarget> sts_thread;
+
+  sts_thread = do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &rv);
+  MOZ_ASSERT(NS_SUCCEEDED(rv));
+
+  return sts_thread;
+}
+
+// These timers must only be used from the STS thread.
+// This function is a helper that enforces that.
+static void CheckSTSThread() {
+  DebugOnly<nsCOMPtr<nsIEventTarget>> sts_thread = GetSTSThread();
+
+  ASSERT_ON_THREAD(sts_thread.value);
+}
+
+static int nr_async_timer_set_zero(NR_async_cb cb, void* arg, char* func, int l,
+                                   nrappkitCallback** handle) {
+  nrappkitScheduledCallback* callback(
+      new nrappkitScheduledCallback(cb, arg, func, l));
+
+  nsresult rv = GetSTSThread()->Dispatch(
+      WrapRunnable(UniquePtr<nrappkitScheduledCallback>(callback),
+                   &nrappkitScheduledCallback::Run),
+      NS_DISPATCH_NORMAL);
+  if (NS_FAILED(rv)) return R_FAILED;
+
+  *handle = callback;
+
+  // On exit to this function, the only strong reference to callback is in
+  // the Runnable. Because we are redispatching to the same thread,
+  // this is always safe.
+  return 0;
+}
+
+static int nr_async_timer_set_nonzero(int timeout, NR_async_cb cb, void* arg,
+                                      char* func, int l,
+                                      nrappkitCallback** handle) {
+  nsresult rv;
+  CheckSTSThread();
+
+  nrappkitTimerCallback* callback = new nrappkitTimerCallback(cb, arg, func, l);
+
+  nsCOMPtr<nsITimer> timer;
+  rv = NS_NewTimerWithCallback(getter_AddRefs(timer), callback, timeout,
+                               nsITimer::TYPE_ONE_SHOT);
+  if (NS_FAILED(rv)) {
+    return R_FAILED;
+  }
+
+  // Move the ownership of the timer to the callback object, which holds the
+  // timer alive per spec.
+  callback->SetTimer(timer.forget());
+
+  *handle = callback;
+
+  return 0;
+}
+
+int NR_async_timer_set(int timeout, NR_async_cb cb, void* arg, char* func,
+                       int l, void** handle) {
+  CheckSTSThread();
+
+  nrappkitCallback* callback;
+  int r;
+
+  if (!timeout) {
+    r = nr_async_timer_set_zero(cb, arg, func, l, &callback);
+  } else {
+    r = nr_async_timer_set_nonzero(timeout, cb, arg, func, l, &callback);
+  }
+
+  if (r) return r;
+
+  if (handle) *handle = callback;
+
+  return 0;
+}
+
+int NR_async_schedule(NR_async_cb cb, void* arg, char* func, int l) {
+  // No need to check the thread because we check it next in the
+  // timer set.
+  return NR_async_timer_set(0, cb, arg, func, l, nullptr);
+}
+
+int NR_async_timer_cancel(void* handle) {
+  // Check for the handle being nonzero because sometimes we get
+  // no-op cancels that aren't on the STS thread. This can be
+  // non-racy as long as the upper-level code is careful.
+  if (!handle) return 0;
+
+  CheckSTSThread();
+
+  nrappkitCallback* callback = static_cast<nrappkitCallback*>(handle);
+  callback->Cancel();
+
+  return 0;
+}
diff --git a/dom/media/webrtc/transport/nricectx.cpp b/dom/media/webrtc/transport/nricectx.cpp
new file mode 100644
index 0000000000..697eecdb07
--- /dev/null
+++ b/dom/media/webrtc/transport/nricectx.cpp
@@ -0,0 +1,1106 @@
+/* -*- mode: c++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+// Some of this code is cut-and-pasted from nICEr. Copyright is:
+
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <string>
+#include <vector>
+
+#include "nr_socket_proxy_config.h"
+#include "nsXULAppAPI.h"
+
+#include "logging.h"
+#include "pk11pub.h"
+#include "plbase64.h"
+
+#include "nsCOMPtr.h"
+#include "nsError.h"
+#include "nsNetCID.h"
+#include "nsServiceManagerUtils.h"
+#include "ScopedNSSTypes.h"
+#include "runnable_utils.h"
+#include "nsIUUIDGenerator.h"
+
+// nICEr includes
+extern "C" {
+#include "nr_api.h"
+#include "registry.h"
+#include "async_timer.h"
+#include "r_crc32.h"
+#include "r_memory.h"
+#include "ice_reg.h"
+#include "transport_addr.h"
+#include "nr_crypto.h"
+#include "nr_socket.h"
+#include "nr_socket_local.h"
+#include "stun_reg.h"
+#include "stun_util.h"
+#include "ice_codeword.h"
+#include "ice_ctx.h"
+#include "ice_candidate.h"
+}
+
+// Local includes
+#include "nricectx.h"
+#include "nricemediastream.h"
+#include "nr_socket_prsock.h"
+#include "nrinterfaceprioritizer.h"
+#include "rlogconnector.h"
+#include "test_nr_socket.h"
+
+namespace mozilla {
+
+using std::shared_ptr;
+
+TimeStamp nr_socket_short_term_violation_time() {
+  return NrSocketBase::short_term_violation_time();
+}
+
+TimeStamp nr_socket_long_term_violation_time() {
+  return NrSocketBase::long_term_violation_time();
+}
+
+MOZ_MTLOG_MODULE("mtransport")
+
+const char kNrIceTransportUdp[] = "udp";
+const char kNrIceTransportTcp[] = "tcp";
+const char kNrIceTransportTls[] = "tls";
+
+static bool initialized = false;
+
+static int noop(void** obj) { return 0; }
+
+static nr_socket_factory_vtbl ctx_socket_factory_vtbl = {nr_socket_local_create,
+                                                         noop};
+
+// Implement NSPR-based crypto algorithms
+static int nr_crypto_nss_random_bytes(UCHAR* buf, size_t len) {
+  UniquePK11SlotInfo slot(PK11_GetInternalSlot());
+  if (!slot) return R_INTERNAL;
+
+  SECStatus rv = PK11_GenerateRandomOnSlot(slot.get(), buf, len);
+  if (rv != SECSuccess) return R_INTERNAL;
+
+  return 0;
+}
+
+static int nr_crypto_nss_hmac(UCHAR* key, size_t keyl, UCHAR* buf, size_t bufl,
+                              UCHAR* result) {
+  CK_MECHANISM_TYPE mech = CKM_SHA_1_HMAC;
+  PK11SlotInfo* slot = nullptr;
+  MOZ_ASSERT(keyl > 0);
+  SECItem keyi = {siBuffer, key, static_cast<unsigned int>(keyl)};
+  PK11SymKey* skey = nullptr;
+  PK11Context* hmac_ctx = nullptr;
+  SECStatus status;
+  unsigned int hmac_len;
+  SECItem param = {siBuffer, nullptr, 0};
+  int err = R_INTERNAL;
+
+  slot = PK11_GetInternalKeySlot();
+  if (!slot) goto abort;
+
+  skey = PK11_ImportSymKey(slot, mech, PK11_OriginUnwrap, CKA_SIGN, &keyi,
+                           nullptr);
+  if (!skey) goto abort;
+
+  hmac_ctx = PK11_CreateContextBySymKey(mech, CKA_SIGN, skey, &param);
+  if (!hmac_ctx) goto abort;
+
+  status = PK11_DigestBegin(hmac_ctx);
+  if (status != SECSuccess) goto abort;
+
+  status = PK11_DigestOp(hmac_ctx, buf, bufl);
+  if (status != SECSuccess) goto abort;
+
+  status = PK11_DigestFinal(hmac_ctx, result, &hmac_len, 20);
+  if (status != SECSuccess) goto abort;
+
+  MOZ_ASSERT(hmac_len == 20);
+
+  err = 0;
+
+abort:
+  if (hmac_ctx) PK11_DestroyContext(hmac_ctx, PR_TRUE);
+  if (skey) PK11_FreeSymKey(skey);
+  if (slot) PK11_FreeSlot(slot);
+
+  return err;
+}
+
+static int nr_crypto_nss_md5(UCHAR* buf, size_t bufl, UCHAR* result) {
+  int err = R_INTERNAL;
+  SECStatus rv;
+
+  const SECHashObject* ho = HASH_GetHashObject(HASH_AlgMD5);
+  MOZ_ASSERT(ho);
+  if (!ho) goto abort;
+
+  MOZ_ASSERT(ho->length == 16);
+
+  rv = HASH_HashBuf(ho->type, result, buf, bufl);
+  if (rv != SECSuccess) goto abort;
+
+  err = 0;
+abort:
+  return err;
+}
+
+static nr_ice_crypto_vtbl nr_ice_crypto_nss_vtbl = {
+    nr_crypto_nss_random_bytes, nr_crypto_nss_hmac, nr_crypto_nss_md5};
+
+nsresult NrIceStunServer::ToNicerStunStruct(nr_ice_stun_server* server) const {
+  int r;
+
+  memset(server, 0, sizeof(nr_ice_stun_server));
+  uint8_t protocol;
+  if (transport_ == kNrIceTransportUdp) {
+    protocol = IPPROTO_UDP;
+  } else if (transport_ == kNrIceTransportTcp) {
+    protocol = IPPROTO_TCP;
+  } else if (transport_ == kNrIceTransportTls) {
+    protocol = IPPROTO_TCP;
+  } else {
+    MOZ_MTLOG(ML_ERROR, "Unsupported STUN server transport: " << transport_);
+    return NS_ERROR_FAILURE;
+  }
+
+  if (has_addr_) {
+    if (transport_ == kNrIceTransportTls) {
+      // Refuse to try TLS without an FQDN
+      return NS_ERROR_INVALID_ARG;
+    }
+    r = nr_praddr_to_transport_addr(&addr_, &server->addr, protocol, 0);
+    if (r) {
+      return NS_ERROR_FAILURE;
+    }
+  } else {
+    MOZ_ASSERT(sizeof(server->addr.fqdn) > host_.size());
+    // Dummy information to keep nICEr happy
+    if (use_ipv6_if_fqdn_) {
+      nr_str_port_to_transport_addr("::", port_, protocol, &server->addr);
+    } else {
+      nr_str_port_to_transport_addr("0.0.0.0", port_, protocol, &server->addr);
+    }
+    PL_strncpyz(server->addr.fqdn, host_.c_str(), sizeof(server->addr.fqdn));
+    if (transport_ == kNrIceTransportTls) {
+      server->addr.tls = 1;
+    }
+  }
+
+  nr_transport_addr_fmt_addr_string(&server->addr);
+
+  return NS_OK;
+}
+
+nsresult NrIceTurnServer::ToNicerTurnStruct(nr_ice_turn_server* server) const {
+  memset(server, 0, sizeof(nr_ice_turn_server));
+
+  nsresult rv = ToNicerStunStruct(&server->turn_server);
+  if (NS_FAILED(rv)) return rv;
+
+  if (!(server->username = r_strdup(username_.c_str())))
+    return NS_ERROR_OUT_OF_MEMORY;
+
+  // TODO(ekr@rtfm.com): handle non-ASCII passwords somehow?
+  // STUN requires they be SASLpreped, but we don't know if
+  // they are at this point.
+
+  // C++03 23.2.4, Paragraph 1 stipulates that the elements
+  // in std::vector must be contiguous, and can therefore be
+  // used as input to functions expecting C arrays.
+  const UCHAR* data = password_.empty() ? nullptr : &password_[0];
+  int r = r_data_create(&server->password, data, password_.size());
+  if (r) {
+    RFREE(server->username);
+    return NS_ERROR_OUT_OF_MEMORY;
+  }
+
+  return NS_OK;
+}
+
+NrIceCtx::NrIceCtx(const std::string& name)
+    : connection_state_(ICE_CTX_INIT),
+      gathering_state_(ICE_CTX_GATHER_INIT),
+      name_(name),
+      ice_controlling_set_(false),
+      streams_(),
+      ctx_(nullptr),
+      peer_(nullptr),
+      ice_handler_vtbl_(nullptr),
+      ice_handler_(nullptr),
+      trickle_(true),
+      config_(),
+      nat_(nullptr),
+      proxy_config_(nullptr) {}
+
+/* static */
+RefPtr<NrIceCtx> NrIceCtx::Create(const std::string& aName) {
+  RefPtr<NrIceCtx> ctx = new NrIceCtx(aName);
+
+  if (!ctx->Initialize()) {
+    return nullptr;
+  }
+
+  return ctx;
+}
+
+nsresult NrIceCtx::SetIceConfig(const Config& aConfig) {
+  config_ = aConfig;
+  switch (config_.mPolicy) {
+    case ICE_POLICY_RELAY:
+      MOZ_MTLOG(ML_DEBUG, "SetIceConfig: relay only");
+      nr_ice_ctx_remove_flags(ctx_, NR_ICE_CTX_FLAGS_DISABLE_HOST_CANDIDATES);
+      nr_ice_ctx_add_flags(ctx_, NR_ICE_CTX_FLAGS_RELAY_ONLY);
+      break;
+    case ICE_POLICY_NO_HOST:
+      MOZ_MTLOG(ML_DEBUG, "SetIceConfig: no host");
+      nr_ice_ctx_add_flags(ctx_, NR_ICE_CTX_FLAGS_DISABLE_HOST_CANDIDATES);
+      nr_ice_ctx_remove_flags(ctx_, NR_ICE_CTX_FLAGS_RELAY_ONLY);
+      break;
+    case ICE_POLICY_ALL:
+      MOZ_MTLOG(ML_DEBUG, "SetIceConfig: all");
+      nr_ice_ctx_remove_flags(ctx_, NR_ICE_CTX_FLAGS_DISABLE_HOST_CANDIDATES);
+      nr_ice_ctx_remove_flags(ctx_, NR_ICE_CTX_FLAGS_RELAY_ONLY);
+      break;
+  }
+
+  // TODO: Support re-configuring the test NAT someday?
+  if (!nat_ && config_.mNatSimulatorConfig.isSome()) {
+    TestNat* test_nat = new TestNat;
+    test_nat->filtering_type_ = TestNat::ToNatBehavior(
+        config_.mNatSimulatorConfig->mFilteringType.get());
+    test_nat->mapping_type_ =
+        TestNat::ToNatBehavior(config_.mNatSimulatorConfig->mMappingType.get());
+    test_nat->block_udp_ = config_.mNatSimulatorConfig->mBlockUdp;
+    test_nat->block_tcp_ = config_.mNatSimulatorConfig->mBlockTcp;
+    test_nat->block_tls_ = config_.mNatSimulatorConfig->mBlockTls;
+    test_nat->error_code_for_drop_ =
+        config_.mNatSimulatorConfig->mErrorCodeForDrop;
+    if (config_.mNatSimulatorConfig->mRedirectAddress.Length()) {
+      test_nat
+          ->stun_redirect_map_[config_.mNatSimulatorConfig->mRedirectAddress] =
+          config_.mNatSimulatorConfig->mRedirectTargets;
+    }
+    test_nat->enabled_ = true;
+    SetNat(test_nat);
+  }
+
+  return NS_OK;
+}
+
+RefPtr<NrIceMediaStream> NrIceCtx::CreateStream(const std::string& id,
+                                                const std::string& name,
+                                                int components) {
+  if (streams_.count(id)) {
+    MOZ_ASSERT(false);
+    return nullptr;
+  }
+
+  RefPtr<NrIceMediaStream> stream =
+      new NrIceMediaStream(this, id, name, components);
+  streams_[id] = stream;
+  return stream;
+}
+
+void NrIceCtx::DestroyStream(const std::string& id) {
+  auto it = streams_.find(id);
+  if (it != streams_.end()) {
+    auto preexisting_stream = it->second;
+    streams_.erase(it);
+    preexisting_stream->Close();
+  }
+
+  if (streams_.empty()) {
+    SetGatheringState(ICE_CTX_GATHER_INIT);
+  }
+}
+
+// Handler callbacks
+int NrIceCtx::select_pair(void* obj, nr_ice_media_stream* stream,
+                          int component_id, nr_ice_cand_pair** potentials,
+                          int potential_ct) {
+  MOZ_MTLOG(ML_DEBUG, "select pair called: potential_ct = " << potential_ct);
+  MOZ_ASSERT(stream->local_stream);
+  MOZ_ASSERT(!stream->local_stream->obsolete);
+
+  return 0;
+}
+
+int NrIceCtx::stream_ready(void* obj, nr_ice_media_stream* stream) {
+  MOZ_MTLOG(ML_DEBUG, "stream_ready called");
+  MOZ_ASSERT(!stream->local_stream);
+  MOZ_ASSERT(!stream->obsolete);
+
+  // Get the ICE ctx.
+  NrIceCtx* ctx = static_cast<NrIceCtx*>(obj);
+
+  RefPtr<NrIceMediaStream> s = ctx->FindStream(stream);
+
+  // Streams which do not exist should never be ready.
+  MOZ_ASSERT(s);
+
+  s->Ready();
+
+  return 0;
+}
+
+int NrIceCtx::stream_failed(void* obj, nr_ice_media_stream* stream) {
+  MOZ_MTLOG(ML_DEBUG, "stream_failed called");
+  MOZ_ASSERT(!stream->local_stream);
+  MOZ_ASSERT(!stream->obsolete);
+
+  // Get the ICE ctx
+  NrIceCtx* ctx = static_cast<NrIceCtx*>(obj);
+  RefPtr<NrIceMediaStream> s = ctx->FindStream(stream);
+
+  // Streams which do not exist should never fail.
+  MOZ_ASSERT(s);
+
+  ctx->SetConnectionState(ICE_CTX_FAILED);
+  s->Failed();
+  return 0;
+}
+
+int NrIceCtx::ice_checking(void* obj, nr_ice_peer_ctx* pctx) {
+  MOZ_MTLOG(ML_DEBUG, "ice_checking called");
+
+  // Get the ICE ctx
+  NrIceCtx* ctx = static_cast<NrIceCtx*>(obj);
+
+  ctx->SetConnectionState(ICE_CTX_CHECKING);
+
+  return 0;
+}
+
+int NrIceCtx::ice_connected(void* obj, nr_ice_peer_ctx* pctx) {
+  MOZ_MTLOG(ML_DEBUG, "ice_connected called");
+
+  // Get the ICE ctx
+  NrIceCtx* ctx = static_cast<NrIceCtx*>(obj);
+
+  // This is called even on failed contexts.
+  if (ctx->connection_state() != ICE_CTX_FAILED) {
+    ctx->SetConnectionState(ICE_CTX_CONNECTED);
+  }
+
+  return 0;
+}
+
+int NrIceCtx::ice_disconnected(void* obj, nr_ice_peer_ctx* pctx) {
+  MOZ_MTLOG(ML_DEBUG, "ice_disconnected called");
+
+  // Get the ICE ctx
+  NrIceCtx* ctx = static_cast<NrIceCtx*>(obj);
+
+  ctx->SetConnectionState(ICE_CTX_DISCONNECTED);
+
+  return 0;
+}
+
+int NrIceCtx::msg_recvd(void* obj, nr_ice_peer_ctx* pctx,
+                        nr_ice_media_stream* stream, int component_id,
+                        UCHAR* msg, int len) {
+  // Get the ICE ctx
+  NrIceCtx* ctx = static_cast<NrIceCtx*>(obj);
+  RefPtr<NrIceMediaStream> s = ctx->FindStream(stream);
+
+  // Streams which do not exist should never have packets.
+  MOZ_ASSERT(s);
+
+  s->SignalPacketReceived(s, component_id, msg, len);
+
+  return 0;
+}
+
+void NrIceCtx::trickle_cb(void* arg, nr_ice_ctx* ice_ctx,
+                          nr_ice_media_stream* stream, int component_id,
+                          nr_ice_candidate* candidate) {
+  if (stream->obsolete) {
+    // Stream was probably just marked obsolete, resulting in this callback
+    return;
+  }
+  // Get the ICE ctx
+  NrIceCtx* ctx = static_cast<NrIceCtx*>(arg);
+  RefPtr<NrIceMediaStream> s = ctx->FindStream(stream);
+
+  if (!s) {
+    // This stream has been removed because it is inactive
+    return;
+  }
+
+  if (!candidate) {
+    s->SignalCandidate(s, "", stream->ufrag, "", "");
+    return;
+  }
+
+  std::string actual_addr;
+  std::string mdns_addr;
+  ctx->GenerateObfuscatedAddress(candidate, &mdns_addr, &actual_addr);
+
+  // Format the candidate.
+  char candidate_str[NR_ICE_MAX_ATTRIBUTE_SIZE];
+  int r = nr_ice_format_candidate_attribute(
+      candidate, candidate_str, sizeof(candidate_str),
+      (ctx->ctx()->flags & NR_ICE_CTX_FLAGS_OBFUSCATE_HOST_ADDRESSES) ? 1 : 0);
+  MOZ_ASSERT(!r);
+  if (r) return;
+
+  MOZ_MTLOG(ML_INFO, "NrIceCtx(" << ctx->name_ << "): trickling candidate "
+                                 << candidate_str);
+
+  s->SignalCandidate(s, candidate_str, stream->ufrag, mdns_addr, actual_addr);
+}
+
+void NrIceCtx::InitializeGlobals(const GlobalConfig& aConfig) {
+  RLogConnector::CreateInstance();
+  // Initialize the crypto callbacks and logging stuff
+  if (!initialized) {
+    NR_reg_init(NR_REG_MODE_LOCAL);
+    nr_crypto_vtbl = &nr_ice_crypto_nss_vtbl;
+    initialized = true;
+
+    // Set the priorites for candidate type preferences.
+    // These numbers come from RFC 5245 S. 4.1.2.2
+    NR_reg_set_uchar((char*)NR_ICE_REG_PREF_TYPE_SRV_RFLX, 100);
+    NR_reg_set_uchar((char*)NR_ICE_REG_PREF_TYPE_PEER_RFLX, 110);
+    NR_reg_set_uchar((char*)NR_ICE_REG_PREF_TYPE_HOST, 126);
+    NR_reg_set_uchar((char*)NR_ICE_REG_PREF_TYPE_RELAYED, 5);
+    NR_reg_set_uchar((char*)NR_ICE_REG_PREF_TYPE_SRV_RFLX_TCP, 99);
+    NR_reg_set_uchar((char*)NR_ICE_REG_PREF_TYPE_PEER_RFLX_TCP, 109);
+    NR_reg_set_uchar((char*)NR_ICE_REG_PREF_TYPE_HOST_TCP, 125);
+    NR_reg_set_uchar((char*)NR_ICE_REG_PREF_TYPE_RELAYED_TCP, 0);
+    NR_reg_set_uint4((char*)"stun.client.maximum_transmits",
+                     aConfig.mStunClientMaxTransmits);
+    NR_reg_set_uint4((char*)NR_ICE_REG_TRICKLE_GRACE_PERIOD,
+                     aConfig.mTrickleIceGracePeriod);
+    NR_reg_set_int4((char*)NR_ICE_REG_ICE_TCP_SO_SOCK_COUNT,
+                    aConfig.mIceTcpSoSockCount);
+    NR_reg_set_int4((char*)NR_ICE_REG_ICE_TCP_LISTEN_BACKLOG,
+                    aConfig.mIceTcpListenBacklog);
+
+    NR_reg_set_char((char*)NR_ICE_REG_ICE_TCP_DISABLE, !aConfig.mTcpEnabled);
+
+    if (aConfig.mAllowLoopback) {
+      NR_reg_set_char((char*)NR_STUN_REG_PREF_ALLOW_LOOPBACK_ADDRS, 1);
+    }
+
+    if (aConfig.mAllowLinkLocal) {
+      NR_reg_set_char((char*)NR_STUN_REG_PREF_ALLOW_LINK_LOCAL_ADDRS, 1);
+    }
+    if (!aConfig.mForceNetInterface.Length()) {
+      NR_reg_set_string((char*)NR_ICE_REG_PREF_FORCE_INTERFACE_NAME,
+                        const_cast<char*>(aConfig.mForceNetInterface.get()));
+    }
+
+    // For now, always use nr_resolver for UDP.
+    NR_reg_set_char((char*)NR_ICE_REG_USE_NR_RESOLVER_FOR_UDP, 1);
+
+    // Use nr_resolver for TCP only when not in e10s mode (for unit-tests)
+    if (XRE_IsParentProcess()) {
+      NR_reg_set_char((char*)NR_ICE_REG_USE_NR_RESOLVER_FOR_TCP, 1);
+    }
+  }
+}
+
+void NrIceCtx::SetTargetForDefaultLocalAddressLookup(
+    const std::string& target_ip, uint16_t target_port) {
+  nr_ice_set_target_for_default_local_address_lookup(ctx_, target_ip.c_str(),
+                                                     target_port);
+}
+
+#define MAXADDRS 100  // mirrors setting in ice_ctx.c
+
+/* static */
+nsTArray<NrIceStunAddr> NrIceCtx::GetStunAddrs() {
+  nsTArray<NrIceStunAddr> addrs;
+
+  nr_local_addr local_addrs[MAXADDRS];
+  int addr_ct = 0;
+
+  // most likely running on parent process and need crypto vtbl
+  // initialized on Windows (Linux and OSX don't seem to care)
+  if (!initialized) {
+    nr_crypto_vtbl = &nr_ice_crypto_nss_vtbl;
+  }
+
+  MOZ_MTLOG(ML_INFO, "NrIceCtx static call to find local stun addresses");
+  if (nr_stun_find_local_addresses(local_addrs, MAXADDRS, &addr_ct)) {
+    MOZ_MTLOG(ML_INFO, "Error finding local stun addresses");
+  } else {
+    for (int i = 0; i < addr_ct; ++i) {
+      NrIceStunAddr addr(&local_addrs[i]);
+      addrs.AppendElement(addr);
+    }
+  }
+
+  return addrs;
+}
+
+void NrIceCtx::SetStunAddrs(const nsTArray<NrIceStunAddr>& addrs) {
+  nr_local_addr* local_addrs;
+  local_addrs = new nr_local_addr[addrs.Length()];
+
+  for (size_t i = 0; i < addrs.Length(); ++i) {
+    nr_local_addr_copy(&local_addrs[i],
+                       const_cast<nr_local_addr*>(&addrs[i].localAddr()));
+  }
+  nr_ice_set_local_addresses(ctx_, local_addrs, addrs.Length());
+
+  delete[] local_addrs;
+}
+
+bool NrIceCtx::Initialize() {
+  // Create the ICE context
+  int r;
+
+  UINT4 flags = NR_ICE_CTX_FLAGS_AGGRESSIVE_NOMINATION;
+  r = nr_ice_ctx_create(const_cast<char*>(name_.c_str()), flags, &ctx_);
+
+  if (r) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't create ICE ctx for '" << name_ << "'");
+    return false;
+  }
+
+  // override default factory to capture optional proxy config when creating
+  // sockets.
+  nr_socket_factory* factory;
+  r = nr_socket_factory_create_int(this, &ctx_socket_factory_vtbl, &factory);
+
+  if (r) {
+    MOZ_MTLOG(LogLevel::Error, "Couldn't create ctx socket factory.");
+    return false;
+  }
+  nr_ice_ctx_set_socket_factory(ctx_, factory);
+
+  nr_interface_prioritizer* prioritizer = CreateInterfacePrioritizer();
+  if (!prioritizer) {
+    MOZ_MTLOG(LogLevel::Error, "Couldn't create interface prioritizer.");
+    return false;
+  }
+
+  r = nr_ice_ctx_set_interface_prioritizer(ctx_, prioritizer);
+  if (r) {
+    MOZ_MTLOG(LogLevel::Error, "Couldn't set interface prioritizer.");
+    return false;
+  }
+
+  if (generating_trickle()) {
+    r = nr_ice_ctx_set_trickle_cb(ctx_, &NrIceCtx::trickle_cb, this);
+    if (r) {
+      MOZ_MTLOG(ML_ERROR, "Couldn't set trickle cb for '" << name_ << "'");
+      return false;
+    }
+  }
+
+  // Create the handler objects
+  ice_handler_vtbl_ = new nr_ice_handler_vtbl();
+  ice_handler_vtbl_->select_pair = &NrIceCtx::select_pair;
+  ice_handler_vtbl_->stream_ready = &NrIceCtx::stream_ready;
+  ice_handler_vtbl_->stream_failed = &NrIceCtx::stream_failed;
+  ice_handler_vtbl_->ice_connected = &NrIceCtx::ice_connected;
+  ice_handler_vtbl_->msg_recvd = &NrIceCtx::msg_recvd;
+  ice_handler_vtbl_->ice_checking = &NrIceCtx::ice_checking;
+  ice_handler_vtbl_->ice_disconnected = &NrIceCtx::ice_disconnected;
+
+  ice_handler_ = new nr_ice_handler();
+  ice_handler_->vtbl = ice_handler_vtbl_;
+  ice_handler_->obj = this;
+
+  // Create the peer ctx. Because we do not support parallel forking, we
+  // only have one peer ctx.
+  std::string peer_name = name_ + ":default";
+  r = nr_ice_peer_ctx_create(ctx_, ice_handler_,
+                             const_cast<char*>(peer_name.c_str()), &peer_);
+  if (r) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't create ICE peer ctx for '" << name_ << "'");
+    return false;
+  }
+
+  nsresult rv;
+  sts_target_ = do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &rv);
+
+  if (!NS_SUCCEEDED(rv)) return false;
+
+  return true;
+}
+
+int NrIceCtx::SetNat(const RefPtr<TestNat>& aNat) {
+  nat_ = aNat;
+  nr_socket_factory* fac;
+  int r = nat_->create_socket_factory(&fac);
+  if (r) {
+    return r;
+  }
+  nr_ice_ctx_set_socket_factory(ctx_, fac);
+  return 0;
+}
+
+// ONLY USE THIS FOR TESTING. Will cause totally unpredictable and possibly very
+// bad effects if ICE is still live.
+void NrIceCtx::internal_DeinitializeGlobal() {
+  NR_reg_del((char*)"stun");
+  NR_reg_del((char*)"ice");
+  RLogConnector::DestroyInstance();
+  nr_crypto_vtbl = nullptr;
+  initialized = false;
+}
+
+void NrIceCtx::internal_SetTimerAccelarator(int divider) {
+  ctx_->test_timer_divider = divider;
+}
+
+void NrIceCtx::AccumulateStats(const NrIceStats& stats) {
+  nr_accumulate_count(&(ctx_->stats.stun_retransmits), stats.stun_retransmits);
+  nr_accumulate_count(&(ctx_->stats.turn_401s), stats.turn_401s);
+  nr_accumulate_count(&(ctx_->stats.turn_403s), stats.turn_403s);
+  nr_accumulate_count(&(ctx_->stats.turn_438s), stats.turn_438s);
+}
+
+NrIceStats NrIceCtx::Destroy() {
+  // designed to be called more than once so if stats are desired, this can be
+  // called just prior to the destructor
+  MOZ_MTLOG(ML_NOTICE, "NrIceCtx(" << name_ << "): " << __func__);
+
+  for (auto& idAndStream : streams_) {
+    idAndStream.second->Close();
+  }
+
+  NrIceStats stats;
+  if (ctx_) {
+    stats.stun_retransmits = ctx_->stats.stun_retransmits;
+    stats.turn_401s = ctx_->stats.turn_401s;
+    stats.turn_403s = ctx_->stats.turn_403s;
+    stats.turn_438s = ctx_->stats.turn_438s;
+  }
+
+  if (peer_) {
+    nr_ice_peer_ctx_destroy(&peer_);
+  }
+  if (ctx_) {
+    nr_ice_ctx_destroy(&ctx_);
+  }
+
+  delete ice_handler_vtbl_;
+  delete ice_handler_;
+
+  ice_handler_vtbl_ = nullptr;
+  ice_handler_ = nullptr;
+  proxy_config_ = nullptr;
+  streams_.clear();
+
+  return stats;
+}
+
+NrIceCtx::~NrIceCtx() = default;
+
+void NrIceCtx::destroy_peer_ctx() { nr_ice_peer_ctx_destroy(&peer_); }
+
+nsresult NrIceCtx::SetControlling(Controlling controlling) {
+  if (!ice_controlling_set_) {
+    peer_->controlling = (controlling == ICE_CONTROLLING) ? 1 : 0;
+    ice_controlling_set_ = true;
+
+    MOZ_MTLOG(ML_DEBUG,
+              "ICE ctx " << name_ << " setting controlling to" << controlling);
+  }
+  return NS_OK;
+}
+
+NrIceCtx::Controlling NrIceCtx::GetControlling() {
+  return (peer_->controlling) ? ICE_CONTROLLING : ICE_CONTROLLED;
+}
+
+nsresult NrIceCtx::SetStunServers(
+    const std::vector<NrIceStunServer>& stun_servers) {
+  MOZ_MTLOG(ML_NOTICE, "NrIceCtx(" << name_ << "): " << __func__);
+  // We assume nr_ice_stun_server is memmoveable. That's true right now.
+  std::vector<nr_ice_stun_server> servers;
+
+  for (size_t i = 0; i < stun_servers.size(); ++i) {
+    nr_ice_stun_server server;
+    nsresult rv = stun_servers[i].ToNicerStunStruct(&server);
+    if (NS_WARN_IF(NS_FAILED(rv))) {
+      MOZ_MTLOG(ML_ERROR, "Couldn't convert STUN server for '" << name_ << "'");
+    } else {
+      servers.push_back(server);
+    }
+  }
+
+  int r = nr_ice_ctx_set_stun_servers(ctx_, servers.data(),
+                                      static_cast<int>(servers.size()));
+  if (r) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't set STUN servers for '" << name_ << "'");
+    return NS_ERROR_FAILURE;
+  }
+
+  return NS_OK;
+}
+
+// TODO(ekr@rtfm.com): This is just SetStunServers with s/Stun/Turn
+// Could we do a template or something?
+nsresult NrIceCtx::SetTurnServers(
+    const std::vector<NrIceTurnServer>& turn_servers) {
+  MOZ_MTLOG(ML_NOTICE, "NrIceCtx(" << name_ << "): " << __func__);
+  // We assume nr_ice_turn_server is memmoveable. That's true right now.
+  std::vector<nr_ice_turn_server> servers;
+
+  for (size_t i = 0; i < turn_servers.size(); ++i) {
+    nr_ice_turn_server server;
+    nsresult rv = turn_servers[i].ToNicerTurnStruct(&server);
+    if (NS_WARN_IF(NS_FAILED(rv))) {
+      MOZ_MTLOG(ML_ERROR, "Couldn't convert TURN server for '" << name_ << "'");
+    } else {
+      servers.push_back(server);
+    }
+  }
+
+  int r = nr_ice_ctx_set_turn_servers(ctx_, servers.data(),
+                                      static_cast<int>(servers.size()));
+  if (r) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't set TURN servers for '" << name_ << "'");
+    // TODO(ekr@rtfm.com): This leaks the username/password. Need to free that.
+    return NS_ERROR_FAILURE;
+  }
+
+  return NS_OK;
+}
+
+nsresult NrIceCtx::SetResolver(nr_resolver* resolver) {
+  int r = nr_ice_ctx_set_resolver(ctx_, resolver);
+
+  if (r) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't set resolver for '" << name_ << "'");
+    return NS_ERROR_FAILURE;
+  }
+
+  return NS_OK;
+}
+
+nsresult NrIceCtx::SetProxyConfig(NrSocketProxyConfig&& config) {
+  proxy_config_.reset(new NrSocketProxyConfig(std::move(config)));
+  if (nat_) {
+    nat_->set_proxy_config(proxy_config_);
+  }
+
+  if (proxy_config_->GetForceProxy()) {
+    nr_ice_ctx_add_flags(ctx_, NR_ICE_CTX_FLAGS_ONLY_PROXY);
+  } else {
+    nr_ice_ctx_remove_flags(ctx_, NR_ICE_CTX_FLAGS_ONLY_PROXY);
+  }
+
+  return NS_OK;
+}
+
+void NrIceCtx::SetCtxFlags(bool default_route_only) {
+  ASSERT_ON_THREAD(sts_target_);
+
+  if (default_route_only) {
+    nr_ice_ctx_add_flags(ctx_, NR_ICE_CTX_FLAGS_ONLY_DEFAULT_ADDRS);
+  } else {
+    nr_ice_ctx_remove_flags(ctx_, NR_ICE_CTX_FLAGS_ONLY_DEFAULT_ADDRS);
+  }
+}
+
+nsresult NrIceCtx::StartGathering(bool default_route_only,
+                                  bool obfuscate_host_addresses) {
+  ASSERT_ON_THREAD(sts_target_);
+  MOZ_MTLOG(ML_NOTICE, "NrIceCtx(" << name_ << "): " << __func__);
+
+  if (obfuscate_host_addresses) {
+    nr_ice_ctx_add_flags(ctx_, NR_ICE_CTX_FLAGS_OBFUSCATE_HOST_ADDRESSES);
+  }
+
+  SetCtxFlags(default_route_only);
+
+  // This might start gathering for the first time, or again after
+  // renegotiation, or might do nothing at all if gathering has already
+  // finished.
+  int r = nr_ice_gather(ctx_, &NrIceCtx::gather_cb, this);
+
+  if (!r) {
+    SetGatheringState(ICE_CTX_GATHER_COMPLETE);
+  } else if (r == R_WOULDBLOCK) {
+    SetGatheringState(ICE_CTX_GATHER_STARTED);
+  } else {
+    SetGatheringState(ICE_CTX_GATHER_COMPLETE);
+    MOZ_MTLOG(ML_ERROR, "ICE FAILED: Couldn't gather ICE candidates for '"
+                            << name_ << "', error=" << r);
+    SetConnectionState(ICE_CTX_FAILED);
+    return NS_ERROR_FAILURE;
+  }
+
+  return NS_OK;
+}
+
+RefPtr<NrIceMediaStream> NrIceCtx::FindStream(nr_ice_media_stream* stream) {
+  for (auto& idAndStream : streams_) {
+    if (idAndStream.second->HasStream(stream)) {
+      return idAndStream.second;
+    }
+  }
+
+  return nullptr;
+}
+
+std::vector<std::string> NrIceCtx::GetGlobalAttributes() {
+  char** attrs = nullptr;
+  int attrct;
+  int r;
+  std::vector<std::string> ret;
+
+  r = nr_ice_get_global_attributes(ctx_, &attrs, &attrct);
+  if (r) {
+    MOZ_MTLOG(ML_ERROR,
+              "Couldn't get ufrag and password for '" << name_ << "'");
+    return ret;
+  }
+
+  for (int i = 0; i < attrct; i++) {
+    ret.push_back(std::string(attrs[i]));
+    RFREE(attrs[i]);
+  }
+  RFREE(attrs);
+
+  return ret;
+}
+
+nsresult NrIceCtx::ParseGlobalAttributes(std::vector<std::string> attrs) {
+  std::vector<char*> attrs_in;
+  attrs_in.reserve(attrs.size());
+  for (auto& attr : attrs) {
+    attrs_in.push_back(const_cast<char*>(attr.c_str()));
+  }
+
+  int r = nr_ice_peer_ctx_parse_global_attributes(
+      peer_, attrs_in.empty() ? nullptr : &attrs_in[0], attrs_in.size());
+  if (r) {
+    MOZ_MTLOG(ML_ERROR,
+              "Couldn't parse global attributes for " << name_ << "'");
+    return NS_ERROR_FAILURE;
+  }
+
+  return NS_OK;
+}
+
+bool NrIceCtx::HasStreamsToConnect() const {
+  for (auto& idAndStream : streams_) {
+    if (idAndStream.second->state() != NrIceMediaStream::ICE_CLOSED) {
+      return true;
+    }
+  }
+  return false;
+}
+
+nsresult NrIceCtx::StartChecks() {
+  int r;
+  MOZ_MTLOG(ML_NOTICE, "NrIceCtx(" << name_ << "): " << __func__);
+
+  if (!HasStreamsToConnect()) {
+    MOZ_MTLOG(ML_NOTICE, "In StartChecks, nothing to do on " << name_);
+    return NS_OK;
+  }
+
+  r = nr_ice_peer_ctx_pair_candidates(peer_);
+  if (r) {
+    MOZ_MTLOG(ML_ERROR, "ICE FAILED: Couldn't pair candidates on " << name_);
+    SetConnectionState(ICE_CTX_FAILED);
+    return NS_ERROR_FAILURE;
+  }
+
+  r = nr_ice_peer_ctx_start_checks2(peer_, 1);
+  if (r) {
+    if (r == R_NOT_FOUND) {
+      MOZ_MTLOG(ML_INFO, "Couldn't start peer checks on "
+                             << name_ << ", assuming trickle ICE");
+    } else {
+      MOZ_MTLOG(ML_ERROR,
+                "ICE FAILED: Couldn't start peer checks on " << name_);
+      SetConnectionState(ICE_CTX_FAILED);
+      return NS_ERROR_FAILURE;
+    }
+  }
+
+  return NS_OK;
+}
+
+void NrIceCtx::gather_cb(NR_SOCKET s, int h, void* arg) {
+  NrIceCtx* ctx = static_cast<NrIceCtx*>(arg);
+
+  ctx->SetGatheringState(ICE_CTX_GATHER_COMPLETE);
+}
+
+void NrIceCtx::UpdateNetworkState(bool online) {
+  MOZ_MTLOG(ML_NOTICE, "NrIceCtx(" << name_ << "): updating network state to "
+                                   << (online ? "online" : "offline"));
+  if (connection_state_ == ICE_CTX_CLOSED) {
+    return;
+  }
+
+  if (online) {
+    nr_ice_peer_ctx_refresh_consent_all_streams(peer_);
+  } else {
+    nr_ice_peer_ctx_disconnect_all_streams(peer_);
+  }
+}
+
+void NrIceCtx::SetConnectionState(ConnectionState state) {
+  if (state == connection_state_) return;
+
+  MOZ_MTLOG(ML_INFO, "NrIceCtx(" << name_ << "): state " << connection_state_
+                                 << "->" << state);
+  connection_state_ = state;
+
+  if (connection_state_ == ICE_CTX_FAILED) {
+    MOZ_MTLOG(ML_INFO,
+              "NrIceCtx(" << name_ << "): dumping r_log ringbuffer... ");
+    std::deque<std::string> logs;
+    RLogConnector::GetInstance()->GetAny(0, &logs);
+    for (auto& log : logs) {
+      MOZ_MTLOG(ML_INFO, log);
+    }
+  }
+
+  SignalConnectionStateChange(this, state);
+}
+
+void NrIceCtx::SetGatheringState(GatheringState state) {
+  if (state == gathering_state_) return;
+
+  MOZ_MTLOG(ML_DEBUG, "NrIceCtx(" << name_ << "): gathering state "
+                                  << gathering_state_ << "->" << state);
+  gathering_state_ = state;
+
+  SignalGatheringStateChange(this, state);
+}
+
+void NrIceCtx::GenerateObfuscatedAddress(nr_ice_candidate* candidate,
+                                         std::string* mdns_address,
+                                         std::string* actual_address) {
+  if (candidate->type == HOST &&
+      (ctx_->flags & NR_ICE_CTX_FLAGS_OBFUSCATE_HOST_ADDRESSES)) {
+    char addr[64];
+    if (nr_transport_addr_get_addrstring(&candidate->addr, addr,
+                                         sizeof(addr))) {
+      return;
+    }
+
+    *actual_address = addr;
+
+    const auto& iter = obfuscated_host_addresses_.find(*actual_address);
+    if (iter != obfuscated_host_addresses_.end()) {
+      *mdns_address = iter->second;
+    } else {
+      nsresult rv;
+      nsCOMPtr<nsIUUIDGenerator> uuidgen =
+          do_GetService("@mozilla.org/uuid-generator;1", &rv);
+      // If this fails, we'll return a zero UUID rather than something
+      // unexpected.
+      nsID id = {};
+      id.Clear();
+      if (NS_SUCCEEDED(rv)) {
+        rv = uuidgen->GenerateUUIDInPlace(&id);
+        if (NS_FAILED(rv)) {
+          id.Clear();
+        }
+      }
+
+      char chars[NSID_LENGTH];
+      id.ToProvidedString(chars);
+      // The string will look like {64888863-a253-424a-9b30-1ed285d20142},
+      // we want to trim off the braces.
+      const char* ptr_to_id = chars;
+      ++ptr_to_id;
+      chars[NSID_LENGTH - 2] = 0;
+
+      std::ostringstream o;
+      o << ptr_to_id << ".local";
+      *mdns_address = o.str();
+
+      obfuscated_host_addresses_[*actual_address] = *mdns_address;
+    }
+    candidate->mdns_addr = r_strdup(mdns_address->c_str());
+  }
+}
+
+}  // namespace mozilla
+
+// Reimplement nr_ice_compute_codeword to avoid copyright issues
+void nr_ice_compute_codeword(char* buf, int len, char* codeword) {
+  UINT4 c;
+
+  r_crc32(buf, len, &c);
+
+  PL_Base64Encode(reinterpret_cast<char*>(&c), 3, codeword);
+  codeword[4] = 0;
+}
+
+int nr_socket_local_create(void* obj, nr_transport_addr* addr,
+                           nr_socket** sockp) {
+  using namespace mozilla;
+
+  RefPtr<NrSocketBase> sock;
+  int r, _status;
+  shared_ptr<NrSocketProxyConfig> config = nullptr;
+
+  if (obj) {
+    config = static_cast<NrIceCtx*>(obj)->GetProxyConfig();
+  }
+
+  r = NrSocketBase::CreateSocket(addr, &sock, config);
+  if (r) {
+    ABORT(r);
+  }
+
+  r = nr_socket_create_int(static_cast<void*>(sock), sock->vtbl(), sockp);
+  if (r) ABORT(r);
+
+  _status = 0;
+
+  {
+    // We will release this reference in destroy(), not exactly the normal
+    // ownership model, but it is what it is.
+    NrSocketBase* dummy = sock.forget().take();
+    (void)dummy;
+  }
+
+abort:
+  return _status;
+}
diff --git a/dom/media/webrtc/transport/nricectx.h b/dom/media/webrtc/transport/nricectx.h
new file mode 100644
index 0000000000..a0a0b5b772
--- /dev/null
+++ b/dom/media/webrtc/transport/nricectx.h
@@ -0,0 +1,421 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+// Some of this code is cut-and-pasted from nICEr. Copyright is:
+
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+// This is a wrapper around the nICEr ICE stack
+#ifndef nricectx_h__
+#define nricectx_h__
+
+#include <memory>
+#include <string>
+#include <vector>
+#include <map>
+
+#include "sigslot.h"
+
+#include "prnetdb.h"
+
+#include "mozilla/RefPtr.h"
+#include "mozilla/UniquePtr.h"
+#include "nsIEventTarget.h"
+#include "nsTArray.h"
+#include "mozilla/Maybe.h"
+
+#include "m_cpp_utils.h"
+#include "nricestunaddr.h"
+#include "nricemediastream.h"
+
+typedef struct nr_ice_ctx_ nr_ice_ctx;
+typedef struct nr_ice_peer_ctx_ nr_ice_peer_ctx;
+typedef struct nr_ice_media_stream_ nr_ice_media_stream;
+typedef struct nr_ice_handler_ nr_ice_handler;
+typedef struct nr_ice_handler_vtbl_ nr_ice_handler_vtbl;
+typedef struct nr_ice_candidate_ nr_ice_candidate;
+typedef struct nr_ice_cand_pair_ nr_ice_cand_pair;
+typedef struct nr_ice_stun_server_ nr_ice_stun_server;
+typedef struct nr_ice_turn_server_ nr_ice_turn_server;
+typedef struct nr_resolver_ nr_resolver;
+typedef struct nr_proxy_tunnel_config_ nr_proxy_tunnel_config;
+
+typedef void* NR_SOCKET;
+
+namespace mozilla {
+
+class NrSocketProxyConfig;
+
+class NrIceMediaStream;
+
+extern const char kNrIceTransportUdp[];
+extern const char kNrIceTransportTcp[];
+extern const char kNrIceTransportTls[];
+
+class NrIceStunServer {
+ public:
+  explicit NrIceStunServer(const PRNetAddr& addr) : has_addr_(true) {
+    memcpy(&addr_, &addr, sizeof(addr));
+  }
+
+  // The main function to use. Will take either an address or a hostname.
+  static UniquePtr<NrIceStunServer> Create(
+      const std::string& addr, uint16_t port,
+      const char* transport = kNrIceTransportUdp) {
+    UniquePtr<NrIceStunServer> server(new NrIceStunServer(transport));
+
+    nsresult rv = server->Init(addr, port);
+    if (NS_FAILED(rv)) return nullptr;
+
+    return server;
+  }
+
+  nsresult ToNicerStunStruct(nr_ice_stun_server* server) const;
+
+  bool HasFqdn() const { return !has_addr_; }
+
+  void SetUseIPv6IfFqdn() {
+    MOZ_ASSERT(HasFqdn());
+    use_ipv6_if_fqdn_ = true;
+  }
+
+ protected:
+  explicit NrIceStunServer(const char* transport)
+      : addr_(), transport_(transport) {}
+
+  nsresult Init(const std::string& addr, uint16_t port) {
+    PRStatus status = PR_StringToNetAddr(addr.c_str(), &addr_);
+    if (status == PR_SUCCESS) {
+      // Parseable as an address
+      addr_.inet.port = PR_htons(port);
+      port_ = port;
+      has_addr_ = true;
+      return NS_OK;
+    } else if (addr.size() < 256) {
+      // Apparently this is a hostname.
+      host_ = addr;
+      port_ = port;
+      has_addr_ = false;
+      return NS_OK;
+    }
+
+    return NS_ERROR_FAILURE;
+  }
+
+  bool has_addr_;
+  std::string host_;
+  uint16_t port_;
+  PRNetAddr addr_;
+  std::string transport_;
+  bool use_ipv6_if_fqdn_ = false;
+};
+
+class NrIceTurnServer : public NrIceStunServer {
+ public:
+  static UniquePtr<NrIceTurnServer> Create(
+      const std::string& addr, uint16_t port, const std::string& username,
+      const std::vector<unsigned char>& password,
+      const char* transport = kNrIceTransportUdp) {
+    UniquePtr<NrIceTurnServer> server(
+        new NrIceTurnServer(username, password, transport));
+
+    nsresult rv = server->Init(addr, port);
+    if (NS_FAILED(rv)) return nullptr;
+
+    return server;
+  }
+
+  nsresult ToNicerTurnStruct(nr_ice_turn_server* server) const;
+
+ private:
+  NrIceTurnServer(const std::string& username,
+                  const std::vector<unsigned char>& password,
+                  const char* transport)
+      : NrIceStunServer(transport), username_(username), password_(password) {}
+
+  std::string username_;
+  std::vector<unsigned char> password_;
+};
+
+class TestNat;
+
+class NrIceStats {
+ public:
+  uint16_t stun_retransmits = 0;
+  uint16_t turn_401s = 0;
+  uint16_t turn_403s = 0;
+  uint16_t turn_438s = 0;
+};
+
+class NrIceCtx {
+ public:
+  enum ConnectionState {
+    ICE_CTX_INIT,
+    ICE_CTX_CHECKING,
+    ICE_CTX_CONNECTED,
+    ICE_CTX_COMPLETED,
+    ICE_CTX_FAILED,
+    ICE_CTX_DISCONNECTED,
+    ICE_CTX_CLOSED
+  };
+
+  enum GatheringState {
+    ICE_CTX_GATHER_INIT,
+    ICE_CTX_GATHER_STARTED,
+    ICE_CTX_GATHER_COMPLETE
+  };
+
+  enum Controlling { ICE_CONTROLLING, ICE_CONTROLLED };
+
+  enum Policy { ICE_POLICY_RELAY, ICE_POLICY_NO_HOST, ICE_POLICY_ALL };
+
+  struct NatSimulatorConfig {
+    bool mBlockTcp = false;
+    bool mBlockUdp = false;
+    bool mBlockTls = false;
+    int mErrorCodeForDrop = 0;
+    nsCString mMappingType = "ENDPOINT_INDEPENDENT"_ns;
+    nsCString mFilteringType = "ENDPOINT_INDEPENDENT"_ns;
+    nsCString mRedirectAddress;
+    CopyableTArray<nsCString> mRedirectTargets;
+  };
+
+  struct Config {
+    NrIceCtx::Policy mPolicy = NrIceCtx::ICE_POLICY_ALL;
+    Maybe<NatSimulatorConfig> mNatSimulatorConfig;
+  };
+
+  static RefPtr<NrIceCtx> Create(const std::string& aName);
+
+  nsresult SetIceConfig(const Config& aConfig);
+
+  RefPtr<NrIceMediaStream> CreateStream(const std::string& id,
+                                        const std::string& name,
+                                        int components);
+  void DestroyStream(const std::string& id);
+
+  struct GlobalConfig {
+    bool mAllowLinkLocal = false;
+    bool mAllowLoopback = false;
+    bool mTcpEnabled = true;
+    int mStunClientMaxTransmits = 7;
+    int mTrickleIceGracePeriod = 5000;
+    int mIceTcpSoSockCount = 3;
+    int mIceTcpListenBacklog = 10;
+    nsCString mForceNetInterface;
+  };
+
+  // initialize ICE globals, crypto, and logging
+  static void InitializeGlobals(const GlobalConfig& aConfig);
+
+  void SetTargetForDefaultLocalAddressLookup(const std::string& target_ip,
+                                             uint16_t target_port);
+
+  // static GetStunAddrs for use in parent process to support
+  // sandboxing restrictions
+  static nsTArray<NrIceStunAddr> GetStunAddrs();
+  void SetStunAddrs(const nsTArray<NrIceStunAddr>& addrs);
+
+  bool Initialize();
+
+  int SetNat(const RefPtr<TestNat>& aNat);
+
+  // Deinitialize all ICE global state. Used only for testing.
+  static void internal_DeinitializeGlobal();
+
+  // Divide some timers to faster testing. Used only for testing.
+  void internal_SetTimerAccelarator(int divider);
+
+  nr_ice_ctx* ctx() { return ctx_; }
+  nr_ice_peer_ctx* peer() { return peer_; }
+
+  // Testing only.
+  void destroy_peer_ctx();
+
+  RefPtr<NrIceMediaStream> GetStream(const std::string& id) {
+    auto it = streams_.find(id);
+    if (it != streams_.end()) {
+      return it->second;
+    }
+    return nullptr;
+  }
+
+  std::vector<RefPtr<NrIceMediaStream>> GetStreams() const {
+    std::vector<RefPtr<NrIceMediaStream>> result;
+    for (auto& idAndStream : streams_) {
+      result.push_back(idAndStream.second);
+    }
+    return result;
+  }
+
+  bool HasStreamsToConnect() const;
+
+  // The name of the ctx
+  const std::string& name() const { return name_; }
+
+  // Current state
+  ConnectionState connection_state() const { return connection_state_; }
+
+  // Current state
+  GatheringState gathering_state() const { return gathering_state_; }
+
+  // Get the global attributes
+  std::vector<std::string> GetGlobalAttributes();
+
+  // Set the other side's global attributes
+  nsresult ParseGlobalAttributes(std::vector<std::string> attrs);
+
+  // Set whether we are controlling or not.
+  nsresult SetControlling(Controlling controlling);
+
+  Controlling GetControlling();
+
+  // Set the STUN servers. Must be called before StartGathering
+  // (if at all).
+  nsresult SetStunServers(const std::vector<NrIceStunServer>& stun_servers);
+
+  // Set the TURN servers. Must be called before StartGathering
+  // (if at all).
+  nsresult SetTurnServers(const std::vector<NrIceTurnServer>& turn_servers);
+
+  // Provide the resolution provider. Must be called before
+  // StartGathering.
+  nsresult SetResolver(nr_resolver* resolver);
+
+  // Provide the proxy address. Must be called before
+  // StartGathering.
+  nsresult SetProxyConfig(NrSocketProxyConfig&& config);
+
+  const std::shared_ptr<NrSocketProxyConfig>& GetProxyConfig() {
+    return proxy_config_;
+  }
+
+  void SetCtxFlags(bool default_route_only);
+
+  // Start ICE gathering
+  nsresult StartGathering(bool default_route_only,
+                          bool obfuscate_host_addresses);
+
+  // Start checking
+  nsresult StartChecks();
+
+  // Notify that the network has gone online/offline
+  void UpdateNetworkState(bool online);
+
+  void AccumulateStats(const NrIceStats& stats);
+  NrIceStats Destroy();
+
+  // Are we trickling?
+  bool generating_trickle() const { return trickle_; }
+
+  // Signals to indicate events. API users can (and should)
+  // register for these.
+  sigslot::signal2<NrIceCtx*, NrIceCtx::GatheringState>
+      SignalGatheringStateChange;
+  sigslot::signal2<NrIceCtx*, NrIceCtx::ConnectionState>
+      SignalConnectionStateChange;
+
+  // The thread to direct method calls to
+  nsCOMPtr<nsIEventTarget> thread() { return sts_target_; }
+
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(NrIceCtx)
+
+ private:
+  explicit NrIceCtx(const std::string& name);
+
+  virtual ~NrIceCtx();
+
+  DISALLOW_COPY_ASSIGN(NrIceCtx);
+
+  // Callbacks for nICEr
+  static void gather_cb(NR_SOCKET s, int h, void* arg);  // ICE gather complete
+
+  // Handler implementation
+  static int select_pair(void* obj, nr_ice_media_stream* stream,
+                         int component_id, nr_ice_cand_pair** potentials,
+                         int potential_ct);
+  static int stream_ready(void* obj, nr_ice_media_stream* stream);
+  static int stream_failed(void* obj, nr_ice_media_stream* stream);
+  static int ice_checking(void* obj, nr_ice_peer_ctx* pctx);
+  static int ice_connected(void* obj, nr_ice_peer_ctx* pctx);
+  static int ice_disconnected(void* obj, nr_ice_peer_ctx* pctx);
+  static int msg_recvd(void* obj, nr_ice_peer_ctx* pctx,
+                       nr_ice_media_stream* stream, int component_id,
+                       unsigned char* msg, int len);
+  static void trickle_cb(void* arg, nr_ice_ctx* ctx,
+                         nr_ice_media_stream* stream, int component_id,
+                         nr_ice_candidate* candidate);
+
+  // Find a media stream by stream ptr. Gross
+  RefPtr<NrIceMediaStream> FindStream(nr_ice_media_stream* stream);
+
+  // Set the state
+  void SetConnectionState(ConnectionState state);
+
+  // Set the state
+  void SetGatheringState(GatheringState state);
+
+  void GenerateObfuscatedAddress(nr_ice_candidate* candidate,
+                                 std::string* mdns_address,
+                                 std::string* actual_address);
+
+  ConnectionState connection_state_;
+  GatheringState gathering_state_;
+  const std::string name_;
+  bool ice_controlling_set_;
+  std::map<std::string, RefPtr<NrIceMediaStream>> streams_;
+  nr_ice_ctx* ctx_;
+  nr_ice_peer_ctx* peer_;
+  nr_ice_handler_vtbl* ice_handler_vtbl_;  // Must be pointer
+  nr_ice_handler* ice_handler_;            // Must be pointer
+  bool trickle_;
+  nsCOMPtr<nsIEventTarget> sts_target_;  // The thread to run on
+  Config config_;
+  RefPtr<TestNat> nat_;
+  std::shared_ptr<NrSocketProxyConfig> proxy_config_;
+  std::map<std::string, std::string> obfuscated_host_addresses_;
+};
+
+}  // namespace mozilla
+#endif
diff --git a/dom/media/webrtc/transport/nricemediastream.cpp b/dom/media/webrtc/transport/nricemediastream.cpp
new file mode 100644
index 0000000000..ef7980dcf3
--- /dev/null
+++ b/dom/media/webrtc/transport/nricemediastream.cpp
@@ -0,0 +1,709 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+// Some of this code is cut-and-pasted from nICEr. Copyright is:
+
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <string>
+#include <vector>
+
+#include "logging.h"
+#include "nsError.h"
+#include "nsThreadUtils.h"
+
+// nICEr includes
+extern "C" {
+#include "nr_api.h"
+#include "transport_addr.h"
+#include "nr_socket.h"
+#include "ice_ctx.h"
+#include "ice_candidate.h"
+#include "ice_handler.h"
+}
+
+// Local includes
+#include "nricectx.h"
+#include "nricemediastream.h"
+
+namespace mozilla {
+
+MOZ_MTLOG_MODULE("mtransport")
+
+static bool ToNrIceAddr(nr_transport_addr& addr, NrIceAddr* out) {
+  int r;
+  char addrstring[INET6_ADDRSTRLEN + 1];
+
+  r = nr_transport_addr_get_addrstring(&addr, addrstring, sizeof(addrstring));
+  if (r) return false;
+  out->host = addrstring;
+
+  int port;
+  r = nr_transport_addr_get_port(&addr, &port);
+  if (r) return false;
+
+  out->port = port;
+
+  switch (addr.protocol) {
+    case IPPROTO_TCP:
+      if (addr.tls) {
+        out->transport = kNrIceTransportTls;
+      } else {
+        out->transport = kNrIceTransportTcp;
+      }
+      break;
+    case IPPROTO_UDP:
+      out->transport = kNrIceTransportUdp;
+      break;
+    default:
+      MOZ_CRASH();
+      return false;
+  }
+
+  return true;
+}
+
+static bool ToNrIceCandidate(const nr_ice_candidate& candc,
+                             NrIceCandidate* out) {
+  MOZ_ASSERT(out);
+  int r;
+  // Const-cast because the internal nICEr code isn't const-correct.
+  nr_ice_candidate* cand = const_cast<nr_ice_candidate*>(&candc);
+
+  if (!ToNrIceAddr(cand->addr, &out->cand_addr)) return false;
+
+  if (cand->mdns_addr) {
+    out->mdns_addr = cand->mdns_addr;
+  }
+
+  if (cand->isock) {
+    nr_transport_addr addr;
+    r = nr_socket_getaddr(cand->isock->sock, &addr);
+    if (r) return false;
+
+    out->is_proxied = addr.is_proxied;
+
+    if (!ToNrIceAddr(addr, &out->local_addr)) return false;
+  }
+
+  NrIceCandidate::Type type;
+
+  switch (cand->type) {
+    case HOST:
+      type = NrIceCandidate::ICE_HOST;
+      break;
+    case SERVER_REFLEXIVE:
+      type = NrIceCandidate::ICE_SERVER_REFLEXIVE;
+      break;
+    case PEER_REFLEXIVE:
+      type = NrIceCandidate::ICE_PEER_REFLEXIVE;
+      break;
+    case RELAYED:
+      type = NrIceCandidate::ICE_RELAYED;
+      break;
+    default:
+      return false;
+  }
+
+  NrIceCandidate::TcpType tcp_type;
+  switch (cand->tcp_type) {
+    case TCP_TYPE_ACTIVE:
+      tcp_type = NrIceCandidate::ICE_ACTIVE;
+      break;
+    case TCP_TYPE_PASSIVE:
+      tcp_type = NrIceCandidate::ICE_PASSIVE;
+      break;
+    case TCP_TYPE_SO:
+      tcp_type = NrIceCandidate::ICE_SO;
+      break;
+    default:
+      tcp_type = NrIceCandidate::ICE_NONE;
+      break;
+  }
+
+  out->type = type;
+  out->tcp_type = tcp_type;
+  out->codeword = candc.codeword;
+  out->label = candc.label;
+  out->trickled = candc.trickled;
+  out->priority = candc.priority;
+  return true;
+}
+
+// Make an NrIceCandidate from the candidate |cand|.
+// This is not a member fxn because we want to hide the
+// defn of nr_ice_candidate but we pass by reference.
+static UniquePtr<NrIceCandidate> MakeNrIceCandidate(
+    const nr_ice_candidate& candc) {
+  UniquePtr<NrIceCandidate> out(new NrIceCandidate());
+
+  if (!ToNrIceCandidate(candc, out.get())) {
+    return nullptr;
+  }
+  return out;
+}
+
+static bool Matches(const nr_ice_media_stream* stream, const std::string& ufrag,
+                    const std::string& pwd) {
+  return stream && (stream->ufrag == ufrag) && (stream->pwd == pwd);
+}
+
+NrIceMediaStream::NrIceMediaStream(NrIceCtx* ctx, const std::string& id,
+                                   const std::string& name, size_t components)
+    : state_(ICE_CONNECTING),
+      ctx_(ctx),
+      name_(name),
+      components_(components),
+      stream_(nullptr),
+      old_stream_(nullptr),
+      id_(id) {}
+
+NrIceMediaStream::~NrIceMediaStream() {
+  // We do not need to destroy anything. All major resources
+  // are attached to the ice ctx.
+}
+
+nsresult NrIceMediaStream::ConnectToPeer(
+    const std::string& ufrag, const std::string& pwd,
+    const std::vector<std::string>& attributes) {
+  MOZ_ASSERT(stream_);
+
+  if (Matches(old_stream_, ufrag, pwd)) {
+    // (We swap before we close so we never have stream_ == nullptr)
+    MOZ_MTLOG(ML_DEBUG,
+              "Rolling back to old stream ufrag=" << ufrag << " " << name_);
+    std::swap(stream_, old_stream_);
+    CloseStream(&old_stream_);
+  } else if (old_stream_) {
+    // Right now we wait for ICE to complete before closing the old stream.
+    // It might be worth it to close it sooner, but we don't want to close it
+    // right away.
+    MOZ_MTLOG(ML_DEBUG,
+              "ICE restart committed, marking old stream as obsolete, "
+              "beginning switchover to ufrag="
+                  << ufrag << " " << name_);
+    nr_ice_media_stream_set_obsolete(old_stream_);
+  }
+
+  nr_ice_media_stream* peer_stream;
+  if (nr_ice_peer_ctx_find_pstream(ctx_->peer(), stream_, &peer_stream)) {
+    // No peer yet
+    std::vector<char*> attributes_in;
+    attributes_in.reserve(attributes.size());
+    for (auto& attribute : attributes) {
+      MOZ_MTLOG(ML_DEBUG, "Setting " << attribute << " on stream " << name_);
+      attributes_in.push_back(const_cast<char*>(attribute.c_str()));
+    }
+
+    // Still need to call nr_ice_ctx_parse_stream_attributes.
+    int r = nr_ice_peer_ctx_parse_stream_attributes(
+        ctx_->peer(), stream_,
+        attributes_in.empty() ? nullptr : &attributes_in[0],
+        attributes_in.size());
+    if (r) {
+      MOZ_MTLOG(ML_ERROR,
+                "Couldn't parse attributes for stream " << name_ << "'");
+      return NS_ERROR_FAILURE;
+    }
+  }
+
+  return NS_OK;
+}
+
+nsresult NrIceMediaStream::SetIceCredentials(const std::string& ufrag,
+                                             const std::string& pwd) {
+  if (Matches(stream_, ufrag, pwd)) {
+    return NS_OK;
+  }
+
+  if (Matches(old_stream_, ufrag, pwd)) {
+    return NS_OK;
+  }
+
+  MOZ_MTLOG(ML_DEBUG, "Setting ICE credentials for " << name_ << " - " << ufrag
+                                                     << ":" << pwd);
+  CloseStream(&old_stream_);
+  old_stream_ = stream_;
+
+  std::string name(name_ + " - " + ufrag + ":" + pwd);
+
+  int r = nr_ice_add_media_stream(ctx_->ctx(), name.c_str(), ufrag.c_str(),
+                                  pwd.c_str(), components_, &stream_);
+  if (r) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't create ICE media stream for '"
+                            << name_ << "': error=" << r);
+    stream_ = old_stream_;
+    old_stream_ = nullptr;
+    return NS_ERROR_FAILURE;
+  }
+
+  state_ = ICE_CONNECTING;
+  return NS_OK;
+}
+
+// Parse trickle ICE candidate
+nsresult NrIceMediaStream::ParseTrickleCandidate(const std::string& candidate,
+                                                 const std::string& ufrag,
+                                                 const std::string& mdns_addr) {
+  nr_ice_media_stream* stream = GetStreamForRemoteUfrag(ufrag);
+  if (!stream) {
+    return NS_ERROR_FAILURE;
+  }
+
+  MOZ_MTLOG(ML_NOTICE, "NrIceCtx(" << ctx_->ctx()->label << ")/STREAM("
+                                   << name() << ") : parsing trickle candidate "
+                                   << candidate);
+
+  int r = nr_ice_peer_ctx_parse_trickle_candidate(
+      ctx_->peer(), stream, const_cast<char*>(candidate.c_str()),
+      mdns_addr.c_str());
+
+  if (r) {
+    if (r == R_ALREADY) {
+      MOZ_MTLOG(ML_INFO, "Trickle candidate is redundant for stream '"
+                             << name_
+                             << "' because it is completed: " << candidate);
+    } else if (r == R_REJECTED) {
+      MOZ_MTLOG(ML_INFO,
+                "Trickle candidate is ignored for stream '"
+                    << name_
+                    << "', probably because it is for an unused component"
+                    << ": " << candidate);
+    } else {
+      MOZ_MTLOG(ML_ERROR, "Couldn't parse trickle candidate for stream '"
+                              << name_ << "': " << candidate);
+      return NS_ERROR_FAILURE;
+    }
+  }
+
+  return NS_OK;
+}
+
+// Returns NS_ERROR_NOT_AVAILABLE if component is unpaired or disabled.
+nsresult NrIceMediaStream::GetActivePair(int component,
+                                         UniquePtr<NrIceCandidate>* localp,
+                                         UniquePtr<NrIceCandidate>* remotep) {
+  int r;
+  nr_ice_candidate* local_int;
+  nr_ice_candidate* remote_int;
+
+  if (!stream_) {
+    return NS_ERROR_NOT_AVAILABLE;
+  }
+
+  r = nr_ice_media_stream_get_active(ctx_->peer(), stream_, component,
+                                     &local_int, &remote_int);
+  // If result is R_REJECTED then component is unpaired or disabled.
+  if (r == R_REJECTED) return NS_ERROR_NOT_AVAILABLE;
+
+  if (r) return NS_ERROR_FAILURE;
+
+  UniquePtr<NrIceCandidate> local(MakeNrIceCandidate(*local_int));
+  if (!local) return NS_ERROR_FAILURE;
+
+  UniquePtr<NrIceCandidate> remote(MakeNrIceCandidate(*remote_int));
+  if (!remote) return NS_ERROR_FAILURE;
+
+  if (localp) *localp = std::move(local);
+  if (remotep) *remotep = std::move(remote);
+
+  return NS_OK;
+}
+
+nsresult NrIceMediaStream::GetCandidatePairs(
+    std::vector<NrIceCandidatePair>* out_pairs) const {
+  MOZ_ASSERT(out_pairs);
+  if (!stream_) {
+    return NS_ERROR_NOT_AVAILABLE;
+  }
+
+  // If we haven't at least started checking then there is nothing to report
+  if (ctx_->peer()->state != NR_ICE_PEER_STATE_PAIRED) {
+    return NS_OK;
+  }
+
+  // Get the check_list on the peer stream (this is where the check_list
+  // actually lives, not in stream_)
+  nr_ice_media_stream* peer_stream;
+  int r = nr_ice_peer_ctx_find_pstream(ctx_->peer(), stream_, &peer_stream);
+  if (r != 0) {
+    return NS_ERROR_FAILURE;
+  }
+
+  nr_ice_cand_pair *p1, *p2;
+  out_pairs->clear();
+
+  TAILQ_FOREACH(p1, &peer_stream->check_list, check_queue_entry) {
+    MOZ_ASSERT(p1);
+    MOZ_ASSERT(p1->local);
+    MOZ_ASSERT(p1->remote);
+    NrIceCandidatePair pair;
+
+    p2 = TAILQ_FIRST(&peer_stream->check_list);
+    while (p2) {
+      if (p1 == p2) {
+        /* Don't compare with our self. */
+        p2 = TAILQ_NEXT(p2, check_queue_entry);
+        continue;
+      }
+      if (strncmp(p1->codeword, p2->codeword, sizeof(p1->codeword)) == 0) {
+        /* In case of duplicate pairs we only report the one winning pair */
+        if (((p2->remote->component && (p2->remote->component->active == p2)) &&
+             !(p1->remote->component &&
+               (p1->remote->component->active == p1))) ||
+            ((p2->peer_nominated || p2->nominated) &&
+             !(p1->peer_nominated || p1->nominated)) ||
+            (p2->priority > p1->priority) ||
+            ((p2->state == NR_ICE_PAIR_STATE_SUCCEEDED) &&
+             (p1->state != NR_ICE_PAIR_STATE_SUCCEEDED)) ||
+            ((p2->state != NR_ICE_PAIR_STATE_CANCELLED) &&
+             (p1->state == NR_ICE_PAIR_STATE_CANCELLED))) {
+          /* p2 is a better pair. */
+          break;
+        }
+      }
+      p2 = TAILQ_NEXT(p2, check_queue_entry);
+    }
+    if (p2) {
+      /* p2 points to a duplicate but better pair so skip this one */
+      continue;
+    }
+
+    switch (p1->state) {
+      case NR_ICE_PAIR_STATE_FROZEN:
+        pair.state = NrIceCandidatePair::State::STATE_FROZEN;
+        break;
+      case NR_ICE_PAIR_STATE_WAITING:
+        pair.state = NrIceCandidatePair::State::STATE_WAITING;
+        break;
+      case NR_ICE_PAIR_STATE_IN_PROGRESS:
+        pair.state = NrIceCandidatePair::State::STATE_IN_PROGRESS;
+        break;
+      case NR_ICE_PAIR_STATE_FAILED:
+        pair.state = NrIceCandidatePair::State::STATE_FAILED;
+        break;
+      case NR_ICE_PAIR_STATE_SUCCEEDED:
+        pair.state = NrIceCandidatePair::State::STATE_SUCCEEDED;
+        break;
+      case NR_ICE_PAIR_STATE_CANCELLED:
+        pair.state = NrIceCandidatePair::State::STATE_CANCELLED;
+        break;
+      default:
+        MOZ_ASSERT(0);
+    }
+
+    pair.priority = p1->priority;
+    pair.nominated = p1->peer_nominated || p1->nominated;
+    pair.component_id = p1->remote->component->component_id;
+
+    // As discussed with drno: a component's can_send field (set to true
+    // by ICE consent) is a very close approximation for writable and
+    // readable. Note: the component for the local candidate never has
+    // the can_send member set to true, remote for both readable and
+    // writable. (mjf)
+    pair.writable = p1->remote->component->can_send;
+    pair.readable = p1->remote->component->can_send;
+    pair.selected =
+        p1->remote->component && p1->remote->component->active == p1;
+    pair.codeword = p1->codeword;
+    pair.bytes_sent = p1->bytes_sent;
+    pair.bytes_recvd = p1->bytes_recvd;
+    pair.ms_since_last_send =
+        p1->last_sent.tv_sec * 1000 + p1->last_sent.tv_usec / 1000;
+    pair.ms_since_last_recv =
+        p1->last_recvd.tv_sec * 1000 + p1->last_recvd.tv_usec / 1000;
+
+    if (!ToNrIceCandidate(*(p1->local), &pair.local) ||
+        !ToNrIceCandidate(*(p1->remote), &pair.remote)) {
+      return NS_ERROR_FAILURE;
+    }
+
+    out_pairs->push_back(pair);
+  }
+
+  return NS_OK;
+}
+
+nsresult NrIceMediaStream::GetDefaultCandidate(
+    int component, NrIceCandidate* candidate) const {
+  if (!stream_) {
+    return NS_ERROR_NOT_AVAILABLE;
+  }
+
+  nr_ice_candidate* cand;
+
+  int r = nr_ice_media_stream_get_default_candidate(stream_, component, &cand);
+  if (r) {
+    if (r == R_NOT_FOUND) {
+      MOZ_MTLOG(ML_INFO, "Couldn't get default ICE candidate for '"
+                             << name_ << "', no candidates.");
+    } else {
+      MOZ_MTLOG(ML_ERROR, "Couldn't get default ICE candidate for '"
+                              << name_ << "', " << r);
+    }
+    return NS_ERROR_FAILURE;
+  }
+
+  if (!ToNrIceCandidate(*cand, candidate)) {
+    MOZ_MTLOG(ML_ERROR,
+              "Failed to convert default ICE candidate for '" << name_ << "'");
+    return NS_ERROR_FAILURE;
+  }
+
+  return NS_OK;
+}
+
+std::vector<std::string> NrIceMediaStream::GetAttributes() const {
+  char** attrs = nullptr;
+  int attrct;
+  int r;
+  std::vector<std::string> ret;
+
+  if (!stream_) {
+    return ret;
+  }
+
+  r = nr_ice_media_stream_get_attributes(stream_, &attrs, &attrct);
+  if (r) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't get ICE candidates for '" << name_ << "'");
+    return ret;
+  }
+
+  for (int i = 0; i < attrct; i++) {
+    ret.push_back(attrs[i]);
+    RFREE(attrs[i]);
+  }
+
+  RFREE(attrs);
+
+  return ret;
+}
+
+static nsresult GetCandidatesFromStream(
+    nr_ice_media_stream* stream, std::vector<NrIceCandidate>* candidates) {
+  MOZ_ASSERT(candidates);
+  nr_ice_component* comp = STAILQ_FIRST(&stream->components);
+  while (comp) {
+    if (comp->state != NR_ICE_COMPONENT_DISABLED) {
+      nr_ice_candidate* cand = TAILQ_FIRST(&comp->candidates);
+      while (cand) {
+        NrIceCandidate new_cand;
+        // This can fail if the candidate is server reflexive or relayed, and
+        // has not yet received a response (ie; it doesn't know its address
+        // yet). For the purposes of this code, this isn't a candidate we're
+        // interested in, since it is not fully baked yet.
+        if (ToNrIceCandidate(*cand, &new_cand)) {
+          candidates->push_back(new_cand);
+        }
+        cand = TAILQ_NEXT(cand, entry_comp);
+      }
+    }
+    comp = STAILQ_NEXT(comp, entry);
+  }
+
+  return NS_OK;
+}
+
+nsresult NrIceMediaStream::GetLocalCandidates(
+    std::vector<NrIceCandidate>* candidates) const {
+  if (!stream_) {
+    return NS_ERROR_NOT_AVAILABLE;
+  }
+
+  return GetCandidatesFromStream(stream_, candidates);
+}
+
+nsresult NrIceMediaStream::GetRemoteCandidates(
+    std::vector<NrIceCandidate>* candidates) const {
+  if (!stream_) {
+    return NS_ERROR_NOT_AVAILABLE;
+  }
+
+  // If we haven't at least started checking then there is nothing to report
+  if (ctx_->peer()->state != NR_ICE_PEER_STATE_PAIRED) {
+    return NS_OK;
+  }
+
+  nr_ice_media_stream* peer_stream;
+  int r = nr_ice_peer_ctx_find_pstream(ctx_->peer(), stream_, &peer_stream);
+  if (r != 0) {
+    return NS_ERROR_FAILURE;
+  }
+
+  return GetCandidatesFromStream(peer_stream, candidates);
+}
+
+nsresult NrIceMediaStream::DisableComponent(int component_id) {
+  if (!stream_) return NS_ERROR_FAILURE;
+
+  int r = nr_ice_media_stream_disable_component(stream_, component_id);
+  if (r) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't disable '" << name_ << "':" << component_id);
+    return NS_ERROR_FAILURE;
+  }
+
+  return NS_OK;
+}
+
+nsresult NrIceMediaStream::GetConsentStatus(int component_id, bool* can_send,
+                                            struct timeval* ts) {
+  if (!stream_) return NS_ERROR_FAILURE;
+
+  nr_ice_media_stream* peer_stream;
+  int r = nr_ice_peer_ctx_find_pstream(ctx_->peer(), stream_, &peer_stream);
+  if (r) {
+    MOZ_MTLOG(ML_ERROR, "Failed to find peer stream for '"
+                            << name_ << "':" << component_id);
+    return NS_ERROR_FAILURE;
+  }
+
+  int send = 0;
+  r = nr_ice_media_stream_get_consent_status(peer_stream, component_id, &send,
+                                             ts);
+  if (r) {
+    MOZ_MTLOG(ML_ERROR, "Failed to get consent status for '"
+                            << name_ << "':" << component_id);
+    return NS_ERROR_FAILURE;
+  }
+  *can_send = !!send;
+
+  return NS_OK;
+}
+
+bool NrIceMediaStream::HasStream(nr_ice_media_stream* stream) const {
+  return (stream == stream_) || (stream == old_stream_);
+}
+
+nsresult NrIceMediaStream::SendPacket(int component_id,
+                                      const unsigned char* data, size_t len) {
+  nr_ice_media_stream* stream = old_stream_ ? old_stream_ : stream_;
+  if (!stream) {
+    return NS_ERROR_FAILURE;
+  }
+
+  int r = nr_ice_media_stream_send(ctx_->peer(), stream, component_id,
+                                   const_cast<unsigned char*>(data), len);
+  if (r) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't send media on '" << name_ << "'");
+    if (r == R_WOULDBLOCK) {
+      return NS_BASE_STREAM_WOULD_BLOCK;
+    }
+
+    return NS_BASE_STREAM_OSERROR;
+  }
+
+  return NS_OK;
+}
+
+void NrIceMediaStream::Ready() {
+  // This function is called whenever a stream becomes ready, but it
+  // gets fired multiple times when a stream gets nominated repeatedly.
+  if (state_ != ICE_OPEN) {
+    MOZ_MTLOG(ML_DEBUG, "Marking stream ready '" << name_ << "'");
+    state_ = ICE_OPEN;
+    NS_DispatchToCurrentThread(NewRunnableMethod<nr_ice_media_stream*>(
+        "NrIceMediaStream::DeferredCloseOldStream", this,
+        &NrIceMediaStream::DeferredCloseOldStream, old_stream_));
+    SignalReady(this);
+  } else {
+    MOZ_MTLOG(ML_DEBUG,
+              "Stream ready callback fired again for '" << name_ << "'");
+  }
+}
+
+void NrIceMediaStream::Failed() {
+  if (state_ != ICE_CLOSED) {
+    MOZ_MTLOG(ML_DEBUG, "Marking stream failed '" << name_ << "'");
+    state_ = ICE_CLOSED;
+    // We don't need the old stream anymore.
+    NS_DispatchToCurrentThread(NewRunnableMethod<nr_ice_media_stream*>(
+        "NrIceMediaStream::DeferredCloseOldStream", this,
+        &NrIceMediaStream::DeferredCloseOldStream, old_stream_));
+    SignalFailed(this);
+  }
+}
+
+void NrIceMediaStream::Close() {
+  MOZ_MTLOG(ML_DEBUG, "Marking stream closed '" << name_ << "'");
+  state_ = ICE_CLOSED;
+
+  CloseStream(&old_stream_);
+  CloseStream(&stream_);
+  ctx_ = nullptr;
+}
+
+void NrIceMediaStream::CloseStream(nr_ice_media_stream** stream) {
+  if (*stream) {
+    int r = nr_ice_remove_media_stream(ctx_->ctx(), stream);
+    if (r) {
+      MOZ_ASSERT(false, "Failed to remove stream");
+      MOZ_MTLOG(ML_ERROR, "Failed to remove stream, error=" << r);
+    }
+    *stream = nullptr;
+  }
+}
+
+void NrIceMediaStream::DeferredCloseOldStream(const nr_ice_media_stream* old) {
+  if (old == old_stream_) {
+    CloseStream(&old_stream_);
+  }
+}
+
+nr_ice_media_stream* NrIceMediaStream::GetStreamForRemoteUfrag(
+    const std::string& aUfrag) {
+  if (aUfrag.empty()) {
+    return stream_;
+  }
+
+  nr_ice_media_stream* peer_stream = nullptr;
+
+  if (!nr_ice_peer_ctx_find_pstream(ctx_->peer(), stream_, &peer_stream) &&
+      aUfrag == peer_stream->ufrag) {
+    return stream_;
+  }
+
+  if (old_stream_ &&
+      !nr_ice_peer_ctx_find_pstream(ctx_->peer(), old_stream_, &peer_stream) &&
+      aUfrag == peer_stream->ufrag) {
+    return old_stream_;
+  }
+
+  return nullptr;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/transport/nricemediastream.h b/dom/media/webrtc/transport/nricemediastream.h
new file mode 100644
index 0000000000..f18b09c47f
--- /dev/null
+++ b/dom/media/webrtc/transport/nricemediastream.h
@@ -0,0 +1,225 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+// Some of this code is cut-and-pasted from nICEr. Copyright is:
+
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+// This is a wrapper around the nICEr ICE stack
+#ifndef nricemediastream_h__
+#define nricemediastream_h__
+
+#include <string>
+#include <vector>
+
+#include "sigslot.h"
+
+#include "mozilla/RefPtr.h"
+#include "mozilla/UniquePtr.h"
+#include "nscore.h"
+
+#include "m_cpp_utils.h"
+
+namespace mozilla {
+
+typedef struct nr_ice_ctx_ nr_ice_ctx;
+typedef struct nr_ice_peer_ctx_ nr_ice_peer_ctx;
+typedef struct nr_ice_media_stream_ nr_ice_media_stream;
+
+class NrIceCtx;
+
+struct NrIceAddr {
+  std::string host;
+  uint16_t port;
+  std::string transport;
+};
+
+/* A summary of a candidate, for use in asking which candidate
+   pair is active */
+struct NrIceCandidate {
+  enum Type { ICE_HOST, ICE_SERVER_REFLEXIVE, ICE_PEER_REFLEXIVE, ICE_RELAYED };
+
+  enum TcpType { ICE_NONE, ICE_ACTIVE, ICE_PASSIVE, ICE_SO };
+
+  NrIceAddr cand_addr;
+  NrIceAddr local_addr;
+  std::string mdns_addr;
+  Type type;
+  TcpType tcp_type;
+  std::string codeword;
+  std::string label;
+  bool trickled;
+  uint32_t priority;
+  bool is_proxied = false;
+};
+
+struct NrIceCandidatePair {
+  enum State {
+    STATE_FROZEN,
+    STATE_WAITING,
+    STATE_IN_PROGRESS,
+    STATE_FAILED,
+    STATE_SUCCEEDED,
+    STATE_CANCELLED
+  };
+
+  State state;
+  uint64_t priority;
+  // Set regardless of who nominated it. Does not necessarily mean that it is
+  // ready to be selected (ie; nominated by peer, but our check has not
+  // succeeded yet.) Note: since this implementation uses aggressive nomination,
+  // when we are the controlling agent, this will always be set if the pair is
+  // in STATE_SUCCEEDED.
+  bool nominated;
+  bool writable;
+  bool readable;
+  // Set if this candidate pair has been selected. Note: Since we are using
+  // aggressive nomination, this could change frequently as ICE runs.
+  bool selected;
+  NrIceCandidate local;
+  NrIceCandidate remote;
+  // TODO(bcampen@mozilla.com): Is it important to put the foundation in here?
+  std::string codeword;
+  uint64_t component_id;
+
+  // for RTCIceCandidatePairStats
+  uint64_t bytes_sent;
+  uint64_t bytes_recvd;
+  uint64_t ms_since_last_send;
+  uint64_t ms_since_last_recv;
+};
+
+class NrIceMediaStream {
+ public:
+  NrIceMediaStream(NrIceCtx* ctx, const std::string& id,
+                   const std::string& name, size_t components);
+
+  nsresult SetIceCredentials(const std::string& ufrag, const std::string& pwd);
+  nsresult ConnectToPeer(const std::string& ufrag, const std::string& pwd,
+                         const std::vector<std::string>& peer_attrs);
+  enum State { ICE_CONNECTING, ICE_OPEN, ICE_CLOSED };
+
+  State state() const { return state_; }
+
+  // The name of the stream
+  const std::string& name() const { return name_; }
+
+  // Get all the ICE attributes; used for testing
+  std::vector<std::string> GetAttributes() const;
+
+  nsresult GetLocalCandidates(std::vector<NrIceCandidate>* candidates) const;
+  nsresult GetRemoteCandidates(std::vector<NrIceCandidate>* candidates) const;
+
+  // Get all candidate pairs, whether in the check list or triggered check
+  // queue, in priority order. |out_pairs| is cleared before being filled.
+  nsresult GetCandidatePairs(std::vector<NrIceCandidatePair>* out_pairs) const;
+
+  nsresult GetDefaultCandidate(int component, NrIceCandidate* candidate) const;
+
+  // Parse trickle ICE candidate
+  nsresult ParseTrickleCandidate(const std::string& candidate,
+                                 const std::string& ufrag,
+                                 const std::string& mdns_addr);
+
+  // Disable a component
+  nsresult DisableComponent(int component);
+
+  // Get the candidate pair currently active. It's the
+  // caller's responsibility to free these.
+  nsresult GetActivePair(int component, UniquePtr<NrIceCandidate>* local,
+                         UniquePtr<NrIceCandidate>* remote);
+
+  // Get the current ICE consent send status plus the timeval of the last
+  // consent update time.
+  nsresult GetConsentStatus(int component, bool* can_send, struct timeval* ts);
+
+  // The number of components
+  size_t components() const { return components_; }
+
+  bool HasStream(nr_ice_media_stream* stream) const;
+  // Signals to indicate events. API users can (and should)
+  // register for these.
+
+  // Send a packet
+  nsresult SendPacket(int component_id, const unsigned char* data, size_t len);
+
+  // Set your state to ready. Called by the NrIceCtx;
+  void Ready();
+  void Failed();
+
+  // Close the stream. Called by the NrIceCtx.
+  // Different from the destructor because other people
+  // might be holding RefPtrs but we want those writes to fail once
+  // the context has been destroyed.
+  void Close();
+
+  // So the receiver of SignalCandidate can determine which transport
+  // the candidate belongs to.
+  const std::string& GetId() const { return id_; }
+
+  sigslot::signal5<NrIceMediaStream*, const std::string&, const std::string&,
+                   const std::string&, const std::string&>
+      SignalCandidate;  // A new ICE candidate:
+
+  sigslot::signal1<NrIceMediaStream*> SignalReady;   // Candidate pair ready.
+  sigslot::signal1<NrIceMediaStream*> SignalFailed;  // Candidate pair failed.
+  sigslot::signal4<NrIceMediaStream*, int, const unsigned char*, int>
+      SignalPacketReceived;  // Incoming packet
+
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(NrIceMediaStream)
+
+ private:
+  ~NrIceMediaStream();
+
+  DISALLOW_COPY_ASSIGN(NrIceMediaStream);
+
+  void CloseStream(nr_ice_media_stream** stream);
+  void DeferredCloseOldStream(const nr_ice_media_stream* old);
+  nr_ice_media_stream* GetStreamForRemoteUfrag(const std::string& ufrag);
+
+  State state_;
+  RefPtr<NrIceCtx> ctx_;
+  const std::string name_;
+  const size_t components_;
+  nr_ice_media_stream* stream_;
+  nr_ice_media_stream* old_stream_;
+  const std::string id_;
+};
+
+}  // namespace mozilla
+#endif
diff --git a/dom/media/webrtc/transport/nriceresolver.cpp b/dom/media/webrtc/transport/nriceresolver.cpp
new file mode 100644
index 0000000000..e1737e0e65
--- /dev/null
+++ b/dom/media/webrtc/transport/nriceresolver.cpp
@@ -0,0 +1,234 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original authors: jib@mozilla.com, ekr@rtfm.com
+
+// Some of this code is cut-and-pasted from nICEr. Copyright is:
+
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include "logging.h"
+#include "nspr.h"
+#include "prnetdb.h"
+
+#include "mozilla/Assertions.h"
+
+extern "C" {
+#include "nr_api.h"
+#include "async_timer.h"
+#include "nr_resolver.h"
+#include "transport_addr.h"
+}
+
+#include "mozilla/net/DNS.h"  // TODO(jib@mozilla.com) down here because bug 848578
+#include "nsThreadUtils.h"
+#include "nsServiceManagerUtils.h"
+#include "nsIDNSService.h"
+#include "nsIDNSListener.h"
+#include "nsIDNSRecord.h"
+#include "nsNetCID.h"
+#include "nsCOMPtr.h"
+#include "nriceresolver.h"
+#include "nr_socket_prsock.h"
+#include "transport/runnable_utils.h"
+
+namespace mozilla {
+
+MOZ_MTLOG_MODULE("mtransport")
+
+NrIceResolver::NrIceResolver()
+    : vtbl_(new nr_resolver_vtbl())
+#ifdef DEBUG
+      ,
+      allocated_resolvers_(0)
+#endif
+{
+  vtbl_->destroy = &NrIceResolver::destroy;
+  vtbl_->resolve = &NrIceResolver::resolve;
+  vtbl_->cancel = &NrIceResolver::cancel;
+}
+
+NrIceResolver::~NrIceResolver() {
+  MOZ_ASSERT(!allocated_resolvers_);
+  delete vtbl_;
+}
+
+nsresult NrIceResolver::Init() {
+  nsresult rv;
+
+  sts_thread_ = do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &rv);
+  MOZ_ASSERT(NS_SUCCEEDED(rv));
+  dns_ = do_GetService(NS_DNSSERVICE_CONTRACTID, &rv);
+  if (NS_WARN_IF(NS_FAILED(rv))) {
+    MOZ_MTLOG(ML_ERROR, "Could not acquire DNS service");
+  }
+  return rv;
+}
+
+nr_resolver* NrIceResolver::AllocateResolver() {
+  nr_resolver* resolver;
+
+  int r = nr_resolver_create_int((void*)this, vtbl_, &resolver);
+  MOZ_ASSERT(!r);
+  if (r) {
+    MOZ_MTLOG(ML_ERROR, "nr_resolver_create_int failed");
+    return nullptr;
+  }
+  // We must be available to allocators until they all call DestroyResolver,
+  // because allocators may (and do) outlive the originator of NrIceResolver.
+  AddRef();
+#ifdef DEBUG
+  ++allocated_resolvers_;
+#endif
+  return resolver;
+}
+
+void NrIceResolver::DestroyResolver() {
+#ifdef DEBUG
+  --allocated_resolvers_;
+#endif
+  // Undoes Addref in AllocateResolver so the NrIceResolver can be freed.
+  Release();
+}
+
+int NrIceResolver::destroy(void** objp) {
+  if (!objp || !*objp) return 0;
+  NrIceResolver* resolver = static_cast<NrIceResolver*>(*objp);
+  *objp = nullptr;
+  resolver->DestroyResolver();
+  return 0;
+}
+
+int NrIceResolver::resolve(void* obj, nr_resolver_resource* resource,
+                           int (*cb)(void* cb_arg, nr_transport_addr* addr),
+                           void* cb_arg, void** handle) {
+  MOZ_ASSERT(obj);
+  return static_cast<NrIceResolver*>(obj)->resolve(resource, cb, cb_arg,
+                                                   handle);
+}
+
+int NrIceResolver::resolve(nr_resolver_resource* resource,
+                           int (*cb)(void* cb_arg, nr_transport_addr* addr),
+                           void* cb_arg, void** handle) {
+  int _status;
+  MOZ_ASSERT(allocated_resolvers_ > 0);
+  ASSERT_ON_THREAD(sts_thread_);
+  RefPtr<PendingResolution> pr;
+  nsIDNSService::DNSFlags resolve_flags = nsIDNSService::RESOLVE_DEFAULT_FLAGS;
+  OriginAttributes attrs;
+
+  if (resource->transport_protocol != IPPROTO_UDP &&
+      resource->transport_protocol != IPPROTO_TCP) {
+    MOZ_MTLOG(ML_ERROR, "Only UDP and TCP are supported.");
+    ABORT(R_NOT_FOUND);
+  }
+  pr = new PendingResolution(
+      sts_thread_, resource->port ? resource->port : 3478,
+      resource->transport_protocol ? resource->transport_protocol : IPPROTO_UDP,
+      cb, cb_arg);
+
+  switch (resource->address_family) {
+    case AF_INET:
+      resolve_flags = nsIDNSService::RESOLVE_DISABLE_IPV6;
+      break;
+    case AF_INET6:
+      resolve_flags = nsIDNSService::RESOLVE_DISABLE_IPV4;
+      break;
+    default:
+      ABORT(R_BAD_ARGS);
+  }
+
+  if (NS_FAILED(dns_->AsyncResolveNative(
+          nsAutoCString(resource->domain_name),
+          nsIDNSService::RESOLVE_TYPE_DEFAULT, resolve_flags, nullptr, pr,
+          sts_thread_, attrs, getter_AddRefs(pr->request_)))) {
+    MOZ_MTLOG(ML_ERROR, "AsyncResolve failed.");
+    ABORT(R_NOT_FOUND);
+  }
+  // Because the C API offers no "finished" method to release the handle we
+  // return, we cannot return the request we got from AsyncResolve directly.
+  //
+  // Instead, we return an addref'ed reference to PendingResolution itself,
+  // which in turn holds the request and coordinates between cancel and
+  // OnLookupComplete to release it only once.
+  pr.forget(handle);
+
+  _status = 0;
+abort:
+  return _status;
+}
+
+nsresult NrIceResolver::PendingResolution::OnLookupComplete(
+    nsICancelable* request, nsIDNSRecord* aRecord, nsresult status) {
+  ASSERT_ON_THREAD(thread_);
+  // First check if we've been canceled. This is single-threaded on the STS
+  // thread, but cancel() cannot guarantee this event isn't on the queue.
+  if (request_) {
+    nr_transport_addr* cb_addr = nullptr;
+    nr_transport_addr ta;
+    // TODO(jib@mozilla.com): Revisit when we do TURN.
+    if (NS_SUCCEEDED(status)) {
+      net::NetAddr na;
+      nsCOMPtr<nsIDNSAddrRecord> record = do_QueryInterface(aRecord);
+      if (record && NS_SUCCEEDED(record->GetNextAddr(port_, &na))) {
+        MOZ_ALWAYS_TRUE(nr_netaddr_to_transport_addr(&na, &ta, transport_) ==
+                        0);
+        cb_addr = &ta;
+      }
+    }
+    cb_(cb_arg_, cb_addr);
+    request_ = nullptr;
+    Release();
+  }
+  return NS_OK;
+}
+
+int NrIceResolver::cancel(void* obj, void* handle) {
+  MOZ_ALWAYS_TRUE(obj);
+  MOZ_ASSERT(handle);
+  ASSERT_ON_THREAD(static_cast<NrIceResolver*>(obj)->sts_thread_);
+  return static_cast<PendingResolution*>(handle)->cancel();
+}
+
+int NrIceResolver::PendingResolution::cancel() {
+  request_->Cancel(NS_ERROR_ABORT);
+  request_ = nullptr;
+  Release();
+  return 0;
+}
+
+NS_IMPL_ISUPPORTS(NrIceResolver::PendingResolution, nsIDNSListener);
+}  // End of namespace mozilla
diff --git a/dom/media/webrtc/transport/nriceresolver.h b/dom/media/webrtc/transport/nriceresolver.h
new file mode 100644
index 0000000000..7399c78050
--- /dev/null
+++ b/dom/media/webrtc/transport/nriceresolver.h
@@ -0,0 +1,119 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original authors: jib@mozilla.com, ekr@rtfm.com
+
+// Some of this code is cut-and-pasted from nICEr. Copyright is:
+
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef nriceresolver_h__
+#define nriceresolver_h__
+
+#include "nsIDNSService.h"
+#include "nsIDNSListener.h"
+#include "nsICancelable.h"
+#include "nricectx.h"
+
+typedef struct nr_resolver_ nr_resolver;
+typedef struct nr_resolver_vtbl_ nr_resolver_vtbl;
+typedef struct nr_transport_addr_ nr_transport_addr;
+typedef struct nr_resolver_resource_ nr_resolver_resource;
+
+namespace mozilla {
+
+class NrIceResolver {
+ private:
+  ~NrIceResolver();
+
+ public:
+  NrIceResolver();
+
+  nsresult Init();
+  nr_resolver* AllocateResolver();
+  void DestroyResolver();
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(NrIceResolver)
+
+  int resolve(nr_resolver_resource* resource,
+              int (*cb)(void* cb_arg, nr_transport_addr* addr), void* cb_arg,
+              void** handle);
+
+ private:
+  // Implementations of vtbl functions
+  static int destroy(void** objp);
+  static int resolve(void* obj, nr_resolver_resource* resource,
+                     int (*cb)(void* cb_arg, nr_transport_addr* addr),
+                     void* cb_arg, void** handle);
+  static void resolve_cb(NR_SOCKET s, int how, void* cb_arg);
+  static int cancel(void* obj, void* handle);
+
+  class PendingResolution : public nsIDNSListener {
+   public:
+    PendingResolution(nsIEventTarget* thread, uint16_t port, int transport,
+                      int (*cb)(void* cb_arg, nr_transport_addr* addr),
+                      void* cb_arg)
+        : thread_(thread),
+          port_(port),
+          transport_(transport),
+          cb_(cb),
+          cb_arg_(cb_arg) {}
+    NS_IMETHOD OnLookupComplete(nsICancelable* request, nsIDNSRecord* record,
+                                nsresult status) override;
+
+    int cancel();
+    nsCOMPtr<nsICancelable> request_;
+    NS_DECL_THREADSAFE_ISUPPORTS
+
+   private:
+    virtual ~PendingResolution() = default;
+
+    nsCOMPtr<nsIEventTarget> thread_;
+    uint16_t port_;
+    int transport_;
+    int (*cb_)(void* cb_arg, nr_transport_addr* addr);
+    void* cb_arg_;
+  };
+
+  nr_resolver_vtbl* vtbl_;
+  nsCOMPtr<nsIEventTarget> sts_thread_;
+  nsCOMPtr<nsIDNSService> dns_;
+#ifdef DEBUG
+  int allocated_resolvers_;
+#endif
+};
+
+}  // End of namespace mozilla
+#endif
diff --git a/dom/media/webrtc/transport/nriceresolverfake.cpp b/dom/media/webrtc/transport/nriceresolverfake.cpp
new file mode 100644
index 0000000000..93ef37efc1
--- /dev/null
+++ b/dom/media/webrtc/transport/nriceresolverfake.cpp
@@ -0,0 +1,174 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+// Some of this code is cut-and-pasted from nICEr. Copyright is:
+
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include "prio.h"
+#include "mozilla/Assertions.h"
+
+extern "C" {
+#include "async_wait.h"
+#include "async_timer.h"
+#include "nr_resolver.h"
+#include "r_macros.h"
+#include "transport_addr.h"
+}
+
+#include "nriceresolverfake.h"
+#include "nr_socket_prsock.h"
+
+namespace mozilla {
+
+NrIceResolverFake::NrIceResolverFake()
+    : vtbl_(new nr_resolver_vtbl),
+      addrs_(),
+      delay_ms_(100),
+      allocated_resolvers_(0) {
+  vtbl_->destroy = &NrIceResolverFake::destroy;
+  vtbl_->resolve = &NrIceResolverFake::resolve;
+  vtbl_->cancel = &NrIceResolverFake::cancel;
+}
+
+NrIceResolverFake::~NrIceResolverFake() {
+  MOZ_ASSERT(allocated_resolvers_ == 0);
+  delete vtbl_;
+}
+
+nr_resolver* NrIceResolverFake::AllocateResolver() {
+  nr_resolver* resolver;
+
+  int r = nr_resolver_create_int((void*)this, vtbl_, &resolver);
+  MOZ_ASSERT(!r);
+  if (r) return nullptr;
+
+  ++allocated_resolvers_;
+
+  return resolver;
+}
+
+void NrIceResolverFake::DestroyResolver() { --allocated_resolvers_; }
+
+int NrIceResolverFake::destroy(void** objp) {
+  if (!objp || !*objp) return 0;
+
+  NrIceResolverFake* fake = static_cast<NrIceResolverFake*>(*objp);
+  *objp = nullptr;
+
+  fake->DestroyResolver();
+
+  return 0;
+}
+
+int NrIceResolverFake::resolve(void* obj, nr_resolver_resource* resource,
+                               int (*cb)(void* cb_arg, nr_transport_addr* addr),
+                               void* cb_arg, void** handle) {
+  int r, _status;
+
+  MOZ_ASSERT(obj);
+  NrIceResolverFake* fake = static_cast<NrIceResolverFake*>(obj);
+
+  MOZ_ASSERT(fake->allocated_resolvers_ > 0);
+
+  PendingResolution* pending = new PendingResolution(
+      fake, resource->domain_name, resource->port ? resource->port : 3478,
+      resource->transport_protocol ? resource->transport_protocol : IPPROTO_UDP,
+      resource->address_family, cb, cb_arg);
+
+  if ((r = NR_ASYNC_TIMER_SET(fake->delay_ms_, NrIceResolverFake::resolve_cb,
+                              (void*)pending, &pending->timer_handle_))) {
+    delete pending;
+    ABORT(r);
+  }
+  *handle = pending;
+
+  _status = 0;
+abort:
+  return (_status);
+}
+
+void NrIceResolverFake::resolve_cb(NR_SOCKET s, int how, void* cb_arg) {
+  MOZ_ASSERT(cb_arg);
+  PendingResolution* pending = static_cast<PendingResolution*>(cb_arg);
+
+  const PRNetAddr* addr =
+      pending->resolver_->Resolve(pending->hostname_, pending->address_family_);
+
+  if (addr) {
+    nr_transport_addr transport_addr;
+
+    int r = nr_praddr_to_transport_addr(addr, &transport_addr,
+                                        pending->transport_, 0);
+    MOZ_ASSERT(!r);
+    if (r) goto abort;
+
+    r = nr_transport_addr_set_port(&transport_addr, pending->port_);
+    MOZ_ASSERT(!r);
+    if (r) goto abort;
+
+    /* Fill in the address string */
+    r = nr_transport_addr_fmt_addr_string(&transport_addr);
+    MOZ_ASSERT(!r);
+    if (r) goto abort;
+
+    pending->cb_(pending->cb_arg_, &transport_addr);
+    delete pending;
+    return;
+  }
+
+abort:
+  // Resolution failed.
+  pending->cb_(pending->cb_arg_, nullptr);
+
+  delete pending;
+}
+
+int NrIceResolverFake::cancel(void* obj, void* handle) {
+  MOZ_ASSERT(obj);
+  MOZ_ASSERT(static_cast<NrIceResolverFake*>(obj)->allocated_resolvers_ > 0);
+
+  PendingResolution* pending = static_cast<PendingResolution*>(handle);
+
+  NR_async_timer_cancel(pending->timer_handle_);
+  delete pending;
+
+  return (0);
+}
+
+}  // End of namespace mozilla
diff --git a/dom/media/webrtc/transport/nriceresolverfake.h b/dom/media/webrtc/transport/nriceresolverfake.h
new file mode 100644
index 0000000000..e46ce603de
--- /dev/null
+++ b/dom/media/webrtc/transport/nriceresolverfake.h
@@ -0,0 +1,137 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+// Some of this code is cut-and-pasted from nICEr. Copyright is:
+
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef nriceresolverfake_h__
+#define nriceresolverfake_h__
+
+#include <map>
+#include <string>
+
+#include "csi_platform.h"
+
+typedef struct nr_resolver_ nr_resolver;
+typedef struct nr_resolver_vtbl_ nr_resolver_vtbl;
+typedef struct nr_transport_addr_ nr_transport_addr;
+typedef struct nr_resolver_resource_ nr_resolver_resource;
+
+namespace mozilla {
+
+class NrIceResolverFake {
+ public:
+  NrIceResolverFake();
+  ~NrIceResolverFake();
+
+  void SetAddr(const std::string& hostname, const PRNetAddr& addr) {
+    switch (addr.raw.family) {
+      case AF_INET:
+        addrs_[hostname] = addr;
+        break;
+      case AF_INET6:
+        addrs6_[hostname] = addr;
+        break;
+      default:
+        MOZ_CRASH();
+    }
+  }
+
+  nr_resolver* AllocateResolver();
+
+  void DestroyResolver();
+
+ private:
+  // Implementations of vtbl functions
+  static int destroy(void** objp);
+  static int resolve(void* obj, nr_resolver_resource* resource,
+                     int (*cb)(void* cb_arg, nr_transport_addr* addr),
+                     void* cb_arg, void** handle);
+  static void resolve_cb(NR_SOCKET s, int how, void* cb_arg);
+  static int cancel(void* obj, void* handle);
+
+  // Get an address.
+  const PRNetAddr* Resolve(const std::string& hostname, int address_family) {
+    switch (address_family) {
+      case AF_INET:
+        if (!addrs_.count(hostname)) return nullptr;
+
+        return &addrs_[hostname];
+      case AF_INET6:
+        if (!addrs6_.count(hostname)) return nullptr;
+
+        return &addrs6_[hostname];
+      default:
+        MOZ_CRASH();
+    }
+  }
+
+  struct PendingResolution {
+    PendingResolution(NrIceResolverFake* resolver, const std::string& hostname,
+                      uint16_t port, int transport, int address_family,
+                      int (*cb)(void* cb_arg, nr_transport_addr* addr),
+                      void* cb_arg)
+        : resolver_(resolver),
+          hostname_(hostname),
+          port_(port),
+          transport_(transport),
+          address_family_(address_family),
+          cb_(cb),
+          cb_arg_(cb_arg) {}
+
+    NrIceResolverFake* resolver_;
+    std::string hostname_;
+    uint16_t port_;
+    int transport_;
+    int address_family_;
+    int (*cb_)(void* cb_arg, nr_transport_addr* addr);
+    void* cb_arg_;
+    void* timer_handle_;
+  };
+
+  nr_resolver_vtbl* vtbl_;
+  std::map<std::string, PRNetAddr> addrs_;
+  std::map<std::string, PRNetAddr> addrs6_;
+  uint32_t delay_ms_;
+  int allocated_resolvers_;
+};
+
+}  // End of namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/transport/nricestunaddr.cpp b/dom/media/webrtc/transport/nricestunaddr.cpp
new file mode 100644
index 0000000000..d1e8d3ac52
--- /dev/null
+++ b/dom/media/webrtc/transport/nricestunaddr.cpp
@@ -0,0 +1,93 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "logging.h"
+
+// nICEr includes
+extern "C" {
+#include "nr_api.h"
+#include "local_addr.h"
+}
+
+// Local includes
+#include "nricestunaddr.h"
+
+namespace mozilla {
+
+MOZ_MTLOG_MODULE("mtransport")
+
+NrIceStunAddr::NrIceStunAddr() : localAddr_(new nr_local_addr) {
+  memset(localAddr_, 0, sizeof(nr_local_addr));
+}
+
+NrIceStunAddr::NrIceStunAddr(const nr_local_addr* addr)
+    : localAddr_(new nr_local_addr) {
+  nr_local_addr_copy(localAddr_, const_cast<nr_local_addr*>(addr));
+}
+
+NrIceStunAddr::NrIceStunAddr(const NrIceStunAddr& rhs)
+    : localAddr_(new nr_local_addr) {
+  nr_local_addr_copy(localAddr_, const_cast<nr_local_addr*>(rhs.localAddr_));
+}
+
+NrIceStunAddr::~NrIceStunAddr() { delete localAddr_; }
+
+size_t NrIceStunAddr::SerializationBufferSize() const {
+  return sizeof(nr_local_addr);
+}
+
+nsresult NrIceStunAddr::Serialize(char* buffer, size_t buffer_size) const {
+  if (buffer_size != sizeof(nr_local_addr)) {
+    MOZ_MTLOG(ML_ERROR,
+              "Failed trying to serialize NrIceStunAddr, "
+              "input buffer length ("
+                  << buffer_size << ") does not match required length ("
+                  << sizeof(nr_local_addr) << ")");
+    MOZ_ASSERT(false, "Failed to serialize NrIceStunAddr, bad buffer size");
+    return NS_ERROR_FAILURE;
+  }
+
+  nr_local_addr* toAddr = (nr_local_addr*)buffer;
+  if (nr_local_addr_copy(toAddr, localAddr_)) {
+    MOZ_MTLOG(ML_ERROR,
+              "Failed trying to serialize NrIceStunAddr, "
+              "could not copy nr_local_addr.");
+    MOZ_ASSERT(false,
+               "Failed to serialize NrIceStunAddr, nr_local_addr_copy failed");
+    return NS_ERROR_FAILURE;
+  }
+
+  return NS_OK;
+}
+
+nsresult NrIceStunAddr::Deserialize(const char* buffer, size_t buffer_size) {
+  if (buffer_size != sizeof(nr_local_addr)) {
+    MOZ_MTLOG(ML_ERROR,
+              "Failed trying to deserialize NrIceStunAddr, "
+              "input buffer length ("
+                  << buffer_size << ") does not match required length ("
+                  << sizeof(nr_local_addr) << ")");
+    MOZ_ASSERT(false, "Failed to deserialize NrIceStunAddr, bad buffer size");
+    return NS_ERROR_FAILURE;
+  }
+
+  nr_local_addr* from_addr =
+      const_cast<nr_local_addr*>((const nr_local_addr*)buffer);
+
+  // At this point, from_addr->addr.addr is invalid (null), but will
+  // be fixed by nr_local_addr_copy.
+  if (nr_local_addr_copy(localAddr_, from_addr)) {
+    MOZ_MTLOG(ML_ERROR,
+              "Failed trying to deserialize NrIceStunAddr, "
+              "could not copy nr_local_addr.");
+    MOZ_ASSERT(
+        false,
+        "Failed to deserialize NrIceStunAddr, nr_local_addr_copy failed");
+    return NS_ERROR_FAILURE;
+  }
+
+  return NS_OK;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/transport/nricestunaddr.h b/dom/media/webrtc/transport/nricestunaddr.h
new file mode 100644
index 0000000000..8ebcdd6fe7
--- /dev/null
+++ b/dom/media/webrtc/transport/nricestunaddr.h
@@ -0,0 +1,36 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef nricestunaddr_h__
+#define nricestunaddr_h__
+
+#include "nsError.h"  // for nsresult
+
+typedef struct nr_local_addr_ nr_local_addr;
+
+namespace mozilla {
+
+class NrIceStunAddr {
+ public:
+  NrIceStunAddr();  // needed for IPC deserialization
+  explicit NrIceStunAddr(const nr_local_addr* addr);
+  NrIceStunAddr(const NrIceStunAddr& rhs);
+
+  ~NrIceStunAddr();
+
+  const nr_local_addr& localAddr() const { return *localAddr_; }
+
+  // serialization/deserialization helper functions for use
+  // in dom/media/webrtc/transport/ipc/NrIceStunAddrMessagUtils.h
+  size_t SerializationBufferSize() const;
+  nsresult Serialize(char* buffer, size_t buffer_size) const;
+  nsresult Deserialize(const char* buffer, size_t buffer_size);
+
+ private:
+  nr_local_addr* localAddr_;
+};
+
+}  // namespace mozilla
+
+#endif  // nricestunaddr_h__
diff --git a/dom/media/webrtc/transport/nrinterfaceprioritizer.cpp b/dom/media/webrtc/transport/nrinterfaceprioritizer.cpp
new file mode 100644
index 0000000000..f022f8c29a
--- /dev/null
+++ b/dom/media/webrtc/transport/nrinterfaceprioritizer.cpp
@@ -0,0 +1,258 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+#include <algorithm>
+#include <map>
+#include <set>
+#include <string>
+#include <vector>
+#include "logging.h"
+#include "nr_api.h"
+#include "nrinterfaceprioritizer.h"
+
+MOZ_MTLOG_MODULE("mtransport")
+
+namespace {
+
+class LocalAddress {
+ public:
+  LocalAddress()
+      : ifname_(),
+        addr_(),
+        key_(),
+        is_vpn_(-1),
+        estimated_speed_(-1),
+        type_preference_(-1),
+        ip_version_(-1) {}
+
+  bool Init(const nr_local_addr& local_addr) {
+    ifname_ = local_addr.addr.ifname;
+
+    char buf[MAXIFNAME + 47];
+    int r = nr_transport_addr_fmt_ifname_addr_string(&local_addr.addr, buf,
+                                                     sizeof(buf));
+    if (r) {
+      MOZ_MTLOG(ML_ERROR, "Error formatting interface key.");
+      return false;
+    }
+    key_ = buf;
+
+    r = nr_transport_addr_get_addrstring(&local_addr.addr, buf, sizeof(buf));
+    if (r) {
+      MOZ_MTLOG(ML_ERROR, "Error formatting address string.");
+      return false;
+    }
+    addr_ = buf;
+
+    is_vpn_ = (local_addr.interface.type & NR_INTERFACE_TYPE_VPN) != 0 ? 1 : 0;
+    estimated_speed_ = local_addr.interface.estimated_speed;
+    type_preference_ = GetNetworkTypePreference(local_addr.interface.type);
+    ip_version_ = local_addr.addr.ip_version;
+    return true;
+  }
+
+  bool operator<(const LocalAddress& rhs) const {
+    // Interface that is "less" here is preferred.
+    // If type preferences are different, we should simply sort by
+    // |type_preference_|.
+    if (type_preference_ != rhs.type_preference_) {
+      return type_preference_ < rhs.type_preference_;
+    }
+
+    // If type preferences are the same, the next thing we use to sort is vpn.
+    // If two LocalAddress are different in |is_vpn_|, the LocalAddress that is
+    // not in vpn gets priority.
+    if (is_vpn_ != rhs.is_vpn_) {
+      return is_vpn_ < rhs.is_vpn_;
+    }
+
+    // Compare estimated speed.
+    if (estimated_speed_ != rhs.estimated_speed_) {
+      return estimated_speed_ > rhs.estimated_speed_;
+    }
+
+    // See if our hard-coded pref list helps us.
+    auto thisindex = std::find(interface_preference_list().begin(),
+                               interface_preference_list().end(), ifname_);
+    auto rhsindex = std::find(interface_preference_list().begin(),
+                              interface_preference_list().end(), rhs.ifname_);
+    if (thisindex != rhsindex) {
+      return thisindex < rhsindex;
+    }
+
+    // Prefer IPV6 over IPV4
+    if (ip_version_ != rhs.ip_version_) {
+      return ip_version_ > rhs.ip_version_;
+    }
+
+    // Now we start getting into arbitrary stuff
+    if (ifname_ != rhs.ifname_) {
+      return ifname_ < rhs.ifname_;
+    }
+
+    return addr_ < rhs.addr_;
+  }
+
+  const std::string& GetKey() const { return key_; }
+
+ private:
+  // Getting the preference corresponding to a type. Getting lower number here
+  // means the type of network is preferred.
+  static inline int GetNetworkTypePreference(int type) {
+    if (type & NR_INTERFACE_TYPE_WIRED) {
+      return 1;
+    }
+    if (type & NR_INTERFACE_TYPE_WIFI) {
+      return 2;
+    }
+    if (type & NR_INTERFACE_TYPE_MOBILE) {
+      return 3;
+    }
+    if (type & NR_INTERFACE_TYPE_TEREDO) {
+      // Teredo gets penalty because it's IP relayed
+      return 5;
+    }
+    return 4;
+  }
+
+  // TODO(bug 895790): Once we can get useful interface properties on Darwin,
+  // we should remove this stuff.
+  static const std::vector<std::string>& interface_preference_list() {
+    static std::vector<std::string> list(build_interface_preference_list());
+    return list;
+  }
+
+  static std::vector<std::string> build_interface_preference_list() {
+    std::vector<std::string> result;
+    result.push_back("rl0");
+    result.push_back("wi0");
+    result.push_back("en0");
+    result.push_back("enp2s0");
+    result.push_back("enp3s0");
+    result.push_back("en1");
+    result.push_back("en2");
+    result.push_back("en3");
+    result.push_back("eth0");
+    result.push_back("eth1");
+    result.push_back("eth2");
+    result.push_back("em1");
+    result.push_back("em0");
+    result.push_back("ppp");
+    result.push_back("ppp0");
+    result.push_back("vmnet1");
+    result.push_back("vmnet0");
+    result.push_back("vmnet3");
+    result.push_back("vmnet4");
+    result.push_back("vmnet5");
+    result.push_back("vmnet6");
+    result.push_back("vmnet7");
+    result.push_back("vmnet8");
+    result.push_back("virbr0");
+    result.push_back("wlan0");
+    result.push_back("lo0");
+    return result;
+  }
+
+  std::string ifname_;
+  std::string addr_;
+  std::string key_;
+  int is_vpn_;
+  int estimated_speed_;
+  int type_preference_;
+  int ip_version_;
+};
+
+class InterfacePrioritizer {
+ public:
+  InterfacePrioritizer() : local_addrs_(), preference_map_(), sorted_(false) {}
+
+  int add(const nr_local_addr* iface) {
+    LocalAddress addr;
+    if (!addr.Init(*iface)) {
+      return R_FAILED;
+    }
+    std::pair<std::set<LocalAddress>::iterator, bool> r =
+        local_addrs_.insert(addr);
+    if (!r.second) {
+      return R_ALREADY;  // This address is already in the set.
+    }
+    sorted_ = false;
+    return 0;
+  }
+
+  int sort() {
+    UCHAR tmp_pref = 127;
+    preference_map_.clear();
+    for (const auto& local_addr : local_addrs_) {
+      if (tmp_pref == 0) {
+        return R_FAILED;
+      }
+      preference_map_.insert(make_pair(local_addr.GetKey(), tmp_pref--));
+    }
+    sorted_ = true;
+    return 0;
+  }
+
+  int getPreference(const char* key, UCHAR* pref) {
+    if (!sorted_) {
+      return R_FAILED;
+    }
+    std::map<std::string, UCHAR>::iterator i = preference_map_.find(key);
+    if (i == preference_map_.end()) {
+      return R_NOT_FOUND;
+    }
+    *pref = i->second;
+    return 0;
+  }
+
+ private:
+  std::set<LocalAddress> local_addrs_;
+  std::map<std::string, UCHAR> preference_map_;
+  bool sorted_;
+};
+
+}  // anonymous namespace
+
+static int add_interface(void* obj, nr_local_addr* iface) {
+  InterfacePrioritizer* ip = static_cast<InterfacePrioritizer*>(obj);
+  return ip->add(iface);
+}
+
+static int get_priority(void* obj, const char* key, UCHAR* pref) {
+  InterfacePrioritizer* ip = static_cast<InterfacePrioritizer*>(obj);
+  return ip->getPreference(key, pref);
+}
+
+static int sort_preference(void* obj) {
+  InterfacePrioritizer* ip = static_cast<InterfacePrioritizer*>(obj);
+  return ip->sort();
+}
+
+static int destroy(void** objp) {
+  if (!objp || !*objp) {
+    return 0;
+  }
+
+  InterfacePrioritizer* ip = static_cast<InterfacePrioritizer*>(*objp);
+  *objp = nullptr;
+  delete ip;
+
+  return 0;
+}
+
+static nr_interface_prioritizer_vtbl priorizer_vtbl = {
+    add_interface, get_priority, sort_preference, destroy};
+
+namespace mozilla {
+
+nr_interface_prioritizer* CreateInterfacePrioritizer() {
+  nr_interface_prioritizer* ip;
+  int r = nr_interface_prioritizer_create_int(new InterfacePrioritizer(),
+                                              &priorizer_vtbl, &ip);
+  if (r != 0) {
+    return nullptr;
+  }
+  return ip;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/transport/nrinterfaceprioritizer.h b/dom/media/webrtc/transport/nrinterfaceprioritizer.h
new file mode 100644
index 0000000000..051b586445
--- /dev/null
+++ b/dom/media/webrtc/transport/nrinterfaceprioritizer.h
@@ -0,0 +1,17 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+#ifndef nrinterfacepriority_h__
+#define nrinterfacepriority_h__
+
+extern "C" {
+#include "nr_interface_prioritizer.h"
+}
+
+namespace mozilla {
+
+nr_interface_prioritizer* CreateInterfacePrioritizer();
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/transport/rlogconnector.cpp b/dom/media/webrtc/transport/rlogconnector.cpp
new file mode 100644
index 0000000000..bdb58aac56
--- /dev/null
+++ b/dom/media/webrtc/transport/rlogconnector.cpp
@@ -0,0 +1,186 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+/* Original author: bcampen@mozilla.com */
+
+#include "rlogconnector.h"
+
+#include <cstdarg>
+#include <deque>
+#include <string>
+#include <utility>  // Pinch hitting for <utility> and std::move
+#include <vector>
+
+#include "logging.h"
+#include "mozilla/Assertions.h"
+#include "mozilla/Mutex.h"
+#include "mozilla/Sprintf.h"
+
+extern "C" {
+#include <csi_platform.h>
+#include "r_log.h"
+#include "registry.h"
+}
+
+/* Matches r_dest_vlog type defined in r_log.h */
+static int ringbuffer_vlog(int facility, int level, const char* format,
+                           va_list ap) {
+  if (mozilla::RLogConnector::GetInstance()->ShouldLog(level)) {
+    // I could be evil and printf right into a std::string, but unless this
+    // shows up in profiling, it is not worth doing.
+    char temp[4096];
+    VsprintfLiteral(temp, format, ap);
+
+    mozilla::RLogConnector::GetInstance()->Log(level, std::string(temp));
+  }
+  return 0;
+}
+
+static mozilla::LogLevel rLogLvlToMozLogLvl(int level) {
+  switch (level) {
+    case LOG_EMERG:
+    case LOG_ALERT:
+    case LOG_CRIT:
+    case LOG_ERR:
+      return mozilla::LogLevel::Error;
+    case LOG_WARNING:
+      return mozilla::LogLevel::Warning;
+    case LOG_NOTICE:
+      return mozilla::LogLevel::Info;
+    case LOG_INFO:
+      return mozilla::LogLevel::Debug;
+    case LOG_DEBUG:
+    default:
+      return mozilla::LogLevel::Verbose;
+  }
+}
+
+MOZ_MTLOG_MODULE("nicer");
+
+namespace mozilla {
+
+RLogConnector* RLogConnector::instance;
+
+RLogConnector::RLogConnector()
+    : log_limit_(4096), mutex_("RLogConnector::mutex_"), disableCount_(0) {}
+
+RLogConnector::~RLogConnector() = default;
+
+void RLogConnector::SetLogLimit(uint32_t new_limit) {
+  OffTheBooksMutexAutoLock lock(mutex_);
+  log_limit_ = new_limit;
+  RemoveOld();
+}
+
+bool RLogConnector::ShouldLog(int level) const {
+  return level <= LOG_INFO ||
+         MOZ_LOG_TEST(getLogModule(), rLogLvlToMozLogLvl(level));
+}
+
+void RLogConnector::Log(int level, std::string&& log) {
+  MOZ_MTLOG(rLogLvlToMozLogLvl(level), log);
+  OffTheBooksMutexAutoLock lock(mutex_);
+  if (disableCount_ == 0) {
+    AddMsg(std::move(log));
+  }
+}
+
+void RLogConnector::AddMsg(std::string&& msg) {
+  log_messages_.push_front(std::move(msg));
+  RemoveOld();
+}
+
+inline void RLogConnector::RemoveOld() {
+  if (log_messages_.size() > log_limit_) {
+    log_messages_.resize(log_limit_);
+  }
+}
+
+RLogConnector* RLogConnector::CreateInstance() {
+  if (!instance) {
+    instance = new RLogConnector;
+    NR_reg_init(NR_REG_MODE_LOCAL);
+    r_log_set_extra_destination(LOG_DEBUG, &ringbuffer_vlog);
+  }
+  return instance;
+}
+
+RLogConnector* RLogConnector::GetInstance() { return instance; }
+
+void RLogConnector::DestroyInstance() {
+  // First param is ignored when passing null
+  r_log_set_extra_destination(LOG_DEBUG, nullptr);
+  delete instance;
+  instance = nullptr;
+}
+
+// As long as at least one PeerConnection exists in a Private Window rlog
+// messages will not be saved in the RLogConnector. This is necessary because
+// the log_messages buffer is shared across all instances of
+// PeerConnectionImpls. There is no way with the current structure of r_log to
+// run separate logs.
+
+void RLogConnector::EnterPrivateMode() {
+  OffTheBooksMutexAutoLock lock(mutex_);
+  ++disableCount_;
+  MOZ_ASSERT(disableCount_ != 0);
+
+  if (disableCount_ == 1) {
+    AddMsg("LOGGING SUSPENDED: a connection is active in a Private Window ***");
+  }
+}
+
+void RLogConnector::ExitPrivateMode() {
+  OffTheBooksMutexAutoLock lock(mutex_);
+  MOZ_ASSERT(disableCount_ != 0);
+
+  if (--disableCount_ == 0) {
+    AddMsg(
+        "LOGGING RESUMED: no connections are active in a Private Window ***");
+  }
+}
+
+void RLogConnector::Clear() {
+  OffTheBooksMutexAutoLock lock(mutex_);
+  log_messages_.clear();
+}
+
+void RLogConnector::Filter(const std::string& substring, uint32_t limit,
+                           std::deque<std::string>* matching_logs) {
+  std::vector<std::string> substrings;
+  substrings.push_back(substring);
+  FilterAny(substrings, limit, matching_logs);
+}
+
+inline bool AnySubstringMatches(const std::vector<std::string>& substrings,
+                                const std::string& string) {
+  for (auto sub = substrings.begin(); sub != substrings.end(); ++sub) {
+    if (string.find(*sub) != std::string::npos) {
+      return true;
+    }
+  }
+  return false;
+}
+
+void RLogConnector::FilterAny(const std::vector<std::string>& substrings,
+                              uint32_t limit,
+                              std::deque<std::string>* matching_logs) {
+  OffTheBooksMutexAutoLock lock(mutex_);
+  if (limit == 0) {
+    // At a max, all of the log messages.
+    limit = log_limit_;
+  }
+
+  for (auto log = log_messages_.begin();
+       log != log_messages_.end() && matching_logs->size() < limit; ++log) {
+    if (AnySubstringMatches(substrings, *log)) {
+      matching_logs->push_front(*log);
+    }
+  }
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/transport/rlogconnector.h b/dom/media/webrtc/transport/rlogconnector.h
new file mode 100644
index 0000000000..8236eb2ab3
--- /dev/null
+++ b/dom/media/webrtc/transport/rlogconnector.h
@@ -0,0 +1,127 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Some of this code is cut-and-pasted from nICEr. Copyright is:
+
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+/* Original author: bcampen@mozilla.com */
+
+/*
+   This file defines an r_dest_vlog that can be used to accumulate log messages
+   for later inspection/filtering. The intent is to use this for interactive
+   debug purposes on an about:webrtc page or similar.
+*/
+
+#ifndef rlogconnector_h__
+#define rlogconnector_h__
+
+#include <stdint.h>
+
+#include <deque>
+#include <string>
+#include <vector>
+
+#include "mozilla/Mutex.h"
+
+#include "m_cpp_utils.h"
+
+namespace mozilla {
+
+class RLogConnector {
+ public:
+  /*
+     NB: These are not threadsafe, nor are they safe to call during static
+     init/deinit.
+  */
+  static RLogConnector* CreateInstance();
+  static RLogConnector* GetInstance();
+  static void DestroyInstance();
+
+  /*
+     Retrieves log statements that match a given substring, subject to a
+     limit. |matching_logs| will be filled in chronological order (front()
+     is oldest, back() is newest). |limit| == 0 will be interpreted as no
+     limit.
+  */
+  void Filter(const std::string& substring, uint32_t limit,
+              std::deque<std::string>* matching_logs);
+
+  void FilterAny(const std::vector<std::string>& substrings, uint32_t limit,
+                 std::deque<std::string>* matching_logs);
+
+  inline void GetAny(uint32_t limit, std::deque<std::string>* matching_logs) {
+    Filter("", limit, matching_logs);
+  }
+
+  void SetLogLimit(uint32_t new_limit);
+  bool ShouldLog(int level) const;
+  void Log(int level, std::string&& log);
+  void Clear();
+
+  // Methods to signal when a PeerConnection exists in a Private Window.
+  void EnterPrivateMode();
+  void ExitPrivateMode();
+
+ private:
+  RLogConnector();
+  ~RLogConnector();
+  void RemoveOld();
+  void AddMsg(std::string&& msg);
+
+  static RLogConnector* instance;
+
+  /*
+   * Might be worthwhile making this a circular buffer, but I think it is
+   * preferable to take up as little space as possible if no logging is
+   * happening/the ringbuffer is not being used.
+   */
+  std::deque<std::string> log_messages_;
+  /* Max size of log buffer (should we use time-depth instead/also?) */
+  uint32_t log_limit_;
+  OffTheBooksMutex mutex_;
+  uint32_t disableCount_;
+
+  DISALLOW_COPY_ASSIGN(RLogConnector);
+};  // class RLogConnector
+
+}  // namespace mozilla
+
+#endif  // rlogconnector_h__
diff --git a/dom/media/webrtc/transport/runnable_utils.h b/dom/media/webrtc/transport/runnable_utils.h
new file mode 100644
index 0000000000..c7248f4f55
--- /dev/null
+++ b/dom/media/webrtc/transport/runnable_utils.h
@@ -0,0 +1,222 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+#ifndef runnable_utils_h__
+#define runnable_utils_h__
+
+#include <utility>
+
+#include "mozilla/RefPtr.h"
+#include "nsThreadUtils.h"
+#include <functional>
+#include <tuple>
+#include <type_traits>
+
+// Abstract base class for all of our templates
+namespace mozilla {
+
+namespace detail {
+
+enum RunnableResult { NoResult, ReturnsResult };
+
+static inline nsresult RunOnThreadInternal(nsIEventTarget* thread,
+                                           nsIRunnable* runnable,
+                                           uint32_t flags) {
+  return thread->Dispatch(runnable, flags);
+}
+
+template <RunnableResult result>
+class runnable_args_base : public Runnable {
+ public:
+  runnable_args_base() : Runnable("media-runnable_args_base") {}
+
+  NS_IMETHOD Run() final {
+    MOZ_ASSERT(!mHasRun, "Can only be run once");
+
+    RunInternal();
+#ifdef DEBUG
+    mHasRun = true;
+#endif
+
+    return NS_OK;
+  }
+
+ protected:
+  virtual void RunInternal() = 0;
+#ifdef DEBUG
+  bool mHasRun = false;
+#endif
+};
+
+}  // namespace detail
+
+template <typename FunType, typename... Args>
+class runnable_args_func : public detail::runnable_args_base<detail::NoResult> {
+ public:
+  // |explicit| to pacify static analysis when there are no |args|.
+  template <typename... Arguments>
+  explicit runnable_args_func(FunType f, Arguments&&... args)
+      : mFunc(f), mArgs(std::forward<Arguments>(args)...) {}
+
+ protected:
+  void RunInternal() override {
+    std::apply(std::move(mFunc), std::move(mArgs));
+  }
+
+ private:
+  FunType mFunc;
+  std::tuple<Args...> mArgs;
+};
+
+template <typename FunType, typename... Args>
+runnable_args_func<FunType, std::decay_t<Args>...>* WrapRunnableNM(
+    FunType f, Args&&... args) {
+  return new runnable_args_func<FunType, std::decay_t<Args>...>(
+      f, std::forward<Args>(args)...);
+}
+
+template <typename Ret, typename FunType, typename... Args>
+class runnable_args_func_ret
+    : public detail::runnable_args_base<detail::ReturnsResult> {
+ public:
+  template <typename... Arguments>
+  runnable_args_func_ret(Ret* ret, FunType f, Arguments&&... args)
+      : mReturn(ret), mFunc(f), mArgs(std::forward<Arguments>(args)...) {}
+
+ protected:
+  void RunInternal() override {
+    *mReturn = std::apply(std::move(mFunc), std::move(mArgs));
+  }
+
+ private:
+  Ret* mReturn;
+  FunType mFunc;
+  std::tuple<Args...> mArgs;
+};
+
+template <typename R, typename FunType, typename... Args>
+runnable_args_func_ret<R, FunType, std::decay_t<Args>...>* WrapRunnableNMRet(
+    R* ret, FunType f, Args&&... args) {
+  return new runnable_args_func_ret<R, FunType, std::decay_t<Args>...>(
+      ret, f, std::forward<Args>(args)...);
+}
+
+template <typename Class, typename M, typename... Args>
+class runnable_args_memfn
+    : public detail::runnable_args_base<detail::NoResult> {
+ public:
+  template <typename... Arguments>
+  runnable_args_memfn(Class&& obj, M method, Arguments&&... args)
+      : mObj(std::forward<Class>(obj)),
+        mMethod(method),
+        mArgs(std::forward<Arguments>(args)...) {}
+
+ protected:
+  void RunInternal() override {
+    std::apply(std::mem_fn(mMethod),
+               std::tuple_cat(std::tie(mObj), std::move(mArgs)));
+  }
+
+ private:
+  // For holders such as RefPtr and UniquePtr make sure concrete copy is held
+  // rather than a potential dangling reference.
+  std::decay_t<Class> mObj;
+  M mMethod;
+  std::tuple<Args...> mArgs;
+};
+
+template <typename Class, typename M, typename... Args>
+runnable_args_memfn<Class, M, std::decay_t<Args>...>* WrapRunnable(
+    Class&& obj, M method, Args&&... args) {
+  return new runnable_args_memfn<Class, M, std::decay_t<Args>...>(
+      std::forward<Class>(obj), method, std::forward<Args>(args)...);
+}
+
+template <typename Ret, typename Class, typename M, typename... Args>
+class runnable_args_memfn_ret
+    : public detail::runnable_args_base<detail::ReturnsResult> {
+ public:
+  template <typename... Arguments>
+  runnable_args_memfn_ret(Ret* ret, Class&& obj, M method, Arguments... args)
+      : mReturn(ret),
+        mObj(std::forward<Class>(obj)),
+        mMethod(method),
+        mArgs(std::forward<Arguments>(args)...) {}
+
+ protected:
+  void RunInternal() override {
+    *mReturn = std::apply(std::mem_fn(mMethod),
+                          std::tuple_cat(std::tie(mObj), std::move(mArgs)));
+  }
+
+ private:
+  Ret* mReturn;
+  // For holders such as RefPtr and UniquePtr make sure concrete copy is held
+  // rather than a potential dangling reference.
+  std::decay_t<Class> mObj;
+  M mMethod;
+  std::tuple<Args...> mArgs;
+};
+
+template <typename R, typename Class, typename M, typename... Args>
+runnable_args_memfn_ret<R, Class, M, std::decay_t<Args>...>* WrapRunnableRet(
+    R* ret, Class&& obj, M method, Args&&... args) {
+  return new runnable_args_memfn_ret<R, Class, M, std::decay_t<Args>...>(
+      ret, std::forward<Class>(obj), method, std::forward<Args>(args)...);
+}
+
+static inline nsresult RUN_ON_THREAD(
+    nsIEventTarget* thread,
+    detail::runnable_args_base<detail::NoResult>* runnable, uint32_t flags) {
+  return detail::RunOnThreadInternal(
+      thread, static_cast<nsIRunnable*>(runnable), flags);
+}
+
+static inline nsresult RUN_ON_THREAD(
+    nsIEventTarget* thread,
+    detail::runnable_args_base<detail::ReturnsResult>* runnable) {
+  return NS_DispatchAndSpinEventLoopUntilComplete(
+      "webrtc RUN_ON_THREAD"_ns, thread,
+      do_AddRef(static_cast<nsIRunnable*>(runnable)));
+}
+
+#ifdef DEBUG
+#  define ASSERT_ON_THREAD(t)           \
+    do {                                \
+      if (t) {                          \
+        bool on;                        \
+        nsresult rv;                    \
+        rv = t->IsOnCurrentThread(&on); \
+        MOZ_ASSERT(NS_SUCCEEDED(rv));   \
+        MOZ_ASSERT(on);                 \
+      }                                 \
+    } while (0)
+#else
+#  define ASSERT_ON_THREAD(t)
+#endif
+
+template <class T>
+class DispatchedRelease : public detail::runnable_args_base<detail::NoResult> {
+ public:
+  explicit DispatchedRelease(already_AddRefed<T>& ref) : ref_(ref) {}
+
+ protected:
+  void RunInternal() override { ref_ = nullptr; }
+
+ private:
+  RefPtr<T> ref_;
+};
+
+template <typename T>
+DispatchedRelease<T>* WrapRelease(already_AddRefed<T>&& ref) {
+  return new DispatchedRelease<T>(ref);
+}
+
+} /* namespace mozilla */
+
+#endif
diff --git a/dom/media/webrtc/transport/sigslot.h b/dom/media/webrtc/transport/sigslot.h
new file mode 100644
index 0000000000..448e8137fe
--- /dev/null
+++ b/dom/media/webrtc/transport/sigslot.h
@@ -0,0 +1,619 @@
+// sigslot.h: Signal/Slot classes
+//
+// Written by Sarah Thompson (sarah@telergy.com) 2002.
+//
+// License: Public domain. You are free to use this code however you like, with
+// the proviso that the author takes on no responsibility or liability for any
+// use.
+//
+// QUICK DOCUMENTATION
+//
+//        (see also the full documentation at http://sigslot.sourceforge.net/)
+//
+//    #define switches
+//      SIGSLOT_PURE_ISO:
+//        Define this to force ISO C++ compliance. This also disables all of
+//        the thread safety support on platforms where it is available.
+//
+//      SIGSLOT_USE_POSIX_THREADS:
+//        Force use of Posix threads when using a C++ compiler other than gcc
+//        on a platform that supports Posix threads. (When using gcc, this is
+//        the default - use SIGSLOT_PURE_ISO to disable this if necessary)
+//
+//      SIGSLOT_DEFAULT_MT_POLICY:
+//        Where thread support is enabled, this defaults to
+//        multi_threaded_global. Otherwise, the default is single_threaded.
+//        #define this yourself to override the default. In pure ISO mode,
+//        anything other than single_threaded will cause a compiler error.
+//
+//    PLATFORM NOTES
+//
+//      Win32:
+//        On Win32, the WEBRTC_WIN symbol must be #defined. Most mainstream
+//        compilers do this by default, but you may need to define it yourself
+//        if your build environment is less standard. This causes the Win32
+//        thread support to be compiled in and used automatically.
+//
+//      Unix/Linux/BSD, etc.:
+//        If you're using gcc, it is assumed that you have Posix threads
+//        available, so they are used automatically. You can override this (as
+//        under Windows) with the SIGSLOT_PURE_ISO switch. If you're using
+//        something other than gcc but still want to use Posix threads, you
+//        need to #define SIGSLOT_USE_POSIX_THREADS.
+//
+//      ISO C++:
+//        If none of the supported platforms are detected, or if
+//        SIGSLOT_PURE_ISO is defined, all multithreading support is turned
+//        off, along with any code that might cause a pure ISO C++ environment
+//        to complain. Before you ask, gcc -ansi -pedantic won't compile this
+//        library, but gcc -ansi is fine. Pedantic mode seems to throw a lot of
+//        errors that aren't really there. If you feel like investigating this,
+//        please contact the author.
+//
+//
+//    THREADING MODES
+//
+//      single_threaded:
+//        Your program is assumed to be single threaded from the point of view
+//        of signal/slot usage (i.e. all objects using signals and slots are
+//        created and destroyed from a single thread). Behaviour if objects are
+//        destroyed concurrently is undefined (i.e. you'll get the occasional
+//        segmentation fault/memory exception).
+//
+//      multi_threaded_global:
+//        Your program is assumed to be multi threaded. Objects using signals
+//        and slots can be safely created and destroyed from any thread, even
+//        when connections exist. In multi_threaded_global mode, this is
+//        achieved by a single global mutex (actually a critical section on
+//        Windows because they are faster). This option uses less OS resources,
+//        but results in more opportunities for contention, possibly resulting
+//        in more context switches than are strictly necessary.
+//
+//      multi_threaded_local:
+//        Behaviour in this mode is essentially the same as
+//        multi_threaded_global, except that each signal, and each object that
+//        inherits has_slots, all have their own mutex/critical section. In
+//        practice, this means that mutex collisions (and hence context
+//        switches) only happen if they are absolutely essential. However, on
+//        some platforms, creating a lot of mutexes can slow down the whole OS,
+//        so use this option with care.
+//
+//    USING THE LIBRARY
+//
+//      See the full documentation at http://sigslot.sourceforge.net/
+//
+// Libjingle specific:
+//
+// This file has been modified such that has_slots and signalx do not have to be
+// using the same threading requirements. E.g. it is possible to connect a
+// has_slots<single_threaded> and signal0<multi_threaded_local> or
+// has_slots<multi_threaded_local> and signal0<single_threaded>.
+// If has_slots is single threaded the user must ensure that it is not trying
+// to connect or disconnect to signalx concurrently or data race may occur.
+// If signalx is single threaded the user must ensure that disconnect, connect
+// or signal is not happening concurrently or data race may occur.
+
+#ifndef RTC_BASE_SIGSLOT_H_
+#define RTC_BASE_SIGSLOT_H_
+
+#include <stdlib.h>
+#include <cstring>
+#include <list>
+#include <set>
+
+// On our copy of sigslot.h, we set single threading as default.
+#define SIGSLOT_DEFAULT_MT_POLICY single_threaded
+
+#if defined(SIGSLOT_PURE_ISO) ||                   \
+    (!defined(WEBRTC_WIN) && !defined(__GNUG__) && \
+     !defined(SIGSLOT_USE_POSIX_THREADS))
+#  define _SIGSLOT_SINGLE_THREADED
+#elif defined(WEBRTC_WIN)
+#  define _SIGSLOT_HAS_WIN32_THREADS
+#  if !defined(WIN32_LEAN_AND_MEAN)
+#    define WIN32_LEAN_AND_MEAN
+#  endif
+#  include "rtc_base/win32.h"
+#elif defined(__GNUG__) || defined(SIGSLOT_USE_POSIX_THREADS)
+#  define _SIGSLOT_HAS_POSIX_THREADS
+#  include <pthread.h>
+#else
+#  define _SIGSLOT_SINGLE_THREADED
+#endif
+
+#ifndef SIGSLOT_DEFAULT_MT_POLICY
+#  ifdef _SIGSLOT_SINGLE_THREADED
+#    define SIGSLOT_DEFAULT_MT_POLICY single_threaded
+#  else
+#    define SIGSLOT_DEFAULT_MT_POLICY multi_threaded_local
+#  endif
+#endif
+
+// TODO: change this namespace to rtc?
+namespace sigslot {
+
+class single_threaded {
+ public:
+  void lock() {}
+  void unlock() {}
+};
+
+#ifdef _SIGSLOT_HAS_WIN32_THREADS
+// The multi threading policies only get compiled in if they are enabled.
+class multi_threaded_global {
+ public:
+  multi_threaded_global() {
+    static bool isinitialised = false;
+
+    if (!isinitialised) {
+      InitializeCriticalSection(get_critsec());
+      isinitialised = true;
+    }
+  }
+
+  void lock() { EnterCriticalSection(get_critsec()); }
+
+  void unlock() { LeaveCriticalSection(get_critsec()); }
+
+ private:
+  CRITICAL_SECTION* get_critsec() {
+    static CRITICAL_SECTION g_critsec;
+    return &g_critsec;
+  }
+};
+
+class multi_threaded_local {
+ public:
+  multi_threaded_local() { InitializeCriticalSection(&m_critsec); }
+
+  multi_threaded_local(const multi_threaded_local&) {
+    InitializeCriticalSection(&m_critsec);
+  }
+
+  ~multi_threaded_local() { DeleteCriticalSection(&m_critsec); }
+
+  void lock() { EnterCriticalSection(&m_critsec); }
+
+  void unlock() { LeaveCriticalSection(&m_critsec); }
+
+ private:
+  CRITICAL_SECTION m_critsec;
+};
+#endif  // _SIGSLOT_HAS_WIN32_THREADS
+
+#ifdef _SIGSLOT_HAS_POSIX_THREADS
+// The multi threading policies only get compiled in if they are enabled.
+class multi_threaded_global {
+ public:
+  void lock() { pthread_mutex_lock(get_mutex()); }
+  void unlock() { pthread_mutex_unlock(get_mutex()); }
+
+ private:
+  static pthread_mutex_t* get_mutex();
+};
+
+class multi_threaded_local {
+ public:
+  multi_threaded_local() { pthread_mutex_init(&m_mutex, nullptr); }
+  multi_threaded_local(const multi_threaded_local&) {
+    pthread_mutex_init(&m_mutex, nullptr);
+  }
+  ~multi_threaded_local() { pthread_mutex_destroy(&m_mutex); }
+  void lock() { pthread_mutex_lock(&m_mutex); }
+  void unlock() { pthread_mutex_unlock(&m_mutex); }
+
+ private:
+  pthread_mutex_t m_mutex;
+};
+#endif  // _SIGSLOT_HAS_POSIX_THREADS
+
+template <class mt_policy>
+class lock_block {
+ public:
+  mt_policy* m_mutex;
+
+  explicit lock_block(mt_policy* mtx) : m_mutex(mtx) { m_mutex->lock(); }
+
+  ~lock_block() { m_mutex->unlock(); }
+};
+
+class _signal_base_interface;
+
+class has_slots_interface {
+ private:
+  typedef void (*signal_connect_t)(has_slots_interface* self,
+                                   _signal_base_interface* sender);
+  typedef void (*signal_disconnect_t)(has_slots_interface* self,
+                                      _signal_base_interface* sender);
+  typedef void (*disconnect_all_t)(has_slots_interface* self);
+
+  const signal_connect_t m_signal_connect;
+  const signal_disconnect_t m_signal_disconnect;
+  const disconnect_all_t m_disconnect_all;
+
+ protected:
+  has_slots_interface(signal_connect_t conn, signal_disconnect_t disc,
+                      disconnect_all_t disc_all)
+      : m_signal_connect(conn),
+        m_signal_disconnect(disc),
+        m_disconnect_all(disc_all) {}
+
+  // Doesn't really need to be virtual, but is for backwards compatibility
+  // (it was virtual in a previous version of sigslot).
+  virtual ~has_slots_interface() = default;
+
+ public:
+  void signal_connect(_signal_base_interface* sender) {
+    m_signal_connect(this, sender);
+  }
+
+  void signal_disconnect(_signal_base_interface* sender) {
+    m_signal_disconnect(this, sender);
+  }
+
+  void disconnect_all() { m_disconnect_all(this); }
+};
+
+class _signal_base_interface {
+ private:
+  typedef void (*slot_disconnect_t)(_signal_base_interface* self,
+                                    has_slots_interface* pslot);
+  typedef void (*slot_duplicate_t)(_signal_base_interface* self,
+                                   const has_slots_interface* poldslot,
+                                   has_slots_interface* pnewslot);
+
+  const slot_disconnect_t m_slot_disconnect;
+  const slot_duplicate_t m_slot_duplicate;
+
+ protected:
+  _signal_base_interface(slot_disconnect_t disc, slot_duplicate_t dupl)
+      : m_slot_disconnect(disc), m_slot_duplicate(dupl) {}
+
+  ~_signal_base_interface() = default;
+
+ public:
+  void slot_disconnect(has_slots_interface* pslot) {
+    m_slot_disconnect(this, pslot);
+  }
+
+  void slot_duplicate(const has_slots_interface* poldslot,
+                      has_slots_interface* pnewslot) {
+    m_slot_duplicate(this, poldslot, pnewslot);
+  }
+};
+
+class _opaque_connection {
+ private:
+  typedef void (*emit_t)(const _opaque_connection*);
+  template <typename FromT, typename ToT>
+  union union_caster {
+    FromT from;
+    ToT to;
+  };
+
+  emit_t pemit;
+  has_slots_interface* pdest;
+  // Pointers to member functions may be up to 16 bytes for virtual classes,
+  // so make sure we have enough space to store it.
+  unsigned char pmethod[16];
+
+ public:
+  template <typename DestT, typename... Args>
+  _opaque_connection(DestT* pd, void (DestT::*pm)(Args...)) : pdest(pd) {
+    typedef void (DestT::*pm_t)(Args...);
+    static_assert(sizeof(pm_t) <= sizeof(pmethod),
+                  "Size of slot function pointer too large.");
+
+    std::memcpy(pmethod, &pm, sizeof(pm_t));
+
+    typedef void (*em_t)(const _opaque_connection* self, Args...);
+    union_caster<em_t, emit_t> caster2;
+    caster2.from = &_opaque_connection::emitter<DestT, Args...>;
+    pemit = caster2.to;
+  }
+
+  has_slots_interface* getdest() const { return pdest; }
+
+  _opaque_connection duplicate(has_slots_interface* newtarget) const {
+    _opaque_connection res = *this;
+    res.pdest = newtarget;
+    return res;
+  }
+
+  // Just calls the stored "emitter" function pointer stored at construction
+  // time.
+  template <typename... Args>
+  void emit(Args... args) const {
+    typedef void (*em_t)(const _opaque_connection*, Args...);
+    union_caster<emit_t, em_t> caster;
+    caster.from = pemit;
+    (caster.to)(this, args...);
+  }
+
+ private:
+  template <typename DestT, typename... Args>
+  static void emitter(const _opaque_connection* self, Args... args) {
+    typedef void (DestT::*pm_t)(Args...);
+    pm_t pm;
+    std::memcpy(&pm, self->pmethod, sizeof(pm_t));
+    (static_cast<DestT*>(self->pdest)->*(pm))(args...);
+  }
+};
+
+template <class mt_policy>
+class _signal_base : public _signal_base_interface, public mt_policy {
+ protected:
+  typedef std::list<_opaque_connection> connections_list;
+
+  _signal_base()
+      : _signal_base_interface(&_signal_base::do_slot_disconnect,
+                               &_signal_base::do_slot_duplicate),
+        m_current_iterator(m_connected_slots.end()) {}
+
+  ~_signal_base() { disconnect_all(); }
+
+ private:
+  _signal_base& operator=(_signal_base const& that);
+
+ public:
+  _signal_base(const _signal_base& o)
+      : _signal_base_interface(&_signal_base::do_slot_disconnect,
+                               &_signal_base::do_slot_duplicate),
+        m_current_iterator(m_connected_slots.end()) {
+    lock_block<mt_policy> lock(this);
+    for (const auto& connection : o.m_connected_slots) {
+      connection.getdest()->signal_connect(this);
+      m_connected_slots.push_back(connection);
+    }
+  }
+
+  bool is_empty() {
+    lock_block<mt_policy> lock(this);
+    return m_connected_slots.empty();
+  }
+
+  void disconnect_all() {
+    lock_block<mt_policy> lock(this);
+
+    while (!m_connected_slots.empty()) {
+      has_slots_interface* pdest = m_connected_slots.front().getdest();
+      m_connected_slots.pop_front();
+      pdest->signal_disconnect(static_cast<_signal_base_interface*>(this));
+    }
+    // If disconnect_all is called while the signal is firing, advance the
+    // current slot iterator to the end to avoid an invalidated iterator from
+    // being dereferenced.
+    m_current_iterator = m_connected_slots.end();
+  }
+
+#if !defined(NDEBUG)
+  bool connected(has_slots_interface* pclass) {
+    lock_block<mt_policy> lock(this);
+    connections_list::const_iterator it = m_connected_slots.begin();
+    connections_list::const_iterator itEnd = m_connected_slots.end();
+    while (it != itEnd) {
+      if (it->getdest() == pclass) return true;
+      ++it;
+    }
+    return false;
+  }
+#endif
+
+  void disconnect(has_slots_interface* pclass) {
+    lock_block<mt_policy> lock(this);
+    connections_list::iterator it = m_connected_slots.begin();
+    connections_list::iterator itEnd = m_connected_slots.end();
+
+    while (it != itEnd) {
+      if (it->getdest() == pclass) {
+        // If we're currently using this iterator because the signal is firing,
+        // advance it to avoid it being invalidated.
+        if (m_current_iterator == it) {
+          m_current_iterator = m_connected_slots.erase(it);
+        } else {
+          m_connected_slots.erase(it);
+        }
+        pclass->signal_disconnect(static_cast<_signal_base_interface*>(this));
+        return;
+      }
+      ++it;
+    }
+  }
+
+ private:
+  static void do_slot_disconnect(_signal_base_interface* p,
+                                 has_slots_interface* pslot) {
+    _signal_base* const self = static_cast<_signal_base*>(p);
+    lock_block<mt_policy> lock(self);
+    connections_list::iterator it = self->m_connected_slots.begin();
+    connections_list::iterator itEnd = self->m_connected_slots.end();
+
+    while (it != itEnd) {
+      connections_list::iterator itNext = it;
+      ++itNext;
+
+      if (it->getdest() == pslot) {
+        // If we're currently using this iterator because the signal is firing,
+        // advance it to avoid it being invalidated.
+        if (self->m_current_iterator == it) {
+          self->m_current_iterator = self->m_connected_slots.erase(it);
+        } else {
+          self->m_connected_slots.erase(it);
+        }
+      }
+
+      it = itNext;
+    }
+  }
+
+  static void do_slot_duplicate(_signal_base_interface* p,
+                                const has_slots_interface* oldtarget,
+                                has_slots_interface* newtarget) {
+    _signal_base* const self = static_cast<_signal_base*>(p);
+    lock_block<mt_policy> lock(self);
+    connections_list::iterator it = self->m_connected_slots.begin();
+    connections_list::iterator itEnd = self->m_connected_slots.end();
+
+    while (it != itEnd) {
+      if (it->getdest() == oldtarget) {
+        self->m_connected_slots.push_back(it->duplicate(newtarget));
+      }
+
+      ++it;
+    }
+  }
+
+ protected:
+  connections_list m_connected_slots;
+
+  // Used to handle a slot being disconnected while a signal is
+  // firing (iterating m_connected_slots).
+  connections_list::iterator m_current_iterator;
+  bool m_erase_current_iterator = false;
+};
+
+template <class mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
+class has_slots : public has_slots_interface, public mt_policy {
+ private:
+  typedef std::set<_signal_base_interface*> sender_set;
+  typedef sender_set::const_iterator const_iterator;
+
+ public:
+  has_slots()
+      : has_slots_interface(&has_slots::do_signal_connect,
+                            &has_slots::do_signal_disconnect,
+                            &has_slots::do_disconnect_all) {}
+
+  has_slots(has_slots const& o)
+      : has_slots_interface(&has_slots::do_signal_connect,
+                            &has_slots::do_signal_disconnect,
+                            &has_slots::do_disconnect_all) {
+    lock_block<mt_policy> lock(this);
+    for (auto* sender : o.m_senders) {
+      sender->slot_duplicate(&o, this);
+      m_senders.insert(sender);
+    }
+  }
+
+  ~has_slots() { this->disconnect_all(); }
+
+ private:
+  has_slots& operator=(has_slots const&);
+
+  static void do_signal_connect(has_slots_interface* p,
+                                _signal_base_interface* sender) {
+    has_slots* const self = static_cast<has_slots*>(p);
+    lock_block<mt_policy> lock(self);
+    self->m_senders.insert(sender);
+  }
+
+  static void do_signal_disconnect(has_slots_interface* p,
+                                   _signal_base_interface* sender) {
+    has_slots* const self = static_cast<has_slots*>(p);
+    lock_block<mt_policy> lock(self);
+    self->m_senders.erase(sender);
+  }
+
+  static void do_disconnect_all(has_slots_interface* p) {
+    has_slots* const self = static_cast<has_slots*>(p);
+    lock_block<mt_policy> lock(self);
+    while (!self->m_senders.empty()) {
+      std::set<_signal_base_interface*> senders;
+      senders.swap(self->m_senders);
+      const_iterator it = senders.begin();
+      const_iterator itEnd = senders.end();
+
+      while (it != itEnd) {
+        _signal_base_interface* s = *it;
+        ++it;
+        s->slot_disconnect(p);
+      }
+    }
+  }
+
+ private:
+  sender_set m_senders;
+};
+
+template <class mt_policy, typename... Args>
+class signal_with_thread_policy : public _signal_base<mt_policy> {
+ private:
+  typedef _signal_base<mt_policy> base;
+
+ protected:
+  typedef typename base::connections_list connections_list;
+
+ public:
+  signal_with_thread_policy() = default;
+
+  template <class desttype>
+  void connect(desttype* pclass, void (desttype::*pmemfun)(Args...)) {
+    lock_block<mt_policy> lock(this);
+    this->m_connected_slots.push_back(_opaque_connection(pclass, pmemfun));
+    pclass->signal_connect(static_cast<_signal_base_interface*>(this));
+  }
+
+  void emit(Args... args) {
+    lock_block<mt_policy> lock(this);
+    this->m_current_iterator = this->m_connected_slots.begin();
+    while (this->m_current_iterator != this->m_connected_slots.end()) {
+      _opaque_connection const& conn = *this->m_current_iterator;
+      ++(this->m_current_iterator);
+      conn.emit<Args...>(args...);
+    }
+  }
+
+  void operator()(Args... args) { emit(args...); }
+};
+
+// Alias with default thread policy. Needed because both default arguments
+// and variadic template arguments must go at the end of the list, so we
+// can't have both at once.
+template <typename... Args>
+using signal = signal_with_thread_policy<SIGSLOT_DEFAULT_MT_POLICY, Args...>;
+
+// The previous verion of sigslot didn't use variadic templates, so you would
+// need to write "sigslot::signal2<Arg1, Arg2>", for example.
+// Now you can just write "sigslot::signal<Arg1, Arg2>", but these aliases
+// exist for backwards compatibility.
+template <typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
+using signal0 = signal_with_thread_policy<mt_policy>;
+
+template <typename A1, typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
+using signal1 = signal_with_thread_policy<mt_policy, A1>;
+
+template <typename A1, typename A2,
+          typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
+using signal2 = signal_with_thread_policy<mt_policy, A1, A2>;
+
+template <typename A1, typename A2, typename A3,
+          typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
+using signal3 = signal_with_thread_policy<mt_policy, A1, A2, A3>;
+
+template <typename A1, typename A2, typename A3, typename A4,
+          typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
+using signal4 = signal_with_thread_policy<mt_policy, A1, A2, A3, A4>;
+
+template <typename A1, typename A2, typename A3, typename A4, typename A5,
+          typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
+using signal5 = signal_with_thread_policy<mt_policy, A1, A2, A3, A4, A5>;
+
+template <typename A1, typename A2, typename A3, typename A4, typename A5,
+          typename A6, typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
+using signal6 = signal_with_thread_policy<mt_policy, A1, A2, A3, A4, A5, A6>;
+
+template <typename A1, typename A2, typename A3, typename A4, typename A5,
+          typename A6, typename A7,
+          typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
+using signal7 =
+    signal_with_thread_policy<mt_policy, A1, A2, A3, A4, A5, A6, A7>;
+
+template <typename A1, typename A2, typename A3, typename A4, typename A5,
+          typename A6, typename A7, typename A8,
+          typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
+using signal8 =
+    signal_with_thread_policy<mt_policy, A1, A2, A3, A4, A5, A6, A7, A8>;
+
+}  // namespace sigslot
+
+#endif  // RTC_BASE_SIGSLOT_H_
diff --git a/dom/media/webrtc/transport/simpletokenbucket.cpp b/dom/media/webrtc/transport/simpletokenbucket.cpp
new file mode 100644
index 0000000000..0509697c7b
--- /dev/null
+++ b/dom/media/webrtc/transport/simpletokenbucket.cpp
@@ -0,0 +1,60 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+/* Original author: bcampen@mozilla.com */
+
+#include "simpletokenbucket.h"
+
+#include <stdint.h>
+
+#include "prinrval.h"
+
+namespace mozilla {
+
+SimpleTokenBucket::SimpleTokenBucket(size_t bucket_size,
+                                     size_t tokens_per_second)
+    : max_tokens_(bucket_size),
+      num_tokens_(bucket_size),
+      tokens_per_second_(tokens_per_second),
+      last_time_tokens_added_(PR_IntervalNow()) {}
+
+size_t SimpleTokenBucket::getTokens(size_t num_requested_tokens) {
+  // Only fill if there isn't enough to satisfy the request.
+  // If we get tokens so seldomly that we are able to roll the timer all
+  // the way around its range, then we lose that entire range of time
+  // for token accumulation. Probably not the end of the world.
+  if (num_requested_tokens > num_tokens_) {
+    PRIntervalTime now = PR_IntervalNow();
+
+    // If we roll over the max, since everything in this calculation is the same
+    // unsigned type, this will still yield the elapsed time (unless we've
+    // wrapped more than once).
+    PRIntervalTime elapsed_ticks = now - last_time_tokens_added_;
+
+    uint32_t elapsed_milli_sec = PR_IntervalToMilliseconds(elapsed_ticks);
+    size_t tokens_to_add = (elapsed_milli_sec * tokens_per_second_) / 1000;
+
+    // Only update our timestamp if we added some tokens
+    // TODO:(bcampen@mozilla.com) Should we attempt to "save" leftover time?
+    if (tokens_to_add) {
+      num_tokens_ += tokens_to_add;
+      if (num_tokens_ > max_tokens_) {
+        num_tokens_ = max_tokens_;
+      }
+
+      last_time_tokens_added_ = now;
+    }
+
+    if (num_requested_tokens > num_tokens_) {
+      return num_tokens_;
+    }
+  }
+
+  num_tokens_ -= num_requested_tokens;
+  return num_requested_tokens;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/transport/simpletokenbucket.h b/dom/media/webrtc/transport/simpletokenbucket.h
new file mode 100644
index 0000000000..7e809535b1
--- /dev/null
+++ b/dom/media/webrtc/transport/simpletokenbucket.h
@@ -0,0 +1,54 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+/* Original author: bcampen@mozilla.com */
+
+/*
+ *  This file defines a dirt-simple token bucket class.
+ */
+
+#ifndef simpletokenbucket_h__
+#define simpletokenbucket_h__
+
+#include <stdint.h>
+
+#include "prinrval.h"
+
+#include "m_cpp_utils.h"
+
+namespace mozilla {
+
+class SimpleTokenBucket {
+ public:
+  /*
+   *  Create a SimpleTokenBucket with a given maximum size and
+   *  token replenishment rate.
+   *  (eg; if you want a maximum rate of 5 per second over a 7 second
+   *  period, call SimpleTokenBucket b(5*7, 5);)
+   */
+  SimpleTokenBucket(size_t bucket_size, size_t tokens_per_second);
+
+  /*
+   *  Attempt to acquire a number of tokens. If successful, returns
+   *  |num_tokens|, otherwise returns the number of tokens currently
+   *  in the bucket.
+   *  Note: To get the number of tokens in the bucket, pass something
+   *  like UINT32_MAX.
+   */
+  size_t getTokens(size_t num_tokens);
+
+ protected:  // Allow testing to touch these.
+  uint64_t max_tokens_;
+  uint64_t num_tokens_;
+  size_t tokens_per_second_;
+  PRIntervalTime last_time_tokens_added_;
+
+  DISALLOW_COPY_ASSIGN(SimpleTokenBucket);
+};
+
+}  // namespace mozilla
+
+#endif  // simpletokenbucket_h__
diff --git a/dom/media/webrtc/transport/srtp/README_MOZILLA b/dom/media/webrtc/transport/srtp/README_MOZILLA
new file mode 100644
index 0000000000..8533b67c53
--- /dev/null
+++ b/dom/media/webrtc/transport/srtp/README_MOZILLA
@@ -0,0 +1,7 @@
+This directory contains build files for libsrtp. The actual library
+source is in $TOPSRCDIR/third_party/libsrtp/
+
+The upstream git repository is https://github.com/cisco/libsrtp
+
+TBD add code and instructions how to do a clean update import without manual
+intervention.
diff --git a/dom/media/webrtc/transport/srtp/moz.build b/dom/media/webrtc/transport/srtp/moz.build
new file mode 100644
index 0000000000..d3d4971a6d
--- /dev/null
+++ b/dom/media/webrtc/transport/srtp/moz.build
@@ -0,0 +1,8 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+if CONFIG["MOZ_SRTP"]:
+    DIRS += ["/third_party/libsrtp/src"]
diff --git a/dom/media/webrtc/transport/stun_socket_filter.cpp b/dom/media/webrtc/transport/stun_socket_filter.cpp
new file mode 100644
index 0000000000..b568f97a40
--- /dev/null
+++ b/dom/media/webrtc/transport/stun_socket_filter.cpp
@@ -0,0 +1,432 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+#include <string>
+#include <set>
+#include <iomanip>
+
+extern "C" {
+#include "nr_api.h"
+#include "transport_addr.h"
+#include "stun.h"
+}
+
+#include "logging.h"
+#include "mozilla/Attributes.h"
+#include "mozilla/net/DNS.h"
+#include "stun_socket_filter.h"
+#include "nr_socket_prsock.h"
+
+namespace {
+
+MOZ_MTLOG_MODULE("mtransport")
+
+class NetAddrCompare {
+ public:
+  bool operator()(const mozilla::net::NetAddr& lhs,
+                  const mozilla::net::NetAddr& rhs) const {
+    if (lhs.raw.family != rhs.raw.family) {
+      return lhs.raw.family < rhs.raw.family;
+    }
+
+    switch (lhs.raw.family) {
+      case AF_INET:
+        if (lhs.inet.port != rhs.inet.port) {
+          return lhs.inet.port < rhs.inet.port;
+        }
+        return lhs.inet.ip < rhs.inet.ip;
+      case AF_INET6:
+        if (lhs.inet6.port != rhs.inet6.port) {
+          return lhs.inet6.port < rhs.inet6.port;
+        }
+        return memcmp(&lhs.inet6.ip, &rhs.inet6.ip, sizeof(lhs.inet6.ip)) < 0;
+      default:
+        MOZ_ASSERT(false);
+    }
+    return false;
+  }
+};
+
+class PendingSTUNRequest {
+ public:
+  PendingSTUNRequest(const mozilla::net::NetAddr& netaddr, const UINT12& id)
+      : id_(id), net_addr_(netaddr), is_id_set_(true) {}
+
+  MOZ_IMPLICIT PendingSTUNRequest(const mozilla::net::NetAddr& netaddr)
+      : id_(), net_addr_(netaddr), is_id_set_(false) {}
+
+  bool operator<(const PendingSTUNRequest& rhs) const {
+    if (NetAddrCompare()(net_addr_, rhs.net_addr_)) {
+      return true;
+    }
+
+    if (NetAddrCompare()(rhs.net_addr_, net_addr_)) {
+      return false;
+    }
+
+    if (!is_id_set_ && !rhs.is_id_set_) {
+      // PendingSTUNRequest can be stored to set only when it has id,
+      // so comparing two PendingSTUNRequst without id is not going
+      // to happen.
+      MOZ_CRASH();
+    }
+
+    if (!(is_id_set_ && rhs.is_id_set_)) {
+      // one of operands doesn't have id, ignore the difference.
+      return false;
+    }
+
+    return memcmp(id_.octet, rhs.id_.octet, sizeof(id_.octet)) < 0;
+  }
+
+ private:
+  const UINT12 id_;
+  const mozilla::net::NetAddr net_addr_;
+  const bool is_id_set_;
+};
+
+static uint16_t GetPortInfallible(const mozilla::net::NetAddr& aAddr) {
+  uint16_t result = 0;
+  (void)aAddr.GetPort(&result);
+  return result;
+}
+
+static std::ostream& operator<<(std::ostream& aStream, UINT12 aId) {
+  for (int octet : aId.octet) {
+    aStream << std::hex << std::setfill('0') << std::setw(2) << octet;
+  }
+  return aStream;
+}
+
+class STUNUDPSocketFilter : public nsISocketFilter {
+ public:
+  STUNUDPSocketFilter() : white_list_(), pending_requests_() {}
+
+  // Allocated/freed and used on the PBackground IPC thread
+  NS_DECL_ISUPPORTS
+  NS_DECL_NSISOCKETFILTER
+
+ private:
+  virtual ~STUNUDPSocketFilter() = default;
+
+  bool filter_incoming_packet(const mozilla::net::NetAddr* remote_addr,
+                              const uint8_t* data, uint32_t len);
+
+  bool filter_outgoing_packet(const mozilla::net::NetAddr* remote_addr,
+                              const uint8_t* data, uint32_t len);
+
+  std::set<mozilla::net::NetAddr, NetAddrCompare> white_list_;
+  std::set<PendingSTUNRequest> pending_requests_;
+  std::set<PendingSTUNRequest> response_allowed_;
+};
+
+NS_IMPL_ISUPPORTS(STUNUDPSocketFilter, nsISocketFilter)
+
+NS_IMETHODIMP
+STUNUDPSocketFilter::FilterPacket(const mozilla::net::NetAddr* remote_addr,
+                                  const uint8_t* data, uint32_t len,
+                                  int32_t direction, bool* result) {
+  switch (direction) {
+    case nsISocketFilter::SF_INCOMING:
+      *result = filter_incoming_packet(remote_addr, data, len);
+      break;
+    case nsISocketFilter::SF_OUTGOING:
+      *result = filter_outgoing_packet(remote_addr, data, len);
+      break;
+    default:
+      MOZ_CRASH("Unknown packet direction");
+  }
+  return NS_OK;
+}
+
+bool STUNUDPSocketFilter::filter_incoming_packet(
+    const mozilla::net::NetAddr* remote_addr, const uint8_t* data,
+    uint32_t len) {
+  // Check white list
+  if (white_list_.find(*remote_addr) != white_list_.end()) {
+    MOZ_MTLOG(ML_DEBUG, __func__ << this << " Address in whitelist: "
+                                 << remote_addr->ToString() << ":"
+                                 << GetPortInfallible(*remote_addr));
+    return true;
+  }
+
+  // If it is a STUN response message and we can match its id with one of the
+  // pending requests, we can add this address into whitelist.
+  if (nr_is_stun_response_message(
+          reinterpret_cast<UCHAR*>(const_cast<uint8_t*>(data)), len)) {
+    const nr_stun_message_header* msg =
+        reinterpret_cast<const nr_stun_message_header*>(data);
+    PendingSTUNRequest pending_req(*remote_addr, msg->id);
+    std::set<PendingSTUNRequest>::iterator it =
+        pending_requests_.find(pending_req);
+    if (it != pending_requests_.end()) {
+      pending_requests_.erase(it);
+      response_allowed_.erase(pending_req);
+      white_list_.insert(*remote_addr);
+      MOZ_MTLOG(ML_DEBUG, __func__ << this
+                                   << " Allowing known STUN response, "
+                                      "remembering address in whitelist: "
+                                   << remote_addr->ToString() << ":"
+                                   << GetPortInfallible(*remote_addr)
+                                   << " id=" << msg->id);
+      return true;
+    }
+  }
+  // If it's an incoming STUN request we let it pass and add it to the list of
+  // pending response for white listing once we answer.
+  if (nr_is_stun_request_message(
+          reinterpret_cast<UCHAR*>(const_cast<uint8_t*>(data)), len)) {
+    const nr_stun_message_header* msg =
+        reinterpret_cast<const nr_stun_message_header*>(data);
+    response_allowed_.insert(PendingSTUNRequest(*remote_addr, msg->id));
+    MOZ_MTLOG(
+        ML_DEBUG,
+        __func__ << this
+                 << " Allowing STUN request, will allow packets in return: "
+                 << remote_addr->ToString() << ":"
+                 << GetPortInfallible(*remote_addr) << " id=" << msg->id);
+    return true;
+  }
+  // Lastly if we have send a STUN request to the destination of this
+  // packet we allow it to send us anything back in case it's for example a
+  // DTLS message (but we don't white list).
+  std::set<PendingSTUNRequest>::iterator it =
+      pending_requests_.find(PendingSTUNRequest(*remote_addr));
+  if (it != pending_requests_.end()) {
+    MOZ_MTLOG(
+        ML_DEBUG,
+        __func__
+            << this
+            << " Allowing packet from source while waiting for a response: "
+            << remote_addr->ToString() << ":"
+            << GetPortInfallible(*remote_addr));
+    return true;
+  }
+
+  MOZ_MTLOG(
+      ML_DEBUG,
+      __func__
+          << " Disallowing packet that is neither a STUN request or response: "
+          << remote_addr->ToString() << ":" << GetPortInfallible(*remote_addr));
+  return false;
+}
+
+bool STUNUDPSocketFilter::filter_outgoing_packet(
+    const mozilla::net::NetAddr* remote_addr, const uint8_t* data,
+    uint32_t len) {
+  // Check white list
+  if (white_list_.find(*remote_addr) != white_list_.end()) {
+    MOZ_MTLOG(ML_DEBUG, __func__ << this << " Address in whitelist: "
+                                 << remote_addr->ToString() << ":"
+                                 << GetPortInfallible(*remote_addr));
+    return true;
+  }
+
+  // Check if it is a stun packet. If yes, we put it into a pending list and
+  // wait for response packet.
+  if (nr_is_stun_request_message(
+          reinterpret_cast<UCHAR*>(const_cast<uint8_t*>(data)), len)) {
+    const nr_stun_message_header* msg =
+        reinterpret_cast<const nr_stun_message_header*>(data);
+    pending_requests_.insert(PendingSTUNRequest(*remote_addr, msg->id));
+    MOZ_MTLOG(
+        ML_DEBUG,
+        __func__ << this
+                 << " Allowing STUN request, will allow packets in return: "
+                 << remote_addr->ToString() << ":"
+                 << GetPortInfallible(*remote_addr) << " id=" << msg->id);
+    return true;
+  }
+
+  // If it is a stun response packet, and we had received the request before, we
+  // can allow it packet to pass filter.
+  if (nr_is_stun_response_message(
+          reinterpret_cast<UCHAR*>(const_cast<uint8_t*>(data)), len)) {
+    const nr_stun_message_header* msg =
+        reinterpret_cast<const nr_stun_message_header*>(data);
+    std::set<PendingSTUNRequest>::iterator it =
+        response_allowed_.find(PendingSTUNRequest(*remote_addr, msg->id));
+    if (it != response_allowed_.end()) {
+      white_list_.insert(*remote_addr);
+      response_allowed_.erase(it);
+      MOZ_MTLOG(ML_DEBUG, __func__ << this
+                                   << " Allowing known STUN response, "
+                                      "remembering address in whitelist: "
+                                   << remote_addr->ToString() << ":"
+                                   << GetPortInfallible(*remote_addr)
+                                   << " id=" << msg->id);
+      return true;
+    }
+
+    MOZ_MTLOG(ML_DEBUG,
+              __func__ << this << " Disallowing unknown STUN response: "
+                       << remote_addr->ToString() << ":"
+                       << GetPortInfallible(*remote_addr) << " id=" << msg->id);
+    return false;
+  }
+
+  MOZ_MTLOG(
+      ML_DEBUG,
+      __func__
+          << " Disallowing packet that is neither a STUN request or response: "
+          << remote_addr->ToString() << ":" << GetPortInfallible(*remote_addr));
+  return false;
+}
+
+class PendingSTUNId {
+ public:
+  explicit PendingSTUNId(const UINT12& id) : id_(id) {}
+
+  bool operator<(const PendingSTUNId& rhs) const {
+    return memcmp(id_.octet, rhs.id_.octet, sizeof(id_.octet)) < 0;
+  }
+
+ private:
+  const UINT12 id_;
+};
+
+class STUNTCPSocketFilter : public nsISocketFilter {
+ public:
+  STUNTCPSocketFilter()
+      : white_listed_(false), pending_request_ids_(), response_allowed_ids_() {}
+
+  // Allocated/freed and used on the PBackground IPC thread
+  NS_DECL_ISUPPORTS
+  NS_DECL_NSISOCKETFILTER
+
+ private:
+  virtual ~STUNTCPSocketFilter() = default;
+
+  bool filter_incoming_packet(const uint8_t* data, uint32_t len);
+
+  bool filter_outgoing_packet(const uint8_t* data, uint32_t len);
+
+  bool white_listed_;
+  std::set<PendingSTUNId> pending_request_ids_;
+  std::set<PendingSTUNId> response_allowed_ids_;
+};
+
+NS_IMPL_ISUPPORTS(STUNTCPSocketFilter, nsISocketFilter)
+
+NS_IMETHODIMP
+STUNTCPSocketFilter::FilterPacket(const mozilla::net::NetAddr* remote_addr,
+                                  const uint8_t* data, uint32_t len,
+                                  int32_t direction, bool* result) {
+  switch (direction) {
+    case nsISocketFilter::SF_INCOMING:
+      *result = filter_incoming_packet(data, len);
+      break;
+    case nsISocketFilter::SF_OUTGOING:
+      *result = filter_outgoing_packet(data, len);
+      break;
+    default:
+      MOZ_CRASH("Unknown packet direction");
+  }
+  return NS_OK;
+}
+
+bool STUNTCPSocketFilter::filter_incoming_packet(const uint8_t* data,
+                                                 uint32_t len) {
+  // check if white listed already
+  if (white_listed_) {
+    return true;
+  }
+
+  UCHAR* stun = const_cast<uint8_t*>(data);
+  uint32_t length = len;
+  if (!nr_is_stun_message(stun, length)) {
+    stun += 2;
+    length -= 2;
+    if (!nr_is_stun_message(stun, length)) {
+      // Note: the UDP filter lets incoming packets pass, because order of
+      // packets is not guaranteed and the next packet is likely an important
+      // packet for DTLS (which is costly in terms of timing to wait for a
+      // retransmit). This does not apply to TCP with its guaranteed order. But
+      // we still let it pass, because otherwise we would have to buffer bytes
+      // here until the minimum STUN request size of bytes has been received.
+      return true;
+    }
+  }
+
+  const nr_stun_message_header* msg =
+      reinterpret_cast<const nr_stun_message_header*>(stun);
+
+  // If it is a STUN response message and we can match its id with one of the
+  // pending requests, we can add this address into whitelist.
+  if (nr_is_stun_response_message(stun, length)) {
+    std::set<PendingSTUNId>::iterator it =
+        pending_request_ids_.find(PendingSTUNId(msg->id));
+    if (it != pending_request_ids_.end()) {
+      pending_request_ids_.erase(it);
+      white_listed_ = true;
+    }
+  } else {
+    // If it is a STUN message, but not a response message, we add it into
+    // response allowed list and allow outgoing filter to send a response back.
+    response_allowed_ids_.insert(PendingSTUNId(msg->id));
+  }
+
+  return true;
+}
+
+bool STUNTCPSocketFilter::filter_outgoing_packet(const uint8_t* data,
+                                                 uint32_t len) {
+  // check if white listed already
+  if (white_listed_) {
+    return true;
+  }
+
+  UCHAR* stun = const_cast<uint8_t*>(data);
+  uint32_t length = len;
+  if (!nr_is_stun_message(stun, length)) {
+    stun += 2;
+    length -= 2;
+    if (!nr_is_stun_message(stun, length)) {
+      return false;
+    }
+  }
+
+  const nr_stun_message_header* msg =
+      reinterpret_cast<const nr_stun_message_header*>(stun);
+
+  // Check if it is a stun request. If yes, we put it into a pending list and
+  // wait for response packet.
+  if (nr_is_stun_request_message(stun, length)) {
+    pending_request_ids_.insert(PendingSTUNId(msg->id));
+    return true;
+  }
+
+  // If it is a stun response packet, and we had received the request before, we
+  // can allow it packet to pass filter.
+  if (nr_is_stun_response_message(stun, length)) {
+    std::set<PendingSTUNId>::iterator it =
+        response_allowed_ids_.find(PendingSTUNId(msg->id));
+    if (it != response_allowed_ids_.end()) {
+      response_allowed_ids_.erase(it);
+      white_listed_ = true;
+      return true;
+    }
+  }
+
+  return false;
+}
+
+}  // anonymous namespace
+
+NS_IMPL_ISUPPORTS(nsStunUDPSocketFilterHandler, nsISocketFilterHandler)
+
+NS_IMETHODIMP nsStunUDPSocketFilterHandler::NewFilter(
+    nsISocketFilter** result) {
+  nsISocketFilter* ret = new STUNUDPSocketFilter();
+  NS_ADDREF(*result = ret);
+  return NS_OK;
+}
+
+NS_IMPL_ISUPPORTS(nsStunTCPSocketFilterHandler, nsISocketFilterHandler)
+
+NS_IMETHODIMP nsStunTCPSocketFilterHandler::NewFilter(
+    nsISocketFilter** result) {
+  nsISocketFilter* ret = new STUNTCPSocketFilter();
+  NS_ADDREF(*result = ret);
+  return NS_OK;
+}
diff --git a/dom/media/webrtc/transport/stun_socket_filter.h b/dom/media/webrtc/transport/stun_socket_filter.h
new file mode 100644
index 0000000000..393257577d
--- /dev/null
+++ b/dom/media/webrtc/transport/stun_socket_filter.h
@@ -0,0 +1,41 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+#ifndef stun_socket_filter_h__
+#define stun_socket_filter_h__
+
+#include "nsISocketFilter.h"
+
+#define NS_STUN_UDP_SOCKET_FILTER_HANDLER_CID \
+  {0x3e43ee93,                                \
+   0x829e,                                    \
+   0x4ea6,                                    \
+   {0xa3, 0x4e, 0x62, 0xd9, 0xe4, 0xc9, 0xf9, 0x93}};
+
+class nsStunUDPSocketFilterHandler : public nsISocketFilterHandler {
+ public:
+  // Threadsafe because we create off-main-thread, but destroy on MainThread
+  // via FreeFactoryEntries()
+  NS_DECL_THREADSAFE_ISUPPORTS
+  NS_DECL_NSISOCKETFILTERHANDLER
+ private:
+  virtual ~nsStunUDPSocketFilterHandler() = default;
+};
+
+#define NS_STUN_TCP_SOCKET_FILTER_HANDLER_CID \
+  {0x9fea635a,                                \
+   0x2fc2,                                    \
+   0x4d08,                                    \
+   {0x97, 0x21, 0xd2, 0x38, 0xd3, 0xf5, 0x2f, 0x92}};
+
+class nsStunTCPSocketFilterHandler : public nsISocketFilterHandler {
+ public:
+  // Threadsafe because we create off-main-thread, but destroy on MainThread
+  // via FreeFactoryEntries()
+  NS_DECL_THREADSAFE_ISUPPORTS
+  NS_DECL_NSISOCKETFILTERHANDLER
+ private:
+  virtual ~nsStunTCPSocketFilterHandler() = default;
+};
+
+#endif  // stun_socket_filter_h__
diff --git a/dom/media/webrtc/transport/test/TestSyncRunnable.cpp b/dom/media/webrtc/transport/test/TestSyncRunnable.cpp
new file mode 100644
index 0000000000..ca671b4e79
--- /dev/null
+++ b/dom/media/webrtc/transport/test/TestSyncRunnable.cpp
@@ -0,0 +1,56 @@
+/* -*- Mode: C++; tab-width: 12; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "nsIThread.h"
+#include "nsThreadUtils.h"
+#include "mozilla/SyncRunnable.h"
+
+#include "gtest/gtest.h"
+
+using namespace mozilla;
+
+nsIThread* gThread = nullptr;
+
+class TestRunnable : public Runnable {
+ public:
+  TestRunnable() : Runnable("TestRunnable"), ran_(false) {}
+
+  NS_IMETHOD Run() override {
+    ran_ = true;
+
+    return NS_OK;
+  }
+
+  bool ran() const { return ran_; }
+
+ private:
+  bool ran_;
+};
+
+class TestSyncRunnable : public ::testing::Test {
+ public:
+  static void SetUpTestCase() {
+    nsresult rv = NS_NewNamedThread("thread", &gThread);
+    ASSERT_TRUE(NS_SUCCEEDED(rv));
+  }
+
+  static void TearDownTestCase() {
+    if (gThread) gThread->Shutdown();
+  }
+};
+
+TEST_F(TestSyncRunnable, TestDispatch) {
+  RefPtr<TestRunnable> r(new TestRunnable());
+  RefPtr<SyncRunnable> s(new SyncRunnable(r));
+  s->DispatchToThread(gThread);
+
+  ASSERT_TRUE(r->ran());
+}
+
+TEST_F(TestSyncRunnable, TestDispatchStatic) {
+  RefPtr<TestRunnable> r(new TestRunnable());
+  SyncRunnable::DispatchToThread(gThread, r);
+  ASSERT_TRUE(r->ran());
+}
diff --git a/dom/media/webrtc/transport/test/buffered_stun_socket_unittest.cpp b/dom/media/webrtc/transport/test/buffered_stun_socket_unittest.cpp
new file mode 100644
index 0000000000..e6a9cd38a2
--- /dev/null
+++ b/dom/media/webrtc/transport/test/buffered_stun_socket_unittest.cpp
@@ -0,0 +1,245 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+extern "C" {
+#include "nr_api.h"
+#include "nr_socket.h"
+#include "nr_socket_buffered_stun.h"
+#include "transport_addr.h"
+}
+
+#include "stun_msg.h"
+
+#include "dummysocket.h"
+
+#include "nr_socket_prsock.h"
+
+#define GTEST_HAS_RTTI 0
+#include "gtest/gtest.h"
+#include "gtest_utils.h"
+
+using namespace mozilla;
+
+static uint8_t kStunMessage[] = {0x00, 0x01, 0x00, 0x08, 0x21, 0x12, 0xa4,
+                                 0x42, 0x9b, 0x90, 0xbe, 0x2c, 0xae, 0x1a,
+                                 0x0c, 0xa8, 0xa0, 0xd6, 0x8b, 0x08, 0x80,
+                                 0x28, 0x00, 0x04, 0xdb, 0x35, 0x5f, 0xaa};
+static size_t kStunMessageLen = sizeof(kStunMessage);
+
+class BufferedStunSocketTest : public MtransportTest {
+ public:
+  BufferedStunSocketTest()
+      : MtransportTest(), dummy_(nullptr), test_socket_(nullptr) {}
+
+  ~BufferedStunSocketTest() { nr_socket_destroy(&test_socket_); }
+
+  void SetUp() override {
+    MtransportTest::SetUp();
+
+    RefPtr<DummySocket> dummy(new DummySocket());
+
+    int r =
+        nr_socket_buffered_stun_create(dummy->get_nr_socket(), kStunMessageLen,
+                                       TURN_TCP_FRAMING, &test_socket_);
+    ASSERT_EQ(0, r);
+    dummy_ = std::move(dummy);  // Now owned by test_socket_.
+
+    r = nr_str_port_to_transport_addr((char*)"192.0.2.133", 3333, IPPROTO_TCP,
+                                      &remote_addr_);
+    ASSERT_EQ(0, r);
+
+    r = nr_socket_connect(test_socket_, &remote_addr_);
+    ASSERT_EQ(0, r);
+  }
+
+  nr_socket* socket() { return test_socket_; }
+
+ protected:
+  RefPtr<DummySocket> dummy_;
+  nr_socket* test_socket_;
+  nr_transport_addr remote_addr_;
+};
+
+TEST_F(BufferedStunSocketTest, TestCreate) {}
+
+TEST_F(BufferedStunSocketTest, TestSendTo) {
+  int r = nr_socket_sendto(test_socket_, kStunMessage, kStunMessageLen, 0,
+                           &remote_addr_);
+  ASSERT_EQ(0, r);
+
+  dummy_->CheckWriteBuffer(kStunMessage, kStunMessageLen);
+}
+
+TEST_F(BufferedStunSocketTest, TestSendToBuffered) {
+  dummy_->SetWritable(0);
+
+  int r = nr_socket_sendto(test_socket_, kStunMessage, kStunMessageLen, 0,
+                           &remote_addr_);
+  ASSERT_EQ(0, r);
+
+  dummy_->CheckWriteBuffer(nullptr, 0);
+
+  dummy_->SetWritable(kStunMessageLen);
+  dummy_->FireWritableCb();
+  dummy_->CheckWriteBuffer(kStunMessage, kStunMessageLen);
+}
+
+TEST_F(BufferedStunSocketTest, TestSendFullThenDrain) {
+  dummy_->SetWritable(0);
+
+  for (;;) {
+    int r = nr_socket_sendto(test_socket_, kStunMessage, kStunMessageLen, 0,
+                             &remote_addr_);
+    if (r == R_WOULDBLOCK) break;
+
+    ASSERT_EQ(0, r);
+  }
+
+  // Nothing was written.
+  dummy_->CheckWriteBuffer(nullptr, 0);
+
+  // Now flush.
+  dummy_->SetWritable(kStunMessageLen);
+  dummy_->FireWritableCb();
+  dummy_->ClearWriteBuffer();
+
+  // Verify we can write something.
+  int r = nr_socket_sendto(test_socket_, kStunMessage, kStunMessageLen, 0,
+                           &remote_addr_);
+  ASSERT_EQ(0, r);
+
+  // And that it appears.
+  dummy_->CheckWriteBuffer(kStunMessage, kStunMessageLen);
+}
+
+TEST_F(BufferedStunSocketTest, TestSendToPartialBuffered) {
+  dummy_->SetWritable(10);
+
+  int r = nr_socket_sendto(test_socket_, kStunMessage, kStunMessageLen, 0,
+                           &remote_addr_);
+  ASSERT_EQ(0, r);
+
+  dummy_->CheckWriteBuffer(kStunMessage, 10);
+  dummy_->ClearWriteBuffer();
+
+  dummy_->SetWritable(kStunMessageLen);
+  dummy_->FireWritableCb();
+  dummy_->CheckWriteBuffer(kStunMessage + 10, kStunMessageLen - 10);
+}
+
+TEST_F(BufferedStunSocketTest, TestSendToReject) {
+  dummy_->SetWritable(0);
+
+  int r = nr_socket_sendto(test_socket_, kStunMessage, kStunMessageLen, 0,
+                           &remote_addr_);
+  ASSERT_EQ(0, r);
+
+  dummy_->CheckWriteBuffer(nullptr, 0);
+
+  r = nr_socket_sendto(test_socket_, kStunMessage, kStunMessageLen, 0,
+                       &remote_addr_);
+  ASSERT_EQ(R_WOULDBLOCK, r);
+
+  dummy_->CheckWriteBuffer(nullptr, 0);
+}
+
+TEST_F(BufferedStunSocketTest, TestSendToWrongAddr) {
+  nr_transport_addr addr;
+
+  int r = nr_str_port_to_transport_addr((char*)"192.0.2.134", 3333, IPPROTO_TCP,
+                                        &addr);
+  ASSERT_EQ(0, r);
+
+  r = nr_socket_sendto(test_socket_, kStunMessage, kStunMessageLen, 0, &addr);
+  ASSERT_EQ(R_BAD_DATA, r);
+}
+
+TEST_F(BufferedStunSocketTest, TestReceiveRecvFrom) {
+  dummy_->SetReadBuffer(kStunMessage, kStunMessageLen);
+
+  unsigned char tmp[2048];
+  size_t len;
+  nr_transport_addr addr;
+
+  int r = nr_socket_recvfrom(test_socket_, tmp, sizeof(tmp), &len, 0, &addr);
+  ASSERT_EQ(0, r);
+  ASSERT_EQ(kStunMessageLen, len);
+  ASSERT_EQ(0, memcmp(kStunMessage, tmp, kStunMessageLen));
+  ASSERT_EQ(0, nr_transport_addr_cmp(&addr, &remote_addr_,
+                                     NR_TRANSPORT_ADDR_CMP_MODE_ALL));
+}
+
+TEST_F(BufferedStunSocketTest, TestReceiveRecvFromPartial) {
+  dummy_->SetReadBuffer(kStunMessage, 15);
+
+  unsigned char tmp[2048];
+  size_t len;
+  nr_transport_addr addr;
+
+  int r = nr_socket_recvfrom(test_socket_, tmp, sizeof(tmp), &len, 0, &addr);
+  ASSERT_EQ(R_WOULDBLOCK, r);
+
+  dummy_->SetReadBuffer(kStunMessage + 15, kStunMessageLen - 15);
+
+  r = nr_socket_recvfrom(test_socket_, tmp, sizeof(tmp), &len, 0, &addr);
+  ASSERT_EQ(0, r);
+  ASSERT_EQ(kStunMessageLen, len);
+  ASSERT_EQ(0, memcmp(kStunMessage, tmp, kStunMessageLen));
+  ASSERT_EQ(0, nr_transport_addr_cmp(&addr, &remote_addr_,
+                                     NR_TRANSPORT_ADDR_CMP_MODE_ALL));
+
+  r = nr_socket_recvfrom(test_socket_, tmp, sizeof(tmp), &len, 0, &addr);
+  ASSERT_EQ(R_WOULDBLOCK, r);
+}
+
+TEST_F(BufferedStunSocketTest, TestReceiveRecvFromGarbage) {
+  uint8_t garbage[50];
+  memset(garbage, 0xff, sizeof(garbage));
+
+  dummy_->SetReadBuffer(garbage, sizeof(garbage));
+
+  unsigned char tmp[2048];
+  size_t len;
+  nr_transport_addr addr;
+  int r = nr_socket_recvfrom(test_socket_, tmp, sizeof(tmp), &len, 0, &addr);
+  ASSERT_EQ(R_BAD_DATA, r);
+
+  r = nr_socket_recvfrom(test_socket_, tmp, sizeof(tmp), &len, 0, &addr);
+  ASSERT_EQ(R_FAILED, r);
+}
+
+TEST_F(BufferedStunSocketTest, TestReceiveRecvFromTooShort) {
+  dummy_->SetReadBuffer(kStunMessage, kStunMessageLen);
+
+  unsigned char tmp[2048];
+  size_t len;
+  nr_transport_addr addr;
+
+  int r = nr_socket_recvfrom(test_socket_, tmp, kStunMessageLen - 1, &len, 0,
+                             &addr);
+  ASSERT_EQ(R_BAD_ARGS, r);
+}
+
+TEST_F(BufferedStunSocketTest, TestReceiveRecvFromReallyLong) {
+  uint8_t garbage[4096];
+  memset(garbage, 0xff, sizeof(garbage));
+  memcpy(garbage, kStunMessage, kStunMessageLen);
+  nr_stun_message_header* hdr =
+      reinterpret_cast<nr_stun_message_header*>(garbage);
+  hdr->length = htons(3000);
+
+  dummy_->SetReadBuffer(garbage, sizeof(garbage));
+
+  unsigned char tmp[4096];
+  size_t len;
+  nr_transport_addr addr;
+
+  int r = nr_socket_recvfrom(test_socket_, tmp, kStunMessageLen - 1, &len, 0,
+                             &addr);
+  ASSERT_EQ(R_BAD_DATA, r);
+}
diff --git a/dom/media/webrtc/transport/test/dummysocket.h b/dom/media/webrtc/transport/test/dummysocket.h
new file mode 100644
index 0000000000..6e20a1f7e7
--- /dev/null
+++ b/dom/media/webrtc/transport/test/dummysocket.h
@@ -0,0 +1,217 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original authors: ekr@rtfm.com; ryan@tokbox.com
+
+#ifndef MTRANSPORT_DUMMY_SOCKET_H_
+#define MTRANSPORT_DUMMY_SOCKET_H_
+
+#include "nr_socket_prsock.h"
+
+extern "C" {
+#include "transport_addr.h"
+}
+
+#include "mediapacket.h"
+#include "mozilla/UniquePtr.h"
+
+#define GTEST_HAS_RTTI 0
+#include "gtest/gtest.h"
+#include "gtest_utils.h"
+
+namespace mozilla {
+
+static UniquePtr<MediaPacket> merge(UniquePtr<MediaPacket> a,
+                                    UniquePtr<MediaPacket> b) {
+  if (a && a->len() && b && b->len()) {
+    UniquePtr<uint8_t[]> data(new uint8_t[a->len() + b->len()]);
+    memcpy(data.get(), a->data(), a->len());
+    memcpy(data.get() + a->len(), b->data(), b->len());
+
+    UniquePtr<MediaPacket> merged(new MediaPacket);
+    merged->Take(std::move(data), a->len() + b->len());
+    return merged;
+  }
+
+  if (a && a->len()) {
+    return a;
+  }
+
+  if (b && b->len()) {
+    return b;
+  }
+
+  return nullptr;
+}
+
+class DummySocket : public NrSocketBase {
+ public:
+  DummySocket()
+      : writable_(UINT_MAX),
+        write_buffer_(nullptr),
+        readable_(UINT_MAX),
+        read_buffer_(nullptr),
+        cb_(nullptr),
+        cb_arg_(nullptr),
+        self_(nullptr) {}
+
+  // the nr_socket APIs
+  virtual int create(nr_transport_addr* addr) override { return 0; }
+
+  virtual int sendto(const void* msg, size_t len, int flags,
+                     const nr_transport_addr* to) override {
+    MOZ_CRASH();
+    return 0;
+  }
+
+  virtual int recvfrom(void* buf, size_t maxlen, size_t* len, int flags,
+                       nr_transport_addr* from) override {
+    MOZ_CRASH();
+    return 0;
+  }
+
+  virtual int getaddr(nr_transport_addr* addrp) override {
+    MOZ_CRASH();
+    return 0;
+  }
+
+  virtual void close() override {}
+
+  virtual int connect(const nr_transport_addr* addr) override {
+    nr_transport_addr_copy(&connect_addr_, addr);
+    return 0;
+  }
+
+  virtual int listen(int backlog) override { return 0; }
+
+  virtual int accept(nr_transport_addr* addrp, nr_socket** sockp) override {
+    return 0;
+  }
+
+  virtual int write(const void* msg, size_t len, size_t* written) override {
+    size_t to_write = std::min(len, writable_);
+
+    if (to_write) {
+      UniquePtr<MediaPacket> msgbuf(new MediaPacket);
+      msgbuf->Copy(static_cast<const uint8_t*>(msg), to_write);
+      write_buffer_ = merge(std::move(write_buffer_), std::move(msgbuf));
+    }
+
+    *written = to_write;
+
+    return 0;
+  }
+
+  virtual int read(void* buf, size_t maxlen, size_t* len) override {
+    if (!read_buffer_.get()) {
+      return R_WOULDBLOCK;
+    }
+
+    size_t to_read = std::min(read_buffer_->len(), std::min(maxlen, readable_));
+
+    memcpy(buf, read_buffer_->data(), to_read);
+    *len = to_read;
+
+    if (to_read < read_buffer_->len()) {
+      MediaPacket* newPacket = new MediaPacket;
+      newPacket->Copy(read_buffer_->data() + to_read,
+                      read_buffer_->len() - to_read);
+      read_buffer_.reset(newPacket);
+    } else {
+      read_buffer_.reset();
+    }
+
+    return 0;
+  }
+
+  // Implementations of the async_event APIs.
+  // These are no-ops because we handle scheduling manually
+  // for test purposes.
+  virtual int async_wait(int how, NR_async_cb cb, void* cb_arg, char* function,
+                         int line) override {
+    EXPECT_EQ(nullptr, cb_);
+    cb_ = cb;
+    cb_arg_ = cb_arg;
+
+    return 0;
+  }
+
+  virtual int cancel(int how) override {
+    cb_ = nullptr;
+    cb_arg_ = nullptr;
+
+    return 0;
+  }
+
+  // Read/Manipulate the current state.
+  void CheckWriteBuffer(const uint8_t* data, size_t len) {
+    if (!len) {
+      EXPECT_EQ(nullptr, write_buffer_.get());
+    } else {
+      EXPECT_NE(nullptr, write_buffer_.get());
+      ASSERT_EQ(len, write_buffer_->len());
+      ASSERT_EQ(0, memcmp(data, write_buffer_->data(), len));
+    }
+  }
+
+  void ClearWriteBuffer() { write_buffer_.reset(); }
+
+  void SetWritable(size_t val) { writable_ = val; }
+
+  void FireWritableCb() {
+    NR_async_cb cb = cb_;
+    void* cb_arg = cb_arg_;
+
+    cb_ = nullptr;
+    cb_arg_ = nullptr;
+
+    cb(this, NR_ASYNC_WAIT_WRITE, cb_arg);
+  }
+
+  void SetReadBuffer(const uint8_t* data, size_t len) {
+    EXPECT_EQ(nullptr, write_buffer_.get());
+    read_buffer_.reset(new MediaPacket);
+    read_buffer_->Copy(data, len);
+  }
+
+  void ClearReadBuffer() { read_buffer_.reset(); }
+
+  void SetReadable(size_t val) { readable_ = val; }
+
+  nr_socket* get_nr_socket() {
+    if (!self_) {
+      int r = nr_socket_create_int(this, vtbl(), &self_);
+      AddRef();
+      if (r) return nullptr;
+    }
+
+    return self_;
+  }
+
+  nr_transport_addr* get_connect_addr() { return &connect_addr_; }
+
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(DummySocket, override);
+
+ private:
+  ~DummySocket() = default;
+
+  DISALLOW_COPY_ASSIGN(DummySocket);
+
+  size_t writable_;  // Amount we allow someone to write.
+  UniquePtr<MediaPacket> write_buffer_;
+  size_t readable_;  // Amount we allow someone to read.
+  UniquePtr<MediaPacket> read_buffer_;
+
+  NR_async_cb cb_;
+  void* cb_arg_;
+  nr_socket* self_;
+
+  nr_transport_addr connect_addr_;
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webrtc/transport/test/gtest_ringbuffer_dumper.h b/dom/media/webrtc/transport/test/gtest_ringbuffer_dumper.h
new file mode 100644
index 0000000000..25e85c2155
--- /dev/null
+++ b/dom/media/webrtc/transport/test/gtest_ringbuffer_dumper.h
@@ -0,0 +1,78 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: bcampen@mozilla.com
+
+#ifndef gtest_ringbuffer_dumper_h__
+#define gtest_ringbuffer_dumper_h__
+
+#include "mozilla/SyncRunnable.h"
+
+#define GTEST_HAS_RTTI 0
+#include "gtest/gtest.h"
+
+#include "mtransport_test_utils.h"
+#include "runnable_utils.h"
+#include "rlogconnector.h"
+
+using mozilla::RLogConnector;
+using mozilla::WrapRunnable;
+
+namespace test {
+class RingbufferDumper : public ::testing::EmptyTestEventListener {
+ public:
+  explicit RingbufferDumper(MtransportTestUtils* test_utils)
+      : test_utils_(test_utils) {}
+
+  void ClearRingBuffer_s() {
+    RLogConnector::CreateInstance();
+    // Set limit to zero to clear the ringbuffer
+    RLogConnector::GetInstance()->SetLogLimit(0);
+    RLogConnector::GetInstance()->SetLogLimit(UINT32_MAX);
+  }
+
+  void DestroyRingBuffer_s() { RLogConnector::DestroyInstance(); }
+
+  void DumpRingBuffer_s() {
+    std::deque<std::string> logs;
+    // Get an unlimited number of log lines, with no filter
+    RLogConnector::GetInstance()->GetAny(0, &logs);
+    for (auto l = logs.begin(); l != logs.end(); ++l) {
+      std::cout << *l << std::endl;
+    }
+    ClearRingBuffer_s();
+  }
+
+  virtual void OnTestStart(const ::testing::TestInfo& testInfo) override {
+    mozilla::SyncRunnable::DispatchToThread(
+        test_utils_->sts_target(),
+        WrapRunnable(this, &RingbufferDumper::ClearRingBuffer_s));
+  }
+
+  virtual void OnTestEnd(const ::testing::TestInfo& testInfo) override {
+    mozilla::SyncRunnable::DispatchToThread(
+        test_utils_->sts_target(),
+        WrapRunnable(this, &RingbufferDumper::DestroyRingBuffer_s));
+  }
+
+  // Called after a failed assertion or a SUCCEED() invocation.
+  virtual void OnTestPartResult(
+      const ::testing::TestPartResult& testResult) override {
+    if (testResult.failed()) {
+      // Dump (and empty) the RLogConnector
+      mozilla::SyncRunnable::DispatchToThread(
+          test_utils_->sts_target(),
+          WrapRunnable(this, &RingbufferDumper::DumpRingBuffer_s));
+    }
+  }
+
+ private:
+  MtransportTestUtils* test_utils_;
+};
+
+}  // namespace test
+
+#endif  // gtest_ringbuffer_dumper_h__
diff --git a/dom/media/webrtc/transport/test/gtest_utils.h b/dom/media/webrtc/transport/test/gtest_utils.h
new file mode 100644
index 0000000000..40c2570ea1
--- /dev/null
+++ b/dom/media/webrtc/transport/test/gtest_utils.h
@@ -0,0 +1,201 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Utilities to wrap gtest, based on libjingle's gunit
+
+// Some sections of this code are under the following license:
+
+/*
+ * libjingle
+ * Copyright 2004--2008, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+// Original author: ekr@rtfm.com
+#ifndef gtest_utils__h__
+#define gtest_utils__h__
+
+#include <iostream>
+
+#include "nspr.h"
+#include "prinrval.h"
+#include "prthread.h"
+
+#define GTEST_HAS_RTTI 0
+#include "gtest/gtest.h"
+
+#include "gtest_ringbuffer_dumper.h"
+#include "mtransport_test_utils.h"
+#include "nss.h"
+#include "ssl.h"
+
+extern "C" {
+#include "registry.h"
+#include "transport_addr.h"
+}
+
+// Wait up to timeout seconds for expression to be true
+#define WAIT(expression, timeout)                                   \
+  do {                                                              \
+    for (PRIntervalTime start = PR_IntervalNow();                   \
+         !(expression) && !((PR_IntervalNow() - start) >            \
+                            PR_MillisecondsToInterval(timeout));) { \
+      PR_Sleep(10);                                                 \
+    }                                                               \
+  } while (0)
+
+// Same as GTEST_WAIT, but stores the result in res. Used when
+// you also want the result of expression but wish to avoid
+// double evaluation.
+#define WAIT_(expression, timeout, res)                                     \
+  do {                                                                      \
+    for (PRIntervalTime start = PR_IntervalNow();                           \
+         !(res = (expression)) && !((PR_IntervalNow() - start) >            \
+                                    PR_MillisecondsToInterval(timeout));) { \
+      PR_Sleep(10);                                                         \
+    }                                                                       \
+  } while (0)
+
+#define ASSERT_TRUE_WAIT(expression, timeout) \
+  do {                                        \
+    bool res;                                 \
+    WAIT_(expression, timeout, res);          \
+    ASSERT_TRUE(res);                         \
+  } while (0)
+
+#define EXPECT_TRUE_WAIT(expression, timeout) \
+  do {                                        \
+    bool res;                                 \
+    WAIT_(expression, timeout, res);          \
+    EXPECT_TRUE(res);                         \
+  } while (0)
+
+#define ASSERT_EQ_WAIT(expected, actual, timeout) \
+  do {                                            \
+    WAIT(expected == actual, timeout);            \
+    ASSERT_EQ(expected, actual);                  \
+  } while (0)
+
+using test::RingbufferDumper;
+
+class MtransportTest : public ::testing::Test {
+ public:
+  MtransportTest() : test_utils_(nullptr), dumper_(nullptr) {}
+
+  void SetUp() override {
+    test_utils_ = new MtransportTestUtils();
+    NSS_NoDB_Init(nullptr);
+    NSS_SetDomesticPolicy();
+
+    NR_reg_init(NR_REG_MODE_LOCAL);
+
+    // Attempt to load env vars used by tests.
+    GetEnvironment("TURN_SERVER_ADDRESS", turn_server_);
+    GetEnvironment("TURN_SERVER_USER", turn_user_);
+    GetEnvironment("TURN_SERVER_PASSWORD", turn_password_);
+    GetEnvironment("STUN_SERVER_ADDRESS", stun_server_address_);
+    GetEnvironment("STUN_SERVER_HOSTNAME", stun_server_hostname_);
+
+    std::string disable_non_local;
+    GetEnvironment("MOZ_DISABLE_NONLOCAL_CONNECTIONS", disable_non_local);
+    std::string upload_dir;
+    GetEnvironment("MOZ_UPLOAD_DIR", upload_dir);
+
+    if ((!disable_non_local.empty() && disable_non_local != "0") ||
+        !upload_dir.empty()) {
+      // We're assuming that MOZ_UPLOAD_DIR is only set on tbpl;
+      // MOZ_DISABLE_NONLOCAL_CONNECTIONS probably should be set when running
+      // the cpp unit-tests, but is not presently.
+      stun_server_address_ = "";
+      stun_server_hostname_ = "";
+      turn_server_ = "";
+    }
+
+    // Some tests are flaky and need to check if they're supposed to run.
+    webrtc_enabled_ = CheckEnvironmentFlag("MOZ_WEBRTC_TESTS");
+
+    ::testing::TestEventListeners& listeners =
+        ::testing::UnitTest::GetInstance()->listeners();
+
+    dumper_ = new RingbufferDumper(test_utils_);
+    listeners.Append(dumper_);
+  }
+
+  void TearDown() override {
+    ::testing::UnitTest::GetInstance()->listeners().Release(dumper_);
+    delete dumper_;
+    delete test_utils_;
+  }
+
+  void GetEnvironment(const char* aVar, std::string& out) {
+    char* value = getenv(aVar);
+    if (value) {
+      out = value;
+    }
+  }
+
+  bool CheckEnvironmentFlag(const char* aVar) {
+    std::string value;
+    GetEnvironment(aVar, value);
+    return value == "1";
+  }
+
+  bool WarnIfTurnNotConfigured() const {
+    bool configured =
+        !turn_server_.empty() && !turn_user_.empty() && !turn_password_.empty();
+
+    if (configured) {
+      nr_transport_addr addr;
+      if (nr_str_port_to_transport_addr(turn_server_.c_str(), 3478, IPPROTO_UDP,
+                                        &addr)) {
+        printf(
+            "Invalid TURN_SERVER_ADDRESS \"%s\". Only IP numbers supported.\n",
+            turn_server_.c_str());
+        configured = false;
+      }
+    } else {
+      printf(
+          "Set TURN_SERVER_ADDRESS, TURN_SERVER_USER, and "
+          "TURN_SERVER_PASSWORD\n"
+          "environment variables to run this test\n");
+    }
+
+    return !configured;
+  }
+
+  MtransportTestUtils* test_utils_;
+  RingbufferDumper* dumper_;
+
+  std::string turn_server_;
+  std::string turn_user_;
+  std::string turn_password_;
+  std::string stun_server_address_;
+  std::string stun_server_hostname_;
+
+  bool webrtc_enabled_;
+};
+#endif
diff --git a/dom/media/webrtc/transport/test/ice_unittest.cpp b/dom/media/webrtc/transport/test/ice_unittest.cpp
new file mode 100644
index 0000000000..d87fa0b0da
--- /dev/null
+++ b/dom/media/webrtc/transport/test/ice_unittest.cpp
@@ -0,0 +1,4400 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+#include <algorithm>
+#include <deque>
+#include <iostream>
+#include <limits>
+#include <map>
+#include <string>
+#include <vector>
+
+#include "sigslot.h"
+
+#include "logging.h"
+#include "ssl.h"
+
+#include "mozilla/Preferences.h"
+#include "nsThreadUtils.h"
+#include "nsXPCOM.h"
+
+extern "C" {
+#include "r_types.h"
+#include "async_wait.h"
+#include "async_timer.h"
+#include "r_data.h"
+#include "util.h"
+#include "r_time.h"
+}
+
+#include "ice_ctx.h"
+#include "ice_peer_ctx.h"
+#include "ice_media_stream.h"
+
+#include "nricectx.h"
+#include "nricemediastream.h"
+#include "nriceresolverfake.h"
+#include "nriceresolver.h"
+#include "nrinterfaceprioritizer.h"
+#include "gtest_ringbuffer_dumper.h"
+#include "rlogconnector.h"
+#include "runnable_utils.h"
+#include "stunserver.h"
+#include "nr_socket_prsock.h"
+#include "test_nr_socket.h"
+#include "nsISocketFilter.h"
+#include "mozilla/net/DNS.h"
+
+#define GTEST_HAS_RTTI 0
+#include "gtest/gtest.h"
+#include "gtest_utils.h"
+
+using namespace mozilla;
+
+static unsigned int kDefaultTimeout = 7000;
+
+// TODO(nils@mozilla.com): This should get replaced with some non-external
+// solution like discussed in bug 860775.
+const std::string kDefaultStunServerHostname((char*)"stun.l.google.com");
+const std::string kBogusStunServerHostname(
+    (char*)"stun-server-nonexistent.invalid");
+const uint16_t kDefaultStunServerPort = 19305;
+const std::string kBogusIceCandidate(
+    (char*)"candidate:0 2 UDP 2113601790 192.168.178.20 50769 typ");
+
+const std::string kUnreachableHostIceCandidate(
+    (char*)"candidate:0 1 UDP 2113601790 192.168.178.20 50769 typ host");
+
+namespace {
+
+// DNS resolution helper code
+static std::string Resolve(const std::string& fqdn, int address_family) {
+  struct addrinfo hints;
+  memset(&hints, 0, sizeof(hints));
+  hints.ai_family = address_family;
+  hints.ai_protocol = IPPROTO_UDP;
+  struct addrinfo* res;
+  int err = getaddrinfo(fqdn.c_str(), nullptr, &hints, &res);
+  if (err) {
+    std::cerr << "Error in getaddrinfo: " << err << std::endl;
+    return "";
+  }
+
+  char str_addr[64] = {0};
+  switch (res->ai_family) {
+    case AF_INET:
+      inet_ntop(AF_INET,
+                &reinterpret_cast<struct sockaddr_in*>(res->ai_addr)->sin_addr,
+                str_addr, sizeof(str_addr));
+      break;
+    case AF_INET6:
+      inet_ntop(
+          AF_INET6,
+          &reinterpret_cast<struct sockaddr_in6*>(res->ai_addr)->sin6_addr,
+          str_addr, sizeof(str_addr));
+      break;
+    default:
+      std::cerr << "Got unexpected address family in DNS lookup: "
+                << res->ai_family << std::endl;
+      freeaddrinfo(res);
+      return "";
+  }
+
+  if (!strlen(str_addr)) {
+    std::cerr << "inet_ntop failed" << std::endl;
+  }
+
+  freeaddrinfo(res);
+  return str_addr;
+}
+
+class StunTest : public MtransportTest {
+ public:
+  StunTest() : MtransportTest() {}
+
+  void SetUp() override {
+    MtransportTest::SetUp();
+
+    stun_server_hostname_ = kDefaultStunServerHostname;
+    // If only a STUN server FQDN was provided, look up its IP address for the
+    // address-only tests.
+    if (stun_server_address_.empty() && !stun_server_hostname_.empty()) {
+      stun_server_address_ = Resolve(stun_server_hostname_, AF_INET);
+      ASSERT_TRUE(!stun_server_address_.empty());
+    }
+
+    // Make sure NrIceCtx is in a testable state.
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnableNM(&NrIceCtx::internal_DeinitializeGlobal));
+
+    // NB: NrIceCtx::internal_DeinitializeGlobal destroys the RLogConnector
+    // singleton.
+    RLogConnector::CreateInstance();
+
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnableNM(&TestStunServer::GetInstance, AF_INET));
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnableNM(&TestStunServer::GetInstance, AF_INET6));
+
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnableNM(&TestStunTcpServer::GetInstance, AF_INET));
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnableNM(&TestStunTcpServer::GetInstance, AF_INET6));
+  }
+
+  void TearDown() override {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnableNM(&NrIceCtx::internal_DeinitializeGlobal));
+
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnableNM(&TestStunServer::ShutdownInstance));
+
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnableNM(&TestStunTcpServer::ShutdownInstance));
+
+    RLogConnector::DestroyInstance();
+
+    MtransportTest::TearDown();
+  }
+};
+
+enum TrickleMode { TRICKLE_NONE, TRICKLE_SIMULATE, TRICKLE_REAL };
+
+enum ConsentStatus { CONSENT_FRESH, CONSENT_STALE, CONSENT_EXPIRED };
+
+typedef std::string (*CandidateFilter)(const std::string& candidate);
+
+std::vector<std::string> split(const std::string& s, char delim) {
+  std::vector<std::string> elems;
+  std::stringstream ss(s);
+  std::string item;
+  while (std::getline(ss, item, delim)) {
+    elems.push_back(item);
+  }
+  return elems;
+}
+
+static std::string IsSrflxCandidate(const std::string& candidate) {
+  std::vector<std::string> tokens = split(candidate, ' ');
+  if ((tokens.at(6) == "typ") && (tokens.at(7) == "srflx")) {
+    return candidate;
+  }
+  return std::string();
+}
+
+static std::string IsRelayCandidate(const std::string& candidate) {
+  if (candidate.find("typ relay") != std::string::npos) {
+    return candidate;
+  }
+  return std::string();
+}
+
+static std::string IsTcpCandidate(const std::string& candidate) {
+  if (candidate.find("TCP") != std::string::npos) {
+    return candidate;
+  }
+  return std::string();
+}
+
+static std::string IsTcpSoCandidate(const std::string& candidate) {
+  if (candidate.find("tcptype so") != std::string::npos) {
+    return candidate;
+  }
+  return std::string();
+}
+
+static std::string IsLoopbackCandidate(const std::string& candidate) {
+  if (candidate.find("127.0.0.") != std::string::npos) {
+    return candidate;
+  }
+  return std::string();
+}
+
+static std::string IsIpv4Candidate(const std::string& candidate) {
+  std::vector<std::string> tokens = split(candidate, ' ');
+  if (tokens.at(4).find(':') == std::string::npos) {
+    return candidate;
+  }
+  return std::string();
+}
+
+static std::string SabotageHostCandidateAndDropReflexive(
+    const std::string& candidate) {
+  if (candidate.find("typ srflx") != std::string::npos) {
+    return std::string();
+  }
+
+  if (candidate.find("typ host") != std::string::npos) {
+    return kUnreachableHostIceCandidate;
+  }
+
+  return candidate;
+}
+
+bool ContainsSucceededPair(const std::vector<NrIceCandidatePair>& pairs) {
+  for (const auto& pair : pairs) {
+    if (pair.state == NrIceCandidatePair::STATE_SUCCEEDED) {
+      return true;
+    }
+  }
+  return false;
+}
+
+// Note: Does not correspond to any notion of prioritization; this is just
+// so we can use stl containers/algorithms that need a comparator
+bool operator<(const NrIceCandidate& lhs, const NrIceCandidate& rhs) {
+  if (lhs.cand_addr.host == rhs.cand_addr.host) {
+    if (lhs.cand_addr.port == rhs.cand_addr.port) {
+      if (lhs.cand_addr.transport == rhs.cand_addr.transport) {
+        if (lhs.type == rhs.type) {
+          return lhs.tcp_type < rhs.tcp_type;
+        }
+        return lhs.type < rhs.type;
+      }
+      return lhs.cand_addr.transport < rhs.cand_addr.transport;
+    }
+    return lhs.cand_addr.port < rhs.cand_addr.port;
+  }
+  return lhs.cand_addr.host < rhs.cand_addr.host;
+}
+
+bool operator==(const NrIceCandidate& lhs, const NrIceCandidate& rhs) {
+  return !((lhs < rhs) || (rhs < lhs));
+}
+
+class IceCandidatePairCompare {
+ public:
+  bool operator()(const NrIceCandidatePair& lhs,
+                  const NrIceCandidatePair& rhs) const {
+    if (lhs.priority == rhs.priority) {
+      if (lhs.local == rhs.local) {
+        if (lhs.remote == rhs.remote) {
+          return lhs.codeword < rhs.codeword;
+        }
+        return lhs.remote < rhs.remote;
+      }
+      return lhs.local < rhs.local;
+    }
+    return lhs.priority < rhs.priority;
+  }
+};
+
+class IceTestPeer;
+
+class SchedulableTrickleCandidate {
+ public:
+  SchedulableTrickleCandidate(IceTestPeer* peer, size_t stream,
+                              const std::string& candidate,
+                              const std::string& ufrag,
+                              MtransportTestUtils* utils)
+      : peer_(peer),
+        stream_(stream),
+        candidate_(candidate),
+        ufrag_(ufrag),
+        timer_handle_(nullptr),
+        test_utils_(utils) {}
+
+  ~SchedulableTrickleCandidate() {
+    if (timer_handle_) NR_async_timer_cancel(timer_handle_);
+  }
+
+  void Schedule(unsigned int ms) {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &SchedulableTrickleCandidate::Schedule_s, ms));
+  }
+
+  void Schedule_s(unsigned int ms) {
+    MOZ_ASSERT(!timer_handle_);
+    NR_ASYNC_TIMER_SET(ms, Trickle_cb, this, &timer_handle_);
+  }
+
+  static void Trickle_cb(NR_SOCKET s, int how, void* cb_arg) {
+    static_cast<SchedulableTrickleCandidate*>(cb_arg)->Trickle();
+  }
+
+  void Trickle();
+
+  std::string& Candidate() { return candidate_; }
+
+  const std::string& Candidate() const { return candidate_; }
+
+  bool IsHost() const {
+    return candidate_.find("typ host") != std::string::npos;
+  }
+
+  bool IsReflexive() const {
+    return candidate_.find("typ srflx") != std::string::npos;
+  }
+
+  bool IsRelay() const {
+    return candidate_.find("typ relay") != std::string::npos;
+  }
+
+ private:
+  IceTestPeer* peer_;
+  size_t stream_;
+  std::string candidate_;
+  std::string ufrag_;
+  void* timer_handle_;
+  MtransportTestUtils* test_utils_;
+
+  DISALLOW_COPY_ASSIGN(SchedulableTrickleCandidate);
+};
+
+class IceTestPeer : public sigslot::has_slots<> {
+ public:
+  IceTestPeer(const std::string& name, MtransportTestUtils* utils, bool offerer,
+              const NrIceCtx::Config& config)
+      : name_(name),
+        ice_ctx_(NrIceCtx::Create(name)),
+        offerer_(offerer),
+        candidates_(),
+        stream_counter_(0),
+        shutting_down_(false),
+        gathering_complete_(false),
+        ready_ct_(0),
+        ice_connected_(false),
+        ice_failed_(false),
+        ice_reached_checking_(false),
+        received_(0),
+        sent_(0),
+        fake_resolver_(),
+        dns_resolver_(new NrIceResolver()),
+        remote_(nullptr),
+        candidate_filter_(nullptr),
+        expected_local_type_(NrIceCandidate::ICE_HOST),
+        expected_local_transport_(kNrIceTransportUdp),
+        expected_remote_type_(NrIceCandidate::ICE_HOST),
+        trickle_mode_(TRICKLE_NONE),
+        simulate_ice_lite_(false),
+        nat_(new TestNat),
+        test_utils_(utils) {
+    ice_ctx_->SignalGatheringStateChange.connect(
+        this, &IceTestPeer::GatheringStateChange);
+    ice_ctx_->SignalConnectionStateChange.connect(
+        this, &IceTestPeer::ConnectionStateChange);
+
+    ice_ctx_->SetIceConfig(config);
+
+    consent_timestamp_.tv_sec = 0;
+    consent_timestamp_.tv_usec = 0;
+    int r = ice_ctx_->SetNat(nat_);
+    (void)r;
+    MOZ_ASSERT(!r);
+  }
+
+  ~IceTestPeer() {
+    test_utils_->SyncDispatchToSTS(WrapRunnable(this, &IceTestPeer::Shutdown));
+
+    // Give the ICE destruction callback time to fire before
+    // we destroy the resolver.
+    PR_Sleep(1000);
+  }
+
+  std::string MakeTransportId(size_t index) const {
+    char id[100];
+    snprintf(id, sizeof(id), "%s:stream%d", name_.c_str(), (int)index);
+    return id;
+  }
+
+  void SetIceCredentials_s(NrIceMediaStream& stream) {
+    static size_t counter = 0;
+    std::ostringstream prefix;
+    prefix << name_ << "-" << counter++;
+    std::string ufrag = prefix.str() + "-ufrag";
+    std::string pwd = prefix.str() + "-pwd";
+    if (mIceCredentials.count(stream.GetId())) {
+      mOldIceCredentials[stream.GetId()] = mIceCredentials[stream.GetId()];
+    }
+    mIceCredentials[stream.GetId()] = std::make_pair(ufrag, pwd);
+    stream.SetIceCredentials(ufrag, pwd);
+  }
+
+  void AddStream_s(int components) {
+    std::string id = MakeTransportId(stream_counter_++);
+
+    RefPtr<NrIceMediaStream> stream =
+        ice_ctx_->CreateStream(id, id, components);
+
+    ASSERT_TRUE(stream);
+    SetIceCredentials_s(*stream);
+
+    stream->SignalCandidate.connect(this, &IceTestPeer::CandidateInitialized);
+    stream->SignalReady.connect(this, &IceTestPeer::StreamReady);
+    stream->SignalFailed.connect(this, &IceTestPeer::StreamFailed);
+    stream->SignalPacketReceived.connect(this, &IceTestPeer::PacketReceived);
+  }
+
+  void AddStream(int components) {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &IceTestPeer::AddStream_s, components));
+  }
+
+  void RemoveStream_s(size_t index) {
+    ice_ctx_->DestroyStream(MakeTransportId(index));
+  }
+
+  void RemoveStream(size_t index) {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &IceTestPeer::RemoveStream_s, index));
+  }
+
+  RefPtr<NrIceMediaStream> GetStream_s(size_t index) {
+    std::string id = MakeTransportId(index);
+    return ice_ctx_->GetStream(id);
+  }
+
+  void SetStunServer(const std::string addr, uint16_t port,
+                     const char* transport = kNrIceTransportUdp) {
+    if (addr.empty()) {
+      // Happens when MOZ_DISABLE_NONLOCAL_CONNECTIONS is set
+      return;
+    }
+
+    std::vector<NrIceStunServer> stun_servers;
+    UniquePtr<NrIceStunServer> server(
+        NrIceStunServer::Create(addr, port, transport));
+    stun_servers.push_back(*server);
+    SetStunServers(stun_servers);
+  }
+
+  void SetStunServers(const std::vector<NrIceStunServer>& servers) {
+    ASSERT_TRUE(NS_SUCCEEDED(ice_ctx_->SetStunServers(servers)));
+  }
+
+  void UseTestStunServer() {
+    SetStunServer(TestStunServer::GetInstance(AF_INET)->addr(),
+                  TestStunServer::GetInstance(AF_INET)->port());
+  }
+
+  void SetTurnServer(const std::string addr, uint16_t port,
+                     const std::string username, const std::string password,
+                     const char* transport) {
+    std::vector<unsigned char> password_vec(password.begin(), password.end());
+    SetTurnServer(addr, port, username, password_vec, transport);
+  }
+
+  void SetTurnServer(const std::string addr, uint16_t port,
+                     const std::string username,
+                     const std::vector<unsigned char> password,
+                     const char* transport) {
+    std::vector<NrIceTurnServer> turn_servers;
+    UniquePtr<NrIceTurnServer> server(
+        NrIceTurnServer::Create(addr, port, username, password, transport));
+    turn_servers.push_back(*server);
+    ASSERT_TRUE(NS_SUCCEEDED(ice_ctx_->SetTurnServers(turn_servers)));
+  }
+
+  void SetTurnServers(const std::vector<NrIceTurnServer> servers) {
+    ASSERT_TRUE(NS_SUCCEEDED(ice_ctx_->SetTurnServers(servers)));
+  }
+
+  void SetFakeResolver(const std::string& ip, const std::string& fqdn) {
+    ASSERT_TRUE(NS_SUCCEEDED(dns_resolver_->Init()));
+    if (!ip.empty() && !fqdn.empty()) {
+      PRNetAddr addr;
+      PRStatus status = PR_StringToNetAddr(ip.c_str(), &addr);
+      addr.inet.port = kDefaultStunServerPort;
+      ASSERT_EQ(PR_SUCCESS, status);
+      fake_resolver_.SetAddr(fqdn, addr);
+    }
+    ASSERT_TRUE(
+        NS_SUCCEEDED(ice_ctx_->SetResolver(fake_resolver_.AllocateResolver())));
+  }
+
+  void SetDNSResolver() {
+    ASSERT_TRUE(NS_SUCCEEDED(dns_resolver_->Init()));
+    ASSERT_TRUE(
+        NS_SUCCEEDED(ice_ctx_->SetResolver(dns_resolver_->AllocateResolver())));
+  }
+
+  void Gather(bool default_route_only = false,
+              bool obfuscate_host_addresses = false) {
+    nsresult res;
+
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnableRet(&res, ice_ctx_, &NrIceCtx::StartGathering,
+                        default_route_only, obfuscate_host_addresses));
+
+    ASSERT_TRUE(NS_SUCCEEDED(res));
+  }
+
+  void SetCtxFlags(bool default_route_only) {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(ice_ctx_, &NrIceCtx::SetCtxFlags, default_route_only));
+  }
+
+  nsTArray<NrIceStunAddr> GetStunAddrs() { return ice_ctx_->GetStunAddrs(); }
+
+  void SetStunAddrs(const nsTArray<NrIceStunAddr>& addrs) {
+    ice_ctx_->SetStunAddrs(addrs);
+  }
+
+  void UseNat() { nat_->enabled_ = true; }
+
+  void SetTimerDivider(int div) { ice_ctx_->internal_SetTimerAccelarator(div); }
+
+  void SetStunResponseDelay(uint32_t delay) {
+    nat_->delay_stun_resp_ms_ = delay;
+  }
+
+  void SetFilteringType(TestNat::NatBehavior type) {
+    MOZ_ASSERT(!nat_->has_port_mappings());
+    nat_->filtering_type_ = type;
+  }
+
+  void SetMappingType(TestNat::NatBehavior type) {
+    MOZ_ASSERT(!nat_->has_port_mappings());
+    nat_->mapping_type_ = type;
+  }
+
+  void SetBlockUdp(bool block) {
+    MOZ_ASSERT(!nat_->has_port_mappings());
+    nat_->block_udp_ = block;
+  }
+
+  void SetBlockStun(bool block) { nat_->block_stun_ = block; }
+
+  // Get various pieces of state
+  std::vector<std::string> GetGlobalAttributes() {
+    std::vector<std::string> attrs(ice_ctx_->GetGlobalAttributes());
+    if (simulate_ice_lite_) {
+      attrs.push_back("ice-lite");
+    }
+    return attrs;
+  }
+
+  std::vector<std::string> GetAttributes(size_t stream) {
+    std::vector<std::string> v;
+
+    RUN_ON_THREAD(
+        test_utils_->sts_target(),
+        WrapRunnableRet(&v, this, &IceTestPeer::GetAttributes_s, stream));
+
+    return v;
+  }
+
+  std::string FilterCandidate(const std::string& candidate) {
+    if (candidate_filter_) {
+      return candidate_filter_(candidate);
+    }
+    return candidate;
+  }
+
+  std::vector<std::string> GetAttributes_s(size_t index) {
+    std::vector<std::string> attributes;
+
+    auto stream = GetStream_s(index);
+    if (!stream) {
+      EXPECT_TRUE(false) << "No such stream " << index;
+      return attributes;
+    }
+
+    std::vector<std::string> attributes_in = stream->GetAttributes();
+
+    for (const auto& attribute : attributes_in) {
+      if (attribute.find("candidate:") != std::string::npos) {
+        std::string candidate(FilterCandidate(attribute));
+        if (!candidate.empty()) {
+          std::cerr << name_ << " Returning candidate: " << candidate
+                    << std::endl;
+          attributes.push_back(candidate);
+        }
+      } else {
+        attributes.push_back(attribute);
+      }
+    }
+
+    return attributes;
+  }
+
+  void SetExpectedTypes(NrIceCandidate::Type local, NrIceCandidate::Type remote,
+                        std::string local_transport = kNrIceTransportUdp) {
+    expected_local_type_ = local;
+    expected_local_transport_ = local_transport;
+    expected_remote_type_ = remote;
+  }
+
+  void SetExpectedRemoteCandidateAddr(const std::string& addr) {
+    expected_remote_addr_ = addr;
+  }
+
+  int GetCandidatesPrivateIpv4Range(size_t stream) {
+    std::vector<std::string> attributes = GetAttributes(stream);
+
+    int host_net = 0;
+    for (const auto& a : attributes) {
+      if (a.find("typ host") != std::string::npos) {
+        nr_transport_addr addr;
+        std::vector<std::string> tokens = split(a, ' ');
+        int r = nr_str_port_to_transport_addr(tokens.at(4).c_str(), 0,
+                                              IPPROTO_UDP, &addr);
+        MOZ_ASSERT(!r);
+        if (!r && (addr.ip_version == NR_IPV4)) {
+          int n = nr_transport_addr_get_private_addr_range(&addr);
+          if (n) {
+            if (host_net) {
+              // TODO: add support for multiple private interfaces
+              std::cerr
+                  << "This test doesn't support multiple private interfaces";
+              return -1;
+            }
+            host_net = n;
+          }
+        }
+      }
+    }
+    return host_net;
+  }
+
+  bool gathering_complete() { return gathering_complete_; }
+  int ready_ct() { return ready_ct_; }
+  bool is_ready_s(size_t index) {
+    auto media_stream = GetStream_s(index);
+    if (!media_stream) {
+      EXPECT_TRUE(false) << "No such stream " << index;
+      return false;
+    }
+    return media_stream->state() == NrIceMediaStream::ICE_OPEN;
+  }
+  bool is_ready(size_t stream) {
+    bool result;
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnableRet(&result, this, &IceTestPeer::is_ready_s, stream));
+    return result;
+  }
+  bool ice_connected() { return ice_connected_; }
+  bool ice_failed() { return ice_failed_; }
+  bool ice_reached_checking() { return ice_reached_checking_; }
+  size_t received() { return received_; }
+  size_t sent() { return sent_; }
+
+  void RestartIce() {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &IceTestPeer::RestartIce_s));
+  }
+
+  void RestartIce_s() {
+    for (auto& stream : ice_ctx_->GetStreams()) {
+      SetIceCredentials_s(*stream);
+    }
+    // take care of some local bookkeeping
+    ready_ct_ = 0;
+    gathering_complete_ = false;
+    ice_connected_ = false;
+    ice_failed_ = false;
+    ice_reached_checking_ = false;
+    remote_ = nullptr;
+  }
+
+  void RollbackIceRestart() {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &IceTestPeer::RollbackIceRestart_s));
+  }
+
+  void RollbackIceRestart_s() {
+    for (auto& stream : ice_ctx_->GetStreams()) {
+      mIceCredentials[stream->GetId()] = mOldIceCredentials[stream->GetId()];
+    }
+  }
+
+  // Start connecting to another peer
+  void Connect_s(IceTestPeer* remote, TrickleMode trickle_mode,
+                 bool start = true) {
+    nsresult res;
+
+    remote_ = remote;
+
+    trickle_mode_ = trickle_mode;
+    ice_connected_ = false;
+    ice_failed_ = false;
+    ice_reached_checking_ = false;
+    res = ice_ctx_->ParseGlobalAttributes(remote->GetGlobalAttributes());
+    ASSERT_FALSE(remote->simulate_ice_lite_ &&
+                 (ice_ctx_->GetControlling() == NrIceCtx::ICE_CONTROLLED));
+    ASSERT_TRUE(NS_SUCCEEDED(res));
+
+    for (size_t i = 0; i < stream_counter_; ++i) {
+      auto aStream = GetStream_s(i);
+      if (aStream) {
+        std::vector<std::string> attributes = remote->GetAttributes(i);
+
+        for (auto it = attributes.begin(); it != attributes.end();) {
+          if (trickle_mode == TRICKLE_SIMULATE &&
+              it->find("candidate:") != std::string::npos) {
+            std::cerr << name_ << " Deferring remote candidate: " << *it
+                      << std::endl;
+            attributes.erase(it);
+          } else {
+            std::cerr << name_ << " Adding remote attribute: " + *it
+                      << std::endl;
+            ++it;
+          }
+        }
+        auto credentials = mIceCredentials[aStream->GetId()];
+        res = aStream->ConnectToPeer(credentials.first, credentials.second,
+                                     attributes);
+        ASSERT_TRUE(NS_SUCCEEDED(res));
+      }
+    }
+
+    if (start) {
+      ice_ctx_->SetControlling(offerer_ ? NrIceCtx::ICE_CONTROLLING
+                                        : NrIceCtx::ICE_CONTROLLED);
+      // Now start checks
+      res = ice_ctx_->StartChecks();
+      ASSERT_TRUE(NS_SUCCEEDED(res));
+    }
+  }
+
+  void Connect(IceTestPeer* remote, TrickleMode trickle_mode,
+               bool start = true) {
+    test_utils_->SyncDispatchToSTS(WrapRunnable(this, &IceTestPeer::Connect_s,
+                                                remote, trickle_mode, start));
+  }
+
+  void SimulateTrickle(size_t stream) {
+    std::cerr << name_ << " Doing trickle for stream " << stream << std::endl;
+    // If we are in trickle deferred mode, now trickle in the candidates
+    // for |stream|
+
+    std::vector<SchedulableTrickleCandidate*>& candidates =
+        ControlTrickle(stream);
+
+    for (auto& candidate : candidates) {
+      candidate->Schedule(0);
+    }
+  }
+
+  // Allows test case to completely control when/if candidates are trickled
+  // (test could also do things like insert extra trickle candidates, or
+  // change existing ones, or insert duplicates, really anything is fair game)
+  std::vector<SchedulableTrickleCandidate*>& ControlTrickle(size_t stream) {
+    std::cerr << "Doing controlled trickle for stream " << stream << std::endl;
+
+    std::vector<std::string> attributes = remote_->GetAttributes(stream);
+
+    for (const auto& attribute : attributes) {
+      if (attribute.find("candidate:") != std::string::npos) {
+        controlled_trickle_candidates_[stream].push_back(
+            new SchedulableTrickleCandidate(this, stream, attribute, "",
+                                            test_utils_));
+      }
+    }
+
+    return controlled_trickle_candidates_[stream];
+  }
+
+  nsresult TrickleCandidate_s(const std::string& candidate,
+                              const std::string& ufrag, size_t index) {
+    auto stream = GetStream_s(index);
+    if (!stream) {
+      // stream might have gone away before the trickle timer popped
+      return NS_OK;
+    }
+    return stream->ParseTrickleCandidate(candidate, ufrag, "");
+  }
+
+  void DumpCandidate(std::string which, const NrIceCandidate& cand) {
+    std::string type;
+    std::string tcp_type;
+
+    std::string addr;
+    int port;
+
+    if (which.find("Remote") != std::string::npos) {
+      addr = cand.cand_addr.host;
+      port = cand.cand_addr.port;
+    } else {
+      addr = cand.local_addr.host;
+      port = cand.local_addr.port;
+    }
+    switch (cand.type) {
+      case NrIceCandidate::ICE_HOST:
+        type = "host";
+        break;
+      case NrIceCandidate::ICE_SERVER_REFLEXIVE:
+        type = "srflx";
+        break;
+      case NrIceCandidate::ICE_PEER_REFLEXIVE:
+        type = "prflx";
+        break;
+      case NrIceCandidate::ICE_RELAYED:
+        type = "relay";
+        if (which.find("Local") != std::string::npos) {
+          type += "(" + cand.local_addr.transport + ")";
+        }
+        break;
+      default:
+        FAIL();
+    };
+
+    switch (cand.tcp_type) {
+      case NrIceCandidate::ICE_NONE:
+        break;
+      case NrIceCandidate::ICE_ACTIVE:
+        tcp_type = " tcptype=active";
+        break;
+      case NrIceCandidate::ICE_PASSIVE:
+        tcp_type = " tcptype=passive";
+        break;
+      case NrIceCandidate::ICE_SO:
+        tcp_type = " tcptype=so";
+        break;
+      default:
+        FAIL();
+    };
+
+    std::cerr << which << " --> " << type << " " << addr << ":" << port << "/"
+              << cand.cand_addr.transport << tcp_type
+              << " codeword=" << cand.codeword << std::endl;
+  }
+
+  void DumpAndCheckActiveCandidates_s() {
+    std::cerr << name_ << " Active candidates:" << std::endl;
+    for (const auto& stream : ice_ctx_->GetStreams()) {
+      for (size_t j = 0; j < stream->components(); ++j) {
+        std::cerr << name_ << " Stream " << stream->GetId() << " component "
+                  << j + 1 << std::endl;
+
+        UniquePtr<NrIceCandidate> local;
+        UniquePtr<NrIceCandidate> remote;
+
+        nsresult res = stream->GetActivePair(j + 1, &local, &remote);
+        if (res == NS_ERROR_NOT_AVAILABLE) {
+          std::cerr << "Component unpaired or disabled." << std::endl;
+        } else {
+          ASSERT_TRUE(NS_SUCCEEDED(res));
+          DumpCandidate("Local  ", *local);
+          /* Depending on timing, and the whims of the network
+           * stack/configuration we're running on top of, prflx is always a
+           * possibility. */
+          if (expected_local_type_ == NrIceCandidate::ICE_HOST) {
+            ASSERT_NE(NrIceCandidate::ICE_SERVER_REFLEXIVE, local->type);
+            ASSERT_NE(NrIceCandidate::ICE_RELAYED, local->type);
+          } else {
+            ASSERT_EQ(expected_local_type_, local->type);
+          }
+          ASSERT_EQ(expected_local_transport_, local->local_addr.transport);
+          DumpCandidate("Remote ", *remote);
+          /* Depending on timing, and the whims of the network
+           * stack/configuration we're running on top of, prflx is always a
+           * possibility. */
+          if (expected_remote_type_ == NrIceCandidate::ICE_HOST) {
+            ASSERT_NE(NrIceCandidate::ICE_SERVER_REFLEXIVE, remote->type);
+            ASSERT_NE(NrIceCandidate::ICE_RELAYED, remote->type);
+          } else {
+            ASSERT_EQ(expected_remote_type_, remote->type);
+          }
+          if (!expected_remote_addr_.empty()) {
+            ASSERT_EQ(expected_remote_addr_, remote->cand_addr.host);
+          }
+        }
+      }
+    }
+  }
+
+  void DumpAndCheckActiveCandidates() {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &IceTestPeer::DumpAndCheckActiveCandidates_s));
+  }
+
+  void Close() {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(ice_ctx_, &NrIceCtx::destroy_peer_ctx));
+  }
+
+  void Shutdown() {
+    std::cerr << name_ << " Shutdown" << std::endl;
+    shutting_down_ = true;
+    for (auto& controlled_trickle_candidate : controlled_trickle_candidates_) {
+      for (auto& cand : controlled_trickle_candidate.second) {
+        delete cand;
+      }
+    }
+
+    ice_ctx_->Destroy();
+    ice_ctx_ = nullptr;
+
+    if (remote_) {
+      remote_->UnsetRemote();
+      remote_ = nullptr;
+    }
+  }
+
+  void UnsetRemote() { remote_ = nullptr; }
+
+  void StartChecks() {
+    nsresult res;
+
+    test_utils_->SyncDispatchToSTS(WrapRunnableRet(
+        &res, ice_ctx_, &NrIceCtx::SetControlling,
+        offerer_ ? NrIceCtx::ICE_CONTROLLING : NrIceCtx::ICE_CONTROLLED));
+    // Now start checks
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnableRet(&res, ice_ctx_, &NrIceCtx::StartChecks));
+    ASSERT_TRUE(NS_SUCCEEDED(res));
+  }
+
+  // Handle events
+  void GatheringStateChange(NrIceCtx* ctx, NrIceCtx::GatheringState state) {
+    if (shutting_down_) {
+      return;
+    }
+    if (state != NrIceCtx::ICE_CTX_GATHER_COMPLETE) {
+      return;
+    }
+
+    std::cerr << name_ << " Gathering complete" << std::endl;
+    gathering_complete_ = true;
+
+    std::cerr << name_ << " ATTRIBUTES:" << std::endl;
+    for (const auto& stream : ice_ctx_->GetStreams()) {
+      std::cerr << "Stream " << stream->GetId() << std::endl;
+
+      std::vector<std::string> attributes = stream->GetAttributes();
+
+      for (const auto& attribute : attributes) {
+        std::cerr << attribute << std::endl;
+      }
+    }
+    std::cerr << std::endl;
+  }
+
+  void CandidateInitialized(NrIceMediaStream* stream,
+                            const std::string& raw_candidate,
+                            const std::string& ufrag,
+                            const std::string& mdns_addr,
+                            const std::string& actual_addr) {
+    std::string candidate(FilterCandidate(raw_candidate));
+    if (candidate.empty()) {
+      return;
+    }
+    std::cerr << "Candidate for stream " << stream->name()
+              << " initialized: " << candidate << std::endl;
+    candidates_[stream->name()].push_back(candidate);
+
+    // If we are connected, then try to trickle to the other side.
+    if (remote_ && remote_->remote_ && (trickle_mode_ != TRICKLE_SIMULATE)) {
+      // first, find the index of the stream we've been given so
+      // we can get the corresponding stream on the remote side
+      for (size_t i = 0; i < stream_counter_; ++i) {
+        if (GetStream_s(i) == stream) {
+          ASSERT_GT(remote_->stream_counter_, i);
+          nsresult res = remote_->GetStream_s(i)->ParseTrickleCandidate(
+              candidate, ufrag, "");
+          ASSERT_TRUE(NS_SUCCEEDED(res));
+          return;
+        }
+      }
+      ADD_FAILURE() << "No matching stream found for " << stream;
+    }
+  }
+
+  nsresult GetCandidatePairs_s(size_t stream_index,
+                               std::vector<NrIceCandidatePair>* pairs) {
+    MOZ_ASSERT(pairs);
+    auto stream = GetStream_s(stream_index);
+    if (!stream) {
+      // Is there a better error for "no such index"?
+      ADD_FAILURE() << "No such media stream index: " << stream_index;
+      return NS_ERROR_INVALID_ARG;
+    }
+
+    return stream->GetCandidatePairs(pairs);
+  }
+
+  nsresult GetCandidatePairs(size_t stream_index,
+                             std::vector<NrIceCandidatePair>* pairs) {
+    nsresult v;
+    test_utils_->SyncDispatchToSTS(WrapRunnableRet(
+        &v, this, &IceTestPeer::GetCandidatePairs_s, stream_index, pairs));
+    return v;
+  }
+
+  void DumpCandidatePair(const NrIceCandidatePair& pair) {
+    std::cerr << std::endl;
+    DumpCandidate("Local", pair.local);
+    DumpCandidate("Remote", pair.remote);
+    std::cerr << "state = " << pair.state << " priority = " << pair.priority
+              << " nominated = " << pair.nominated
+              << " selected = " << pair.selected
+              << " codeword = " << pair.codeword << std::endl;
+  }
+
+  void DumpCandidatePairs_s(NrIceMediaStream* stream) {
+    std::vector<NrIceCandidatePair> pairs;
+    nsresult res = stream->GetCandidatePairs(&pairs);
+    ASSERT_TRUE(NS_SUCCEEDED(res));
+
+    std::cerr << "Begin list of candidate pairs [" << std::endl;
+
+    for (auto& pair : pairs) {
+      DumpCandidatePair(pair);
+    }
+    std::cerr << "]" << std::endl;
+  }
+
+  void DumpCandidatePairs_s() {
+    std::cerr << "Dumping candidate pairs for all streams [" << std::endl;
+    for (const auto& stream : ice_ctx_->GetStreams()) {
+      DumpCandidatePairs_s(stream.get());
+    }
+    std::cerr << "]" << std::endl;
+  }
+
+  bool CandidatePairsPriorityDescending(
+      const std::vector<NrIceCandidatePair>& pairs) {
+    // Verify that priority is descending
+    uint64_t priority = std::numeric_limits<uint64_t>::max();
+
+    for (size_t p = 0; p < pairs.size(); ++p) {
+      if (priority < pairs[p].priority) {
+        std::cerr << "Priority increased in subsequent pairs:" << std::endl;
+        DumpCandidatePair(pairs[p - 1]);
+        DumpCandidatePair(pairs[p]);
+        return false;
+      }
+      if (priority == pairs[p].priority) {
+        if (!IceCandidatePairCompare()(pairs[p], pairs[p - 1]) &&
+            !IceCandidatePairCompare()(pairs[p - 1], pairs[p])) {
+          std::cerr << "Ignoring identical pair from trigger check"
+                    << std::endl;
+        } else {
+          std::cerr << "Duplicate priority in subseqent pairs:" << std::endl;
+          DumpCandidatePair(pairs[p - 1]);
+          DumpCandidatePair(pairs[p]);
+          return false;
+        }
+      }
+      priority = pairs[p].priority;
+    }
+    return true;
+  }
+
+  void UpdateAndValidateCandidatePairs(
+      size_t stream_index, std::vector<NrIceCandidatePair>* new_pairs) {
+    std::vector<NrIceCandidatePair> old_pairs = *new_pairs;
+    GetCandidatePairs(stream_index, new_pairs);
+    ASSERT_TRUE(CandidatePairsPriorityDescending(*new_pairs))
+    << "New list of "
+       "candidate pairs is either not sorted in priority order, or has "
+       "duplicate priorities.";
+    ASSERT_TRUE(CandidatePairsPriorityDescending(old_pairs))
+    << "Old list of "
+       "candidate pairs is either not sorted in priority order, or has "
+       "duplicate priorities. This indicates some bug in the test case.";
+    std::vector<NrIceCandidatePair> added_pairs;
+    std::vector<NrIceCandidatePair> removed_pairs;
+
+    // set_difference computes the set of elements that are present in the
+    // first set, but not the second
+    // NrIceCandidatePair::operator< compares based on the priority, local
+    // candidate, and remote candidate in that order. This means this will
+    // catch cases where the priority has remained the same, but one of the
+    // candidates has changed.
+    std::set_difference((*new_pairs).begin(), (*new_pairs).end(),
+                        old_pairs.begin(), old_pairs.end(),
+                        std::inserter(added_pairs, added_pairs.begin()),
+                        IceCandidatePairCompare());
+
+    std::set_difference(old_pairs.begin(), old_pairs.end(),
+                        (*new_pairs).begin(), (*new_pairs).end(),
+                        std::inserter(removed_pairs, removed_pairs.begin()),
+                        IceCandidatePairCompare());
+
+    for (auto& added_pair : added_pairs) {
+      std::cerr << "Found new candidate pair." << std::endl;
+      DumpCandidatePair(added_pair);
+    }
+
+    for (auto& removed_pair : removed_pairs) {
+      std::cerr << "Pre-existing candidate pair is now missing:" << std::endl;
+      DumpCandidatePair(removed_pair);
+    }
+
+    ASSERT_TRUE(removed_pairs.empty())
+    << "At least one candidate pair has "
+       "gone missing.";
+  }
+
+  void StreamReady(NrIceMediaStream* stream) {
+    ++ready_ct_;
+    std::cerr << name_ << " Stream ready for " << stream->name()
+              << " ct=" << ready_ct_ << std::endl;
+    DumpCandidatePairs_s(stream);
+  }
+  void StreamFailed(NrIceMediaStream* stream) {
+    std::cerr << name_ << " Stream failed for " << stream->name()
+              << " ct=" << ready_ct_ << std::endl;
+    DumpCandidatePairs_s(stream);
+  }
+
+  void ConnectionStateChange(NrIceCtx* ctx, NrIceCtx::ConnectionState state) {
+    (void)ctx;
+    switch (state) {
+      case NrIceCtx::ICE_CTX_INIT:
+        break;
+      case NrIceCtx::ICE_CTX_CHECKING:
+        std::cerr << name_ << " ICE reached checking" << std::endl;
+        ice_reached_checking_ = true;
+        break;
+      case NrIceCtx::ICE_CTX_CONNECTED:
+        std::cerr << name_ << " ICE connected" << std::endl;
+        ice_connected_ = true;
+        break;
+      case NrIceCtx::ICE_CTX_COMPLETED:
+        std::cerr << name_ << " ICE completed" << std::endl;
+        break;
+      case NrIceCtx::ICE_CTX_FAILED:
+        std::cerr << name_ << " ICE failed" << std::endl;
+        ice_failed_ = true;
+        break;
+      case NrIceCtx::ICE_CTX_DISCONNECTED:
+        std::cerr << name_ << " ICE disconnected" << std::endl;
+        ice_connected_ = false;
+        break;
+      default:
+        MOZ_CRASH();
+    }
+  }
+
+  void PacketReceived(NrIceMediaStream* stream, int component,
+                      const unsigned char* data, int len) {
+    std::cerr << name_ << ": received " << len << " bytes" << std::endl;
+    ++received_;
+  }
+
+  void SendPacket(int stream, int component, const unsigned char* data,
+                  int len) {
+    auto media_stream = GetStream_s(stream);
+    if (!media_stream) {
+      ADD_FAILURE() << "No such stream " << stream;
+      return;
+    }
+
+    ASSERT_TRUE(NS_SUCCEEDED(media_stream->SendPacket(component, data, len)));
+
+    ++sent_;
+    std::cerr << name_ << ": sent " << len << " bytes" << std::endl;
+  }
+
+  void SendFailure(int stream, int component) {
+    auto media_stream = GetStream_s(stream);
+    if (!media_stream) {
+      ADD_FAILURE() << "No such stream " << stream;
+      return;
+    }
+
+    const std::string d("FAIL");
+    ASSERT_TRUE(NS_FAILED(media_stream->SendPacket(
+        component, reinterpret_cast<const unsigned char*>(d.c_str()),
+        d.length())));
+
+    std::cerr << name_ << ": send failed as expected" << std::endl;
+  }
+
+  void SetCandidateFilter(CandidateFilter filter) {
+    candidate_filter_ = filter;
+  }
+
+  void ParseCandidate_s(size_t i, const std::string& candidate,
+                        const std::string& mdns_addr) {
+    auto media_stream = GetStream_s(i);
+    ASSERT_TRUE(media_stream.get())
+    << "No such stream " << i;
+    media_stream->ParseTrickleCandidate(candidate, "", mdns_addr);
+  }
+
+  void ParseCandidate(size_t i, const std::string& candidate,
+                      const std::string& mdns_addr) {
+    test_utils_->SyncDispatchToSTS(WrapRunnable(
+        this, &IceTestPeer::ParseCandidate_s, i, candidate, mdns_addr));
+  }
+
+  void DisableComponent_s(size_t index, int component_id) {
+    ASSERT_LT(index, stream_counter_);
+    auto stream = GetStream_s(index);
+    ASSERT_TRUE(stream.get())
+    << "No such stream " << index;
+    nsresult res = stream->DisableComponent(component_id);
+    ASSERT_TRUE(NS_SUCCEEDED(res));
+  }
+
+  void DisableComponent(size_t stream, int component_id) {
+    test_utils_->SyncDispatchToSTS(WrapRunnable(
+        this, &IceTestPeer::DisableComponent_s, stream, component_id));
+  }
+
+  void AssertConsentRefresh_s(size_t index, int component_id,
+                              ConsentStatus status) {
+    ASSERT_LT(index, stream_counter_);
+    auto stream = GetStream_s(index);
+    ASSERT_TRUE(stream.get())
+    << "No such stream " << index;
+    bool can_send;
+    struct timeval timestamp;
+    nsresult res =
+        stream->GetConsentStatus(component_id, &can_send, &timestamp);
+    ASSERT_TRUE(NS_SUCCEEDED(res));
+    if (status == CONSENT_EXPIRED) {
+      ASSERT_EQ(can_send, 0);
+    } else {
+      ASSERT_EQ(can_send, 1);
+    }
+    if (consent_timestamp_.tv_sec) {
+      if (status == CONSENT_FRESH) {
+        ASSERT_EQ(r_timeval_cmp(&timestamp, &consent_timestamp_), 1);
+      } else {
+        ASSERT_EQ(r_timeval_cmp(&timestamp, &consent_timestamp_), 0);
+      }
+    }
+    consent_timestamp_.tv_sec = timestamp.tv_sec;
+    consent_timestamp_.tv_usec = timestamp.tv_usec;
+    std::cerr << name_
+              << ": new consent timestamp = " << consent_timestamp_.tv_sec
+              << "." << consent_timestamp_.tv_usec << std::endl;
+  }
+
+  void AssertConsentRefresh(ConsentStatus status) {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &IceTestPeer::AssertConsentRefresh_s, 0, 1, status));
+  }
+
+  void ChangeNetworkState_s(bool online) {
+    ice_ctx_->UpdateNetworkState(online);
+  }
+
+  void ChangeNetworkStateToOffline() {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &IceTestPeer::ChangeNetworkState_s, false));
+  }
+
+  void ChangeNetworkStateToOnline() {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &IceTestPeer::ChangeNetworkState_s, true));
+  }
+
+  void SetControlling(NrIceCtx::Controlling controlling) {
+    nsresult res;
+    test_utils_->SyncDispatchToSTS(WrapRunnableRet(
+        &res, ice_ctx_, &NrIceCtx::SetControlling, controlling));
+    ASSERT_TRUE(NS_SUCCEEDED(res));
+  }
+
+  NrIceCtx::Controlling GetControlling() { return ice_ctx_->GetControlling(); }
+
+  void SetTiebreaker(uint64_t tiebreaker) {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &IceTestPeer::SetTiebreaker_s, tiebreaker));
+  }
+
+  void SetTiebreaker_s(uint64_t tiebreaker) {
+    ice_ctx_->peer()->tiebreaker = tiebreaker;
+  }
+
+  void SimulateIceLite() {
+    simulate_ice_lite_ = true;
+    SetControlling(NrIceCtx::ICE_CONTROLLED);
+  }
+
+  nsresult GetDefaultCandidate(unsigned int stream, NrIceCandidate* cand) {
+    nsresult rv;
+
+    test_utils_->SyncDispatchToSTS(WrapRunnableRet(
+        &rv, this, &IceTestPeer::GetDefaultCandidate_s, stream, cand));
+
+    return rv;
+  }
+
+  nsresult GetDefaultCandidate_s(unsigned int index, NrIceCandidate* cand) {
+    return GetStream_s(index)->GetDefaultCandidate(1, cand);
+  }
+
+ private:
+  std::string name_;
+  RefPtr<NrIceCtx> ice_ctx_;
+  bool offerer_;
+  std::map<std::string, std::vector<std::string>> candidates_;
+  // Maps from stream id to list of remote trickle candidates
+  std::map<size_t, std::vector<SchedulableTrickleCandidate*>>
+      controlled_trickle_candidates_;
+  std::map<std::string, std::pair<std::string, std::string>> mIceCredentials;
+  std::map<std::string, std::pair<std::string, std::string>> mOldIceCredentials;
+  size_t stream_counter_;
+  bool shutting_down_;
+  bool gathering_complete_;
+  int ready_ct_;
+  bool ice_connected_;
+  bool ice_failed_;
+  bool ice_reached_checking_;
+  size_t received_;
+  size_t sent_;
+  struct timeval consent_timestamp_;
+  NrIceResolverFake fake_resolver_;
+  RefPtr<NrIceResolver> dns_resolver_;
+  IceTestPeer* remote_;
+  CandidateFilter candidate_filter_;
+  NrIceCandidate::Type expected_local_type_;
+  std::string expected_local_transport_;
+  NrIceCandidate::Type expected_remote_type_;
+  std::string expected_remote_addr_;
+  TrickleMode trickle_mode_;
+  bool simulate_ice_lite_;
+  RefPtr<mozilla::TestNat> nat_;
+  MtransportTestUtils* test_utils_;
+};
+
+void SchedulableTrickleCandidate::Trickle() {
+  timer_handle_ = nullptr;
+  nsresult res = peer_->TrickleCandidate_s(candidate_, ufrag_, stream_);
+  ASSERT_TRUE(NS_SUCCEEDED(res));
+}
+
+class WebRtcIceGatherTest : public StunTest {
+ public:
+  void SetUp() override {
+    StunTest::SetUp();
+
+    Preferences::SetInt("media.peerconnection.ice.tcp_so_sock_count", 3);
+
+    test_utils_->SyncDispatchToSTS(WrapRunnable(
+        TestStunServer::GetInstance(AF_INET), &TestStunServer::Reset));
+    if (TestStunServer::GetInstance(AF_INET6)) {
+      test_utils_->SyncDispatchToSTS(WrapRunnable(
+          TestStunServer::GetInstance(AF_INET6), &TestStunServer::Reset));
+    }
+  }
+
+  void TearDown() override {
+    peer_ = nullptr;
+    StunTest::TearDown();
+  }
+
+  void EnsurePeer() {
+    if (!peer_) {
+      peer_ =
+          MakeUnique<IceTestPeer>("P1", test_utils_, true, NrIceCtx::Config());
+    }
+  }
+
+  void Gather(unsigned int waitTime = kDefaultTimeout,
+              bool default_route_only = false,
+              bool obfuscate_host_addresses = false) {
+    EnsurePeer();
+    peer_->Gather(default_route_only, obfuscate_host_addresses);
+
+    if (waitTime) {
+      WaitForGather(waitTime);
+    }
+  }
+
+  void WaitForGather(unsigned int waitTime = kDefaultTimeout) {
+    ASSERT_TRUE_WAIT(peer_->gathering_complete(), waitTime);
+  }
+
+  void AddStunServerWithResponse(const std::string& fake_addr,
+                                 uint16_t fake_port, const std::string& fqdn,
+                                 const std::string& proto,
+                                 std::vector<NrIceStunServer>* stun_servers) {
+    int family;
+    if (fake_addr.find(':') != std::string::npos) {
+      family = AF_INET6;
+    } else {
+      family = AF_INET;
+    }
+
+    std::string stun_addr;
+    uint16_t stun_port;
+    if (proto == kNrIceTransportUdp) {
+      TestStunServer::GetInstance(family)->SetResponseAddr(fake_addr,
+                                                           fake_port);
+      stun_addr = TestStunServer::GetInstance(family)->addr();
+      stun_port = TestStunServer::GetInstance(family)->port();
+    } else if (proto == kNrIceTransportTcp) {
+      TestStunTcpServer::GetInstance(family)->SetResponseAddr(fake_addr,
+                                                              fake_port);
+      stun_addr = TestStunTcpServer::GetInstance(family)->addr();
+      stun_port = TestStunTcpServer::GetInstance(family)->port();
+    } else {
+      MOZ_CRASH();
+    }
+
+    if (!fqdn.empty()) {
+      peer_->SetFakeResolver(stun_addr, fqdn);
+      stun_addr = fqdn;
+    }
+
+    stun_servers->push_back(
+        *NrIceStunServer::Create(stun_addr, stun_port, proto.c_str()));
+
+    if (family == AF_INET6 && !fqdn.empty()) {
+      stun_servers->back().SetUseIPv6IfFqdn();
+    }
+  }
+
+  void UseFakeStunUdpServerWithResponse(
+      const std::string& fake_addr, uint16_t fake_port,
+      const std::string& fqdn = std::string()) {
+    EnsurePeer();
+    std::vector<NrIceStunServer> stun_servers;
+    AddStunServerWithResponse(fake_addr, fake_port, fqdn, "udp", &stun_servers);
+    peer_->SetStunServers(stun_servers);
+  }
+
+  void UseFakeStunTcpServerWithResponse(
+      const std::string& fake_addr, uint16_t fake_port,
+      const std::string& fqdn = std::string()) {
+    EnsurePeer();
+    std::vector<NrIceStunServer> stun_servers;
+    AddStunServerWithResponse(fake_addr, fake_port, fqdn, "tcp", &stun_servers);
+    peer_->SetStunServers(stun_servers);
+  }
+
+  void UseFakeStunUdpTcpServersWithResponse(const std::string& fake_udp_addr,
+                                            uint16_t fake_udp_port,
+                                            const std::string& fake_tcp_addr,
+                                            uint16_t fake_tcp_port) {
+    EnsurePeer();
+    std::vector<NrIceStunServer> stun_servers;
+    AddStunServerWithResponse(fake_udp_addr, fake_udp_port,
+                              "",  // no fqdn
+                              "udp", &stun_servers);
+    AddStunServerWithResponse(fake_tcp_addr, fake_tcp_port,
+                              "",  // no fqdn
+                              "tcp", &stun_servers);
+
+    peer_->SetStunServers(stun_servers);
+  }
+
+  void UseTestStunServer() {
+    TestStunServer::GetInstance(AF_INET)->Reset();
+    peer_->SetStunServer(TestStunServer::GetInstance(AF_INET)->addr(),
+                         TestStunServer::GetInstance(AF_INET)->port());
+  }
+
+  // NB: Only does substring matching, watch out for stuff like "1.2.3.4"
+  // matching "21.2.3.47". " 1.2.3.4 " should not have false positives.
+  bool StreamHasMatchingCandidate(unsigned int stream, const std::string& match,
+                                  const std::string& match2 = "") {
+    std::vector<std::string> attributes = peer_->GetAttributes(stream);
+    for (auto& attribute : attributes) {
+      if (std::string::npos != attribute.find(match)) {
+        if (!match2.length() || std::string::npos != attribute.find(match2)) {
+          return true;
+        }
+      }
+    }
+    return false;
+  }
+
+  void DumpAttributes(unsigned int stream) {
+    std::vector<std::string> attributes = peer_->GetAttributes(stream);
+
+    std::cerr << "Attributes for stream " << stream << "->" << attributes.size()
+              << std::endl;
+
+    for (const auto& a : attributes) {
+      std::cerr << "Attribute: " << a << std::endl;
+    }
+  }
+
+ protected:
+  mozilla::UniquePtr<IceTestPeer> peer_;
+};
+
+class WebRtcIceConnectTest : public StunTest {
+ public:
+  WebRtcIceConnectTest()
+      : initted_(false),
+        test_stun_server_inited_(false),
+        use_nat_(false),
+        filtering_type_(TestNat::ENDPOINT_INDEPENDENT),
+        mapping_type_(TestNat::ENDPOINT_INDEPENDENT),
+        block_udp_(false) {}
+
+  void SetUp() override {
+    StunTest::SetUp();
+
+    nsresult rv;
+    target_ = do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &rv);
+    ASSERT_TRUE(NS_SUCCEEDED(rv));
+  }
+
+  void TearDown() override {
+    p1_ = nullptr;
+    p2_ = nullptr;
+
+    StunTest::TearDown();
+  }
+
+  void AddStream(int components) {
+    Init();
+    p1_->AddStream(components);
+    p2_->AddStream(components);
+  }
+
+  void RemoveStream(size_t index) {
+    p1_->RemoveStream(index);
+    p2_->RemoveStream(index);
+  }
+
+  void Init(bool setup_stun_servers = true,
+            NrIceCtx::Policy ice_policy = NrIceCtx::ICE_POLICY_ALL) {
+    if (initted_) {
+      return;
+    }
+
+    NrIceCtx::Config config;
+    config.mPolicy = ice_policy;
+
+    p1_ = MakeUnique<IceTestPeer>("P1", test_utils_, true, config);
+    p2_ = MakeUnique<IceTestPeer>("P2", test_utils_, false, config);
+    InitPeer(p1_.get(), setup_stun_servers);
+    InitPeer(p2_.get(), setup_stun_servers);
+
+    initted_ = true;
+  }
+
+  void InitPeer(IceTestPeer* peer, bool setup_stun_servers = true) {
+    if (use_nat_) {
+      // If we enable nat simulation, but still use a real STUN server somewhere
+      // on the internet, we will see failures if there is a real NAT in
+      // addition to our simulated one, particularly if it disallows
+      // hairpinning.
+      if (setup_stun_servers) {
+        InitTestStunServer();
+        peer->UseTestStunServer();
+      }
+      peer->UseNat();
+      peer->SetFilteringType(filtering_type_);
+      peer->SetMappingType(mapping_type_);
+      peer->SetBlockUdp(block_udp_);
+    } else if (setup_stun_servers) {
+      std::vector<NrIceStunServer> stun_servers;
+
+      stun_servers.push_back(*NrIceStunServer::Create(
+          stun_server_address_, kDefaultStunServerPort, kNrIceTransportUdp));
+
+      peer->SetStunServers(stun_servers);
+    }
+  }
+
+  bool Gather(unsigned int waitTime = kDefaultTimeout,
+              bool default_route_only = false) {
+    Init();
+
+    return GatherCallerAndCallee(p1_.get(), p2_.get(), waitTime,
+                                 default_route_only);
+  }
+
+  bool GatherCallerAndCallee(IceTestPeer* caller, IceTestPeer* callee,
+                             unsigned int waitTime = kDefaultTimeout,
+                             bool default_route_only = false) {
+    caller->Gather(default_route_only);
+    callee->Gather(default_route_only);
+
+    if (waitTime) {
+      EXPECT_TRUE_WAIT(caller->gathering_complete(), waitTime);
+      if (!caller->gathering_complete()) return false;
+      EXPECT_TRUE_WAIT(callee->gathering_complete(), waitTime);
+      if (!callee->gathering_complete()) return false;
+    }
+    return true;
+  }
+
+  void UseNat() {
+    // to be useful, this method should be called before Init
+    ASSERT_FALSE(initted_);
+    use_nat_ = true;
+  }
+
+  void SetFilteringType(TestNat::NatBehavior type) {
+    // to be useful, this method should be called before Init
+    ASSERT_FALSE(initted_);
+    filtering_type_ = type;
+  }
+
+  void SetMappingType(TestNat::NatBehavior type) {
+    // to be useful, this method should be called before Init
+    ASSERT_FALSE(initted_);
+    mapping_type_ = type;
+  }
+
+  void BlockUdp() {
+    // note: |block_udp_| is used only in InitPeer.
+    // Use IceTestPeer::SetBlockUdp to act on the peer directly.
+    block_udp_ = true;
+  }
+
+  void SetupAndCheckConsent() {
+    p1_->SetTimerDivider(10);
+    p2_->SetTimerDivider(10);
+    ASSERT_TRUE(Gather());
+    Connect();
+    p1_->AssertConsentRefresh(CONSENT_FRESH);
+    p2_->AssertConsentRefresh(CONSENT_FRESH);
+    SendReceive();
+  }
+
+  void AssertConsentRefresh(ConsentStatus status = CONSENT_FRESH) {
+    p1_->AssertConsentRefresh(status);
+    p2_->AssertConsentRefresh(status);
+  }
+
+  void InitTestStunServer() {
+    if (test_stun_server_inited_) {
+      return;
+    }
+
+    std::cerr << "Resetting TestStunServer" << std::endl;
+    TestStunServer::GetInstance(AF_INET)->Reset();
+    test_stun_server_inited_ = true;
+  }
+
+  void UseTestStunServer() {
+    InitTestStunServer();
+    p1_->UseTestStunServer();
+    p2_->UseTestStunServer();
+  }
+
+  void SetTurnServer(const std::string addr, uint16_t port,
+                     const std::string username, const std::string password,
+                     const char* transport = kNrIceTransportUdp) {
+    p1_->SetTurnServer(addr, port, username, password, transport);
+    p2_->SetTurnServer(addr, port, username, password, transport);
+  }
+
+  void SetTurnServers(const std::vector<NrIceTurnServer>& servers) {
+    p1_->SetTurnServers(servers);
+    p2_->SetTurnServers(servers);
+  }
+
+  void SetCandidateFilter(CandidateFilter filter, bool both = true) {
+    p1_->SetCandidateFilter(filter);
+    if (both) {
+      p2_->SetCandidateFilter(filter);
+    }
+  }
+
+  void Connect() { ConnectCallerAndCallee(p1_.get(), p2_.get()); }
+
+  void ConnectCallerAndCallee(IceTestPeer* caller, IceTestPeer* callee,
+                              TrickleMode mode = TRICKLE_NONE) {
+    ASSERT_TRUE(caller->ready_ct() == 0);
+    ASSERT_TRUE(caller->ice_connected() == 0);
+    ASSERT_TRUE(caller->ice_reached_checking() == 0);
+    ASSERT_TRUE(callee->ready_ct() == 0);
+    ASSERT_TRUE(callee->ice_connected() == 0);
+    ASSERT_TRUE(callee->ice_reached_checking() == 0);
+
+    // IceTestPeer::Connect grabs attributes from the first arg, and
+    // gives them to |this|, meaning that callee->Connect(caller, ...)
+    // simulates caller sending an offer to callee. Order matters here
+    // because it determines which peer is controlling.
+    callee->Connect(caller, mode);
+    caller->Connect(callee, mode);
+
+    if (mode != TRICKLE_SIMULATE) {
+      ASSERT_TRUE_WAIT(caller->ice_connected() && callee->ice_connected(),
+                       kDefaultTimeout);
+      ASSERT_TRUE(caller->ready_ct() >= 1 && callee->ready_ct() >= 1);
+      ASSERT_TRUE(caller->ice_reached_checking());
+      ASSERT_TRUE(callee->ice_reached_checking());
+
+      caller->DumpAndCheckActiveCandidates();
+      callee->DumpAndCheckActiveCandidates();
+    }
+  }
+
+  void SetExpectedTypes(NrIceCandidate::Type local, NrIceCandidate::Type remote,
+                        std::string transport = kNrIceTransportUdp) {
+    p1_->SetExpectedTypes(local, remote, transport);
+    p2_->SetExpectedTypes(local, remote, transport);
+  }
+
+  void SetExpectedRemoteCandidateAddr(const std::string& addr) {
+    p1_->SetExpectedRemoteCandidateAddr(addr);
+    p2_->SetExpectedRemoteCandidateAddr(addr);
+  }
+
+  void ConnectP1(TrickleMode mode = TRICKLE_NONE) {
+    p1_->Connect(p2_.get(), mode);
+  }
+
+  void ConnectP2(TrickleMode mode = TRICKLE_NONE) {
+    p2_->Connect(p1_.get(), mode);
+  }
+
+  void WaitForConnectedStreams(int expected_streams = 1) {
+    ASSERT_TRUE_WAIT(p1_->ready_ct() == expected_streams &&
+                         p2_->ready_ct() == expected_streams,
+                     kDefaultTimeout);
+    ASSERT_TRUE_WAIT(p1_->ice_connected() && p2_->ice_connected(),
+                     kDefaultTimeout);
+  }
+
+  void AssertCheckingReached() {
+    ASSERT_TRUE(p1_->ice_reached_checking());
+    ASSERT_TRUE(p2_->ice_reached_checking());
+  }
+
+  void WaitForConnected(unsigned int timeout = kDefaultTimeout) {
+    ASSERT_TRUE_WAIT(p1_->ice_connected(), timeout);
+    ASSERT_TRUE_WAIT(p2_->ice_connected(), timeout);
+  }
+
+  void WaitForGather() {
+    ASSERT_TRUE_WAIT(p1_->gathering_complete(), kDefaultTimeout);
+    ASSERT_TRUE_WAIT(p2_->gathering_complete(), kDefaultTimeout);
+  }
+
+  void WaitForDisconnected(unsigned int timeout = kDefaultTimeout) {
+    ASSERT_TRUE(p1_->ice_connected());
+    ASSERT_TRUE(p2_->ice_connected());
+    ASSERT_TRUE_WAIT(p1_->ice_connected() == 0 && p2_->ice_connected() == 0,
+                     timeout);
+  }
+
+  void WaitForFailed(unsigned int timeout = kDefaultTimeout) {
+    ASSERT_TRUE_WAIT(p1_->ice_failed() && p2_->ice_failed(), timeout);
+  }
+
+  void ConnectTrickle(TrickleMode trickle = TRICKLE_SIMULATE) {
+    p2_->Connect(p1_.get(), trickle);
+    p1_->Connect(p2_.get(), trickle);
+  }
+
+  void SimulateTrickle(size_t stream) {
+    p1_->SimulateTrickle(stream);
+    p2_->SimulateTrickle(stream);
+    ASSERT_TRUE_WAIT(p1_->is_ready(stream), kDefaultTimeout);
+    ASSERT_TRUE_WAIT(p2_->is_ready(stream), kDefaultTimeout);
+  }
+
+  void SimulateTrickleP1(size_t stream) { p1_->SimulateTrickle(stream); }
+
+  void SimulateTrickleP2(size_t stream) { p2_->SimulateTrickle(stream); }
+
+  void CloseP1() { p1_->Close(); }
+
+  void ConnectThenDelete() {
+    p2_->Connect(p1_.get(), TRICKLE_NONE, false);
+    p1_->Connect(p2_.get(), TRICKLE_NONE, true);
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &WebRtcIceConnectTest::CloseP1));
+    p2_->StartChecks();
+
+    // Wait to see if we crash
+    PR_Sleep(PR_MillisecondsToInterval(kDefaultTimeout));
+  }
+
+  // default is p1_ sending to p2_
+  void SendReceive() { SendReceive(p1_.get(), p2_.get()); }
+
+  void SendReceive(IceTestPeer* p1, IceTestPeer* p2,
+                   bool expect_tx_failure = false,
+                   bool expect_rx_failure = false) {
+    size_t previousSent = p1->sent();
+    size_t previousReceived = p2->received();
+
+    if (expect_tx_failure) {
+      test_utils_->SyncDispatchToSTS(
+          WrapRunnable(p1, &IceTestPeer::SendFailure, 0, 1));
+      ASSERT_EQ(previousSent, p1->sent());
+    } else {
+      test_utils_->SyncDispatchToSTS(
+          WrapRunnable(p1, &IceTestPeer::SendPacket, 0, 1,
+                       reinterpret_cast<const unsigned char*>("TEST"), 4));
+      ASSERT_EQ(previousSent + 1, p1->sent());
+    }
+    if (expect_rx_failure) {
+      usleep(1000);
+      ASSERT_EQ(previousReceived, p2->received());
+    } else {
+      ASSERT_TRUE_WAIT(p2->received() == previousReceived + 1, 1000);
+    }
+  }
+
+  void SendFailure() {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(p1_.get(), &IceTestPeer::SendFailure, 0, 1));
+  }
+
+ protected:
+  bool initted_;
+  bool test_stun_server_inited_;
+  nsCOMPtr<nsIEventTarget> target_;
+  mozilla::UniquePtr<IceTestPeer> p1_;
+  mozilla::UniquePtr<IceTestPeer> p2_;
+  bool use_nat_;
+  TestNat::NatBehavior filtering_type_;
+  TestNat::NatBehavior mapping_type_;
+  bool block_udp_;
+};
+
+class WebRtcIcePrioritizerTest : public StunTest {
+ public:
+  WebRtcIcePrioritizerTest() : prioritizer_(nullptr) {}
+
+  ~WebRtcIcePrioritizerTest() {
+    if (prioritizer_) {
+      nr_interface_prioritizer_destroy(&prioritizer_);
+    }
+  }
+
+  void SetPriorizer(nr_interface_prioritizer* prioritizer) {
+    prioritizer_ = prioritizer;
+  }
+
+  void AddInterface(const std::string& num, int type, int estimated_speed) {
+    std::string str_addr = "10.0.0." + num;
+    std::string ifname = "eth" + num;
+    nr_local_addr local_addr;
+    local_addr.interface.type = type;
+    local_addr.interface.estimated_speed = estimated_speed;
+
+    int r = nr_str_port_to_transport_addr(str_addr.c_str(), 0, IPPROTO_UDP,
+                                          &(local_addr.addr));
+    ASSERT_EQ(0, r);
+    strncpy(local_addr.addr.ifname, ifname.c_str(), MAXIFNAME - 1);
+    local_addr.addr.ifname[MAXIFNAME - 1] = '\0';
+
+    r = nr_interface_prioritizer_add_interface(prioritizer_, &local_addr);
+    ASSERT_EQ(0, r);
+    r = nr_interface_prioritizer_sort_preference(prioritizer_);
+    ASSERT_EQ(0, r);
+  }
+
+  void HasLowerPreference(const std::string& num1, const std::string& num2) {
+    std::string key1 = "eth" + num1 + ":10.0.0." + num1;
+    std::string key2 = "eth" + num2 + ":10.0.0." + num2;
+    UCHAR pref1, pref2;
+    int r = nr_interface_prioritizer_get_priority(prioritizer_, key1.c_str(),
+                                                  &pref1);
+    ASSERT_EQ(0, r);
+    r = nr_interface_prioritizer_get_priority(prioritizer_, key2.c_str(),
+                                              &pref2);
+    ASSERT_EQ(0, r);
+    ASSERT_LE(pref1, pref2);
+  }
+
+ private:
+  nr_interface_prioritizer* prioritizer_;
+};
+
+class WebRtcIcePacketFilterTest : public StunTest {
+ public:
+  WebRtcIcePacketFilterTest() : udp_filter_(nullptr), tcp_filter_(nullptr) {}
+
+  void SetUp() {
+    StunTest::SetUp();
+
+    NrIceCtx::InitializeGlobals(NrIceCtx::GlobalConfig());
+
+    // Set up enough of the ICE ctx to allow the packet filter to work
+    ice_ctx_ = NrIceCtx::Create("test");
+
+    nsCOMPtr<nsISocketFilterHandler> udp_handler =
+        do_GetService(NS_STUN_UDP_SOCKET_FILTER_HANDLER_CONTRACTID);
+    ASSERT_TRUE(udp_handler);
+    udp_handler->NewFilter(getter_AddRefs(udp_filter_));
+
+    nsCOMPtr<nsISocketFilterHandler> tcp_handler =
+        do_GetService(NS_STUN_TCP_SOCKET_FILTER_HANDLER_CONTRACTID);
+    ASSERT_TRUE(tcp_handler);
+    tcp_handler->NewFilter(getter_AddRefs(tcp_filter_));
+  }
+
+  void TearDown() {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &WebRtcIcePacketFilterTest::TearDown_s));
+    StunTest::TearDown();
+  }
+
+  void TearDown_s() { ice_ctx_ = nullptr; }
+
+  void TestIncoming(const uint8_t* data, uint32_t len, uint8_t from_addr,
+                    int from_port, bool expected_result) {
+    mozilla::net::NetAddr addr;
+    MakeNetAddr(&addr, from_addr, from_port);
+    bool result;
+    nsresult rv = udp_filter_->FilterPacket(
+        &addr, data, len, nsISocketFilter::SF_INCOMING, &result);
+    ASSERT_EQ(NS_OK, rv);
+    ASSERT_EQ(expected_result, result);
+  }
+
+  void TestIncomingTcp(const uint8_t* data, uint32_t len,
+                       bool expected_result) {
+    mozilla::net::NetAddr addr;
+    bool result;
+    nsresult rv = tcp_filter_->FilterPacket(
+        &addr, data, len, nsISocketFilter::SF_INCOMING, &result);
+    ASSERT_EQ(NS_OK, rv);
+    ASSERT_EQ(expected_result, result);
+  }
+
+  void TestIncomingTcpFramed(const uint8_t* data, uint32_t len,
+                             bool expected_result) {
+    mozilla::net::NetAddr addr;
+    bool result;
+    uint8_t* framed_data = new uint8_t[len + 2];
+    framed_data[0] = htons(len);
+    memcpy(&framed_data[2], data, len);
+    nsresult rv = tcp_filter_->FilterPacket(
+        &addr, framed_data, len + 2, nsISocketFilter::SF_INCOMING, &result);
+    ASSERT_EQ(NS_OK, rv);
+    ASSERT_EQ(expected_result, result);
+    delete[] framed_data;
+  }
+
+  void TestOutgoing(const uint8_t* data, uint32_t len, uint8_t to_addr,
+                    int to_port, bool expected_result) {
+    mozilla::net::NetAddr addr;
+    MakeNetAddr(&addr, to_addr, to_port);
+    bool result;
+    nsresult rv = udp_filter_->FilterPacket(
+        &addr, data, len, nsISocketFilter::SF_OUTGOING, &result);
+    ASSERT_EQ(NS_OK, rv);
+    ASSERT_EQ(expected_result, result);
+  }
+
+  void TestOutgoingTcp(const uint8_t* data, uint32_t len,
+                       bool expected_result) {
+    mozilla::net::NetAddr addr;
+    bool result;
+    nsresult rv = tcp_filter_->FilterPacket(
+        &addr, data, len, nsISocketFilter::SF_OUTGOING, &result);
+    ASSERT_EQ(NS_OK, rv);
+    ASSERT_EQ(expected_result, result);
+  }
+
+  void TestOutgoingTcpFramed(const uint8_t* data, uint32_t len,
+                             bool expected_result) {
+    mozilla::net::NetAddr addr;
+    bool result;
+    uint8_t* framed_data = new uint8_t[len + 2];
+    framed_data[0] = htons(len);
+    memcpy(&framed_data[2], data, len);
+    nsresult rv = tcp_filter_->FilterPacket(
+        &addr, framed_data, len + 2, nsISocketFilter::SF_OUTGOING, &result);
+    ASSERT_EQ(NS_OK, rv);
+    ASSERT_EQ(expected_result, result);
+    delete[] framed_data;
+  }
+
+ private:
+  void MakeNetAddr(mozilla::net::NetAddr* net_addr, uint8_t last_digit,
+                   uint16_t port) {
+    net_addr->inet.family = AF_INET;
+    net_addr->inet.ip = 192 << 24 | 168 << 16 | 1 << 8 | last_digit;
+    net_addr->inet.port = port;
+  }
+
+  nsCOMPtr<nsISocketFilter> udp_filter_;
+  nsCOMPtr<nsISocketFilter> tcp_filter_;
+  RefPtr<NrIceCtx> ice_ctx_;
+};
+}  // end namespace
+
+TEST_F(WebRtcIceGatherTest, TestGatherFakeStunServerHostnameNoResolver) {
+  if (stun_server_hostname_.empty()) {
+    return;
+  }
+
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  EnsurePeer();
+  peer_->SetStunServer(stun_server_hostname_, kDefaultStunServerPort);
+  peer_->AddStream(1);
+  Gather();
+}
+
+// Disabled because google isn't running any TCP stun servers right now
+TEST_F(WebRtcIceGatherTest,
+       DISABLED_TestGatherFakeStunServerTcpHostnameNoResolver) {
+  if (stun_server_hostname_.empty()) {
+    return;
+  }
+
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = true;
+  NrIceCtx::InitializeGlobals(config);
+  EnsurePeer();
+  peer_->SetStunServer(stun_server_hostname_, kDefaultStunServerPort,
+                       kNrIceTransportTcp);
+  peer_->AddStream(1);
+  Gather();
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, " TCP "));
+}
+
+TEST_F(WebRtcIceGatherTest, TestGatherFakeStunServerIpAddress) {
+  if (stun_server_address_.empty()) {
+    return;
+  }
+
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  EnsurePeer();
+  peer_->SetStunServer(stun_server_address_, kDefaultStunServerPort);
+  peer_->SetFakeResolver(stun_server_address_, stun_server_hostname_);
+  peer_->AddStream(1);
+  Gather();
+}
+
+TEST_F(WebRtcIceGatherTest, TestGatherStunServerIpAddressNoHost) {
+  if (stun_server_address_.empty()) {
+    return;
+  }
+
+  {
+    NrIceCtx::GlobalConfig config;
+    config.mTcpEnabled = false;
+    NrIceCtx::InitializeGlobals(config);
+  }
+
+  NrIceCtx::Config config;
+  config.mPolicy = NrIceCtx::ICE_POLICY_NO_HOST;
+  peer_ = MakeUnique<IceTestPeer>("P1", test_utils_, true, config);
+  peer_->AddStream(1);
+  peer_->SetStunServer(stun_server_address_, kDefaultStunServerPort);
+  peer_->SetFakeResolver(stun_server_address_, stun_server_hostname_);
+  Gather();
+  ASSERT_FALSE(StreamHasMatchingCandidate(0, " host "));
+}
+
+TEST_F(WebRtcIceGatherTest, TestGatherFakeStunServerHostname) {
+  if (stun_server_hostname_.empty()) {
+    return;
+  }
+
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  EnsurePeer();
+  peer_->SetStunServer(stun_server_hostname_, kDefaultStunServerPort);
+  peer_->SetFakeResolver(stun_server_address_, stun_server_hostname_);
+  peer_->AddStream(1);
+  Gather();
+}
+
+TEST_F(WebRtcIceGatherTest, TestGatherFakeStunBogusHostname) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  EnsurePeer();
+  peer_->SetStunServer(kBogusStunServerHostname, kDefaultStunServerPort);
+  peer_->SetFakeResolver(stun_server_address_, stun_server_hostname_);
+  peer_->AddStream(1);
+  Gather();
+}
+
+TEST_F(WebRtcIceGatherTest, TestGatherDNSStunServerIpAddress) {
+  if (stun_server_address_.empty()) {
+    return;
+  }
+
+  {
+    NrIceCtx::GlobalConfig config;
+    config.mTcpEnabled = false;
+    NrIceCtx::InitializeGlobals(config);
+  }
+
+  // A srflx candidate is considered redundant and discarded if its address
+  // equals that of a host candidate. (Frequently, a srflx candidate and a host
+  // candidate have equal addresses when the agent is not behind a NAT.) So set
+  // ICE_POLICY_NO_HOST here to ensure that a srflx candidate is not falsely
+  // discarded in this test.
+  NrIceCtx::Config config;
+  config.mPolicy = NrIceCtx::ICE_POLICY_NO_HOST;
+  peer_ = MakeUnique<IceTestPeer>("P1", test_utils_, true, config);
+
+  peer_->SetStunServer(stun_server_address_, kDefaultStunServerPort);
+  peer_->SetDNSResolver();
+  peer_->AddStream(1);
+  Gather();
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, " UDP "));
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, "typ srflx raddr"));
+}
+
+// Disabled because google isn't running any TCP stun servers right now
+TEST_F(WebRtcIceGatherTest, DISABLED_TestGatherDNSStunServerIpAddressTcp) {
+  if (stun_server_address_.empty()) {
+    return;
+  }
+
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = true;
+  NrIceCtx::InitializeGlobals(config);
+  EnsurePeer();
+  peer_->SetStunServer(stun_server_address_, kDefaultStunServerPort,
+                       kNrIceTransportTcp);
+  peer_->SetDNSResolver();
+  peer_->AddStream(1);
+  Gather();
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, "tcptype passive"));
+  ASSERT_FALSE(StreamHasMatchingCandidate(0, "tcptype passive", " 9 "));
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, "tcptype so"));
+  ASSERT_FALSE(StreamHasMatchingCandidate(0, "tcptype so", " 9 "));
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, "tcptype active", " 9 "));
+}
+
+TEST_F(WebRtcIceGatherTest, TestGatherDNSStunServerHostname) {
+  if (stun_server_hostname_.empty()) {
+    return;
+  }
+
+  {
+    NrIceCtx::GlobalConfig config;
+    config.mTcpEnabled = false;
+    NrIceCtx::InitializeGlobals(config);
+  }
+
+  // A srflx candidate is considered redundant and discarded if its address
+  // equals that of a host candidate. (Frequently, a srflx candidate and a host
+  // candidate have equal addresses when the agent is not behind a NAT.) So set
+  // ICE_POLICY_NO_HOST here to ensure that a srflx candidate is not falsely
+  // discarded in this test.
+  NrIceCtx::Config config;
+  config.mPolicy = NrIceCtx::ICE_POLICY_NO_HOST;
+  peer_ = MakeUnique<IceTestPeer>("P1", test_utils_, true, config);
+
+  peer_->SetStunServer(stun_server_hostname_, kDefaultStunServerPort);
+  peer_->SetDNSResolver();
+  peer_->AddStream(1);
+  Gather();
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, " UDP "));
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, "typ srflx raddr"));
+}
+
+// Disabled because google isn't running any TCP stun servers right now
+TEST_F(WebRtcIceGatherTest, DISABLED_TestGatherDNSStunServerHostnameTcp) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = true;
+  NrIceCtx::InitializeGlobals(config);
+  EnsurePeer();
+  peer_->SetStunServer(stun_server_hostname_, kDefaultStunServerPort,
+                       kNrIceTransportTcp);
+  peer_->SetDNSResolver();
+  peer_->AddStream(1);
+  Gather();
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, "tcptype passive"));
+  ASSERT_FALSE(StreamHasMatchingCandidate(0, "tcptype passive", " 9 "));
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, "tcptype so"));
+  ASSERT_FALSE(StreamHasMatchingCandidate(0, "tcptype so", " 9 "));
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, "tcptype active", " 9 "));
+}
+
+// Disabled because google isn't running any TCP stun servers right now
+TEST_F(WebRtcIceGatherTest,
+       DISABLED_TestGatherDNSStunServerHostnameBothUdpTcp) {
+  if (stun_server_hostname_.empty()) {
+    return;
+  }
+
+  std::vector<NrIceStunServer> stun_servers;
+
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = true;
+  NrIceCtx::InitializeGlobals(config);
+  EnsurePeer();
+  stun_servers.push_back(*NrIceStunServer::Create(
+      stun_server_hostname_, kDefaultStunServerPort, kNrIceTransportUdp));
+  stun_servers.push_back(*NrIceStunServer::Create(
+      stun_server_hostname_, kDefaultStunServerPort, kNrIceTransportTcp));
+  peer_->SetStunServers(stun_servers);
+  peer_->SetDNSResolver();
+  peer_->AddStream(1);
+  Gather();
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, " UDP "));
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, " TCP "));
+}
+
+// Disabled because google isn't running any TCP stun servers right now
+TEST_F(WebRtcIceGatherTest,
+       DISABLED_TestGatherDNSStunServerIpAddressBothUdpTcp) {
+  if (stun_server_address_.empty()) {
+    return;
+  }
+
+  std::vector<NrIceStunServer> stun_servers;
+
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = true;
+  NrIceCtx::InitializeGlobals(config);
+  EnsurePeer();
+  stun_servers.push_back(*NrIceStunServer::Create(
+      stun_server_address_, kDefaultStunServerPort, kNrIceTransportUdp));
+  stun_servers.push_back(*NrIceStunServer::Create(
+      stun_server_address_, kDefaultStunServerPort, kNrIceTransportTcp));
+  peer_->SetStunServers(stun_servers);
+  peer_->SetDNSResolver();
+  peer_->AddStream(1);
+  Gather();
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, " UDP "));
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, " TCP "));
+}
+
+TEST_F(WebRtcIceGatherTest, TestGatherDNSStunBogusHostname) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  EnsurePeer();
+  peer_->SetStunServer(kBogusStunServerHostname, kDefaultStunServerPort);
+  peer_->SetDNSResolver();
+  peer_->AddStream(1);
+  Gather();
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, " UDP "));
+}
+
+// Disabled because google isn't running any TCP stun servers right now
+TEST_F(WebRtcIceGatherTest, DISABLED_TestGatherDNSStunBogusHostnameTcp) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = true;
+  NrIceCtx::InitializeGlobals(config);
+  EnsurePeer();
+  peer_->SetStunServer(kBogusStunServerHostname, kDefaultStunServerPort,
+                       kNrIceTransportTcp);
+  peer_->SetDNSResolver();
+  peer_->AddStream(1);
+  Gather();
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, " TCP "));
+}
+
+TEST_F(WebRtcIceGatherTest, TestDefaultCandidate) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  EnsurePeer();
+  peer_->SetStunServer(stun_server_hostname_, kDefaultStunServerPort);
+  peer_->AddStream(1);
+  Gather();
+  NrIceCandidate default_candidate;
+  ASSERT_TRUE(NS_SUCCEEDED(peer_->GetDefaultCandidate(0, &default_candidate)));
+}
+
+TEST_F(WebRtcIceGatherTest, TestGatherTurn) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  EnsurePeer();
+  if (turn_server_.empty()) return;
+  peer_->SetTurnServer(turn_server_, kDefaultStunServerPort, turn_user_,
+                       turn_password_, kNrIceTransportUdp);
+  peer_->AddStream(1);
+  Gather();
+}
+
+TEST_F(WebRtcIceGatherTest, TestGatherTurnTcp) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  EnsurePeer();
+  if (turn_server_.empty()) return;
+  peer_->SetTurnServer(turn_server_, kDefaultStunServerPort, turn_user_,
+                       turn_password_, kNrIceTransportTcp);
+  peer_->AddStream(1);
+  Gather();
+}
+
+TEST_F(WebRtcIceGatherTest, TestGatherDisableComponent) {
+  if (stun_server_hostname_.empty()) {
+    return;
+  }
+
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  EnsurePeer();
+  peer_->SetStunServer(stun_server_hostname_, kDefaultStunServerPort);
+  peer_->AddStream(1);
+  peer_->AddStream(2);
+  peer_->DisableComponent(1, 2);
+  Gather();
+  std::vector<std::string> attributes = peer_->GetAttributes(1);
+
+  for (auto& attribute : attributes) {
+    if (attribute.find("candidate:") != std::string::npos) {
+      size_t sp1 = attribute.find(' ');
+      ASSERT_EQ(0, attribute.compare(sp1 + 1, 1, "1", 1));
+    }
+  }
+}
+
+TEST_F(WebRtcIceGatherTest, TestGatherVerifyNoLoopback) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  EnsurePeer();
+  peer_->AddStream(1);
+  Gather();
+  ASSERT_FALSE(StreamHasMatchingCandidate(0, "127.0.0.1"));
+}
+
+TEST_F(WebRtcIceGatherTest, TestGatherAllowLoopback) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  config.mAllowLoopback = true;
+  NrIceCtx::InitializeGlobals(config);
+
+  // Set up peer with loopback allowed.
+  peer_ = MakeUnique<IceTestPeer>("P1", test_utils_, true, NrIceCtx::Config());
+  peer_->AddStream(1);
+  Gather();
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, "127.0.0.1"));
+}
+
+TEST_F(WebRtcIceGatherTest, TestGatherTcpDisabledNoStun) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  EnsurePeer();
+  peer_->AddStream(1);
+  Gather();
+  ASSERT_FALSE(StreamHasMatchingCandidate(0, " TCP "));
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, " UDP "));
+}
+
+TEST_F(WebRtcIceGatherTest, VerifyTestStunServer) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  UseFakeStunUdpServerWithResponse("192.0.2.133", 3333);
+  peer_->AddStream(1);
+  Gather();
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, " 192.0.2.133 3333 "));
+}
+
+TEST_F(WebRtcIceGatherTest, VerifyTestStunTcpServer) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = true;
+  NrIceCtx::InitializeGlobals(config);
+  UseFakeStunTcpServerWithResponse("192.0.2.233", 3333);
+  peer_->AddStream(1);
+  Gather();
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, " 192.0.2.233 3333 typ srflx",
+                                         " tcptype "));
+}
+
+TEST_F(WebRtcIceGatherTest, VerifyTestStunServerV6) {
+  if (!TestStunServer::GetInstance(AF_INET6)) {
+    // No V6 addresses
+    return;
+  }
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  UseFakeStunUdpServerWithResponse("beef::", 3333);
+  peer_->AddStream(1);
+  Gather();
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, " beef:: 3333 "));
+}
+
+TEST_F(WebRtcIceGatherTest, VerifyTestStunServerFQDN) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  UseFakeStunUdpServerWithResponse("192.0.2.133", 3333, "stun.example.com");
+  peer_->AddStream(1);
+  Gather();
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, " 192.0.2.133 3333 "));
+}
+
+TEST_F(WebRtcIceGatherTest, VerifyTestStunServerV6FQDN) {
+  if (!TestStunServer::GetInstance(AF_INET6)) {
+    // No V6 addresses
+    return;
+  }
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  UseFakeStunUdpServerWithResponse("beef::", 3333, "stun.example.com");
+  peer_->AddStream(1);
+  Gather();
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, " beef:: 3333 "));
+}
+
+TEST_F(WebRtcIceGatherTest, TestStunServerReturnsWildcardAddr) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  UseFakeStunUdpServerWithResponse("0.0.0.0", 3333);
+  peer_->AddStream(1);
+  Gather(kDefaultTimeout * 3);
+  ASSERT_FALSE(StreamHasMatchingCandidate(0, " 0.0.0.0 "));
+}
+
+TEST_F(WebRtcIceGatherTest, TestStunServerReturnsWildcardAddrV6) {
+  if (!TestStunServer::GetInstance(AF_INET6)) {
+    // No V6 addresses
+    return;
+  }
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  UseFakeStunUdpServerWithResponse("::", 3333);
+  peer_->AddStream(1);
+  Gather(kDefaultTimeout * 3);
+  ASSERT_FALSE(StreamHasMatchingCandidate(0, " :: "));
+}
+
+TEST_F(WebRtcIceGatherTest, TestStunServerReturnsPort0) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  UseFakeStunUdpServerWithResponse("192.0.2.133", 0);
+  peer_->AddStream(1);
+  Gather(kDefaultTimeout * 3);
+  ASSERT_FALSE(StreamHasMatchingCandidate(0, " 192.0.2.133 0 "));
+}
+
+TEST_F(WebRtcIceGatherTest, TestStunServerReturnsLoopbackAddr) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  UseFakeStunUdpServerWithResponse("127.0.0.133", 3333);
+  peer_->AddStream(1);
+  Gather(kDefaultTimeout * 3);
+  ASSERT_FALSE(StreamHasMatchingCandidate(0, " 127.0.0.133 "));
+}
+
+TEST_F(WebRtcIceGatherTest, TestStunServerReturnsLoopbackAddrV6) {
+  if (!TestStunServer::GetInstance(AF_INET6)) {
+    // No V6 addresses
+    return;
+  }
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  UseFakeStunUdpServerWithResponse("::1", 3333);
+  peer_->AddStream(1);
+  Gather(kDefaultTimeout * 3);
+  ASSERT_FALSE(StreamHasMatchingCandidate(0, " ::1 "));
+}
+
+TEST_F(WebRtcIceGatherTest, TestStunServerTrickle) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  UseFakeStunUdpServerWithResponse("192.0.2.1", 3333);
+  peer_->AddStream(1);
+  TestStunServer::GetInstance(AF_INET)->SetDropInitialPackets(3);
+  Gather(0);
+  ASSERT_FALSE(StreamHasMatchingCandidate(0, "192.0.2.1"));
+  WaitForGather();
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, "192.0.2.1"));
+}
+
+// Test no host with our fake STUN server and apparently NATted.
+TEST_F(WebRtcIceGatherTest, TestFakeStunServerNatedNoHost) {
+  {
+    NrIceCtx::GlobalConfig config;
+    config.mTcpEnabled = false;
+    NrIceCtx::InitializeGlobals(config);
+  }
+
+  NrIceCtx::Config config;
+  config.mPolicy = NrIceCtx::ICE_POLICY_NO_HOST;
+  peer_ = MakeUnique<IceTestPeer>("P1", test_utils_, true, config);
+  UseFakeStunUdpServerWithResponse("192.0.2.1", 3333);
+  peer_->AddStream(1);
+  Gather(0);
+  WaitForGather();
+  DumpAttributes(0);
+  ASSERT_FALSE(StreamHasMatchingCandidate(0, "host"));
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, "srflx"));
+  NrIceCandidate default_candidate;
+  nsresult rv = peer_->GetDefaultCandidate(0, &default_candidate);
+  if (NS_SUCCEEDED(rv)) {
+    ASSERT_NE(NrIceCandidate::ICE_HOST, default_candidate.type);
+  }
+}
+
+// Test no host with our fake STUN server and apparently non-NATted.
+TEST_F(WebRtcIceGatherTest, TestFakeStunServerNoNatNoHost) {
+  {
+    NrIceCtx::GlobalConfig config;
+    config.mTcpEnabled = false;
+    NrIceCtx::InitializeGlobals(config);
+  }
+
+  NrIceCtx::Config config;
+  config.mPolicy = NrIceCtx::ICE_POLICY_NO_HOST;
+  peer_ = MakeUnique<IceTestPeer>("P1", test_utils_, true, config);
+  UseTestStunServer();
+  peer_->AddStream(1);
+  Gather(0);
+  WaitForGather();
+  DumpAttributes(0);
+  ASSERT_FALSE(StreamHasMatchingCandidate(0, "host"));
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, "srflx"));
+}
+
+// Test that srflx candidate is discarded in non-NATted environment if host
+// address obfuscation is not enabled.
+TEST_F(WebRtcIceGatherTest,
+       TestSrflxCandidateDiscardedWithObfuscateHostAddressesNotEnabled) {
+  {
+    NrIceCtx::GlobalConfig config;
+    config.mTcpEnabled = false;
+    NrIceCtx::InitializeGlobals(config);
+  }
+
+  NrIceCtx::Config config;
+  peer_ = MakeUnique<IceTestPeer>("P1", test_utils_, true, config);
+  UseTestStunServer();
+  peer_->AddStream(1);
+  Gather(0, false, false);
+  WaitForGather();
+  DumpAttributes(0);
+  EXPECT_TRUE(StreamHasMatchingCandidate(0, "host"));
+  EXPECT_FALSE(StreamHasMatchingCandidate(0, "srflx"));
+}
+
+// Test that srflx candidate is generated in non-NATted environment if host
+// address obfuscation is enabled.
+TEST_F(WebRtcIceGatherTest,
+       TestSrflxCandidateGeneratedWithObfuscateHostAddressesEnabled) {
+  {
+    NrIceCtx::GlobalConfig config;
+    config.mTcpEnabled = false;
+    NrIceCtx::InitializeGlobals(config);
+  }
+
+  NrIceCtx::Config config;
+  peer_ = MakeUnique<IceTestPeer>("P1", test_utils_, true, config);
+  UseTestStunServer();
+  peer_->AddStream(1);
+  Gather(0, false, true);
+  WaitForGather();
+  DumpAttributes(0);
+  EXPECT_TRUE(StreamHasMatchingCandidate(0, "host"));
+  EXPECT_TRUE(StreamHasMatchingCandidate(0, "srflx"));
+}
+
+TEST_F(WebRtcIceGatherTest, TestStunTcpServerTrickle) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = true;
+  NrIceCtx::InitializeGlobals(config);
+  UseFakeStunTcpServerWithResponse("192.0.3.1", 3333);
+  TestStunTcpServer::GetInstance(AF_INET)->SetDelay(500);
+  peer_->AddStream(1);
+  Gather(0);
+  ASSERT_FALSE(StreamHasMatchingCandidate(0, " 192.0.3.1 ", " tcptype "));
+  WaitForGather();
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, " 192.0.3.1 ", " tcptype "));
+}
+
+TEST_F(WebRtcIceGatherTest, TestStunTcpAndUdpServerTrickle) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = true;
+  NrIceCtx::InitializeGlobals(config);
+  UseFakeStunUdpTcpServersWithResponse("192.0.2.1", 3333, "192.0.3.1", 3333);
+  TestStunServer::GetInstance(AF_INET)->SetDropInitialPackets(3);
+  TestStunTcpServer::GetInstance(AF_INET)->SetDelay(500);
+  peer_->AddStream(1);
+  Gather(0);
+  ASSERT_FALSE(StreamHasMatchingCandidate(0, "192.0.2.1", "UDP"));
+  ASSERT_FALSE(StreamHasMatchingCandidate(0, " 192.0.3.1 ", " tcptype "));
+  WaitForGather();
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, "192.0.2.1", "UDP"));
+  ASSERT_TRUE(StreamHasMatchingCandidate(0, " 192.0.3.1 ", " tcptype "));
+}
+
+TEST_F(WebRtcIceGatherTest, TestSetIceControlling) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  EnsurePeer();
+  peer_->AddStream(1);
+  peer_->SetControlling(NrIceCtx::ICE_CONTROLLING);
+  NrIceCtx::Controlling controlling = peer_->GetControlling();
+  ASSERT_EQ(NrIceCtx::ICE_CONTROLLING, controlling);
+  // SetControlling should only allow setting this once
+  peer_->SetControlling(NrIceCtx::ICE_CONTROLLED);
+  controlling = peer_->GetControlling();
+  ASSERT_EQ(NrIceCtx::ICE_CONTROLLING, controlling);
+}
+
+TEST_F(WebRtcIceGatherTest, TestSetIceControlled) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  EnsurePeer();
+  peer_->AddStream(1);
+  peer_->SetControlling(NrIceCtx::ICE_CONTROLLED);
+  NrIceCtx::Controlling controlling = peer_->GetControlling();
+  ASSERT_EQ(NrIceCtx::ICE_CONTROLLED, controlling);
+  // SetControlling should only allow setting this once
+  peer_->SetControlling(NrIceCtx::ICE_CONTROLLING);
+  controlling = peer_->GetControlling();
+  ASSERT_EQ(NrIceCtx::ICE_CONTROLLED, controlling);
+}
+
+TEST_F(WebRtcIceConnectTest, TestGather) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+}
+
+TEST_F(WebRtcIceConnectTest, TestGatherTcp) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = true;
+  NrIceCtx::InitializeGlobals(config);
+  Init();
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+}
+
+TEST_F(WebRtcIceConnectTest, TestGatherAutoPrioritize) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  Init();
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnect) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  Connect();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectRestartIce) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  Connect();
+  SendReceive(p1_.get(), p2_.get());
+
+  p2_->RestartIce();
+  ASSERT_FALSE(p2_->gathering_complete());
+
+  // verify p1 and p2 streams are still connected after restarting ice on p2
+  SendReceive(p1_.get(), p2_.get());
+
+  mozilla::UniquePtr<IceTestPeer> p3_;
+  p3_ = MakeUnique<IceTestPeer>("P3", test_utils_, true, NrIceCtx::Config());
+  InitPeer(p3_.get());
+  p3_->AddStream(1);
+
+  ASSERT_TRUE(GatherCallerAndCallee(p2_.get(), p3_.get()));
+  std::cout << "-------------------------------------------------" << std::endl;
+  ConnectCallerAndCallee(p3_.get(), p2_.get(), TRICKLE_SIMULATE);
+  SendReceive(p1_.get(), p2_.get());  // p1 and p2 are still connected
+  SendReceive(p3_.get(), p2_.get(), true, true);  // p3 and p2 not yet connected
+  p2_->SimulateTrickle(0);
+  p3_->SimulateTrickle(0);
+  ASSERT_TRUE_WAIT(p3_->is_ready(0), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(p2_->is_ready(0), kDefaultTimeout);
+  SendReceive(p1_.get(), p2_.get(), false, true);  // p1 and p2 not connected
+  SendReceive(p3_.get(), p2_.get());  // p3 and p2 are now connected
+
+  p3_ = nullptr;
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectRestartIceThenAbort) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  Connect();
+  SendReceive(p1_.get(), p2_.get());
+
+  p2_->RestartIce();
+  ASSERT_FALSE(p2_->gathering_complete());
+
+  // verify p1 and p2 streams are still connected after restarting ice on p2
+  SendReceive(p1_.get(), p2_.get());
+
+  mozilla::UniquePtr<IceTestPeer> p3_;
+  p3_ = MakeUnique<IceTestPeer>("P3", test_utils_, true, NrIceCtx::Config());
+  InitPeer(p3_.get());
+  p3_->AddStream(1);
+
+  ASSERT_TRUE(GatherCallerAndCallee(p2_.get(), p3_.get()));
+  std::cout << "-------------------------------------------------" << std::endl;
+  p2_->RollbackIceRestart();
+  p2_->Connect(p1_.get(), TRICKLE_NONE);
+  SendReceive(p1_.get(), p2_.get());
+  p3_ = nullptr;
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectIceRestartRoleConflict) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  // Just for fun lets do this with switched rolls
+  p1_->SetControlling(NrIceCtx::ICE_CONTROLLED);
+  p2_->SetControlling(NrIceCtx::ICE_CONTROLLING);
+  Connect();
+  SendReceive(p1_.get(), p2_.get());
+  // Set rolls should not switch by connecting
+  ASSERT_EQ(NrIceCtx::ICE_CONTROLLED, p1_->GetControlling());
+  ASSERT_EQ(NrIceCtx::ICE_CONTROLLING, p2_->GetControlling());
+
+  p2_->RestartIce();
+  ASSERT_FALSE(p2_->gathering_complete());
+  p2_->SetControlling(NrIceCtx::ICE_CONTROLLED);
+  ASSERT_EQ(NrIceCtx::ICE_CONTROLLING, p2_->GetControlling())
+      << "ICE restart should not allow role to change, unless ice-lite happens";
+
+  mozilla::UniquePtr<IceTestPeer> p3_;
+  p3_ = MakeUnique<IceTestPeer>("P3", test_utils_, true, NrIceCtx::Config());
+  InitPeer(p3_.get());
+  p3_->AddStream(1);
+  // Set control role for p3 accordingly (with role conflict)
+  p3_->SetControlling(NrIceCtx::ICE_CONTROLLING);
+  ASSERT_EQ(NrIceCtx::ICE_CONTROLLING, p3_->GetControlling());
+
+  ASSERT_TRUE(GatherCallerAndCallee(p2_.get(), p3_.get()));
+  std::cout << "-------------------------------------------------" << std::endl;
+  ConnectCallerAndCallee(p3_.get(), p2_.get());
+  auto p2role = p2_->GetControlling();
+  ASSERT_NE(p2role, p3_->GetControlling()) << "Conflict should be resolved";
+  ASSERT_EQ(NrIceCtx::ICE_CONTROLLED, p1_->GetControlling())
+      << "P1 should be unaffected by role conflict";
+
+  // And again we are not allowed to switch roles at this point any more
+  p1_->SetControlling(NrIceCtx::ICE_CONTROLLING);
+  ASSERT_EQ(NrIceCtx::ICE_CONTROLLED, p1_->GetControlling());
+  p3_->SetControlling(p2role);
+  ASSERT_NE(p2role, p3_->GetControlling());
+
+  p3_ = nullptr;
+}
+
+TEST_F(WebRtcIceConnectTest,
+       TestIceRestartWithMultipleInterfacesAndUserStartingScreenSharing) {
+  const char* FAKE_WIFI_ADDR = "10.0.0.1";
+  const char* FAKE_WIFI_IF_NAME = "wlan9";
+
+  // prepare a fake wifi interface
+  nr_local_addr wifi_addr;
+  wifi_addr.interface.type = NR_INTERFACE_TYPE_WIFI;
+  wifi_addr.interface.estimated_speed = 1000;
+
+  int r = nr_str_port_to_transport_addr(FAKE_WIFI_ADDR, 0, IPPROTO_UDP,
+                                        &(wifi_addr.addr));
+  ASSERT_EQ(0, r);
+  strncpy(wifi_addr.addr.ifname, FAKE_WIFI_IF_NAME, MAXIFNAME);
+
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  // setup initial ICE connection between p1_ and p2_
+  UseNat();
+  AddStream(1);
+  SetExpectedTypes(NrIceCandidate::Type::ICE_SERVER_REFLEXIVE,
+                   NrIceCandidate::Type::ICE_SERVER_REFLEXIVE);
+  ASSERT_TRUE(Gather(kDefaultTimeout, true));
+  Connect();
+
+  // verify the connection is working
+  SendReceive(p1_.get(), p2_.get());
+
+  // simulate user accepting permissions for screen sharing
+  p2_->SetCtxFlags(false);
+
+  // and having an additional non-default interface
+  nsTArray<NrIceStunAddr> stunAddr = p2_->GetStunAddrs();
+  stunAddr.InsertElementAt(0, NrIceStunAddr(&wifi_addr));
+  p2_->SetStunAddrs(stunAddr);
+
+  std::cout << "-------------------------------------------------" << std::endl;
+
+  // now restart ICE
+  p2_->RestartIce();
+  ASSERT_FALSE(p2_->gathering_complete());
+
+  // verify that we can successfully gather candidates
+  p2_->Gather();
+  EXPECT_TRUE_WAIT(p2_->gathering_complete(), kDefaultTimeout);
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectTcp) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = true;
+  NrIceCtx::InitializeGlobals(config);
+  Init();
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  SetCandidateFilter(IsTcpCandidate);
+  SetExpectedTypes(NrIceCandidate::Type::ICE_HOST,
+                   NrIceCandidate::Type::ICE_HOST, kNrIceTransportTcp);
+  Connect();
+}
+
+// TCP SO tests works on localhost only with delay applied:
+//  tc qdisc add dev lo root netem delay 10ms
+TEST_F(WebRtcIceConnectTest, DISABLED_TestConnectTcpSo) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = true;
+  NrIceCtx::InitializeGlobals(config);
+  Init();
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  SetCandidateFilter(IsTcpSoCandidate);
+  SetExpectedTypes(NrIceCandidate::Type::ICE_HOST,
+                   NrIceCandidate::Type::ICE_HOST, kNrIceTransportTcp);
+  Connect();
+}
+
+// Disabled because this breaks with hairpinning.
+TEST_F(WebRtcIceConnectTest, DISABLED_TestConnectNoHost) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  Init(false, NrIceCtx::ICE_POLICY_NO_HOST);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  SetExpectedTypes(NrIceCandidate::Type::ICE_SERVER_REFLEXIVE,
+                   NrIceCandidate::Type::ICE_SERVER_REFLEXIVE,
+                   kNrIceTransportTcp);
+  Connect();
+}
+
+TEST_F(WebRtcIceConnectTest, TestLoopbackOnlySortOf) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  config.mAllowLoopback = true;
+  NrIceCtx::InitializeGlobals(config);
+  Init(false);
+  AddStream(1);
+  SetCandidateFilter(IsLoopbackCandidate);
+  ASSERT_TRUE(Gather());
+  SetExpectedRemoteCandidateAddr("127.0.0.1");
+  Connect();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectBothControllingP1Wins) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  p1_->SetTiebreaker(1);
+  p2_->SetTiebreaker(0);
+  ASSERT_TRUE(Gather());
+  p1_->SetControlling(NrIceCtx::ICE_CONTROLLING);
+  p2_->SetControlling(NrIceCtx::ICE_CONTROLLING);
+  Connect();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectBothControllingP2Wins) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  p1_->SetTiebreaker(0);
+  p2_->SetTiebreaker(1);
+  ASSERT_TRUE(Gather());
+  p1_->SetControlling(NrIceCtx::ICE_CONTROLLING);
+  p2_->SetControlling(NrIceCtx::ICE_CONTROLLING);
+  Connect();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectIceLiteOfferer) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  p1_->SimulateIceLite();
+  Connect();
+}
+
+TEST_F(WebRtcIceConnectTest, TestTrickleBothControllingP1Wins) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  p1_->SetTiebreaker(1);
+  p2_->SetTiebreaker(0);
+  ASSERT_TRUE(Gather());
+  p1_->SetControlling(NrIceCtx::ICE_CONTROLLING);
+  p2_->SetControlling(NrIceCtx::ICE_CONTROLLING);
+  ConnectTrickle();
+  SimulateTrickle(0);
+  WaitForConnected(1000);
+  AssertCheckingReached();
+}
+
+TEST_F(WebRtcIceConnectTest, TestTrickleBothControllingP2Wins) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  p1_->SetTiebreaker(0);
+  p2_->SetTiebreaker(1);
+  ASSERT_TRUE(Gather());
+  p1_->SetControlling(NrIceCtx::ICE_CONTROLLING);
+  p2_->SetControlling(NrIceCtx::ICE_CONTROLLING);
+  ConnectTrickle();
+  SimulateTrickle(0);
+  WaitForConnected(1000);
+  AssertCheckingReached();
+}
+
+TEST_F(WebRtcIceConnectTest, TestTrickleIceLiteOfferer) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  p1_->SimulateIceLite();
+  ConnectTrickle();
+  SimulateTrickle(0);
+  WaitForConnected(1000);
+  AssertCheckingReached();
+}
+
+TEST_F(WebRtcIceConnectTest, TestGatherFullCone) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  UseNat();
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+}
+
+TEST_F(WebRtcIceConnectTest, TestGatherFullConeAutoPrioritize) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  UseNat();
+  Init();
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectFullCone) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  UseNat();
+  AddStream(1);
+  SetExpectedTypes(NrIceCandidate::Type::ICE_SERVER_REFLEXIVE,
+                   NrIceCandidate::Type::ICE_SERVER_REFLEXIVE);
+  ASSERT_TRUE(Gather());
+  Connect();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectNoNatNoHost) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  Init(false, NrIceCtx::ICE_POLICY_NO_HOST);
+  UseTestStunServer();
+  // Because we are connecting from our host candidate to the
+  // other side's apparent srflx (which is also their host)
+  // we see a host/srflx pair.
+  SetExpectedTypes(NrIceCandidate::Type::ICE_HOST,
+                   NrIceCandidate::Type::ICE_SERVER_REFLEXIVE);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  Connect();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectFullConeNoHost) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  UseNat();
+  Init(false, NrIceCtx::ICE_POLICY_NO_HOST);
+  UseTestStunServer();
+  SetExpectedTypes(NrIceCandidate::Type::ICE_SERVER_REFLEXIVE,
+                   NrIceCandidate::Type::ICE_SERVER_REFLEXIVE);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  Connect();
+}
+
+TEST_F(WebRtcIceConnectTest, TestGatherAddressRestrictedCone) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  UseNat();
+  SetFilteringType(TestNat::ADDRESS_DEPENDENT);
+  SetMappingType(TestNat::ENDPOINT_INDEPENDENT);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectAddressRestrictedCone) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  UseNat();
+  SetFilteringType(TestNat::ADDRESS_DEPENDENT);
+  SetMappingType(TestNat::ENDPOINT_INDEPENDENT);
+  AddStream(1);
+  SetExpectedTypes(NrIceCandidate::Type::ICE_SERVER_REFLEXIVE,
+                   NrIceCandidate::Type::ICE_SERVER_REFLEXIVE);
+  ASSERT_TRUE(Gather());
+  Connect();
+}
+
+TEST_F(WebRtcIceConnectTest, TestGatherPortRestrictedCone) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  UseNat();
+  SetFilteringType(TestNat::PORT_DEPENDENT);
+  SetMappingType(TestNat::ENDPOINT_INDEPENDENT);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectPortRestrictedCone) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  UseNat();
+  SetFilteringType(TestNat::PORT_DEPENDENT);
+  SetMappingType(TestNat::ENDPOINT_INDEPENDENT);
+  AddStream(1);
+  SetExpectedTypes(NrIceCandidate::Type::ICE_SERVER_REFLEXIVE,
+                   NrIceCandidate::Type::ICE_SERVER_REFLEXIVE);
+  ASSERT_TRUE(Gather());
+  Connect();
+}
+
+TEST_F(WebRtcIceConnectTest, TestGatherSymmetricNat) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  UseNat();
+  SetFilteringType(TestNat::PORT_DEPENDENT);
+  SetMappingType(TestNat::PORT_DEPENDENT);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectSymmetricNat) {
+  if (turn_server_.empty()) return;
+
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  UseNat();
+  SetFilteringType(TestNat::PORT_DEPENDENT);
+  SetMappingType(TestNat::PORT_DEPENDENT);
+  p1_->SetExpectedTypes(NrIceCandidate::Type::ICE_RELAYED,
+                        NrIceCandidate::Type::ICE_RELAYED);
+  p2_->SetExpectedTypes(NrIceCandidate::Type::ICE_RELAYED,
+                        NrIceCandidate::Type::ICE_RELAYED);
+  SetTurnServer(turn_server_, kDefaultStunServerPort, turn_user_,
+                turn_password_);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  Connect();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectSymmetricNatAndNoNat) {
+  {
+    NrIceCtx::GlobalConfig config;
+    config.mTcpEnabled = true;
+    NrIceCtx::InitializeGlobals(config);
+  }
+
+  NrIceCtx::Config config;
+  p1_ = MakeUnique<IceTestPeer>("P1", test_utils_, true, config);
+  p1_->UseNat();
+  p1_->SetFilteringType(TestNat::PORT_DEPENDENT);
+  p1_->SetMappingType(TestNat::PORT_DEPENDENT);
+
+  p2_ = MakeUnique<IceTestPeer>("P2", test_utils_, false, config);
+  initted_ = true;
+
+  AddStream(1);
+  p1_->SetExpectedTypes(NrIceCandidate::Type::ICE_PEER_REFLEXIVE,
+                        NrIceCandidate::Type::ICE_HOST);
+  p2_->SetExpectedTypes(NrIceCandidate::Type::ICE_HOST,
+                        NrIceCandidate::Type::ICE_PEER_REFLEXIVE);
+  ASSERT_TRUE(Gather());
+  Connect();
+}
+
+TEST_F(WebRtcIceConnectTest, TestGatherNatBlocksUDP) {
+  if (turn_server_.empty()) return;
+
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  UseNat();
+  BlockUdp();
+  std::vector<NrIceTurnServer> turn_servers;
+  std::vector<unsigned char> password_vec(turn_password_.begin(),
+                                          turn_password_.end());
+  turn_servers.push_back(
+      *NrIceTurnServer::Create(turn_server_, kDefaultStunServerPort, turn_user_,
+                               password_vec, kNrIceTransportTcp));
+  turn_servers.push_back(
+      *NrIceTurnServer::Create(turn_server_, kDefaultStunServerPort, turn_user_,
+                               password_vec, kNrIceTransportUdp));
+  SetTurnServers(turn_servers);
+  AddStream(1);
+  // We have to wait for the UDP-based stuff to time out.
+  ASSERT_TRUE(Gather(kDefaultTimeout * 3));
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectNatBlocksUDP) {
+  if (turn_server_.empty()) return;
+
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  UseNat();
+  BlockUdp();
+  std::vector<NrIceTurnServer> turn_servers;
+  std::vector<unsigned char> password_vec(turn_password_.begin(),
+                                          turn_password_.end());
+  turn_servers.push_back(
+      *NrIceTurnServer::Create(turn_server_, kDefaultStunServerPort, turn_user_,
+                               password_vec, kNrIceTransportTcp));
+  turn_servers.push_back(
+      *NrIceTurnServer::Create(turn_server_, kDefaultStunServerPort, turn_user_,
+                               password_vec, kNrIceTransportUdp));
+  SetTurnServers(turn_servers);
+  p1_->SetExpectedTypes(NrIceCandidate::Type::ICE_RELAYED,
+                        NrIceCandidate::Type::ICE_RELAYED, kNrIceTransportTcp);
+  p2_->SetExpectedTypes(NrIceCandidate::Type::ICE_RELAYED,
+                        NrIceCandidate::Type::ICE_RELAYED, kNrIceTransportTcp);
+  AddStream(1);
+  ASSERT_TRUE(Gather(kDefaultTimeout * 3));
+  Connect();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectTwoComponents) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(2);
+  ASSERT_TRUE(Gather());
+  Connect();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectTwoComponentsDisableSecond) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(2);
+  ASSERT_TRUE(Gather());
+  p1_->DisableComponent(0, 2);
+  p2_->DisableComponent(0, 2);
+  Connect();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectP2ThenP1) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  ConnectP2();
+  PR_Sleep(1000);
+  ConnectP1();
+  WaitForConnectedStreams();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectP2ThenP1Trickle) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  ConnectP2();
+  PR_Sleep(1000);
+  ConnectP1(TRICKLE_SIMULATE);
+  SimulateTrickleP1(0);
+  WaitForConnectedStreams();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectP2ThenP1TrickleTwoComponents) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  AddStream(2);
+  ASSERT_TRUE(Gather());
+  ConnectP2();
+  PR_Sleep(1000);
+  ConnectP1(TRICKLE_SIMULATE);
+  SimulateTrickleP1(0);
+  std::cerr << "Sleeping between trickle streams" << std::endl;
+  PR_Sleep(1000);  // Give this some time to settle but not complete
+                   // all of ICE.
+  SimulateTrickleP1(1);
+  WaitForConnectedStreams(2);
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectAutoPrioritize) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  Init();
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  Connect();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectTrickleOneStreamOneComponent) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  ConnectTrickle();
+  SimulateTrickle(0);
+  WaitForConnected(1000);
+  AssertCheckingReached();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectTrickleTwoStreamsOneComponent) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  ConnectTrickle();
+  SimulateTrickle(0);
+  SimulateTrickle(1);
+  WaitForConnected(1000);
+  AssertCheckingReached();
+}
+
+void RealisticTrickleDelay(
+    std::vector<SchedulableTrickleCandidate*>& candidates) {
+  for (size_t i = 0; i < candidates.size(); ++i) {
+    SchedulableTrickleCandidate* cand = candidates[i];
+    if (cand->IsHost()) {
+      cand->Schedule(i * 10);
+    } else if (cand->IsReflexive()) {
+      cand->Schedule(i * 10 + 100);
+    } else if (cand->IsRelay()) {
+      cand->Schedule(i * 10 + 200);
+    }
+  }
+}
+
+void DelayRelayCandidates(std::vector<SchedulableTrickleCandidate*>& candidates,
+                          unsigned int ms) {
+  for (auto& candidate : candidates) {
+    if (candidate->IsRelay()) {
+      candidate->Schedule(ms);
+    } else {
+      candidate->Schedule(0);
+    }
+  }
+}
+
+void AddNonPairableCandidates(
+    std::vector<SchedulableTrickleCandidate*>& candidates, IceTestPeer* peer,
+    size_t stream, int net_type, MtransportTestUtils* test_utils_) {
+  for (int i = 1; i < 5; i++) {
+    if (net_type == i) continue;
+    switch (i) {
+      case 1:
+        candidates.push_back(new SchedulableTrickleCandidate(
+            peer, stream,
+            "candidate:0 1 UDP 2113601790 10.0.0.1 12345 typ host", "",
+            test_utils_));
+        break;
+      case 2:
+        candidates.push_back(new SchedulableTrickleCandidate(
+            peer, stream,
+            "candidate:0 1 UDP 2113601791 172.16.1.1 12345 typ host", "",
+            test_utils_));
+        break;
+      case 3:
+        candidates.push_back(new SchedulableTrickleCandidate(
+            peer, stream,
+            "candidate:0 1 UDP 2113601792 192.168.0.1 12345 typ host", "",
+            test_utils_));
+        break;
+      case 4:
+        candidates.push_back(new SchedulableTrickleCandidate(
+            peer, stream,
+            "candidate:0 1 UDP 2113601793 100.64.1.1 12345 typ host", "",
+            test_utils_));
+        break;
+      default:
+        NR_UNIMPLEMENTED;
+    }
+  }
+
+  for (auto i = candidates.rbegin(); i != candidates.rend(); ++i) {
+    std::cerr << "Scheduling candidate: " << (*i)->Candidate().c_str()
+              << std::endl;
+    (*i)->Schedule(0);
+  }
+}
+
+void DropTrickleCandidates(
+    std::vector<SchedulableTrickleCandidate*>& candidates) {}
+
+TEST_F(WebRtcIceConnectTest, TestConnectTrickleAddStreamDuringICE) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  ConnectTrickle();
+  RealisticTrickleDelay(p1_->ControlTrickle(0));
+  RealisticTrickleDelay(p2_->ControlTrickle(0));
+  AddStream(1);
+  RealisticTrickleDelay(p1_->ControlTrickle(1));
+  RealisticTrickleDelay(p2_->ControlTrickle(1));
+  WaitForConnected(1000);
+  AssertCheckingReached();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectTrickleAddStreamAfterICE) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  ConnectTrickle();
+  RealisticTrickleDelay(p1_->ControlTrickle(0));
+  RealisticTrickleDelay(p2_->ControlTrickle(0));
+  WaitForConnected(1000);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  ConnectTrickle();
+  RealisticTrickleDelay(p1_->ControlTrickle(1));
+  RealisticTrickleDelay(p2_->ControlTrickle(1));
+  WaitForConnected(1000);
+  AssertCheckingReached();
+}
+
+TEST_F(WebRtcIceConnectTest, RemoveStream) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  ConnectTrickle();
+  RealisticTrickleDelay(p1_->ControlTrickle(0));
+  RealisticTrickleDelay(p2_->ControlTrickle(0));
+  RealisticTrickleDelay(p1_->ControlTrickle(1));
+  RealisticTrickleDelay(p2_->ControlTrickle(1));
+  WaitForConnected(1000);
+
+  RemoveStream(0);
+  ASSERT_TRUE(Gather());
+  ConnectTrickle();
+}
+
+TEST_F(WebRtcIceConnectTest, P1NoTrickle) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  ConnectTrickle();
+  DropTrickleCandidates(p1_->ControlTrickle(0));
+  RealisticTrickleDelay(p2_->ControlTrickle(0));
+  WaitForConnected(1000);
+}
+
+TEST_F(WebRtcIceConnectTest, P2NoTrickle) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  ConnectTrickle();
+  RealisticTrickleDelay(p1_->ControlTrickle(0));
+  DropTrickleCandidates(p2_->ControlTrickle(0));
+  WaitForConnected(1000);
+}
+
+TEST_F(WebRtcIceConnectTest, RemoveAndAddStream) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  ConnectTrickle();
+  RealisticTrickleDelay(p1_->ControlTrickle(0));
+  RealisticTrickleDelay(p2_->ControlTrickle(0));
+  RealisticTrickleDelay(p1_->ControlTrickle(1));
+  RealisticTrickleDelay(p2_->ControlTrickle(1));
+  WaitForConnected(1000);
+
+  RemoveStream(0);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  ConnectTrickle();
+  RealisticTrickleDelay(p1_->ControlTrickle(2));
+  RealisticTrickleDelay(p2_->ControlTrickle(2));
+  WaitForConnected(1000);
+}
+
+TEST_F(WebRtcIceConnectTest, RemoveStreamBeforeGather) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  AddStream(1);
+  ASSERT_TRUE(Gather(0));
+  RemoveStream(0);
+  WaitForGather();
+  ConnectTrickle();
+  RealisticTrickleDelay(p1_->ControlTrickle(1));
+  RealisticTrickleDelay(p2_->ControlTrickle(1));
+  WaitForConnected(1000);
+}
+
+TEST_F(WebRtcIceConnectTest, RemoveStreamDuringGather) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  AddStream(1);
+  RemoveStream(0);
+  ASSERT_TRUE(Gather());
+  ConnectTrickle();
+  RealisticTrickleDelay(p1_->ControlTrickle(1));
+  RealisticTrickleDelay(p2_->ControlTrickle(1));
+  WaitForConnected(1000);
+}
+
+TEST_F(WebRtcIceConnectTest, RemoveStreamDuringConnect) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  ConnectTrickle();
+  RealisticTrickleDelay(p1_->ControlTrickle(0));
+  RealisticTrickleDelay(p2_->ControlTrickle(0));
+  RealisticTrickleDelay(p1_->ControlTrickle(1));
+  RealisticTrickleDelay(p2_->ControlTrickle(1));
+  RemoveStream(0);
+  WaitForConnected(1000);
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectRealTrickleOneStreamOneComponent) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  AddStream(1);
+  ASSERT_TRUE(Gather(0));
+  ConnectTrickle(TRICKLE_REAL);
+  WaitForConnected();
+  WaitForGather();  // ICE can complete before we finish gathering.
+  AssertCheckingReached();
+}
+
+TEST_F(WebRtcIceConnectTest, TestSendReceive) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  Connect();
+  SendReceive();
+}
+
+TEST_F(WebRtcIceConnectTest, TestSendReceiveTcp) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = true;
+  NrIceCtx::InitializeGlobals(config);
+  Init();
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  SetCandidateFilter(IsTcpCandidate);
+  SetExpectedTypes(NrIceCandidate::Type::ICE_HOST,
+                   NrIceCandidate::Type::ICE_HOST, kNrIceTransportTcp);
+  Connect();
+  SendReceive();
+}
+
+// TCP SO tests works on localhost only with delay applied:
+//  tc qdisc add dev lo root netem delay 10ms
+TEST_F(WebRtcIceConnectTest, DISABLED_TestSendReceiveTcpSo) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = true;
+  NrIceCtx::InitializeGlobals(config);
+  Init();
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  SetCandidateFilter(IsTcpSoCandidate);
+  SetExpectedTypes(NrIceCandidate::Type::ICE_HOST,
+                   NrIceCandidate::Type::ICE_HOST, kNrIceTransportTcp);
+  Connect();
+  SendReceive();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConsent) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  SetupAndCheckConsent();
+  PR_Sleep(1500);
+  AssertConsentRefresh();
+  SendReceive();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConsentTcp) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = true;
+  NrIceCtx::InitializeGlobals(config);
+  Init();
+  AddStream(1);
+  SetCandidateFilter(IsTcpCandidate);
+  SetExpectedTypes(NrIceCandidate::Type::ICE_HOST,
+                   NrIceCandidate::Type::ICE_HOST, kNrIceTransportTcp);
+  SetupAndCheckConsent();
+  PR_Sleep(1500);
+  AssertConsentRefresh();
+  SendReceive();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConsentIntermittent) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  SetupAndCheckConsent();
+  p1_->SetBlockStun(true);
+  p2_->SetBlockStun(true);
+  WaitForDisconnected();
+  AssertConsentRefresh(CONSENT_STALE);
+  SendReceive();
+  p1_->SetBlockStun(false);
+  p2_->SetBlockStun(false);
+  WaitForConnected();
+  AssertConsentRefresh();
+  SendReceive();
+  p1_->SetBlockStun(true);
+  p2_->SetBlockStun(true);
+  WaitForDisconnected();
+  AssertConsentRefresh(CONSENT_STALE);
+  SendReceive();
+  p1_->SetBlockStun(false);
+  p2_->SetBlockStun(false);
+  WaitForConnected();
+  AssertConsentRefresh();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConsentTimeout) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  SetupAndCheckConsent();
+  p1_->SetBlockStun(true);
+  p2_->SetBlockStun(true);
+  WaitForDisconnected();
+  AssertConsentRefresh(CONSENT_STALE);
+  SendReceive();
+  WaitForFailed();
+  AssertConsentRefresh(CONSENT_EXPIRED);
+  SendFailure();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConsentDelayed) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  SetupAndCheckConsent();
+  /* Note: We don't have a list of STUN transaction IDs of the previously timed
+           out consent requests. Thus responses after sending the next consent
+           request are ignored. */
+  p1_->SetStunResponseDelay(200);
+  p2_->SetStunResponseDelay(200);
+  PR_Sleep(1000);
+  AssertConsentRefresh();
+  SendReceive();
+}
+
+TEST_F(WebRtcIceConnectTest, TestNetworkForcedOfflineAndRecovery) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  SetupAndCheckConsent();
+  p1_->ChangeNetworkStateToOffline();
+  ASSERT_TRUE_WAIT(p1_->ice_connected() == 0, kDefaultTimeout);
+  // Next round of consent check should switch it back to online
+  ASSERT_TRUE_WAIT(p1_->ice_connected(), kDefaultTimeout);
+}
+
+TEST_F(WebRtcIceConnectTest, TestNetworkForcedOfflineTwice) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  SetupAndCheckConsent();
+  p2_->ChangeNetworkStateToOffline();
+  ASSERT_TRUE_WAIT(p2_->ice_connected() == 0, kDefaultTimeout);
+  p2_->ChangeNetworkStateToOffline();
+  ASSERT_TRUE_WAIT(p2_->ice_connected() == 0, kDefaultTimeout);
+}
+
+TEST_F(WebRtcIceConnectTest, TestNetworkOnlineDoesntChangeState) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  SetupAndCheckConsent();
+  p2_->ChangeNetworkStateToOnline();
+  ASSERT_TRUE(p2_->ice_connected());
+  PR_Sleep(1500);
+  p2_->ChangeNetworkStateToOnline();
+  ASSERT_TRUE(p2_->ice_connected());
+}
+
+TEST_F(WebRtcIceConnectTest, TestNetworkOnlineTriggersConsent) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  // Let's emulate audio + video w/o rtcp-mux
+  AddStream(2);
+  AddStream(2);
+  SetupAndCheckConsent();
+  p1_->ChangeNetworkStateToOffline();
+  p1_->SetBlockStun(true);
+  ASSERT_TRUE_WAIT(p1_->ice_connected() == 0, kDefaultTimeout);
+  PR_Sleep(1500);
+  ASSERT_TRUE(p1_->ice_connected() == 0);
+  p1_->SetBlockStun(false);
+  p1_->ChangeNetworkStateToOnline();
+  ASSERT_TRUE_WAIT(p1_->ice_connected(), 500);
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectTurn) {
+  if (turn_server_.empty()) return;
+
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  SetTurnServer(turn_server_, kDefaultStunServerPort, turn_user_,
+                turn_password_);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  Connect();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectTurnWithDelay) {
+  if (turn_server_.empty()) return;
+
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  SetTurnServer(turn_server_, kDefaultStunServerPort, turn_user_,
+                turn_password_);
+  SetCandidateFilter(SabotageHostCandidateAndDropReflexive);
+  AddStream(1);
+  p1_->Gather();
+  PR_Sleep(500);
+  p2_->Gather();
+  ConnectTrickle(TRICKLE_REAL);
+  WaitForGather();
+  WaitForConnectedStreams();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectTurnWithNormalTrickleDelay) {
+  if (turn_server_.empty()) return;
+
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  SetTurnServer(turn_server_, kDefaultStunServerPort, turn_user_,
+                turn_password_);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  ConnectTrickle();
+  RealisticTrickleDelay(p1_->ControlTrickle(0));
+  RealisticTrickleDelay(p2_->ControlTrickle(0));
+
+  WaitForConnected();
+  AssertCheckingReached();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectTurnWithNormalTrickleDelayOneSided) {
+  if (turn_server_.empty()) return;
+
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  SetTurnServer(turn_server_, kDefaultStunServerPort, turn_user_,
+                turn_password_);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  ConnectTrickle();
+  RealisticTrickleDelay(p1_->ControlTrickle(0));
+  p2_->SimulateTrickle(0);
+
+  WaitForConnected();
+  AssertCheckingReached();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectTurnWithLargeTrickleDelay) {
+  if (turn_server_.empty()) return;
+
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  SetTurnServer(turn_server_, kDefaultStunServerPort, turn_user_,
+                turn_password_);
+  SetCandidateFilter(SabotageHostCandidateAndDropReflexive);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  ConnectTrickle();
+  // Trickle host candidates immediately, but delay relay candidates
+  DelayRelayCandidates(p1_->ControlTrickle(0), 3700);
+  DelayRelayCandidates(p2_->ControlTrickle(0), 3700);
+
+  WaitForConnected();
+  AssertCheckingReached();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectTurnTcp) {
+  if (turn_server_.empty()) return;
+
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  SetTurnServer(turn_server_, kDefaultStunServerPort, turn_user_,
+                turn_password_, kNrIceTransportTcp);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  Connect();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectTurnOnly) {
+  if (turn_server_.empty()) return;
+
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  SetTurnServer(turn_server_, kDefaultStunServerPort, turn_user_,
+                turn_password_);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  SetCandidateFilter(IsRelayCandidate);
+  SetExpectedTypes(NrIceCandidate::Type::ICE_RELAYED,
+                   NrIceCandidate::Type::ICE_RELAYED);
+  Connect();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectTurnTcpOnly) {
+  if (turn_server_.empty()) return;
+
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  SetTurnServer(turn_server_, kDefaultStunServerPort, turn_user_,
+                turn_password_, kNrIceTransportTcp);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  SetCandidateFilter(IsRelayCandidate);
+  SetExpectedTypes(NrIceCandidate::Type::ICE_RELAYED,
+                   NrIceCandidate::Type::ICE_RELAYED, kNrIceTransportTcp);
+  Connect();
+}
+
+TEST_F(WebRtcIceConnectTest, TestSendReceiveTurnOnly) {
+  if (turn_server_.empty()) return;
+
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  SetTurnServer(turn_server_, kDefaultStunServerPort, turn_user_,
+                turn_password_);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  SetCandidateFilter(IsRelayCandidate);
+  SetExpectedTypes(NrIceCandidate::Type::ICE_RELAYED,
+                   NrIceCandidate::Type::ICE_RELAYED);
+  Connect();
+  SendReceive();
+}
+
+TEST_F(WebRtcIceConnectTest, TestSendReceiveTurnTcpOnly) {
+  if (turn_server_.empty()) return;
+
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  SetTurnServer(turn_server_, kDefaultStunServerPort, turn_user_,
+                turn_password_, kNrIceTransportTcp);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  SetCandidateFilter(IsRelayCandidate);
+  SetExpectedTypes(NrIceCandidate::Type::ICE_RELAYED,
+                   NrIceCandidate::Type::ICE_RELAYED, kNrIceTransportTcp);
+  Connect();
+  SendReceive();
+}
+
+TEST_F(WebRtcIceConnectTest, TestSendReceiveTurnBothOnly) {
+  if (turn_server_.empty()) return;
+
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  std::vector<NrIceTurnServer> turn_servers;
+  std::vector<unsigned char> password_vec(turn_password_.begin(),
+                                          turn_password_.end());
+  turn_servers.push_back(
+      *NrIceTurnServer::Create(turn_server_, kDefaultStunServerPort, turn_user_,
+                               password_vec, kNrIceTransportTcp));
+  turn_servers.push_back(
+      *NrIceTurnServer::Create(turn_server_, kDefaultStunServerPort, turn_user_,
+                               password_vec, kNrIceTransportUdp));
+  SetTurnServers(turn_servers);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  SetCandidateFilter(IsRelayCandidate);
+  // UDP is preferred.
+  SetExpectedTypes(NrIceCandidate::Type::ICE_RELAYED,
+                   NrIceCandidate::Type::ICE_RELAYED, kNrIceTransportUdp);
+  Connect();
+  SendReceive();
+}
+
+TEST_F(WebRtcIceConnectTest, TestConnectShutdownOneSide) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  ConnectThenDelete();
+}
+
+TEST_F(WebRtcIceConnectTest, TestPollCandPairsBeforeConnect) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+
+  std::vector<NrIceCandidatePair> pairs;
+  nsresult res = p1_->GetCandidatePairs(0, &pairs);
+  // There should be no candidate pairs prior to calling Connect()
+  ASSERT_EQ(NS_OK, res);
+  ASSERT_EQ(0U, pairs.size());
+
+  res = p2_->GetCandidatePairs(0, &pairs);
+  ASSERT_EQ(NS_OK, res);
+  ASSERT_EQ(0U, pairs.size());
+}
+
+TEST_F(WebRtcIceConnectTest, TestPollCandPairsAfterConnect) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  Connect();
+
+  std::vector<NrIceCandidatePair> pairs;
+  nsresult r = p1_->GetCandidatePairs(0, &pairs);
+  ASSERT_EQ(NS_OK, r);
+  // How detailed of a check do we want to do here? If the turn server is
+  // functioning, we'll get at least two pairs, but this is probably not
+  // something we should assume.
+  ASSERT_NE(0U, pairs.size());
+  ASSERT_TRUE(p1_->CandidatePairsPriorityDescending(pairs));
+  ASSERT_TRUE(ContainsSucceededPair(pairs));
+  pairs.clear();
+
+  r = p2_->GetCandidatePairs(0, &pairs);
+  ASSERT_EQ(NS_OK, r);
+  ASSERT_NE(0U, pairs.size());
+  ASSERT_TRUE(p2_->CandidatePairsPriorityDescending(pairs));
+  ASSERT_TRUE(ContainsSucceededPair(pairs));
+}
+
+// TODO Bug 1259842 - disabled until we find a better way to handle two
+// candidates from different RFC1918 ranges
+TEST_F(WebRtcIceConnectTest, DISABLED_TestHostCandPairingFilter) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  Init(false);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  SetCandidateFilter(IsIpv4Candidate);
+
+  int host_net = p1_->GetCandidatesPrivateIpv4Range(0);
+  if (host_net <= 0) {
+    // TODO bug 1226838: make this work with multiple private IPs
+    FAIL() << "This test needs exactly one private IPv4 host candidate to work"
+           << std::endl;
+  }
+
+  ConnectTrickle();
+  AddNonPairableCandidates(p1_->ControlTrickle(0), p1_.get(), 0, host_net,
+                           test_utils_);
+  AddNonPairableCandidates(p2_->ControlTrickle(0), p2_.get(), 0, host_net,
+                           test_utils_);
+
+  std::vector<NrIceCandidatePair> pairs;
+  p1_->GetCandidatePairs(0, &pairs);
+  for (auto p : pairs) {
+    std::cerr << "Verifying pair:" << std::endl;
+    p1_->DumpCandidatePair(p);
+    nr_transport_addr addr;
+    nr_str_port_to_transport_addr(p.local.local_addr.host.c_str(), 0,
+                                  IPPROTO_UDP, &addr);
+    ASSERT_TRUE(nr_transport_addr_get_private_addr_range(&addr) == host_net);
+    nr_str_port_to_transport_addr(p.remote.cand_addr.host.c_str(), 0,
+                                  IPPROTO_UDP, &addr);
+    ASSERT_TRUE(nr_transport_addr_get_private_addr_range(&addr) == host_net);
+  }
+}
+
+// TODO Bug 1226838 - See Comment 2 - this test can't work as written
+TEST_F(WebRtcIceConnectTest, DISABLED_TestSrflxCandPairingFilter) {
+  if (stun_server_address_.empty()) {
+    return;
+  }
+
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  Init(false);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+  SetCandidateFilter(IsSrflxCandidate);
+
+  if (p1_->GetCandidatesPrivateIpv4Range(0) <= 0) {
+    // TODO bug 1226838: make this work with public IP addresses
+    std::cerr << "Don't run this test at IETF meetings!" << std::endl;
+    FAIL() << "This test needs one private IPv4 host candidate to work"
+           << std::endl;
+  }
+
+  ConnectTrickle();
+  SimulateTrickleP1(0);
+  SimulateTrickleP2(0);
+
+  std::vector<NrIceCandidatePair> pairs;
+  p1_->GetCandidatePairs(0, &pairs);
+  for (auto p : pairs) {
+    std::cerr << "Verifying P1 pair:" << std::endl;
+    p1_->DumpCandidatePair(p);
+    nr_transport_addr addr;
+    nr_str_port_to_transport_addr(p.local.local_addr.host.c_str(), 0,
+                                  IPPROTO_UDP, &addr);
+    ASSERT_TRUE(nr_transport_addr_get_private_addr_range(&addr) != 0);
+    nr_str_port_to_transport_addr(p.remote.cand_addr.host.c_str(), 0,
+                                  IPPROTO_UDP, &addr);
+    ASSERT_TRUE(nr_transport_addr_get_private_addr_range(&addr) == 0);
+  }
+  p2_->GetCandidatePairs(0, &pairs);
+  for (auto p : pairs) {
+    std::cerr << "Verifying P2 pair:" << std::endl;
+    p2_->DumpCandidatePair(p);
+    nr_transport_addr addr;
+    nr_str_port_to_transport_addr(p.local.local_addr.host.c_str(), 0,
+                                  IPPROTO_UDP, &addr);
+    ASSERT_TRUE(nr_transport_addr_get_private_addr_range(&addr) != 0);
+    nr_str_port_to_transport_addr(p.remote.cand_addr.host.c_str(), 0,
+                                  IPPROTO_UDP, &addr);
+    ASSERT_TRUE(nr_transport_addr_get_private_addr_range(&addr) == 0);
+  }
+}
+
+TEST_F(WebRtcIceConnectTest, TestPollCandPairsDuringConnect) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+
+  p2_->Connect(p1_.get(), TRICKLE_NONE, false);
+  p1_->Connect(p2_.get(), TRICKLE_NONE, false);
+
+  std::vector<NrIceCandidatePair> pairs1;
+  std::vector<NrIceCandidatePair> pairs2;
+
+  p1_->StartChecks();
+  p1_->UpdateAndValidateCandidatePairs(0, &pairs1);
+  p2_->UpdateAndValidateCandidatePairs(0, &pairs2);
+
+  p2_->StartChecks();
+  p1_->UpdateAndValidateCandidatePairs(0, &pairs1);
+  p2_->UpdateAndValidateCandidatePairs(0, &pairs2);
+
+  WaitForConnectedStreams();
+  p1_->UpdateAndValidateCandidatePairs(0, &pairs1);
+  p2_->UpdateAndValidateCandidatePairs(0, &pairs2);
+  ASSERT_TRUE(ContainsSucceededPair(pairs1));
+  ASSERT_TRUE(ContainsSucceededPair(pairs2));
+}
+
+TEST_F(WebRtcIceConnectTest, TestRLogConnector) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  ASSERT_TRUE(Gather());
+
+  p2_->Connect(p1_.get(), TRICKLE_NONE, false);
+  p1_->Connect(p2_.get(), TRICKLE_NONE, false);
+
+  std::vector<NrIceCandidatePair> pairs1;
+  std::vector<NrIceCandidatePair> pairs2;
+
+  p1_->StartChecks();
+  p1_->UpdateAndValidateCandidatePairs(0, &pairs1);
+  p2_->UpdateAndValidateCandidatePairs(0, &pairs2);
+
+  p2_->StartChecks();
+  p1_->UpdateAndValidateCandidatePairs(0, &pairs1);
+  p2_->UpdateAndValidateCandidatePairs(0, &pairs2);
+
+  WaitForConnectedStreams();
+  p1_->UpdateAndValidateCandidatePairs(0, &pairs1);
+  p2_->UpdateAndValidateCandidatePairs(0, &pairs2);
+  ASSERT_TRUE(ContainsSucceededPair(pairs1));
+  ASSERT_TRUE(ContainsSucceededPair(pairs2));
+
+  for (auto& p : pairs1) {
+    std::deque<std::string> logs;
+    std::string substring("CAND-PAIR(");
+    substring += p.codeword;
+    RLogConnector::GetInstance()->Filter(substring, 0, &logs);
+    ASSERT_NE(0U, logs.size());
+  }
+
+  for (auto& p : pairs2) {
+    std::deque<std::string> logs;
+    std::string substring("CAND-PAIR(");
+    substring += p.codeword;
+    RLogConnector::GetInstance()->Filter(substring, 0, &logs);
+    ASSERT_NE(0U, logs.size());
+  }
+}
+
+// Verify that a bogus candidate doesn't cause crashes on the
+// main thread. See bug 856433.
+TEST_F(WebRtcIceConnectTest, TestBogusCandidate) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  Gather();
+  ConnectTrickle();
+  p1_->ParseCandidate(0, kBogusIceCandidate, "");
+
+  std::vector<NrIceCandidatePair> pairs;
+  nsresult res = p1_->GetCandidatePairs(0, &pairs);
+  ASSERT_EQ(NS_OK, res);
+  ASSERT_EQ(0U, pairs.size());
+}
+
+TEST_F(WebRtcIceConnectTest, TestNonMDNSCandidate) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  Gather();
+  ConnectTrickle();
+  p1_->ParseCandidate(0, kUnreachableHostIceCandidate, "");
+
+  std::vector<NrIceCandidatePair> pairs;
+  nsresult res = p1_->GetCandidatePairs(0, &pairs);
+  ASSERT_EQ(NS_OK, res);
+  ASSERT_EQ(1U, pairs.size());
+  ASSERT_EQ(pairs[0].remote.mdns_addr, "");
+}
+
+TEST_F(WebRtcIceConnectTest, TestMDNSCandidate) {
+  NrIceCtx::GlobalConfig config;
+  config.mTcpEnabled = false;
+  NrIceCtx::InitializeGlobals(config);
+  AddStream(1);
+  Gather();
+  ConnectTrickle();
+  p1_->ParseCandidate(0, kUnreachableHostIceCandidate, "host.local");
+
+  std::vector<NrIceCandidatePair> pairs;
+  nsresult res = p1_->GetCandidatePairs(0, &pairs);
+  ASSERT_EQ(NS_OK, res);
+  ASSERT_EQ(1U, pairs.size());
+  ASSERT_EQ(pairs[0].remote.mdns_addr, "host.local");
+}
+
+TEST_F(WebRtcIcePrioritizerTest, TestPrioritizer) {
+  SetPriorizer(::mozilla::CreateInterfacePrioritizer());
+
+  AddInterface("0", NR_INTERFACE_TYPE_VPN, 100);  // unknown vpn
+  AddInterface("1", NR_INTERFACE_TYPE_VPN | NR_INTERFACE_TYPE_WIRED,
+               100);  // wired vpn
+  AddInterface("2", NR_INTERFACE_TYPE_VPN | NR_INTERFACE_TYPE_WIFI,
+               100);  // wifi vpn
+  AddInterface("3", NR_INTERFACE_TYPE_VPN | NR_INTERFACE_TYPE_MOBILE,
+               100);                                    // wifi vpn
+  AddInterface("4", NR_INTERFACE_TYPE_WIRED, 1000);     // wired, high speed
+  AddInterface("5", NR_INTERFACE_TYPE_WIRED, 10);       // wired, low speed
+  AddInterface("6", NR_INTERFACE_TYPE_WIFI, 10);        // wifi, low speed
+  AddInterface("7", NR_INTERFACE_TYPE_WIFI, 1000);      // wifi, high speed
+  AddInterface("8", NR_INTERFACE_TYPE_MOBILE, 10);      // mobile, low speed
+  AddInterface("9", NR_INTERFACE_TYPE_MOBILE, 1000);    // mobile, high speed
+  AddInterface("10", NR_INTERFACE_TYPE_UNKNOWN, 10);    // unknown, low speed
+  AddInterface("11", NR_INTERFACE_TYPE_UNKNOWN, 1000);  // unknown, high speed
+
+  // expected preference "4" > "5" > "1" > "7" > "6" > "2" > "9" > "8" > "3" >
+  // "11" > "10" > "0"
+
+  HasLowerPreference("0", "10");
+  HasLowerPreference("10", "11");
+  HasLowerPreference("11", "3");
+  HasLowerPreference("3", "8");
+  HasLowerPreference("8", "9");
+  HasLowerPreference("9", "2");
+  HasLowerPreference("2", "6");
+  HasLowerPreference("6", "7");
+  HasLowerPreference("7", "1");
+  HasLowerPreference("1", "5");
+  HasLowerPreference("5", "4");
+}
+
+TEST_F(WebRtcIcePacketFilterTest, TestSendNonStunPacket) {
+  const unsigned char data[] = "12345abcde";
+  TestOutgoing(data, sizeof(data), 123, 45, false);
+  TestOutgoingTcp(data, sizeof(data), false);
+}
+
+TEST_F(WebRtcIcePacketFilterTest, TestRecvNonStunPacket) {
+  const unsigned char data[] = "12345abcde";
+  TestIncoming(data, sizeof(data), 123, 45, false);
+  TestIncomingTcp(data, sizeof(data), true);
+}
+
+TEST_F(WebRtcIcePacketFilterTest, TestSendStunPacket) {
+  nr_stun_message* msg;
+  ASSERT_EQ(0, nr_stun_build_req_no_auth(nullptr, &msg));
+  msg->header.type = NR_STUN_MSG_BINDING_REQUEST;
+  ASSERT_EQ(0, nr_stun_encode_message(msg));
+  TestOutgoing(msg->buffer, msg->length, 123, 45, true);
+  TestOutgoingTcp(msg->buffer, msg->length, true);
+  TestOutgoingTcpFramed(msg->buffer, msg->length, true);
+  ASSERT_EQ(0, nr_stun_message_destroy(&msg));
+}
+
+TEST_F(WebRtcIcePacketFilterTest, TestRecvStunPacketWithoutAPendingId) {
+  nr_stun_message* msg;
+  ASSERT_EQ(0, nr_stun_build_req_no_auth(nullptr, &msg));
+
+  msg->header.id.octet[0] = 1;
+  msg->header.type = NR_STUN_MSG_BINDING_REQUEST;
+  ASSERT_EQ(0, nr_stun_encode_message(msg));
+  TestOutgoing(msg->buffer, msg->length, 123, 45, true);
+  TestOutgoingTcp(msg->buffer, msg->length, true);
+
+  msg->header.id.octet[0] = 0;
+  msg->header.type = NR_STUN_MSG_BINDING_RESPONSE;
+  ASSERT_EQ(0, nr_stun_encode_message(msg));
+  TestIncoming(msg->buffer, msg->length, 123, 45, true);
+  TestIncomingTcp(msg->buffer, msg->length, true);
+
+  ASSERT_EQ(0, nr_stun_message_destroy(&msg));
+}
+
+TEST_F(WebRtcIcePacketFilterTest, TestRecvStunBindingRequestWithoutAPendingId) {
+  nr_stun_message* msg;
+  ASSERT_EQ(0, nr_stun_build_req_no_auth(nullptr, &msg));
+
+  msg->header.id.octet[0] = 1;
+  msg->header.type = NR_STUN_MSG_BINDING_REQUEST;
+  ASSERT_EQ(0, nr_stun_encode_message(msg));
+  TestIncoming(msg->buffer, msg->length, 123, 45, true);
+  TestIncomingTcp(msg->buffer, msg->length, true);
+
+  msg->header.id.octet[0] = 1;
+  msg->header.type = NR_STUN_MSG_BINDING_RESPONSE;
+  ASSERT_EQ(0, nr_stun_encode_message(msg));
+  TestOutgoing(msg->buffer, msg->length, 123, 45, true);
+  TestOutgoingTcp(msg->buffer, msg->length, true);
+
+  ASSERT_EQ(0, nr_stun_message_destroy(&msg));
+}
+
+TEST_F(WebRtcIcePacketFilterTest,
+       TestRecvStunPacketWithoutAPendingIdTcpFramed) {
+  nr_stun_message* msg;
+  ASSERT_EQ(0, nr_stun_build_req_no_auth(nullptr, &msg));
+
+  msg->header.id.octet[0] = 1;
+  msg->header.type = NR_STUN_MSG_BINDING_REQUEST;
+  ASSERT_EQ(0, nr_stun_encode_message(msg));
+  TestOutgoingTcpFramed(msg->buffer, msg->length, true);
+
+  msg->header.id.octet[0] = 0;
+  msg->header.type = NR_STUN_MSG_BINDING_RESPONSE;
+  ASSERT_EQ(0, nr_stun_encode_message(msg));
+  TestIncomingTcpFramed(msg->buffer, msg->length, true);
+
+  ASSERT_EQ(0, nr_stun_message_destroy(&msg));
+}
+
+TEST_F(WebRtcIcePacketFilterTest, TestRecvStunPacketWithoutAPendingAddress) {
+  nr_stun_message* msg;
+  ASSERT_EQ(0, nr_stun_build_req_no_auth(nullptr, &msg));
+
+  msg->header.type = NR_STUN_MSG_BINDING_REQUEST;
+  ASSERT_EQ(0, nr_stun_encode_message(msg));
+  TestOutgoing(msg->buffer, msg->length, 123, 45, true);
+  // nothing to test here for the TCP filter
+
+  msg->header.type = NR_STUN_MSG_BINDING_RESPONSE;
+  ASSERT_EQ(0, nr_stun_encode_message(msg));
+  TestIncoming(msg->buffer, msg->length, 123, 46, false);
+  TestIncoming(msg->buffer, msg->length, 124, 45, false);
+
+  ASSERT_EQ(0, nr_stun_message_destroy(&msg));
+}
+
+TEST_F(WebRtcIcePacketFilterTest, TestRecvStunPacketWithPendingIdAndAddress) {
+  nr_stun_message* msg;
+  ASSERT_EQ(0, nr_stun_build_req_no_auth(nullptr, &msg));
+
+  msg->header.type = NR_STUN_MSG_BINDING_REQUEST;
+  ASSERT_EQ(0, nr_stun_encode_message(msg));
+  TestOutgoing(msg->buffer, msg->length, 123, 45, true);
+  TestOutgoingTcp(msg->buffer, msg->length, true);
+
+  msg->header.type = NR_STUN_MSG_BINDING_RESPONSE;
+  ASSERT_EQ(0, nr_stun_encode_message(msg));
+  TestIncoming(msg->buffer, msg->length, 123, 45, true);
+  TestIncomingTcp(msg->buffer, msg->length, true);
+
+  // Test whitelist by filtering non-stun packets.
+  const unsigned char data[] = "12345abcde";
+
+  // 123:45 is white-listed.
+  TestOutgoing(data, sizeof(data), 123, 45, true);
+  TestOutgoingTcp(data, sizeof(data), true);
+  TestIncoming(data, sizeof(data), 123, 45, true);
+  TestIncomingTcp(data, sizeof(data), true);
+
+  // Indications pass as well.
+  msg->header.type = NR_STUN_MSG_BINDING_INDICATION;
+  ASSERT_EQ(0, nr_stun_encode_message(msg));
+  TestOutgoing(msg->buffer, msg->length, 123, 45, true);
+  TestOutgoingTcp(msg->buffer, msg->length, true);
+  TestIncoming(msg->buffer, msg->length, 123, 45, true);
+  TestIncomingTcp(msg->buffer, msg->length, true);
+
+  // Packets from and to other address are still disallowed.
+  // Note: this doesn't apply for TCP connections
+  TestOutgoing(data, sizeof(data), 123, 46, false);
+  TestIncoming(data, sizeof(data), 123, 46, false);
+  TestOutgoing(data, sizeof(data), 124, 45, false);
+  TestIncoming(data, sizeof(data), 124, 45, false);
+
+  ASSERT_EQ(0, nr_stun_message_destroy(&msg));
+}
+
+TEST_F(WebRtcIcePacketFilterTest, TestRecvStunPacketWithPendingIdTcpFramed) {
+  nr_stun_message* msg;
+  ASSERT_EQ(0, nr_stun_build_req_no_auth(nullptr, &msg));
+
+  msg->header.type = NR_STUN_MSG_BINDING_REQUEST;
+  ASSERT_EQ(0, nr_stun_encode_message(msg));
+  TestOutgoingTcpFramed(msg->buffer, msg->length, true);
+
+  msg->header.type = NR_STUN_MSG_BINDING_RESPONSE;
+  ASSERT_EQ(0, nr_stun_encode_message(msg));
+  TestIncomingTcpFramed(msg->buffer, msg->length, true);
+
+  // Test whitelist by filtering non-stun packets.
+  const unsigned char data[] = "12345abcde";
+
+  TestOutgoingTcpFramed(data, sizeof(data), true);
+  TestIncomingTcpFramed(data, sizeof(data), true);
+
+  ASSERT_EQ(0, nr_stun_message_destroy(&msg));
+}
+
+TEST_F(WebRtcIcePacketFilterTest, TestSendNonRequestStunPacket) {
+  nr_stun_message* msg;
+  ASSERT_EQ(0, nr_stun_build_req_no_auth(nullptr, &msg));
+
+  msg->header.type = NR_STUN_MSG_BINDING_RESPONSE;
+  ASSERT_EQ(0, nr_stun_encode_message(msg));
+  TestOutgoing(msg->buffer, msg->length, 123, 45, false);
+  TestOutgoingTcp(msg->buffer, msg->length, false);
+
+  // Send a packet so we allow the incoming request.
+  msg->header.type = NR_STUN_MSG_BINDING_REQUEST;
+  ASSERT_EQ(0, nr_stun_encode_message(msg));
+  TestOutgoing(msg->buffer, msg->length, 123, 45, true);
+  TestOutgoingTcp(msg->buffer, msg->length, true);
+
+  // This packet makes us able to send a response.
+  msg->header.type = NR_STUN_MSG_BINDING_REQUEST;
+  ASSERT_EQ(0, nr_stun_encode_message(msg));
+  TestIncoming(msg->buffer, msg->length, 123, 45, true);
+  TestIncomingTcp(msg->buffer, msg->length, true);
+
+  msg->header.type = NR_STUN_MSG_BINDING_RESPONSE;
+  ASSERT_EQ(0, nr_stun_encode_message(msg));
+  TestOutgoing(msg->buffer, msg->length, 123, 45, true);
+  TestOutgoingTcp(msg->buffer, msg->length, true);
+
+  ASSERT_EQ(0, nr_stun_message_destroy(&msg));
+}
+
+TEST_F(WebRtcIcePacketFilterTest, TestRecvDataPacketWithAPendingAddress) {
+  nr_stun_message* msg;
+  ASSERT_EQ(0, nr_stun_build_req_no_auth(nullptr, &msg));
+
+  msg->header.type = NR_STUN_MSG_BINDING_REQUEST;
+  ASSERT_EQ(0, nr_stun_encode_message(msg));
+  TestOutgoing(msg->buffer, msg->length, 123, 45, true);
+  TestOutgoingTcp(msg->buffer, msg->length, true);
+
+  const unsigned char data[] = "12345abcde";
+  TestIncoming(data, sizeof(data), 123, 45, true);
+  TestIncomingTcp(data, sizeof(data), true);
+
+  ASSERT_EQ(0, nr_stun_message_destroy(&msg));
+}
+
+TEST(WebRtcIceInternalsTest, TestAddBogusAttribute)
+{
+  nr_stun_message* req;
+  ASSERT_EQ(0, nr_stun_message_create(&req));
+  Data* data;
+  ASSERT_EQ(0, r_data_alloc(&data, 3000));
+  memset(data->data, 'A', data->len);
+  ASSERT_TRUE(nr_stun_message_add_message_integrity_attribute(req, data));
+  ASSERT_EQ(0, r_data_destroy(&data));
+  ASSERT_EQ(0, nr_stun_message_destroy(&req));
+}
diff --git a/dom/media/webrtc/transport/test/moz.build b/dom/media/webrtc/transport/test/moz.build
new file mode 100644
index 0000000000..69d3a587a5
--- /dev/null
+++ b/dom/media/webrtc/transport/test/moz.build
@@ -0,0 +1,104 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+include("/ipc/chromium/chromium-config.mozbuild")
+
+if CONFIG["OS_TARGET"] != "WINNT":
+
+    if CONFIG["OS_TARGET"] != "Android":
+        SOURCES += [
+            "ice_unittest.cpp",
+        ]
+
+    SOURCES += [
+        "buffered_stun_socket_unittest.cpp",
+        "multi_tcp_socket_unittest.cpp",
+        "nrappkit_unittest.cpp",
+        "proxy_tunnel_socket_unittest.cpp",
+        "rlogconnector_unittest.cpp",
+        "runnable_utils_unittest.cpp",
+        "simpletokenbucket_unittest.cpp",
+        "sockettransportservice_unittest.cpp",
+        "stunserver.cpp",
+        "test_nr_socket_ice_unittest.cpp",
+        "test_nr_socket_unittest.cpp",
+        "TestSyncRunnable.cpp",
+        "transport_unittests.cpp",
+        "turn_unittest.cpp",
+        "webrtcproxychannel_unittest.cpp",
+    ]
+
+    if CONFIG["MOZ_SCTP"]:
+        SOURCES += [
+            "sctp_unittest.cpp",
+        ]
+
+
+for var in ("HAVE_STRDUP", "NR_SOCKET_IS_VOID_PTR", "SCTP_DEBUG"):
+    DEFINES[var] = True
+
+if CONFIG["OS_TARGET"] == "Android":
+    DEFINES["LINUX"] = True
+    DEFINES["ANDROID"] = True
+    LOCAL_INCLUDES += [
+        "/dom/media/webrtc/transport/third_party/nrappkit/src/port/android/include",
+    ]
+
+if CONFIG["OS_TARGET"] == "Linux":
+    DEFINES["LINUX"] = True
+    LOCAL_INCLUDES += [
+        "/dom/media/webrtc/transport/third_party/nrappkit/src/port/linux/include",
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+    LOCAL_INCLUDES += [
+        "/dom/media/webrtc/transport/third_party/nrappkit/src/port/darwin/include",
+    ]
+
+if CONFIG["OS_TARGET"] in ("DragonFly", "FreeBSD", "NetBSD", "OpenBSD"):
+    if CONFIG["OS_TARGET"] == "Darwin":
+        DEFINES["DARWIN"] = True
+    else:
+        DEFINES["BSD"] = True
+    LOCAL_INCLUDES += [
+        "/dom/media/webrtc/transport/third_party/nrappkit/src/port/darwin/include",
+    ]
+
+# SCTP DEFINES
+if CONFIG["OS_TARGET"] == "WINNT":
+    DEFINES["WIN"] = True
+    # for stun.h
+    DEFINES["WIN32"] = True
+    DEFINES["__Userspace_os_Windows"] = 1
+else:
+    # Works for Darwin, Linux, Android. Probably doesn't work for others.
+    DEFINES["__Userspace_os_%s" % CONFIG["OS_TARGET"]] = 1
+
+if CONFIG["OS_TARGET"] in ("Darwin", "Android"):
+    DEFINES["GTEST_USE_OWN_TR1_TUPLE"] = 1
+
+LOCAL_INCLUDES += [
+    "/dom/media/webrtc/transport/",
+    "/dom/media/webrtc/transport/third_party/",
+    "/dom/media/webrtc/transport/third_party/nICEr/src/crypto",
+    "/dom/media/webrtc/transport/third_party/nICEr/src/ice",
+    "/dom/media/webrtc/transport/third_party/nICEr/src/net",
+    "/dom/media/webrtc/transport/third_party/nICEr/src/stun",
+    "/dom/media/webrtc/transport/third_party/nICEr/src/util",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/event",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/log",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/plugin",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/port/generic/include",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/registry",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/share",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/stats",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/util/",
+    "/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr",
+    "/netwerk/sctp/src/",
+    "/xpcom/tests/",
+]
+
+FINAL_LIBRARY = "xul-gtest"
diff --git a/dom/media/webrtc/transport/test/mtransport_test_utils.h b/dom/media/webrtc/transport/test/mtransport_test_utils.h
new file mode 100644
index 0000000000..04031c0dc2
--- /dev/null
+++ b/dom/media/webrtc/transport/test/mtransport_test_utils.h
@@ -0,0 +1,57 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+#ifndef mtransport_test_utils_h__
+#define mtransport_test_utils_h__
+
+#include "nsCOMPtr.h"
+#include "nsNetCID.h"
+
+#include "nsISerialEventTarget.h"
+#include "nsPISocketTransportService.h"
+#include "nsServiceManagerUtils.h"
+#include "nsThreadUtils.h"
+
+class MtransportTestUtils {
+ public:
+  MtransportTestUtils() { InitServices(); }
+
+  ~MtransportTestUtils() = default;
+
+  void InitServices() {
+    nsresult rv;
+    sts_target_ = do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &rv);
+    MOZ_ASSERT(NS_SUCCEEDED(rv));
+    sts_ = do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &rv);
+    MOZ_ASSERT(NS_SUCCEEDED(rv));
+  }
+
+  nsISerialEventTarget* sts_target() { return sts_target_; }
+
+  nsresult SyncDispatchToSTS(nsIRunnable* aRunnable) {
+    return SyncDispatchToSTS(do_AddRef(aRunnable));
+  }
+  nsresult SyncDispatchToSTS(already_AddRefed<nsIRunnable>&& aRunnable) {
+    return NS_DispatchAndSpinEventLoopUntilComplete(
+        "MtransportTestUtils::SyncDispatchToSts"_ns, sts_target_,
+        std::move(aRunnable));
+  }
+
+ private:
+  nsCOMPtr<nsISerialEventTarget> sts_target_;
+  nsCOMPtr<nsPISocketTransportService> sts_;
+};
+
+#define CHECK_ENVIRONMENT_FLAG(envname)                                 \
+  char* test_flag = getenv(envname);                                    \
+  if (!test_flag || strcmp(test_flag, "1")) {                           \
+    printf("To run this test set %s=1 in your environment\n", envname); \
+    exit(0);                                                            \
+  }
+
+#endif
diff --git a/dom/media/webrtc/transport/test/multi_tcp_socket_unittest.cpp b/dom/media/webrtc/transport/test/multi_tcp_socket_unittest.cpp
new file mode 100644
index 0000000000..d0c3ae6e53
--- /dev/null
+++ b/dom/media/webrtc/transport/test/multi_tcp_socket_unittest.cpp
@@ -0,0 +1,501 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include <iostream>
+#include <vector>
+
+#include "mozilla/Atomics.h"
+#include "runnable_utils.h"
+#include "pk11pub.h"
+
+extern "C" {
+#include "nr_api.h"
+#include "nr_socket.h"
+#include "transport_addr.h"
+#include "nr_socket_multi_tcp.h"
+}
+
+#include "stunserver.h"
+
+#include "nricectx.h"
+
+#define GTEST_HAS_RTTI 0
+#include "gtest/gtest.h"
+#include "gtest_utils.h"
+
+using namespace mozilla;
+
+namespace {
+
+class MultiTcpSocketTest : public MtransportTest {
+ public:
+  MultiTcpSocketTest()
+      : MtransportTest(), socks(3, nullptr), readable(false), ice_ctx_() {}
+
+  void SetUp() {
+    MtransportTest::SetUp();
+
+    NrIceCtx::InitializeGlobals(NrIceCtx::GlobalConfig());
+    ice_ctx_ = NrIceCtx::Create("stun");
+
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnableNM(&TestStunTcpServer::GetInstance, AF_INET));
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnableNM(&TestStunTcpServer::GetInstance, AF_INET6));
+  }
+
+  void TearDown() {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &MultiTcpSocketTest::Shutdown_s));
+
+    MtransportTest::TearDown();
+  }
+
+  DISALLOW_COPY_ASSIGN(MultiTcpSocketTest);
+
+  static void SockReadable(NR_SOCKET s, int how, void* arg) {
+    MultiTcpSocketTest* obj = static_cast<MultiTcpSocketTest*>(arg);
+    obj->SetReadable(true);
+  }
+
+  void Shutdown_s() {
+    ice_ctx_ = nullptr;
+    for (auto& sock : socks) {
+      nr_socket_destroy(&sock);
+    }
+  }
+
+  static uint16_t GetRandomPort() {
+    uint16_t result;
+    if (PK11_GenerateRandom((unsigned char*)&result, 2) != SECSuccess) {
+      MOZ_ASSERT(false);
+      return 0;
+    }
+    return result;
+  }
+
+  static uint16_t EnsureEphemeral(uint16_t port) {
+    // IANA ephemeral port range (49152 to 65535)
+    return port | 49152;
+  }
+
+  void Create_s(nr_socket_tcp_type tcp_type, std::string stun_server_addr,
+                uint16_t stun_server_port, nr_socket** sock) {
+    nr_transport_addr local;
+    // Get start of port range for test
+    static unsigned short port_s = GetRandomPort();
+    int r;
+
+    if (!stun_server_addr.empty()) {
+      std::vector<NrIceStunServer> stun_servers;
+      UniquePtr<NrIceStunServer> server(NrIceStunServer::Create(
+          stun_server_addr, stun_server_port, kNrIceTransportTcp));
+      stun_servers.push_back(*server);
+
+      ASSERT_TRUE(NS_SUCCEEDED(ice_ctx_->SetStunServers(stun_servers)));
+    }
+
+    r = 1;
+    for (int tries = 10; tries && r; --tries) {
+      r = nr_str_port_to_transport_addr(
+          (char*)"127.0.0.1", EnsureEphemeral(port_s++), IPPROTO_TCP, &local);
+      ASSERT_EQ(0, r);
+
+      r = nr_socket_multi_tcp_create(ice_ctx_->ctx(), nullptr, &local, tcp_type,
+                                     1, 2048, sock);
+    }
+
+    ASSERT_EQ(0, r);
+    printf("Creating socket on %s\n", local.as_string);
+    r = nr_socket_multi_tcp_set_readable_cb(
+        *sock, &MultiTcpSocketTest::SockReadable, this);
+    ASSERT_EQ(0, r);
+  }
+
+  nr_socket* Create(nr_socket_tcp_type tcp_type,
+                    std::string stun_server_addr = "",
+                    uint16_t stun_server_port = 0) {
+    nr_socket* sock = nullptr;
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &MultiTcpSocketTest::Create_s, tcp_type,
+                     stun_server_addr, stun_server_port, &sock));
+    return sock;
+  }
+
+  void Listen_s(nr_socket* sock) {
+    nr_transport_addr addr;
+    int r = nr_socket_getaddr(sock, &addr);
+    ASSERT_EQ(0, r);
+    printf("Listening on %s\n", addr.as_string);
+    r = nr_socket_listen(sock, 5);
+    ASSERT_EQ(0, r);
+  }
+
+  void Listen(nr_socket* sock) {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &MultiTcpSocketTest::Listen_s, sock));
+  }
+
+  void Destroy_s(nr_socket* sock) {
+    int r = nr_socket_destroy(&sock);
+    ASSERT_EQ(0, r);
+  }
+
+  void Destroy(nr_socket* sock) {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &MultiTcpSocketTest::Destroy_s, sock));
+  }
+
+  void Connect_s(nr_socket* from, nr_socket* to) {
+    nr_transport_addr addr_to;
+    nr_transport_addr addr_from;
+    int r = nr_socket_getaddr(to, &addr_to);
+    ASSERT_EQ(0, r);
+    r = nr_socket_getaddr(from, &addr_from);
+    ASSERT_EQ(0, r);
+    printf("Connecting from %s to %s\n", addr_from.as_string,
+           addr_to.as_string);
+    r = nr_socket_connect(from, &addr_to);
+    ASSERT_EQ(0, r);
+  }
+
+  void Connect(nr_socket* from, nr_socket* to) {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &MultiTcpSocketTest::Connect_s, from, to));
+  }
+
+  void ConnectSo_s(nr_socket* so1, nr_socket* so2) {
+    nr_transport_addr addr_so1;
+    nr_transport_addr addr_so2;
+    int r = nr_socket_getaddr(so1, &addr_so1);
+    ASSERT_EQ(0, r);
+    r = nr_socket_getaddr(so2, &addr_so2);
+    ASSERT_EQ(0, r);
+    printf("Connecting SO %s <-> %s\n", addr_so1.as_string, addr_so2.as_string);
+    r = nr_socket_connect(so1, &addr_so2);
+    ASSERT_EQ(0, r);
+    r = nr_socket_connect(so2, &addr_so1);
+    ASSERT_EQ(0, r);
+  }
+
+  void ConnectSo(nr_socket* from, nr_socket* to) {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &MultiTcpSocketTest::ConnectSo_s, from, to));
+  }
+
+  void SendDataToAddress_s(nr_socket* from, nr_transport_addr* to,
+                           const char* data, size_t len) {
+    nr_transport_addr addr_from;
+
+    int r = nr_socket_getaddr(from, &addr_from);
+    ASSERT_EQ(0, r);
+    printf("Sending %lu bytes %s -> %s\n", (unsigned long)len,
+           addr_from.as_string, to->as_string);
+    r = nr_socket_sendto(from, data, len, 0, to);
+    ASSERT_EQ(0, r);
+  }
+
+  void SendData(nr_socket* from, nr_transport_addr* to, const char* data,
+                size_t len) {
+    test_utils_->SyncDispatchToSTS(WrapRunnable(
+        this, &MultiTcpSocketTest::SendDataToAddress_s, from, to, data, len));
+  }
+
+  void SendDataToSocket_s(nr_socket* from, nr_socket* to, const char* data,
+                          size_t len) {
+    nr_transport_addr addr_to;
+
+    int r = nr_socket_getaddr(to, &addr_to);
+    ASSERT_EQ(0, r);
+    SendDataToAddress_s(from, &addr_to, data, len);
+  }
+
+  void SendData(nr_socket* from, nr_socket* to, const char* data, size_t len) {
+    test_utils_->SyncDispatchToSTS(WrapRunnable(
+        this, &MultiTcpSocketTest::SendDataToSocket_s, from, to, data, len));
+  }
+
+  void RecvDataFromAddress_s(nr_transport_addr* expected_from,
+                             nr_socket* sent_to, const char* expected_data,
+                             size_t expected_len) {
+    SetReadable(false);
+    size_t buflen = expected_len ? expected_len + 1 : 100;
+    char received_data[buflen];
+    nr_transport_addr addr_to;
+    nr_transport_addr retaddr;
+    size_t retlen;
+
+    int r = nr_socket_getaddr(sent_to, &addr_to);
+    ASSERT_EQ(0, r);
+    printf("Receiving %lu bytes %s <- %s\n", (unsigned long)expected_len,
+           addr_to.as_string, expected_from->as_string);
+    r = nr_socket_recvfrom(sent_to, received_data, buflen, &retlen, 0,
+                           &retaddr);
+    ASSERT_EQ(0, r);
+    r = nr_transport_addr_cmp(&retaddr, expected_from,
+                              NR_TRANSPORT_ADDR_CMP_MODE_ALL);
+    ASSERT_EQ(0, r);
+    // expected_len == 0 means we just expected some data
+    if (expected_len == 0) {
+      ASSERT_GT(retlen, 0U);
+    } else {
+      ASSERT_EQ(expected_len, retlen);
+      r = memcmp(expected_data, received_data, retlen);
+      ASSERT_EQ(0, r);
+    }
+  }
+
+  void RecvData(nr_transport_addr* expected_from, nr_socket* sent_to,
+                const char* expected_data = nullptr, size_t expected_len = 0) {
+    ASSERT_TRUE_WAIT(IsReadable(), 1000);
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &MultiTcpSocketTest::RecvDataFromAddress_s,
+                     expected_from, sent_to, expected_data, expected_len));
+  }
+
+  void RecvDataFromSocket_s(nr_socket* expected_from, nr_socket* sent_to,
+                            const char* expected_data, size_t expected_len) {
+    nr_transport_addr addr_from;
+
+    int r = nr_socket_getaddr(expected_from, &addr_from);
+    ASSERT_EQ(0, r);
+
+    RecvDataFromAddress_s(&addr_from, sent_to, expected_data, expected_len);
+  }
+
+  void RecvData(nr_socket* expected_from, nr_socket* sent_to,
+                const char* expected_data, size_t expected_len) {
+    ASSERT_TRUE_WAIT(IsReadable(), 1000);
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &MultiTcpSocketTest::RecvDataFromSocket_s,
+                     expected_from, sent_to, expected_data, expected_len));
+  }
+
+  void RecvDataFailed_s(nr_socket* sent_to, size_t expected_len,
+                        int expected_err) {
+    SetReadable(false);
+    char received_data[expected_len + 1];
+    nr_transport_addr addr_to;
+    nr_transport_addr retaddr;
+    size_t retlen;
+
+    int r = nr_socket_getaddr(sent_to, &addr_to);
+    ASSERT_EQ(0, r);
+    r = nr_socket_recvfrom(sent_to, received_data, expected_len + 1, &retlen, 0,
+                           &retaddr);
+    ASSERT_EQ(expected_err, r) << "Expecting receive failure " << expected_err
+                               << " on " << addr_to.as_string;
+  }
+
+  void RecvDataFailed(nr_socket* sent_to, size_t expected_len,
+                      int expected_err) {
+    ASSERT_TRUE_WAIT(IsReadable(), 1000);
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &MultiTcpSocketTest::RecvDataFailed_s, sent_to,
+                     expected_len, expected_err));
+  }
+
+  void TransferData(nr_socket* from, nr_socket* to, const char* data,
+                    size_t len) {
+    SendData(from, to, data, len);
+    RecvData(from, to, data, len);
+  }
+
+ protected:
+  bool IsReadable() const { return readable; }
+  void SetReadable(bool r) { readable = r; }
+  std::vector<nr_socket*> socks;
+  Atomic<bool> readable;
+  RefPtr<NrIceCtx> ice_ctx_;
+};
+}  // namespace
+
+TEST_F(MultiTcpSocketTest, TestListen) {
+  socks[0] = Create(TCP_TYPE_PASSIVE);
+  Listen(socks[0]);
+}
+
+TEST_F(MultiTcpSocketTest, TestConnect) {
+  socks[0] = Create(TCP_TYPE_PASSIVE);
+  socks[1] = Create(TCP_TYPE_ACTIVE);
+  socks[2] = Create(TCP_TYPE_ACTIVE);
+  Listen(socks[0]);
+  Connect(socks[1], socks[0]);
+  Connect(socks[2], socks[0]);
+}
+
+TEST_F(MultiTcpSocketTest, TestTransmit) {
+  const char data[] = "TestTransmit";
+  socks[0] = Create(TCP_TYPE_ACTIVE);
+  socks[1] = Create(TCP_TYPE_PASSIVE);
+  Listen(socks[1]);
+  Connect(socks[0], socks[1]);
+
+  TransferData(socks[0], socks[1], data, sizeof(data));
+  TransferData(socks[1], socks[0], data, sizeof(data));
+}
+
+TEST_F(MultiTcpSocketTest, TestClosePassive) {
+  const char data[] = "TestClosePassive";
+  socks[0] = Create(TCP_TYPE_ACTIVE);
+  socks[1] = Create(TCP_TYPE_PASSIVE);
+  Listen(socks[1]);
+  Connect(socks[0], socks[1]);
+
+  TransferData(socks[0], socks[1], data, sizeof(data));
+  TransferData(socks[1], socks[0], data, sizeof(data));
+
+  /* We have to destroy as only that calls PR_Close() */
+  std::cerr << "Destructing socket" << std::endl;
+  Destroy(socks[1]);
+
+  RecvDataFailed(socks[0], sizeof(data), R_EOD);
+
+  socks[1] = nullptr;
+}
+
+TEST_F(MultiTcpSocketTest, TestCloseActive) {
+  const char data[] = "TestCloseActive";
+  socks[0] = Create(TCP_TYPE_ACTIVE);
+  socks[1] = Create(TCP_TYPE_PASSIVE);
+  Listen(socks[1]);
+  Connect(socks[0], socks[1]);
+
+  TransferData(socks[0], socks[1], data, sizeof(data));
+  TransferData(socks[1], socks[0], data, sizeof(data));
+
+  /* We have to destroy as only that calls PR_Close() */
+  std::cerr << "Destructing socket" << std::endl;
+  Destroy(socks[0]);
+
+  RecvDataFailed(socks[1], sizeof(data), R_EOD);
+
+  socks[0] = nullptr;
+}
+
+TEST_F(MultiTcpSocketTest, TestTwoSendsBeforeReceives) {
+  const char data1[] = "TestTwoSendsBeforeReceives";
+  const char data2[] = "2nd data";
+  socks[0] = Create(TCP_TYPE_ACTIVE);
+  socks[1] = Create(TCP_TYPE_PASSIVE);
+  Listen(socks[1]);
+  Connect(socks[0], socks[1]);
+
+  SendData(socks[0], socks[1], data1, sizeof(data1));
+  SendData(socks[0], socks[1], data2, sizeof(data2));
+  RecvData(socks[0], socks[1], data1, sizeof(data1));
+  /* ICE TCP framing turns TCP effectively into datagram mode */
+  RecvData(socks[0], socks[1], data2, sizeof(data2));
+}
+
+TEST_F(MultiTcpSocketTest, TestTwoActiveBidirectionalTransmit) {
+  const char data1[] = "TestTwoActiveBidirectionalTransmit";
+  const char data2[] = "ReplyToTheFirstSocket";
+  const char data3[] = "TestMessageFromTheSecondSocket";
+  const char data4[] = "ThisIsAReplyToTheSecondSocket";
+  socks[0] = Create(TCP_TYPE_PASSIVE);
+  socks[1] = Create(TCP_TYPE_ACTIVE);
+  socks[2] = Create(TCP_TYPE_ACTIVE);
+  Listen(socks[0]);
+  Connect(socks[1], socks[0]);
+  Connect(socks[2], socks[0]);
+
+  TransferData(socks[1], socks[0], data1, sizeof(data1));
+  TransferData(socks[0], socks[1], data2, sizeof(data2));
+  TransferData(socks[2], socks[0], data3, sizeof(data3));
+  TransferData(socks[0], socks[2], data4, sizeof(data4));
+}
+
+TEST_F(MultiTcpSocketTest, TestTwoPassiveBidirectionalTransmit) {
+  const char data1[] = "TestTwoPassiveBidirectionalTransmit";
+  const char data2[] = "FirstReply";
+  const char data3[] = "TestTwoPassiveBidirectionalTransmitToTheSecondSock";
+  const char data4[] = "SecondReply";
+  socks[0] = Create(TCP_TYPE_PASSIVE);
+  socks[1] = Create(TCP_TYPE_PASSIVE);
+  socks[2] = Create(TCP_TYPE_ACTIVE);
+  Listen(socks[0]);
+  Listen(socks[1]);
+  Connect(socks[2], socks[0]);
+  Connect(socks[2], socks[1]);
+
+  TransferData(socks[2], socks[0], data1, sizeof(data1));
+  TransferData(socks[0], socks[2], data2, sizeof(data2));
+  TransferData(socks[2], socks[1], data3, sizeof(data3));
+  TransferData(socks[1], socks[2], data4, sizeof(data4));
+}
+
+TEST_F(MultiTcpSocketTest, TestActivePassiveWithStunServerMockup) {
+  /* Fake STUN message able to pass the nr_is_stun_msg check
+     used in nr_socket_buffered_stun */
+  const char stunMessage[] = {'\x00', '\x01', '\x00', '\x04', '\x21', '\x12',
+                              '\xa4', '\x42', '\x00', '\x00', '\x00', '\x00',
+                              '\x00', '\x00', '\x0c', '\x00', '\x00', '\x00',
+                              '\x00', '\x00', '\x1c', '\xed', '\xca', '\xfe'};
+  const char data[] = "TestActivePassiveWithStunServerMockup";
+
+  nr_transport_addr stun_srv_addr;
+  std::string stun_addr;
+  uint16_t stun_port;
+  stun_addr = TestStunTcpServer::GetInstance(AF_INET)->addr();
+  stun_port = TestStunTcpServer::GetInstance(AF_INET)->port();
+  int r = nr_str_port_to_transport_addr(stun_addr.c_str(), stun_port,
+                                        IPPROTO_TCP, &stun_srv_addr);
+  ASSERT_EQ(0, r);
+
+  socks[0] = Create(TCP_TYPE_PASSIVE, stun_addr, stun_port);
+  Listen(socks[0]);
+  socks[1] = Create(TCP_TYPE_ACTIVE, stun_addr, stun_port);
+
+  /* Send a fake STUN request and expect a STUN error response */
+  SendData(socks[0], &stun_srv_addr, stunMessage, sizeof(stunMessage));
+  RecvData(&stun_srv_addr, socks[0]);
+
+  Connect(socks[1], socks[0]);
+  TransferData(socks[1], socks[0], data, sizeof(data));
+  TransferData(socks[0], socks[1], data, sizeof(data));
+}
+
+TEST_F(MultiTcpSocketTest, TestConnectTwoSo) {
+  socks[0] = Create(TCP_TYPE_SO);
+  socks[1] = Create(TCP_TYPE_SO);
+  ConnectSo(socks[0], socks[1]);
+}
+
+// test works on localhost only with delay applied:
+//   tc qdisc add dev lo root netem delay 5ms
+TEST_F(MultiTcpSocketTest, DISABLED_TestTwoSoBidirectionalTransmit) {
+  const char data[] = "TestTwoSoBidirectionalTransmit";
+  socks[0] = Create(TCP_TYPE_SO);
+  socks[1] = Create(TCP_TYPE_SO);
+  ConnectSo(socks[0], socks[1]);
+  TransferData(socks[0], socks[1], data, sizeof(data));
+  TransferData(socks[1], socks[0], data, sizeof(data));
+}
+
+TEST_F(MultiTcpSocketTest, TestBigData) {
+  char buf1[2048];
+  char buf2[1024];
+
+  for (unsigned i = 0; i < sizeof(buf1); ++i) {
+    buf1[i] = i & 0xff;
+  }
+  for (unsigned i = 0; i < sizeof(buf2); ++i) {
+    buf2[i] = (i + 0x80) & 0xff;
+  }
+  socks[0] = Create(TCP_TYPE_ACTIVE);
+  socks[1] = Create(TCP_TYPE_PASSIVE);
+  Listen(socks[1]);
+  Connect(socks[0], socks[1]);
+
+  TransferData(socks[0], socks[1], buf1, sizeof(buf1));
+  TransferData(socks[0], socks[1], buf2, sizeof(buf2));
+  // opposite dir
+  SendData(socks[1], socks[0], buf2, sizeof(buf2));
+  SendData(socks[1], socks[0], buf1, sizeof(buf1));
+  RecvData(socks[1], socks[0], buf2, sizeof(buf2));
+  RecvData(socks[1], socks[0], buf1, sizeof(buf1));
+}
diff --git a/dom/media/webrtc/transport/test/nrappkit_unittest.cpp b/dom/media/webrtc/transport/test/nrappkit_unittest.cpp
new file mode 100644
index 0000000000..b6a63fb993
--- /dev/null
+++ b/dom/media/webrtc/transport/test/nrappkit_unittest.cpp
@@ -0,0 +1,123 @@
+
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+#include <iostream>
+
+// nrappkit includes
+extern "C" {
+#include "nr_api.h"
+#include "async_timer.h"
+}
+
+#include "runnable_utils.h"
+
+#define GTEST_HAS_RTTI 0
+#include "gtest/gtest.h"
+#include "gtest_utils.h"
+
+using namespace mozilla;
+
+namespace {
+
+class TimerTest : public MtransportTest {
+ public:
+  TimerTest() : MtransportTest(), handle_(nullptr), fired_(false) {}
+  virtual ~TimerTest() = default;
+
+  int ArmTimer(int timeout) {
+    int ret;
+
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnableRet(&ret, this, &TimerTest::ArmTimer_w, timeout));
+
+    return ret;
+  }
+
+  int ArmCancelTimer(int timeout) {
+    int ret;
+
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnableRet(&ret, this, &TimerTest::ArmCancelTimer_w, timeout));
+
+    return ret;
+  }
+
+  int ArmTimer_w(int timeout) {
+    return NR_ASYNC_TIMER_SET(timeout, cb, this, &handle_);
+  }
+
+  int ArmCancelTimer_w(int timeout) {
+    int r;
+    r = ArmTimer_w(timeout);
+    if (r) return r;
+
+    return CancelTimer_w();
+  }
+
+  int CancelTimer() {
+    int ret;
+
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnableRet(&ret, this, &TimerTest::CancelTimer_w));
+
+    return ret;
+  }
+
+  int CancelTimer_w() { return NR_async_timer_cancel(handle_); }
+
+  int Schedule() {
+    int ret;
+
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnableRet(&ret, this, &TimerTest::Schedule_w));
+
+    return ret;
+  }
+
+  int Schedule_w() {
+    NR_ASYNC_SCHEDULE(cb, this);
+
+    return 0;
+  }
+
+  static void cb(NR_SOCKET r, int how, void* arg) {
+    std::cerr << "Timer fired " << std::endl;
+
+    TimerTest* t = static_cast<TimerTest*>(arg);
+
+    t->fired_ = true;
+  }
+
+ protected:
+  void* handle_;
+  bool fired_;
+};
+}  // namespace
+
+TEST_F(TimerTest, SimpleTimer) {
+  ArmTimer(100);
+  ASSERT_TRUE_WAIT(fired_, 1000);
+}
+
+TEST_F(TimerTest, CancelTimer) {
+  ArmTimer(1000);
+  CancelTimer();
+  PR_Sleep(2000);
+  ASSERT_FALSE(fired_);
+}
+
+TEST_F(TimerTest, CancelTimer0) {
+  ArmCancelTimer(0);
+  PR_Sleep(100);
+  ASSERT_FALSE(fired_);
+}
+
+TEST_F(TimerTest, ScheduleTest) {
+  Schedule();
+  ASSERT_TRUE_WAIT(fired_, 1000);
+}
diff --git a/dom/media/webrtc/transport/test/proxy_tunnel_socket_unittest.cpp b/dom/media/webrtc/transport/test/proxy_tunnel_socket_unittest.cpp
new file mode 100644
index 0000000000..1b54126dd6
--- /dev/null
+++ b/dom/media/webrtc/transport/test/proxy_tunnel_socket_unittest.cpp
@@ -0,0 +1,277 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original authors: ekr@rtfm.com; ryan@tokbox.com
+
+#include <vector>
+#include <numeric>
+
+#include "nr_socket_tcp.h"
+#include "WebrtcTCPSocketWrapper.h"
+
+#define GTEST_HAS_RTTI 0
+#include "gtest/gtest.h"
+#include "gtest_utils.h"
+
+using namespace mozilla;
+
+// update TestReadMultipleSizes if you change this
+const std::string kHelloMessage = "HELLO IS IT ME YOU'RE LOOKING FOR?";
+
+class NrTcpSocketTest : public MtransportTest {
+ public:
+  NrTcpSocketTest()
+      : mSProxy(nullptr),
+        nr_socket_(nullptr),
+        mEmptyArray(0),
+        mReadChunkSize(0),
+        mReadChunkSizeIncrement(1),
+        mReadAllowance(-1),
+        mConnected(false) {}
+
+  void SetUp() override {
+    mSProxy = new NrTcpSocket(nullptr);
+    int r = nr_socket_create_int((void*)mSProxy.get(), mSProxy->vtbl(),
+                                 &nr_socket_);
+    ASSERT_EQ(0, r);
+
+    // fake calling AsyncOpen() due to IPC calls. must be non-null
+    mSProxy->AssignChannel_DoNotUse(new WebrtcTCPSocketWrapper(nullptr));
+  }
+
+  void TearDown() override { mSProxy->close(); }
+
+  static void readable_cb(NR_SOCKET s, int how, void* cb_arg) {
+    NrTcpSocketTest* test = (NrTcpSocketTest*)cb_arg;
+    size_t capacity = std::min(test->mReadChunkSize, test->mReadAllowance);
+    nsTArray<uint8_t> array(capacity);
+    size_t read;
+
+    nr_socket_read(test->nr_socket_, (char*)array.Elements(), array.Capacity(),
+                   &read, 0);
+
+    ASSERT_TRUE(read <= array.Capacity());
+    ASSERT_TRUE(test->mReadAllowance >= read);
+
+    array.SetLength(read);
+    test->mData.AppendElements(array);
+    test->mReadAllowance -= read;
+
+    // We may read more bytes each time we're called. This way we can ensure we
+    // consume buffers partially and across multiple buffers.
+    test->mReadChunkSize += test->mReadChunkSizeIncrement;
+
+    if (test->mReadAllowance > 0) {
+      NR_ASYNC_WAIT(s, how, &NrTcpSocketTest::readable_cb, cb_arg);
+    }
+  }
+
+  static void writable_cb(NR_SOCKET s, int how, void* cb_arg) {
+    NrTcpSocketTest* test = (NrTcpSocketTest*)cb_arg;
+    test->mConnected = true;
+  }
+
+  const std::string DataString() {
+    return std::string((char*)mData.Elements(), mData.Length());
+  }
+
+ protected:
+  RefPtr<NrTcpSocket> mSProxy;
+  nr_socket* nr_socket_;
+
+  nsTArray<uint8_t> mData;
+  nsTArray<uint8_t> mEmptyArray;
+
+  uint32_t mReadChunkSize;
+  uint32_t mReadChunkSizeIncrement;
+  uint32_t mReadAllowance;
+
+  bool mConnected;
+};
+
+TEST_F(NrTcpSocketTest, TestCreate) {}
+
+TEST_F(NrTcpSocketTest, TestConnected) {
+  ASSERT_TRUE(!mConnected);
+
+  NR_ASYNC_WAIT(mSProxy, NR_ASYNC_WAIT_WRITE, &NrTcpSocketTest::writable_cb,
+                this);
+
+  // still not connected just registered for writes...
+  ASSERT_TRUE(!mConnected);
+
+  mSProxy->OnConnected("http"_ns);
+
+  ASSERT_TRUE(mConnected);
+}
+
+TEST_F(NrTcpSocketTest, TestRead) {
+  nsTArray<uint8_t> array;
+  array.AppendElements(kHelloMessage.c_str(), kHelloMessage.length());
+
+  NR_ASYNC_WAIT(mSProxy, NR_ASYNC_WAIT_READ, &NrTcpSocketTest::readable_cb,
+                this);
+  // this will read 0 bytes here
+  mSProxy->OnRead(std::move(array));
+
+  ASSERT_EQ(kHelloMessage.length(), mSProxy->CountUnreadBytes());
+
+  // callback is still set but terminated due to 0 byte read
+  // start callbacks again (first read is 0 then 1,2,3,...)
+  mSProxy->OnRead(std::move(mEmptyArray));
+
+  ASSERT_EQ(kHelloMessage.length(), mData.Length());
+  ASSERT_EQ(kHelloMessage, DataString());
+}
+
+TEST_F(NrTcpSocketTest, TestReadConstantConsumeSize) {
+  std::string data;
+
+  // triangle number
+  const int kCount = 32;
+
+  //  ~17kb
+  // triangle number formula n*(n+1)/2
+  for (int i = 0; i < kCount * (kCount + 1) / 2; ++i) {
+    data += kHelloMessage;
+  }
+
+  // decreasing buffer sizes
+  for (int i = 0, start = 0; i < kCount; ++i) {
+    int length = (kCount - i) * kHelloMessage.length();
+
+    nsTArray<uint8_t> array;
+    array.AppendElements(data.c_str() + start, length);
+    start += length;
+
+    mSProxy->OnRead(std::move(array));
+  }
+
+  ASSERT_EQ(data.length(), mSProxy->CountUnreadBytes());
+
+  // read same amount each callback
+  mReadChunkSize = 128;
+  mReadChunkSizeIncrement = 0;
+  NR_ASYNC_WAIT(mSProxy, NR_ASYNC_WAIT_READ, &NrTcpSocketTest::readable_cb,
+                this);
+
+  ASSERT_EQ(data.length(), mSProxy->CountUnreadBytes());
+
+  // start callbacks
+  mSProxy->OnRead(std::move(mEmptyArray));
+
+  ASSERT_EQ(data.length(), mData.Length());
+  ASSERT_EQ(data, DataString());
+}
+
+TEST_F(NrTcpSocketTest, TestReadNone) {
+  char buf[4096];
+  size_t read = 0;
+  int r = nr_socket_read(nr_socket_, buf, sizeof(buf), &read, 0);
+
+  ASSERT_EQ(R_WOULDBLOCK, r);
+
+  nsTArray<uint8_t> array;
+  array.AppendElements(kHelloMessage.c_str(), kHelloMessage.length());
+  mSProxy->OnRead(std::move(array));
+
+  ASSERT_EQ(kHelloMessage.length(), mSProxy->CountUnreadBytes());
+
+  r = nr_socket_read(nr_socket_, buf, sizeof(buf), &read, 0);
+
+  ASSERT_EQ(0, r);
+  ASSERT_EQ(kHelloMessage.length(), read);
+  ASSERT_EQ(kHelloMessage, std::string(buf, read));
+}
+
+TEST_F(NrTcpSocketTest, TestReadMultipleSizes) {
+  using namespace std;
+
+  string data;
+  // 515 * kHelloMessage.length() == 17510
+  const size_t kCount = 515;
+  // randomly generated numbers, sums to 17510, 20 numbers
+  vector<int> varyingSizes = {404,  622, 1463, 1597, 1676, 389, 389,
+                              1272, 781, 81,   1030, 1450, 256, 812,
+                              1571, 29,  1045, 911,  643,  1089};
+
+  // changing varyingSizes or the test message breaks this so check here
+  ASSERT_EQ(kCount, 17510 / kHelloMessage.length());
+  ASSERT_EQ(17510, accumulate(varyingSizes.begin(), varyingSizes.end(), 0));
+
+  // ~17kb
+  for (size_t i = 0; i < kCount; ++i) {
+    data += kHelloMessage;
+  }
+
+  nsTArray<uint8_t> array;
+  array.AppendElements(data.c_str(), data.length());
+
+  for (int amountToRead : varyingSizes) {
+    nsTArray<uint8_t> buffer;
+    buffer.AppendElements(array.Elements(), amountToRead);
+    array.RemoveElementsAt(0, amountToRead);
+    mSProxy->OnRead(std::move(buffer));
+  }
+
+  ASSERT_EQ(data.length(), mSProxy->CountUnreadBytes());
+
+  // don't need to read 0 on the first read, so start at 1 and keep going
+  mReadChunkSize = 1;
+  NR_ASYNC_WAIT(mSProxy, NR_ASYNC_WAIT_READ, &NrTcpSocketTest::readable_cb,
+                this);
+  // start callbacks
+  mSProxy->OnRead(std::move(mEmptyArray));
+
+  ASSERT_EQ(data.length(), mData.Length());
+  ASSERT_EQ(data, DataString());
+}
+
+TEST_F(NrTcpSocketTest, TestReadConsumeReadDrain) {
+  std::string data;
+  // ~26kb total; should be even
+  const int kCount = 512;
+
+  // there's some division by 2 here so check that kCount is even
+  ASSERT_EQ(0, kCount % 2);
+
+  for (int i = 0; i < kCount; ++i) {
+    data += kHelloMessage;
+    nsTArray<uint8_t> array;
+    array.AppendElements(kHelloMessage.c_str(), kHelloMessage.length());
+    mSProxy->OnRead(std::move(array));
+  }
+
+  // read half at first
+  mReadAllowance = kCount / 2 * kHelloMessage.length();
+  // start by reading 1 byte
+  mReadChunkSize = 1;
+  NR_ASYNC_WAIT(mSProxy, NR_ASYNC_WAIT_READ, &NrTcpSocketTest::readable_cb,
+                this);
+  mSProxy->OnRead(std::move(mEmptyArray));
+
+  ASSERT_EQ(data.length() / 2, mSProxy->CountUnreadBytes());
+  ASSERT_EQ(data.length() / 2, mData.Length());
+
+  // fill read buffer back up
+  for (int i = 0; i < kCount / 2; ++i) {
+    data += kHelloMessage;
+    nsTArray<uint8_t> array;
+    array.AppendElements(kHelloMessage.c_str(), kHelloMessage.length());
+    mSProxy->OnRead(std::move(array));
+  }
+
+  // remove read limit
+  mReadAllowance = -1;
+  // used entire read allowance so we need to setup a new await
+  NR_ASYNC_WAIT(mSProxy, NR_ASYNC_WAIT_READ, &NrTcpSocketTest::readable_cb,
+                this);
+  // start callbacks
+  mSProxy->OnRead(std::move(mEmptyArray));
+
+  ASSERT_EQ(data.length(), mData.Length());
+  ASSERT_EQ(data, DataString());
+}
diff --git a/dom/media/webrtc/transport/test/rlogconnector_unittest.cpp b/dom/media/webrtc/transport/test/rlogconnector_unittest.cpp
new file mode 100644
index 0000000000..93fabae481
--- /dev/null
+++ b/dom/media/webrtc/transport/test/rlogconnector_unittest.cpp
@@ -0,0 +1,255 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+/* Original author: bcampen@mozilla.com */
+
+#include "rlogconnector.h"
+
+extern "C" {
+#include "registry.h"
+#include "r_log.h"
+}
+
+#define GTEST_HAS_RTTI 0
+#include "gtest/gtest.h"
+
+#include <deque>
+#include <string>
+#include <vector>
+
+using mozilla::RLogConnector;
+
+int NR_LOG_TEST = 0;
+
+class RLogConnectorTest : public ::testing::Test {
+ public:
+  RLogConnectorTest() { Init(); }
+
+  ~RLogConnectorTest() { Free(); }
+
+  static void SetUpTestCase() {
+    NR_reg_init(NR_REG_MODE_LOCAL);
+    r_log_init();
+    /* Would be nice to be able to unregister in the fixture */
+    const char* facility = "rlogconnector_test";
+    r_log_register(const_cast<char*>(facility), &NR_LOG_TEST);
+  }
+
+  void Init() { RLogConnector::CreateInstance(); }
+
+  void Free() { RLogConnector::DestroyInstance(); }
+
+  void ReInit() {
+    Free();
+    Init();
+  }
+};
+
+TEST_F(RLogConnectorTest, TestGetFree) {
+  RLogConnector* instance = RLogConnector::GetInstance();
+  ASSERT_NE(nullptr, instance);
+}
+
+TEST_F(RLogConnectorTest, TestFilterEmpty) {
+  std::deque<std::string> logs;
+  RLogConnector::GetInstance()->GetAny(0, &logs);
+  ASSERT_EQ(0U, logs.size());
+}
+
+TEST_F(RLogConnectorTest, TestBasicFilter) {
+  r_log(NR_LOG_TEST, LOG_INFO, "Test");
+  std::deque<std::string> logs;
+  RLogConnector::GetInstance()->Filter("Test", 0, &logs);
+  ASSERT_EQ(1U, logs.size());
+}
+
+TEST_F(RLogConnectorTest, TestBasicFilterContent) {
+  r_log(NR_LOG_TEST, LOG_INFO, "Test");
+  std::deque<std::string> logs;
+  RLogConnector::GetInstance()->Filter("Test", 0, &logs);
+  ASSERT_EQ("Test", logs.back());
+}
+
+TEST_F(RLogConnectorTest, TestFilterAnyFrontMatch) {
+  r_log(NR_LOG_TEST, LOG_INFO, "Test");
+  std::vector<std::string> substrings;
+  substrings.push_back("foo");
+  substrings.push_back("Test");
+  std::deque<std::string> logs;
+  RLogConnector::GetInstance()->FilterAny(substrings, 0, &logs);
+  ASSERT_EQ("Test", logs.back());
+}
+
+TEST_F(RLogConnectorTest, TestFilterAnyBackMatch) {
+  r_log(NR_LOG_TEST, LOG_INFO, "Test");
+  std::vector<std::string> substrings;
+  substrings.push_back("Test");
+  substrings.push_back("foo");
+  std::deque<std::string> logs;
+  RLogConnector::GetInstance()->FilterAny(substrings, 0, &logs);
+  ASSERT_EQ("Test", logs.back());
+}
+
+TEST_F(RLogConnectorTest, TestFilterAnyBothMatch) {
+  r_log(NR_LOG_TEST, LOG_INFO, "Test");
+  std::vector<std::string> substrings;
+  substrings.push_back("Tes");
+  substrings.push_back("est");
+  std::deque<std::string> logs;
+  RLogConnector::GetInstance()->FilterAny(substrings, 0, &logs);
+  ASSERT_EQ("Test", logs.back());
+}
+
+TEST_F(RLogConnectorTest, TestFilterAnyNeitherMatch) {
+  r_log(NR_LOG_TEST, LOG_INFO, "Test");
+  std::vector<std::string> substrings;
+  substrings.push_back("tes");
+  substrings.push_back("esT");
+  std::deque<std::string> logs;
+  RLogConnector::GetInstance()->FilterAny(substrings, 0, &logs);
+  ASSERT_EQ(0U, logs.size());
+}
+
+TEST_F(RLogConnectorTest, TestAllMatch) {
+  r_log(NR_LOG_TEST, LOG_INFO, "Test1");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test2");
+  std::deque<std::string> logs;
+  RLogConnector::GetInstance()->GetAny(0, &logs);
+  ASSERT_EQ(2U, logs.size());
+}
+
+TEST_F(RLogConnectorTest, TestOrder) {
+  r_log(NR_LOG_TEST, LOG_INFO, "Test1");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test2");
+  std::deque<std::string> logs;
+  RLogConnector::GetInstance()->GetAny(0, &logs);
+  ASSERT_EQ("Test2", logs.back());
+  ASSERT_EQ("Test1", logs.front());
+}
+
+TEST_F(RLogConnectorTest, TestNoMatch) {
+  r_log(NR_LOG_TEST, LOG_INFO, "Test1");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test2");
+  std::deque<std::string> logs;
+  RLogConnector::GetInstance()->Filter("foo", 0, &logs);
+  ASSERT_EQ(0U, logs.size());
+}
+
+TEST_F(RLogConnectorTest, TestSubstringFilter) {
+  r_log(NR_LOG_TEST, LOG_INFO, "Test1");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test2");
+  std::deque<std::string> logs;
+  RLogConnector::GetInstance()->Filter("t1", 0, &logs);
+  ASSERT_EQ(1U, logs.size());
+  ASSERT_EQ("Test1", logs.back());
+}
+
+TEST_F(RLogConnectorTest, TestFilterLimit) {
+  r_log(NR_LOG_TEST, LOG_INFO, "Test1");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test2");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test3");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test4");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test5");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test6");
+  std::deque<std::string> logs;
+  RLogConnector::GetInstance()->Filter("Test", 2, &logs);
+  ASSERT_EQ(2U, logs.size());
+  ASSERT_EQ("Test6", logs.back());
+  ASSERT_EQ("Test5", logs.front());
+}
+
+TEST_F(RLogConnectorTest, TestFilterAnyLimit) {
+  r_log(NR_LOG_TEST, LOG_INFO, "TestOne");
+  r_log(NR_LOG_TEST, LOG_INFO, "TestTwo");
+  r_log(NR_LOG_TEST, LOG_INFO, "TestThree");
+  r_log(NR_LOG_TEST, LOG_INFO, "TestFour");
+  r_log(NR_LOG_TEST, LOG_INFO, "TestFive");
+  r_log(NR_LOG_TEST, LOG_INFO, "TestSix");
+  std::vector<std::string> substrings;
+  // Matches Two, Three, Four, and Six
+  substrings.push_back("tT");
+  substrings.push_back("o");
+  substrings.push_back("r");
+  substrings.push_back("S");
+  std::deque<std::string> logs;
+  RLogConnector::GetInstance()->FilterAny(substrings, 2, &logs);
+  ASSERT_EQ(2U, logs.size());
+  ASSERT_EQ("TestSix", logs.back());
+  ASSERT_EQ("TestFour", logs.front());
+}
+
+TEST_F(RLogConnectorTest, TestLimit) {
+  RLogConnector::GetInstance()->SetLogLimit(3);
+  r_log(NR_LOG_TEST, LOG_INFO, "Test1");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test2");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test3");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test4");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test5");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test6");
+  std::deque<std::string> logs;
+  RLogConnector::GetInstance()->GetAny(0, &logs);
+  ASSERT_EQ(3U, logs.size());
+  ASSERT_EQ("Test6", logs.back());
+  ASSERT_EQ("Test4", logs.front());
+}
+
+TEST_F(RLogConnectorTest, TestLimitBulkDiscard) {
+  r_log(NR_LOG_TEST, LOG_INFO, "Test1");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test2");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test3");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test4");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test5");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test6");
+  RLogConnector::GetInstance()->SetLogLimit(3);
+  std::deque<std::string> logs;
+  RLogConnector::GetInstance()->GetAny(0, &logs);
+  ASSERT_EQ(3U, logs.size());
+  ASSERT_EQ("Test6", logs.back());
+  ASSERT_EQ("Test4", logs.front());
+}
+
+TEST_F(RLogConnectorTest, TestIncreaseLimit) {
+  RLogConnector::GetInstance()->SetLogLimit(3);
+  r_log(NR_LOG_TEST, LOG_INFO, "Test1");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test2");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test3");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test4");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test5");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test6");
+  RLogConnector::GetInstance()->SetLogLimit(300);
+  std::deque<std::string> logs;
+  RLogConnector::GetInstance()->GetAny(0, &logs);
+  ASSERT_EQ(3U, logs.size());
+  ASSERT_EQ("Test6", logs.back());
+  ASSERT_EQ("Test4", logs.front());
+}
+
+TEST_F(RLogConnectorTest, TestClear) {
+  r_log(NR_LOG_TEST, LOG_INFO, "Test1");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test2");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test3");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test4");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test5");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test6");
+  RLogConnector::GetInstance()->SetLogLimit(0);
+  RLogConnector::GetInstance()->SetLogLimit(4096);
+  std::deque<std::string> logs;
+  RLogConnector::GetInstance()->GetAny(0, &logs);
+  ASSERT_EQ(0U, logs.size());
+}
+
+TEST_F(RLogConnectorTest, TestReInit) {
+  r_log(NR_LOG_TEST, LOG_INFO, "Test1");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test2");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test3");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test4");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test5");
+  r_log(NR_LOG_TEST, LOG_INFO, "Test6");
+  ReInit();
+  std::deque<std::string> logs;
+  RLogConnector::GetInstance()->GetAny(0, &logs);
+  ASSERT_EQ(0U, logs.size());
+}
diff --git a/dom/media/webrtc/transport/test/runnable_utils_unittest.cpp b/dom/media/webrtc/transport/test/runnable_utils_unittest.cpp
new file mode 100644
index 0000000000..70707b148f
--- /dev/null
+++ b/dom/media/webrtc/transport/test/runnable_utils_unittest.cpp
@@ -0,0 +1,353 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+#include <iostream>
+
+#include "nsCOMPtr.h"
+#include "nsNetCID.h"
+
+#include "mozilla/RefPtr.h"
+#include "mozilla/UniquePtr.h"
+
+#include "nsServiceManagerUtils.h"
+#include "nsThreadUtils.h"
+
+#include "runnable_utils.h"
+
+#define GTEST_HAS_RTTI 0
+#include "gtest/gtest.h"
+#include "gtest_utils.h"
+
+using namespace mozilla;
+
+namespace {
+
+// Helper used to make sure args are properly copied and/or moved.
+struct CtorDtorState {
+  enum class State { Empty, Copy, Explicit, Move, Moved };
+
+  static const char* ToStr(const State& state) {
+    switch (state) {
+      case State::Empty:
+        return "empty";
+      case State::Copy:
+        return "copy";
+      case State::Explicit:
+        return "explicit";
+      case State::Move:
+        return "move";
+      case State::Moved:
+        return "moved";
+      default:
+        return "unknown";
+    }
+  }
+
+  void DumpState() const { std::cerr << ToStr(state_) << std::endl; }
+
+  CtorDtorState() { DumpState(); }
+
+  explicit CtorDtorState(int* destroyed)
+      : dtor_count_(destroyed), state_(State::Explicit) {
+    DumpState();
+  }
+
+  CtorDtorState(const CtorDtorState& other)
+      : dtor_count_(other.dtor_count_), state_(State::Copy) {
+    DumpState();
+  }
+
+  // Clear the other's dtor counter so it's not counted if moved.
+  CtorDtorState(CtorDtorState&& other)
+      : dtor_count_(std::exchange(other.dtor_count_, nullptr)),
+        state_(State::Move) {
+    other.state_ = State::Moved;
+    DumpState();
+  }
+
+  ~CtorDtorState() {
+    const char* const state = ToStr(state_);
+    std::cerr << "Destructor called with end state: " << state << std::endl;
+
+    if (dtor_count_) {
+      ++*dtor_count_;
+    }
+  }
+
+  int* dtor_count_ = nullptr;
+  State state_ = State::Empty;
+};
+
+class Destructor {
+ private:
+  ~Destructor() {
+    std::cerr << "Destructor called" << std::endl;
+    *destroyed_ = true;
+  }
+
+ public:
+  explicit Destructor(bool* destroyed) : destroyed_(destroyed) {}
+
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(Destructor)
+
+ private:
+  bool* destroyed_;
+};
+
+class TargetClass {
+ public:
+  explicit TargetClass(int* ran) : ran_(ran) {}
+
+  void m1(int x) {
+    std::cerr << __FUNCTION__ << " " << x << std::endl;
+    *ran_ = 1;
+  }
+
+  void m2(int x, int y) {
+    std::cerr << __FUNCTION__ << " " << x << " " << y << std::endl;
+    *ran_ = 2;
+  }
+
+  void m1set(bool* z) {
+    std::cerr << __FUNCTION__ << std::endl;
+    *z = true;
+  }
+  int return_int(int x) {
+    std::cerr << __FUNCTION__ << std::endl;
+    return x;
+  }
+
+  void destructor_target_ref(RefPtr<Destructor> destructor) {}
+
+  int* ran_;
+};
+
+class RunnableArgsTest : public MtransportTest {
+ public:
+  RunnableArgsTest() : MtransportTest(), ran_(0), cl_(&ran_) {}
+
+  void Test1Arg() {
+    Runnable* r = WrapRunnable(&cl_, &TargetClass::m1, 1);
+    r->Run();
+    ASSERT_EQ(1, ran_);
+  }
+
+  void Test2Args() {
+    Runnable* r = WrapRunnable(&cl_, &TargetClass::m2, 1, 2);
+    r->Run();
+    ASSERT_EQ(2, ran_);
+  }
+
+ private:
+  int ran_;
+  TargetClass cl_;
+};
+
+class DispatchTest : public MtransportTest {
+ public:
+  DispatchTest() : MtransportTest(), ran_(0), cl_(&ran_) {}
+
+  void SetUp() {
+    MtransportTest::SetUp();
+
+    nsresult rv;
+    target_ = do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &rv);
+    ASSERT_TRUE(NS_SUCCEEDED(rv));
+  }
+
+  void Test1Arg() {
+    Runnable* r = WrapRunnable(&cl_, &TargetClass::m1, 1);
+    NS_DispatchAndSpinEventLoopUntilComplete("DispatchTest::Test1Arg"_ns,
+                                             target_, do_AddRef(r));
+    ASSERT_EQ(1, ran_);
+  }
+
+  void Test2Args() {
+    Runnable* r = WrapRunnable(&cl_, &TargetClass::m2, 1, 2);
+    NS_DispatchAndSpinEventLoopUntilComplete("DispatchTest::Test2Args"_ns,
+                                             target_, do_AddRef(r));
+    ASSERT_EQ(2, ran_);
+  }
+
+  void Test1Set() {
+    bool x = false;
+    NS_DispatchAndSpinEventLoopUntilComplete(
+        "DispatchTest::Test1Set"_ns, target_,
+        do_AddRef(WrapRunnable(&cl_, &TargetClass::m1set, &x)));
+    ASSERT_TRUE(x);
+  }
+
+  void TestRet() {
+    int z;
+    int x = 10;
+
+    NS_DispatchAndSpinEventLoopUntilComplete(
+        "DispatchTest::TestRet"_ns, target_,
+        do_AddRef(WrapRunnableRet(&z, &cl_, &TargetClass::return_int, x)));
+    ASSERT_EQ(10, z);
+  }
+
+ protected:
+  int ran_;
+  TargetClass cl_;
+  nsCOMPtr<nsIEventTarget> target_;
+};
+
+TEST_F(RunnableArgsTest, OneArgument) { Test1Arg(); }
+
+TEST_F(RunnableArgsTest, TwoArguments) { Test2Args(); }
+
+TEST_F(DispatchTest, OneArgument) { Test1Arg(); }
+
+TEST_F(DispatchTest, TwoArguments) { Test2Args(); }
+
+TEST_F(DispatchTest, Test1Set) { Test1Set(); }
+
+TEST_F(DispatchTest, TestRet) { TestRet(); }
+
+void SetNonMethod(TargetClass* cl, int x) { cl->m1(x); }
+
+int SetNonMethodRet(TargetClass* cl, int x) {
+  cl->m1(x);
+
+  return x;
+}
+
+TEST_F(DispatchTest, TestNonMethod) {
+  test_utils_->SyncDispatchToSTS(WrapRunnableNM(SetNonMethod, &cl_, 10));
+
+  ASSERT_EQ(1, ran_);
+}
+
+TEST_F(DispatchTest, TestNonMethodRet) {
+  int z;
+
+  test_utils_->SyncDispatchToSTS(
+      WrapRunnableNMRet(&z, SetNonMethodRet, &cl_, 10));
+
+  ASSERT_EQ(1, ran_);
+  ASSERT_EQ(10, z);
+}
+
+TEST_F(DispatchTest, TestDestructorRef) {
+  bool destroyed = false;
+  {
+    RefPtr<Destructor> destructor = new Destructor(&destroyed);
+    NS_DispatchAndSpinEventLoopUntilComplete(
+        "DispatchTest::TestDestructorRef"_ns, target_,
+        do_AddRef(WrapRunnable(&cl_, &TargetClass::destructor_target_ref,
+                               destructor)));
+    ASSERT_FALSE(destroyed);
+  }
+  ASSERT_TRUE(destroyed);
+
+  // Now try with a move.
+  destroyed = false;
+  {
+    RefPtr<Destructor> destructor = new Destructor(&destroyed);
+    NS_DispatchAndSpinEventLoopUntilComplete(
+        "DispatchTest::TestDestructorRef"_ns, target_,
+        do_AddRef(WrapRunnable(&cl_, &TargetClass::destructor_target_ref,
+                               std::move(destructor))));
+    ASSERT_TRUE(destroyed);
+  }
+}
+
+TEST_F(DispatchTest, TestMove) {
+  int destroyed = 0;
+  {
+    CtorDtorState state(&destroyed);
+
+    // Dispatch with:
+    //   - moved arg
+    //   - by-val capture in function should consume a move
+    //   - expect destruction in the function scope
+    NS_DispatchAndSpinEventLoopUntilComplete(
+        "DispatchTest::TestMove"_ns, target_,
+        do_AddRef(WrapRunnableNM([](CtorDtorState s) {}, std::move(state))));
+    ASSERT_EQ(1, destroyed);
+  }
+  // Still shouldn't count when we go out of scope as it was moved.
+  ASSERT_EQ(1, destroyed);
+
+  {
+    CtorDtorState state(&destroyed);
+
+    // Dispatch with:
+    //   - copied arg
+    //   - by-val capture in function should consume a move
+    //   - expect destruction in the function scope and call scope
+    NS_DispatchAndSpinEventLoopUntilComplete(
+        "DispatchTest::TestMove"_ns, target_,
+        do_AddRef(WrapRunnableNM([](CtorDtorState s) {}, state)));
+    ASSERT_EQ(2, destroyed);
+  }
+  // Original state should be destroyed
+  ASSERT_EQ(3, destroyed);
+
+  {
+    CtorDtorState state(&destroyed);
+
+    // Dispatch with:
+    //   - moved arg
+    //   - by-ref in function should accept a moved arg
+    //   - expect destruction in the wrapper invocation scope
+    NS_DispatchAndSpinEventLoopUntilComplete(
+        "DispatchTest::TestMove"_ns, target_,
+        do_AddRef(
+            WrapRunnableNM([](const CtorDtorState& s) {}, std::move(state))));
+    ASSERT_EQ(4, destroyed);
+  }
+  // Still shouldn't count when we go out of scope as it was moved.
+  ASSERT_EQ(4, destroyed);
+
+  {
+    CtorDtorState state(&destroyed);
+
+    // Dispatch with:
+    //   - moved arg
+    //   - r-value function should accept a moved arg
+    //   - expect destruction in the wrapper invocation scope
+    NS_DispatchAndSpinEventLoopUntilComplete(
+        "DispatchTest::TestMove"_ns, target_,
+        do_AddRef(WrapRunnableNM([](CtorDtorState&& s) {}, std::move(state))));
+    ASSERT_EQ(5, destroyed);
+  }
+  // Still shouldn't count when we go out of scope as it was moved.
+  ASSERT_EQ(5, destroyed);
+}
+
+TEST_F(DispatchTest, TestUniquePtr) {
+  // Test that holding the class in UniquePtr works
+  int ran = 0;
+  auto cl = MakeUnique<TargetClass>(&ran);
+
+  NS_DispatchAndSpinEventLoopUntilComplete(
+      "DispatchTest::TestUniquePtr"_ns, target_,
+      do_AddRef(WrapRunnable(std::move(cl), &TargetClass::m1, 1)));
+  ASSERT_EQ(1, ran);
+
+  // Test that UniquePtr works as a param to the runnable
+  int destroyed = 0;
+  {
+    auto state = MakeUnique<CtorDtorState>(&destroyed);
+
+    // Dispatch with:
+    //   - moved arg
+    //   - Function should move construct from arg
+    //   - expect destruction in the wrapper invocation scope
+    NS_DispatchAndSpinEventLoopUntilComplete(
+        "DispatchTest::TestUniquePtr"_ns, target_,
+        do_AddRef(WrapRunnableNM([](UniquePtr<CtorDtorState> s) {},
+                                 std::move(state))));
+    ASSERT_EQ(1, destroyed);
+  }
+  // Still shouldn't count when we go out of scope as it was moved.
+  ASSERT_EQ(1, destroyed);
+}
+
+}  // end of namespace
diff --git a/dom/media/webrtc/transport/test/sctp_unittest.cpp b/dom/media/webrtc/transport/test/sctp_unittest.cpp
new file mode 100644
index 0000000000..ea32565fb2
--- /dev/null
+++ b/dom/media/webrtc/transport/test/sctp_unittest.cpp
@@ -0,0 +1,381 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+#include <iostream>
+#include <string>
+
+#include "sigslot.h"
+
+#include "nsITimer.h"
+
+#include "transportflow.h"
+#include "transportlayer.h"
+#include "transportlayerloopback.h"
+
+#include "runnable_utils.h"
+#include "usrsctp.h"
+
+#define GTEST_HAS_RTTI 0
+#include "gtest/gtest.h"
+#include "gtest_utils.h"
+
+using namespace mozilla;
+
+static bool sctp_logging = false;
+static int port_number = 5000;
+
+namespace {
+
+class TransportTestPeer;
+
+class SendPeriodic : public nsITimerCallback, public nsINamed {
+ public:
+  SendPeriodic(TransportTestPeer* peer, int to_send)
+      : peer_(peer), to_send_(to_send) {}
+
+  NS_DECL_THREADSAFE_ISUPPORTS
+  NS_DECL_NSITIMERCALLBACK
+  NS_DECL_NSINAMED
+
+ protected:
+  virtual ~SendPeriodic() = default;
+
+  TransportTestPeer* peer_;
+  int to_send_;
+};
+
+NS_IMPL_ISUPPORTS(SendPeriodic, nsITimerCallback, nsINamed)
+
+class TransportTestPeer : public sigslot::has_slots<> {
+ public:
+  TransportTestPeer(std::string name, int local_port, int remote_port,
+                    MtransportTestUtils* utils)
+      : name_(name),
+        connected_(false),
+        sent_(0),
+        received_(0),
+        flow_(new TransportFlow()),
+        loopback_(new TransportLayerLoopback()),
+        sctp_(usrsctp_socket(AF_CONN, SOCK_STREAM, IPPROTO_SCTP, receive_cb,
+                             nullptr, 0, nullptr)),
+        timer_(NS_NewTimer()),
+        periodic_(nullptr),
+        test_utils_(utils) {
+    std::cerr << "Creating TransportTestPeer; flow="
+              << static_cast<void*>(flow_.get()) << " local=" << local_port
+              << " remote=" << remote_port << std::endl;
+
+    usrsctp_register_address(static_cast<void*>(this));
+    int r = usrsctp_set_non_blocking(sctp_, 1);
+    EXPECT_GE(r, 0);
+
+    struct linger l;
+    l.l_onoff = 1;
+    l.l_linger = 0;
+    r = usrsctp_setsockopt(sctp_, SOL_SOCKET, SO_LINGER, &l,
+                           (socklen_t)sizeof(l));
+    EXPECT_GE(r, 0);
+
+    struct sctp_event subscription;
+    memset(&subscription, 0, sizeof(subscription));
+    subscription.se_assoc_id = SCTP_ALL_ASSOC;
+    subscription.se_on = 1;
+    subscription.se_type = SCTP_ASSOC_CHANGE;
+    r = usrsctp_setsockopt(sctp_, IPPROTO_SCTP, SCTP_EVENT, &subscription,
+                           sizeof(subscription));
+    EXPECT_GE(r, 0);
+
+    memset(&local_addr_, 0, sizeof(local_addr_));
+    local_addr_.sconn_family = AF_CONN;
+#if !defined(__Userspace_os_Linux) && !defined(__Userspace_os_Windows) && \
+    !defined(__Userspace_os_Android)
+    local_addr_.sconn_len = sizeof(struct sockaddr_conn);
+#endif
+    local_addr_.sconn_port = htons(local_port);
+    local_addr_.sconn_addr = static_cast<void*>(this);
+
+    memset(&remote_addr_, 0, sizeof(remote_addr_));
+    remote_addr_.sconn_family = AF_CONN;
+#if !defined(__Userspace_os_Linux) && !defined(__Userspace_os_Windows) && \
+    !defined(__Userspace_os_Android)
+    remote_addr_.sconn_len = sizeof(struct sockaddr_conn);
+#endif
+    remote_addr_.sconn_port = htons(remote_port);
+    remote_addr_.sconn_addr = static_cast<void*>(this);
+
+    nsresult res;
+    res = loopback_->Init();
+    EXPECT_EQ((nsresult)NS_OK, res);
+  }
+
+  ~TransportTestPeer() {
+    std::cerr << "Destroying sctp connection flow="
+              << static_cast<void*>(flow_.get()) << std::endl;
+    usrsctp_close(sctp_);
+    usrsctp_deregister_address(static_cast<void*>(this));
+
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &TransportTestPeer::Disconnect_s));
+
+    std::cerr << "~TransportTestPeer() completed" << std::endl;
+  }
+
+  void ConnectSocket(TransportTestPeer* peer) {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &TransportTestPeer::ConnectSocket_s, peer));
+  }
+
+  void ConnectSocket_s(TransportTestPeer* peer) {
+    loopback_->Connect(peer->loopback_);
+    ASSERT_EQ((nsresult)NS_OK, loopback_->Init());
+    flow_->PushLayer(loopback_);
+
+    loopback_->SignalPacketReceived.connect(this,
+                                            &TransportTestPeer::PacketReceived);
+
+    // SCTP here!
+    ASSERT_TRUE(sctp_);
+    std::cerr << "Calling usrsctp_bind()" << std::endl;
+    int r =
+        usrsctp_bind(sctp_, reinterpret_cast<struct sockaddr*>(&local_addr_),
+                     sizeof(local_addr_));
+    ASSERT_GE(0, r);
+
+    std::cerr << "Calling usrsctp_connect()" << std::endl;
+    r = usrsctp_connect(sctp_,
+                        reinterpret_cast<struct sockaddr*>(&remote_addr_),
+                        sizeof(remote_addr_));
+    ASSERT_GE(0, r);
+  }
+
+  void Disconnect_s() {
+    disconnect_all();
+    if (flow_) {
+      flow_ = nullptr;
+    }
+  }
+
+  void Disconnect() { loopback_->Disconnect(); }
+
+  void StartTransfer(size_t to_send) {
+    periodic_ = new SendPeriodic(this, to_send);
+    timer_->SetTarget(test_utils_->sts_target());
+    timer_->InitWithCallback(periodic_, 10, nsITimer::TYPE_REPEATING_SLACK);
+  }
+
+  void SendOne() {
+    unsigned char buf[100];
+    memset(buf, sent_ & 0xff, sizeof(buf));
+
+    struct sctp_sndinfo info;
+    info.snd_sid = 1;
+    info.snd_flags = 0;
+    info.snd_ppid = 50;  // What the heck is this?
+    info.snd_context = 0;
+    info.snd_assoc_id = 0;
+
+    int r = usrsctp_sendv(sctp_, buf, sizeof(buf), nullptr, 0,
+                          static_cast<void*>(&info), sizeof(info),
+                          SCTP_SENDV_SNDINFO, 0);
+    ASSERT_TRUE(r >= 0);
+    ASSERT_EQ(sizeof(buf), (size_t)r);
+
+    ++sent_;
+  }
+
+  int sent() const { return sent_; }
+  int received() const { return received_; }
+  bool connected() const { return connected_; }
+
+  static TransportResult SendPacket_s(UniquePtr<MediaPacket> packet,
+                                      const RefPtr<TransportFlow>& flow,
+                                      TransportLayer* layer) {
+    return layer->SendPacket(*packet);
+  }
+
+  TransportResult SendPacket(const unsigned char* data, size_t len) {
+    UniquePtr<MediaPacket> packet(new MediaPacket);
+    packet->Copy(data, len);
+
+    // Uses DISPATCH_NORMAL to avoid possible deadlocks when we're called
+    // from MainThread especially during shutdown (same as DataChannels).
+    // RUN_ON_THREAD short-circuits if already on the STS thread, which is
+    // normal for most transfers outside of connect() and close().  Passes
+    // a refptr to flow_ to avoid any async deletion issues (since we can't
+    // make 'this' into a refptr as it isn't refcounted)
+    RUN_ON_THREAD(test_utils_->sts_target(),
+                  WrapRunnableNM(&TransportTestPeer::SendPacket_s,
+                                 std::move(packet), flow_, loopback_),
+                  NS_DISPATCH_NORMAL);
+
+    return 0;
+  }
+
+  void PacketReceived(TransportLayer* layer, MediaPacket& packet) {
+    std::cerr << "Received " << packet.len() << " bytes" << std::endl;
+
+    // Pass the data to SCTP
+
+    usrsctp_conninput(static_cast<void*>(this), packet.data(), packet.len(), 0);
+  }
+
+  // Process SCTP notification
+  void Notification(union sctp_notification* msg, size_t len) {
+    ASSERT_EQ(msg->sn_header.sn_length, len);
+
+    if (msg->sn_header.sn_type == SCTP_ASSOC_CHANGE) {
+      struct sctp_assoc_change* change = &msg->sn_assoc_change;
+
+      if (change->sac_state == SCTP_COMM_UP) {
+        std::cerr << "Connection up" << std::endl;
+        SetConnected(true);
+      } else {
+        std::cerr << "Connection down" << std::endl;
+        SetConnected(false);
+      }
+    }
+  }
+
+  void SetConnected(bool state) { connected_ = state; }
+
+  static int conn_output(void* addr, void* buffer, size_t length, uint8_t tos,
+                         uint8_t set_df) {
+    TransportTestPeer* peer = static_cast<TransportTestPeer*>(addr);
+
+    peer->SendPacket(static_cast<unsigned char*>(buffer), length);
+
+    return 0;
+  }
+
+  static int receive_cb(struct socket* sock, union sctp_sockstore addr,
+                        void* data, size_t datalen, struct sctp_rcvinfo rcv,
+                        int flags, void* ulp_info) {
+    TransportTestPeer* me =
+        static_cast<TransportTestPeer*>(addr.sconn.sconn_addr);
+    MOZ_ASSERT(me);
+
+    if (flags & MSG_NOTIFICATION) {
+      union sctp_notification* notif =
+          static_cast<union sctp_notification*>(data);
+
+      me->Notification(notif, datalen);
+      return 0;
+    }
+
+    me->received_ += datalen;
+
+    std::cerr << "receive_cb: sock " << sock << " data " << data << "("
+              << datalen << ") total received bytes = " << me->received_
+              << std::endl;
+
+    return 0;
+  }
+
+ private:
+  std::string name_;
+  bool connected_;
+  size_t sent_;
+  size_t received_;
+  // Owns the TransportLayerLoopback, but basically does nothing else.
+  RefPtr<TransportFlow> flow_;
+  TransportLayerLoopback* loopback_;
+
+  struct sockaddr_conn local_addr_;
+  struct sockaddr_conn remote_addr_;
+  struct socket* sctp_;
+  nsCOMPtr<nsITimer> timer_;
+  RefPtr<SendPeriodic> periodic_;
+  MtransportTestUtils* test_utils_;
+};
+
+// Implemented here because it calls a method of TransportTestPeer
+NS_IMETHODIMP SendPeriodic::Notify(nsITimer* timer) {
+  peer_->SendOne();
+  --to_send_;
+  if (!to_send_) {
+    timer->Cancel();
+  }
+  return NS_OK;
+}
+
+NS_IMETHODIMP
+SendPeriodic::GetName(nsACString& aName) {
+  aName.AssignLiteral("SendPeriodic");
+  return NS_OK;
+}
+
+class SctpTransportTest : public MtransportTest {
+ public:
+  SctpTransportTest() = default;
+
+  ~SctpTransportTest() = default;
+
+  static void debug_printf(const char* format, ...) {
+    va_list ap;
+
+    va_start(ap, format);
+    vprintf(format, ap);
+    va_end(ap);
+  }
+
+  static void SetUpTestCase() {
+    if (sctp_logging) {
+      usrsctp_init(0, &TransportTestPeer::conn_output, debug_printf);
+      usrsctp_sysctl_set_sctp_debug_on(0xffffffff);
+    } else {
+      usrsctp_init(0, &TransportTestPeer::conn_output, nullptr);
+    }
+  }
+
+  void TearDown() override {
+    if (p1_) p1_->Disconnect();
+    if (p2_) p2_->Disconnect();
+    delete p1_;
+    delete p2_;
+
+    MtransportTest::TearDown();
+  }
+
+  void ConnectSocket(int p1port = 0, int p2port = 0) {
+    if (!p1port) p1port = port_number++;
+    if (!p2port) p2port = port_number++;
+
+    p1_ = new TransportTestPeer("P1", p1port, p2port, test_utils_);
+    p2_ = new TransportTestPeer("P2", p2port, p1port, test_utils_);
+
+    p1_->ConnectSocket(p2_);
+    p2_->ConnectSocket(p1_);
+    ASSERT_TRUE_WAIT(p1_->connected(), 2000);
+    ASSERT_TRUE_WAIT(p2_->connected(), 2000);
+  }
+
+  void TestTransfer(int expected = 1) {
+    std::cerr << "Starting trasnsfer test" << std::endl;
+    p1_->StartTransfer(expected);
+    ASSERT_TRUE_WAIT(p1_->sent() == expected, 10000);
+    ASSERT_TRUE_WAIT(p2_->received() == (expected * 100), 10000);
+    std::cerr << "P2 received " << p2_->received() << std::endl;
+  }
+
+ protected:
+  TransportTestPeer* p1_ = nullptr;
+  TransportTestPeer* p2_ = nullptr;
+};
+
+TEST_F(SctpTransportTest, TestConnect) { ConnectSocket(); }
+
+TEST_F(SctpTransportTest, TestConnectSymmetricalPorts) {
+  ConnectSocket(5002, 5002);
+}
+
+TEST_F(SctpTransportTest, TestTransfer) {
+  ConnectSocket();
+  TestTransfer(50);
+}
+
+}  // end namespace
diff --git a/dom/media/webrtc/transport/test/simpletokenbucket_unittest.cpp b/dom/media/webrtc/transport/test/simpletokenbucket_unittest.cpp
new file mode 100644
index 0000000000..66622d795b
--- /dev/null
+++ b/dom/media/webrtc/transport/test/simpletokenbucket_unittest.cpp
@@ -0,0 +1,114 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+/* Original author: bcampen@mozilla.com */
+
+#include "simpletokenbucket.h"
+
+#define GTEST_HAS_RTTI 0
+#include "gtest/gtest.h"
+
+using mozilla::SimpleTokenBucket;
+
+class TestSimpleTokenBucket : public SimpleTokenBucket {
+ public:
+  TestSimpleTokenBucket(size_t bucketSize, size_t tokensPerSecond)
+      : SimpleTokenBucket(bucketSize, tokensPerSecond) {}
+
+  void fastForward(int32_t timeMilliSeconds) {
+    if (timeMilliSeconds >= 0) {
+      last_time_tokens_added_ -= PR_MillisecondsToInterval(timeMilliSeconds);
+    } else {
+      last_time_tokens_added_ += PR_MillisecondsToInterval(-timeMilliSeconds);
+    }
+  }
+};
+
+TEST(SimpleTokenBucketTest, TestConstruct)
+{ TestSimpleTokenBucket b(10, 1); }
+
+TEST(SimpleTokenBucketTest, TestGet)
+{
+  TestSimpleTokenBucket b(10, 1);
+  ASSERT_EQ(5U, b.getTokens(5));
+}
+
+TEST(SimpleTokenBucketTest, TestGetAll)
+{
+  TestSimpleTokenBucket b(10, 1);
+  ASSERT_EQ(10U, b.getTokens(10));
+}
+
+TEST(SimpleTokenBucketTest, TestGetInsufficient)
+{
+  TestSimpleTokenBucket b(10, 1);
+  ASSERT_EQ(5U, b.getTokens(5));
+  ASSERT_EQ(5U, b.getTokens(6));
+}
+
+TEST(SimpleTokenBucketTest, TestGetBucketCount)
+{
+  TestSimpleTokenBucket b(10, 1);
+  ASSERT_EQ(10U, b.getTokens(UINT32_MAX));
+  ASSERT_EQ(5U, b.getTokens(5));
+  ASSERT_EQ(5U, b.getTokens(UINT32_MAX));
+}
+
+TEST(SimpleTokenBucketTest, TestTokenRefill)
+{
+  TestSimpleTokenBucket b(10, 1);
+  ASSERT_EQ(5U, b.getTokens(5));
+  b.fastForward(1000);
+  ASSERT_EQ(6U, b.getTokens(6));
+}
+
+TEST(SimpleTokenBucketTest, TestNoTimeWasted)
+{
+  // Makes sure that when the time elapsed is insufficient to add any
+  // tokens to the bucket, the internal timestamp that is used in this
+  // calculation is not updated (ie; two subsequent 0.5 second elapsed times
+  // counts as a full second)
+  TestSimpleTokenBucket b(10, 1);
+  ASSERT_EQ(5U, b.getTokens(5));
+  b.fastForward(500);
+  ASSERT_EQ(5U, b.getTokens(6));
+  b.fastForward(500);
+  ASSERT_EQ(6U, b.getTokens(6));
+}
+
+TEST(SimpleTokenBucketTest, TestNegativeTime)
+{
+  TestSimpleTokenBucket b(10, 1);
+  b.fastForward(-1000);
+  // Make sure we don't end up with an invalid number of tokens, but otherwise
+  // permit anything.
+  ASSERT_GT(11U, b.getTokens(100));
+}
+
+TEST(SimpleTokenBucketTest, TestEmptyBucket)
+{
+  TestSimpleTokenBucket b(10, 1);
+  ASSERT_EQ(10U, b.getTokens(10));
+  ASSERT_EQ(0U, b.getTokens(10));
+}
+
+TEST(SimpleTokenBucketTest, TestEmptyThenFillBucket)
+{
+  TestSimpleTokenBucket b(10, 1);
+  ASSERT_EQ(10U, b.getTokens(10));
+  ASSERT_EQ(0U, b.getTokens(1));
+  b.fastForward(50000);
+  ASSERT_EQ(10U, b.getTokens(10));
+}
+
+TEST(SimpleTokenBucketTest, TestNoOverflow)
+{
+  TestSimpleTokenBucket b(10, 1);
+  ASSERT_EQ(10U, b.getTokens(10));
+  ASSERT_EQ(0U, b.getTokens(1));
+  b.fastForward(50000);
+  ASSERT_EQ(10U, b.getTokens(11));
+}
diff --git a/dom/media/webrtc/transport/test/sockettransportservice_unittest.cpp b/dom/media/webrtc/transport/test/sockettransportservice_unittest.cpp
new file mode 100644
index 0000000000..ffa87fe91f
--- /dev/null
+++ b/dom/media/webrtc/transport/test/sockettransportservice_unittest.cpp
@@ -0,0 +1,181 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+#include <iostream>
+
+#include "prio.h"
+
+#include "nsCOMPtr.h"
+#include "nsNetCID.h"
+
+#include "nsISocketTransportService.h"
+
+#include "nsASocketHandler.h"
+#include "nsServiceManagerUtils.h"
+#include "nsThreadUtils.h"
+
+#define GTEST_HAS_RTTI 0
+#include "gtest/gtest.h"
+#include "gtest_utils.h"
+
+using namespace mozilla;
+
+namespace {
+class SocketTransportServiceTest : public MtransportTest {
+ public:
+  SocketTransportServiceTest()
+      : MtransportTest(),
+        received_(0),
+        readpipe_(nullptr),
+        writepipe_(nullptr),
+        registered_(false) {}
+
+  ~SocketTransportServiceTest() {
+    if (readpipe_) PR_Close(readpipe_);
+    if (writepipe_) PR_Close(writepipe_);
+  }
+
+  void SetUp();
+  void RegisterHandler();
+  void SendEvent();
+  void SendPacket();
+
+  void ReceivePacket() { ++received_; }
+
+  void ReceiveEvent() { ++received_; }
+
+  size_t Received() { return received_; }
+
+ private:
+  nsCOMPtr<nsISocketTransportService> stservice_;
+  nsCOMPtr<nsIEventTarget> target_;
+  size_t received_;
+  PRFileDesc* readpipe_;
+  PRFileDesc* writepipe_;
+  bool registered_;
+};
+
+// Received an event.
+class EventReceived : public Runnable {
+ public:
+  explicit EventReceived(SocketTransportServiceTest* test)
+      : Runnable("EventReceived"), test_(test) {}
+
+  NS_IMETHOD Run() override {
+    test_->ReceiveEvent();
+    return NS_OK;
+  }
+
+  SocketTransportServiceTest* test_;
+};
+
+// Register our listener on the socket
+class RegisterEvent : public Runnable {
+ public:
+  explicit RegisterEvent(SocketTransportServiceTest* test)
+      : Runnable("RegisterEvent"), test_(test) {}
+
+  NS_IMETHOD Run() override {
+    test_->RegisterHandler();
+    return NS_OK;
+  }
+
+  SocketTransportServiceTest* test_;
+};
+
+class SocketHandler : public nsASocketHandler {
+ public:
+  explicit SocketHandler(SocketTransportServiceTest* test) : test_(test) {}
+
+  void OnSocketReady(PRFileDesc* fd, int16_t outflags) override {
+    unsigned char buf[1600];
+
+    int32_t rv;
+    rv = PR_Recv(fd, buf, sizeof(buf), 0, PR_INTERVAL_NO_WAIT);
+    if (rv > 0) {
+      std::cerr << "Read " << rv << " bytes" << std::endl;
+      test_->ReceivePacket();
+    }
+  }
+
+  void OnSocketDetached(PRFileDesc* fd) override {}
+
+  void IsLocal(bool* aIsLocal) override {
+    // TODO(jesup): better check? Does it matter? (likely no)
+    *aIsLocal = false;
+  }
+
+  virtual uint64_t ByteCountSent() override { return 0; }
+  virtual uint64_t ByteCountReceived() override { return 0; }
+
+  NS_DECL_ISUPPORTS
+
+ protected:
+  virtual ~SocketHandler() = default;
+
+ private:
+  SocketTransportServiceTest* test_;
+};
+
+NS_IMPL_ISUPPORTS0(SocketHandler)
+
+void SocketTransportServiceTest::SetUp() {
+  MtransportTest::SetUp();
+
+  // Get the transport service as a dispatch target
+  nsresult rv;
+  target_ = do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &rv);
+  ASSERT_TRUE(NS_SUCCEEDED(rv));
+
+  // Get the transport service as a transport service
+  stservice_ = do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &rv);
+  ASSERT_TRUE(NS_SUCCEEDED(rv));
+
+  // Create a loopback pipe
+  PRStatus status = PR_CreatePipe(&readpipe_, &writepipe_);
+  ASSERT_EQ(status, PR_SUCCESS);
+
+  // Register ourselves as a listener for the read side of the
+  // socket. The registration has to happen on the STS thread,
+  // hence this event stuff.
+  rv = target_->Dispatch(new RegisterEvent(this), 0);
+  ASSERT_TRUE(NS_SUCCEEDED(rv));
+  ASSERT_TRUE_WAIT(registered_, 10000);
+}
+
+void SocketTransportServiceTest::RegisterHandler() {
+  nsresult rv;
+
+  rv = stservice_->AttachSocket(readpipe_, new SocketHandler(this));
+  ASSERT_TRUE(NS_SUCCEEDED(rv));
+
+  registered_ = true;
+}
+
+void SocketTransportServiceTest::SendEvent() {
+  nsresult rv;
+
+  rv = target_->Dispatch(new EventReceived(this), 0);
+  ASSERT_TRUE(NS_SUCCEEDED(rv));
+  ASSERT_TRUE_WAIT(Received() == 1, 10000);
+}
+
+void SocketTransportServiceTest::SendPacket() {
+  unsigned char buffer[1024];
+  memset(buffer, 0, sizeof(buffer));
+
+  int32_t status = PR_Write(writepipe_, buffer, sizeof(buffer));
+  uint32_t size = status & 0xffff;
+  ASSERT_EQ(sizeof(buffer), size);
+}
+
+// The unit tests themselves
+TEST_F(SocketTransportServiceTest, SendEvent) { SendEvent(); }
+
+TEST_F(SocketTransportServiceTest, SendPacket) { SendPacket(); }
+
+}  // end namespace
diff --git a/dom/media/webrtc/transport/test/stunserver.cpp b/dom/media/webrtc/transport/test/stunserver.cpp
new file mode 100644
index 0000000000..b5fce21e19
--- /dev/null
+++ b/dom/media/webrtc/transport/test/stunserver.cpp
@@ -0,0 +1,652 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+/*
+Original code from nICEr and nrappkit.
+
+nICEr copyright:
+
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+
+nrappkit copyright:
+
+   Copyright (C) 2001-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   ekr@rtfm.com  Thu Dec 20 20:14:49 2001
+*/
+#include "logging.h"
+#include "mozilla/UniquePtr.h"
+#include "mozilla/Unused.h"
+#include "mediapacket.h"
+
+// mozilla/utils.h defines this as well
+#ifdef UNIMPLEMENTED
+#  undef UNIMPLEMENTED
+#endif
+
+extern "C" {
+#include "nr_api.h"
+#include "async_wait.h"
+#include "async_timer.h"
+#include "nr_socket.h"
+#include "nr_socket_local.h"
+#include "transport_addr.h"
+#include "stun_util.h"
+#include "registry.h"
+#include "nr_socket_buffered_stun.h"
+}
+
+#include "stunserver.h"
+
+#include <string>
+
+MOZ_MTLOG_MODULE("stunserver");
+
+namespace mozilla {
+
+// Wrapper nr_socket which allows us to lie to the stun server about the
+// IP address.
+struct nr_socket_wrapped {
+  nr_socket* sock_;
+  nr_transport_addr addr_;
+};
+
+static int nr_socket_wrapped_destroy(void** objp) {
+  if (!objp || !*objp) return 0;
+
+  nr_socket_wrapped* wrapped = static_cast<nr_socket_wrapped*>(*objp);
+  *objp = nullptr;
+
+  delete wrapped;
+
+  return 0;
+}
+
+static int nr_socket_wrapped_sendto(void* obj, const void* msg, size_t len,
+                                    int flags, const nr_transport_addr* addr) {
+  nr_socket_wrapped* wrapped = static_cast<nr_socket_wrapped*>(obj);
+
+  return nr_socket_sendto(wrapped->sock_, msg, len, flags, &wrapped->addr_);
+}
+
+static int nr_socket_wrapped_recvfrom(void* obj, void* restrict buf,
+                                      size_t maxlen, size_t* len, int flags,
+                                      nr_transport_addr* addr) {
+  nr_socket_wrapped* wrapped = static_cast<nr_socket_wrapped*>(obj);
+
+  return nr_socket_recvfrom(wrapped->sock_, buf, maxlen, len, flags, addr);
+}
+
+static int nr_socket_wrapped_getfd(void* obj, NR_SOCKET* fd) {
+  nr_socket_wrapped* wrapped = static_cast<nr_socket_wrapped*>(obj);
+
+  return nr_socket_getfd(wrapped->sock_, fd);
+}
+
+static int nr_socket_wrapped_getaddr(void* obj, nr_transport_addr* addrp) {
+  nr_socket_wrapped* wrapped = static_cast<nr_socket_wrapped*>(obj);
+
+  return nr_socket_getaddr(wrapped->sock_, addrp);
+}
+
+static int nr_socket_wrapped_close(void* obj) { MOZ_CRASH(); }
+
+static int nr_socket_wrapped_set_send_addr(nr_socket* sock,
+                                           nr_transport_addr* addr) {
+  nr_socket_wrapped* wrapped = static_cast<nr_socket_wrapped*>(sock->obj);
+
+  return nr_transport_addr_copy(&wrapped->addr_, addr);
+}
+
+static nr_socket_vtbl nr_socket_wrapped_vtbl = {2,
+                                                nr_socket_wrapped_destroy,
+                                                nr_socket_wrapped_sendto,
+                                                nr_socket_wrapped_recvfrom,
+                                                nr_socket_wrapped_getfd,
+                                                nr_socket_wrapped_getaddr,
+                                                nullptr,
+                                                nullptr,
+                                                nullptr,
+                                                nr_socket_wrapped_close,
+                                                nullptr,
+                                                nullptr};
+
+int nr_socket_wrapped_create(nr_socket* inner, nr_socket** outp) {
+  auto wrapped = MakeUnique<nr_socket_wrapped>();
+
+  wrapped->sock_ = inner;
+
+  int r = nr_socket_create_int(wrapped.get(), &nr_socket_wrapped_vtbl, outp);
+  if (r) return r;
+
+  Unused << wrapped.release();
+  return 0;
+}
+
+// Instance static.
+// Note: Calling Create() at static init time is not going to be safe, since
+// we have no reason to expect this will be initted to a nullptr yet.
+TestStunServer* TestStunServer::instance;
+TestStunTcpServer* TestStunTcpServer::instance;
+TestStunServer* TestStunServer::instance6;
+TestStunTcpServer* TestStunTcpServer::instance6;
+uint16_t TestStunServer::instance_port = 3478;
+uint16_t TestStunTcpServer::instance_port = 3478;
+
+TestStunServer::~TestStunServer() {
+  // TODO(ekr@rtfm.com): Put this on the right thread.
+
+  // Unhook callback from our listen socket.
+  if (listen_sock_) {
+    NR_SOCKET fd;
+    if (!nr_socket_getfd(listen_sock_, &fd)) {
+      NR_ASYNC_CANCEL(fd, NR_ASYNC_WAIT_READ);
+    }
+  }
+
+  // Free up stun context and network resources
+  nr_stun_server_ctx_destroy(&stun_server_);
+  nr_socket_destroy(&listen_sock_);
+  nr_socket_destroy(&send_sock_);
+
+  // Make sure we aren't still waiting on a deferred response timer to pop
+  if (timer_handle_) NR_async_timer_cancel(timer_handle_);
+
+  delete response_addr_;
+}
+
+int TestStunServer::SetInternalPort(nr_local_addr* addr, uint16_t port) {
+  if (nr_transport_addr_set_port(&addr->addr, port)) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't set port");
+    return R_INTERNAL;
+  }
+
+  if (nr_transport_addr_fmt_addr_string(&addr->addr)) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't re-set addr string");
+    return R_INTERNAL;
+  }
+
+  return 0;
+}
+
+int TestStunServer::TryOpenListenSocket(nr_local_addr* addr, uint16_t port) {
+  int r = SetInternalPort(addr, port);
+
+  if (r) return r;
+
+  if (nr_socket_local_create(nullptr, &addr->addr, &listen_sock_)) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't create listen socket");
+    return R_ALREADY;
+  }
+
+  return 0;
+}
+
+static int addressFamilyToIpVersion(int address_family) {
+  switch (address_family) {
+    case AF_INET:
+      return NR_IPV4;
+    case AF_INET6:
+      return NR_IPV6;
+    default:
+      MOZ_CRASH();
+  }
+  return NR_IPV4;
+}
+
+int TestStunServer::Initialize(int address_family) {
+  static const size_t max_addrs = 100;
+  nr_local_addr addrs[max_addrs];
+  int addr_ct;
+  int r;
+  int i;
+
+  r = nr_stun_find_local_addresses(addrs, max_addrs, &addr_ct);
+  if (r) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't retrieve addresses");
+    return R_INTERNAL;
+  }
+
+  // removes duplicates and, based on prefs, loopback and link_local addrs
+  r = nr_stun_filter_local_addresses(addrs, &addr_ct);
+  if (r) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't filter addresses");
+    return R_INTERNAL;
+  }
+
+  if (addr_ct < 1) {
+    MOZ_MTLOG(ML_ERROR, "No local addresses");
+    return R_INTERNAL;
+  }
+
+  for (i = 0; i < addr_ct; ++i) {
+    if (addrs[i].addr.ip_version == addressFamilyToIpVersion(address_family)) {
+      break;
+    }
+  }
+
+  if (i == addr_ct) {
+    MOZ_MTLOG(ML_ERROR, "No local addresses of the configured IP version");
+    return R_INTERNAL;
+  }
+
+  int tries = 100;
+  while (tries--) {
+    // Bind on configured port (default 3478)
+    r = TryOpenListenSocket(&addrs[i], instance_port);
+    // We interpret R_ALREADY to mean the addr is probably in use. Try another.
+    // Otherwise, it either worked or it didn't, and we check below.
+    if (r != R_ALREADY) {
+      break;
+    }
+    ++instance_port;
+  }
+
+  if (r) {
+    return R_INTERNAL;
+  }
+
+  r = nr_socket_wrapped_create(listen_sock_, &send_sock_);
+  if (r) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't create send socket");
+    return R_INTERNAL;
+  }
+
+  r = nr_stun_server_ctx_create(const_cast<char*>("Test STUN server"),
+                                &stun_server_);
+  if (r) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't create STUN server");
+    return R_INTERNAL;
+  }
+
+  // Cache the address and port.
+  char addr_string[INET6_ADDRSTRLEN];
+  r = nr_transport_addr_get_addrstring(&addrs[i].addr, addr_string,
+                                       sizeof(addr_string));
+  if (r) {
+    MOZ_MTLOG(ML_ERROR,
+              "Failed to convert listen addr to a string representation");
+    return R_INTERNAL;
+  }
+
+  listen_addr_ = addr_string;
+  listen_port_ = instance_port;
+
+  return 0;
+}
+
+UniquePtr<TestStunServer> TestStunServer::Create(int address_family) {
+  NR_reg_init(NR_REG_MODE_LOCAL);
+
+  UniquePtr<TestStunServer> server(new TestStunServer());
+
+  if (server->Initialize(address_family)) return nullptr;
+
+  NR_SOCKET fd;
+  int r = nr_socket_getfd(server->listen_sock_, &fd);
+  if (r) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't get fd");
+    return nullptr;
+  }
+
+  NR_ASYNC_WAIT(fd, NR_ASYNC_WAIT_READ, &TestStunServer::readable_cb,
+                server.get());
+
+  return server;
+}
+
+void TestStunServer::ConfigurePort(uint16_t port) { instance_port = port; }
+
+TestStunServer* TestStunServer::GetInstance(int address_family) {
+  switch (address_family) {
+    case AF_INET:
+      if (!instance) instance = Create(address_family).release();
+
+      MOZ_ASSERT(instance);
+      return instance;
+    case AF_INET6:
+      if (!instance6) instance6 = Create(address_family).release();
+
+      return instance6;
+    default:
+      MOZ_CRASH();
+  }
+}
+
+void TestStunServer::ShutdownInstance() {
+  delete instance;
+  instance = nullptr;
+  delete instance6;
+  instance6 = nullptr;
+}
+
+struct DeferredStunOperation {
+  DeferredStunOperation(TestStunServer* server, const char* data, size_t len,
+                        nr_transport_addr* addr, nr_socket* sock)
+      : server_(server), buffer_(), sock_(sock) {
+    buffer_.Copy(reinterpret_cast<const uint8_t*>(data), len);
+    nr_transport_addr_copy(&addr_, addr);
+  }
+
+  TestStunServer* server_;
+  MediaPacket buffer_;
+  nr_transport_addr addr_;
+  nr_socket* sock_;
+};
+
+void TestStunServer::Process(const uint8_t* msg, size_t len,
+                             nr_transport_addr* addr, nr_socket* sock) {
+  if (!sock) {
+    sock = send_sock_;
+  }
+
+  // Set the wrapped address so that the response goes to the right place.
+  nr_socket_wrapped_set_send_addr(sock, addr);
+
+  nr_stun_server_process_request(
+      stun_server_, sock, const_cast<char*>(reinterpret_cast<const char*>(msg)),
+      len, response_addr_ ? response_addr_ : addr, NR_STUN_AUTH_RULE_OPTIONAL);
+}
+
+void TestStunServer::process_cb(NR_SOCKET s, int how, void* cb_arg) {
+  DeferredStunOperation* op = static_cast<DeferredStunOperation*>(cb_arg);
+  op->server_->timer_handle_ = nullptr;
+  op->server_->Process(op->buffer_.data(), op->buffer_.len(), &op->addr_,
+                       op->sock_);
+
+  delete op;
+}
+
+nr_socket* TestStunServer::GetReceivingSocket(NR_SOCKET s) {
+  return listen_sock_;
+}
+
+nr_socket* TestStunServer::GetSendingSocket(nr_socket* sock) {
+  return send_sock_;
+}
+
+void TestStunServer::readable_cb(NR_SOCKET s, int how, void* cb_arg) {
+  TestStunServer* server = static_cast<TestStunServer*>(cb_arg);
+
+  char message[max_stun_message_size];
+  size_t message_len;
+  nr_transport_addr addr;
+  nr_socket* recv_sock = server->GetReceivingSocket(s);
+  if (!recv_sock) {
+    MOZ_MTLOG(ML_ERROR, "Failed to lookup receiving socket");
+    return;
+  }
+  nr_socket* send_sock = server->GetSendingSocket(recv_sock);
+
+  /* Re-arm. */
+  NR_ASYNC_WAIT(s, NR_ASYNC_WAIT_READ, &TestStunServer::readable_cb, server);
+
+  if (nr_socket_recvfrom(recv_sock, message, sizeof(message), &message_len, 0,
+                         &addr)) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't read STUN message");
+    return;
+  }
+
+  MOZ_MTLOG(ML_DEBUG, "Received data of length " << message_len);
+
+  // If we have initial dropping set, check at this point.
+  std::string key(addr.as_string);
+
+  if (server->received_ct_.count(key) == 0) {
+    server->received_ct_[key] = 0;
+  }
+
+  ++server->received_ct_[key];
+
+  if (!server->active_ || (server->received_ct_[key] <= server->initial_ct_)) {
+    MOZ_MTLOG(ML_DEBUG, "Dropping message #" << server->received_ct_[key]
+                                             << " from " << key);
+    return;
+  }
+
+  if (server->delay_ms_) {
+    NR_ASYNC_TIMER_SET(server->delay_ms_, process_cb,
+                       new DeferredStunOperation(server, message, message_len,
+                                                 &addr, send_sock),
+                       &server->timer_handle_);
+  } else {
+    server->Process(reinterpret_cast<const uint8_t*>(message), message_len,
+                    &addr, send_sock);
+  }
+}
+
+void TestStunServer::SetActive(bool active) { active_ = active; }
+
+void TestStunServer::SetDelay(uint32_t delay_ms) { delay_ms_ = delay_ms; }
+
+void TestStunServer::SetDropInitialPackets(uint32_t count) {
+  initial_ct_ = count;
+}
+
+nsresult TestStunServer::SetResponseAddr(nr_transport_addr* addr) {
+  delete response_addr_;
+
+  response_addr_ = new nr_transport_addr();
+
+  int r = nr_transport_addr_copy(response_addr_, addr);
+  if (r) return NS_ERROR_FAILURE;
+
+  return NS_OK;
+}
+
+nsresult TestStunServer::SetResponseAddr(const std::string& addr,
+                                         uint16_t port) {
+  nr_transport_addr addr2;
+
+  int r =
+      nr_str_port_to_transport_addr(addr.c_str(), port, IPPROTO_UDP, &addr2);
+  if (r) return NS_ERROR_FAILURE;
+
+  return SetResponseAddr(&addr2);
+}
+
+void TestStunServer::Reset() {
+  delay_ms_ = 0;
+  if (timer_handle_) {
+    NR_async_timer_cancel(timer_handle_);
+    timer_handle_ = nullptr;
+  }
+  delete response_addr_;
+  response_addr_ = nullptr;
+  received_ct_.clear();
+}
+
+// TestStunTcpServer
+
+void TestStunTcpServer::ConfigurePort(uint16_t port) { instance_port = port; }
+
+TestStunTcpServer* TestStunTcpServer::GetInstance(int address_family) {
+  switch (address_family) {
+    case AF_INET:
+      if (!instance) instance = Create(address_family).release();
+
+      MOZ_ASSERT(instance);
+      return instance;
+    case AF_INET6:
+      if (!instance6) instance6 = Create(address_family).release();
+
+      return instance6;
+    default:
+      MOZ_CRASH();
+  }
+}
+
+void TestStunTcpServer::ShutdownInstance() {
+  delete instance;
+  instance = nullptr;
+  delete instance6;
+  instance6 = nullptr;
+}
+
+int TestStunTcpServer::TryOpenListenSocket(nr_local_addr* addr, uint16_t port) {
+  addr->addr.protocol = IPPROTO_TCP;
+
+  int r = SetInternalPort(addr, port);
+
+  if (r) return r;
+
+  nr_socket* sock;
+  if (nr_socket_local_create(nullptr, &addr->addr, &sock)) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't create listen tcp socket");
+    return R_ALREADY;
+  }
+
+  if (nr_socket_buffered_stun_create(sock, 2048, TURN_TCP_FRAMING,
+                                     &listen_sock_)) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't create listen tcp socket");
+    return R_ALREADY;
+  }
+
+  if (nr_socket_listen(listen_sock_, 10)) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't listen on socket");
+    return R_ALREADY;
+  }
+
+  return 0;
+}
+
+nr_socket* TestStunTcpServer::GetReceivingSocket(NR_SOCKET s) {
+  return connections_[s];
+}
+
+nr_socket* TestStunTcpServer::GetSendingSocket(nr_socket* sock) { return sock; }
+
+void TestStunTcpServer::accept_cb(NR_SOCKET s, int how, void* cb_arg) {
+  TestStunTcpServer* server = static_cast<TestStunTcpServer*>(cb_arg);
+  nr_socket *newsock, *bufsock, *wrapsock;
+  nr_transport_addr remote_addr;
+  NR_SOCKET fd;
+
+  /* rearm */
+  NR_ASYNC_WAIT(s, NR_ASYNC_WAIT_READ, &TestStunTcpServer::accept_cb, cb_arg);
+
+  /* accept */
+  if (nr_socket_accept(server->listen_sock_, &remote_addr, &newsock)) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't accept incoming tcp connection");
+    return;
+  }
+
+  if (nr_socket_buffered_stun_create(newsock, 2048, TURN_TCP_FRAMING,
+                                     &bufsock)) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't create connected tcp socket");
+    nr_socket_destroy(&newsock);
+    return;
+  }
+
+  nr_socket_buffered_set_connected_to(bufsock, &remote_addr);
+
+  if (nr_socket_wrapped_create(bufsock, &wrapsock)) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't wrap connected tcp socket");
+    nr_socket_destroy(&bufsock);
+    return;
+  }
+
+  if (nr_socket_getfd(wrapsock, &fd)) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't get fd from connected tcp socket");
+    nr_socket_destroy(&wrapsock);
+    return;
+  }
+
+  server->connections_[fd] = wrapsock;
+
+  NR_ASYNC_WAIT(fd, NR_ASYNC_WAIT_READ, &TestStunServer::readable_cb, server);
+}
+
+UniquePtr<TestStunTcpServer> TestStunTcpServer::Create(int address_family) {
+  NR_reg_init(NR_REG_MODE_LOCAL);
+
+  UniquePtr<TestStunTcpServer> server(new TestStunTcpServer());
+
+  if (server->Initialize(address_family)) {
+    return nullptr;
+  }
+
+  NR_SOCKET fd;
+  if (nr_socket_getfd(server->listen_sock_, &fd)) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't get tcp fd");
+    return nullptr;
+  }
+
+  NR_ASYNC_WAIT(fd, NR_ASYNC_WAIT_READ, &TestStunTcpServer::accept_cb,
+                server.get());
+
+  return server;
+}
+
+TestStunTcpServer::~TestStunTcpServer() {
+  for (auto it = connections_.begin(); it != connections_.end();) {
+    NR_ASYNC_CANCEL(it->first, NR_ASYNC_WAIT_READ);
+    nr_socket_destroy(&it->second);
+    connections_.erase(it++);
+  }
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/transport/test/stunserver.h b/dom/media/webrtc/transport/test/stunserver.h
new file mode 100644
index 0000000000..4903cb89ad
--- /dev/null
+++ b/dom/media/webrtc/transport/test/stunserver.h
@@ -0,0 +1,123 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+#ifndef stunserver_h__
+#define stunserver_h__
+
+#include <map>
+#include <string>
+#include "nsError.h"
+#include "mozilla/UniquePtr.h"
+
+typedef struct nr_stun_server_ctx_ nr_stun_server_ctx;
+typedef struct nr_socket_ nr_socket;
+typedef struct nr_local_addr_ nr_local_addr;
+
+namespace mozilla {
+
+class TestStunServer {
+ public:
+  // Generally, you should only call API in this class from the same thread that
+  // the initial |GetInstance| call was made from.
+  static TestStunServer* GetInstance(int address_family = AF_INET);
+  static void ShutdownInstance();
+  // |ConfigurePort| will only have an effect if called before the first call
+  // to |GetInstance| (possibly following a |ShutdownInstance| call)
+  static void ConfigurePort(uint16_t port);
+  // AF_INET, AF_INET6
+  static UniquePtr<TestStunServer> Create(int address_family);
+
+  virtual ~TestStunServer();
+
+  void SetActive(bool active);
+  void SetDelay(uint32_t delay_ms);
+  void SetDropInitialPackets(uint32_t count);
+  const std::string& addr() const { return listen_addr_; }
+  uint16_t port() const { return listen_port_; }
+
+  // These should only be called from the same thread as the initial
+  // |GetInstance| call.
+  nsresult SetResponseAddr(nr_transport_addr* addr);
+  nsresult SetResponseAddr(const std::string& addr, uint16_t port);
+
+  void Reset();
+
+  static const size_t max_stun_message_size = 4096;
+
+  virtual nr_socket* GetReceivingSocket(NR_SOCKET s);
+  virtual nr_socket* GetSendingSocket(nr_socket* sock);
+
+ protected:
+  TestStunServer()
+      : listen_port_(0),
+        listen_sock_(nullptr),
+        send_sock_(nullptr),
+        stun_server_(nullptr),
+        active_(true),
+        delay_ms_(0),
+        initial_ct_(0),
+        response_addr_(nullptr),
+        timer_handle_(nullptr) {}
+
+  int SetInternalPort(nr_local_addr* addr, uint16_t port);
+  int Initialize(int address_family);
+
+  static void readable_cb(NR_SOCKET sock, int how, void* cb_arg);
+
+ private:
+  void Process(const uint8_t* msg, size_t len, nr_transport_addr* addr_in,
+               nr_socket* sock);
+  virtual int TryOpenListenSocket(nr_local_addr* addr, uint16_t port);
+  static void process_cb(NR_SOCKET sock, int how, void* cb_arg);
+
+ protected:
+  std::string listen_addr_;
+  uint16_t listen_port_;
+  nr_socket* listen_sock_;
+  nr_socket* send_sock_;
+  nr_stun_server_ctx* stun_server_;
+
+ private:
+  bool active_;
+  uint32_t delay_ms_;
+  uint32_t initial_ct_;
+  nr_transport_addr* response_addr_;
+  void* timer_handle_;
+  std::map<std::string, uint32_t> received_ct_;
+
+  static TestStunServer* instance;
+  static TestStunServer* instance6;
+  static uint16_t instance_port;
+};
+
+class TestStunTcpServer : public TestStunServer {
+ public:
+  static TestStunTcpServer* GetInstance(int address_family);
+  static void ShutdownInstance();
+  static void ConfigurePort(uint16_t port);
+  virtual ~TestStunTcpServer();
+
+  virtual nr_socket* GetReceivingSocket(NR_SOCKET s);
+  virtual nr_socket* GetSendingSocket(nr_socket* sock);
+
+ protected:
+  TestStunTcpServer() = default;
+  static void accept_cb(NR_SOCKET sock, int how, void* cb_arg);
+
+ private:
+  virtual int TryOpenListenSocket(nr_local_addr* addr, uint16_t port);
+  static UniquePtr<TestStunTcpServer> Create(int address_family);
+
+  static TestStunTcpServer* instance;
+  static TestStunTcpServer* instance6;
+  static uint16_t instance_port;
+
+  std::map<NR_SOCKET, nr_socket*> connections_;
+};
+}  // End of namespace mozilla
+#endif
diff --git a/dom/media/webrtc/transport/test/test_nr_socket_ice_unittest.cpp b/dom/media/webrtc/transport/test/test_nr_socket_ice_unittest.cpp
new file mode 100644
index 0000000000..b55b05f10c
--- /dev/null
+++ b/dom/media/webrtc/transport/test/test_nr_socket_ice_unittest.cpp
@@ -0,0 +1,409 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Some of this code is taken from nricectx.cpp and nricemediastream.cpp
+// which in turn contains code cut-and-pasted from nICEr. Copyright is:
+
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include "gtest/gtest.h"
+#include "gtest_utils.h"
+
+extern "C" {
+#include "ice_ctx.h"
+#include "ice_peer_ctx.h"
+#include "nICEr/src/net/transport_addr.h"
+}
+
+#include "mtransport_test_utils.h"
+#include "nricectx.h"
+#include "nricemediastream.h"
+#include "runnable_utils.h"
+#include "test_nr_socket.h"
+
+namespace mozilla {
+
+static unsigned int kDefaultTimeout = 7000;
+
+class IcePeer {
+ public:
+  IcePeer(const char* name, TestNat* nat, UINT4 flags,
+          MtransportTestUtils* test_utils)
+      : name_(name),
+        ice_checking_(false),
+        ice_connected_(false),
+        ice_disconnected_(false),
+        gather_cb_(false),
+        stream_ready_(false),
+        stream_failed_(false),
+        ice_ctx_(nullptr),
+        peer_ctx_(nullptr),
+        nat_(nat),
+        test_utils_(test_utils) {
+    nr_ice_ctx_create(const_cast<char*>(name_.c_str()), flags, &ice_ctx_);
+
+    if (nat_) {
+      nr_socket_factory* factory;
+      nat_->create_socket_factory(&factory);
+      nr_ice_ctx_set_socket_factory(ice_ctx_, factory);
+    }
+
+    // Create the handler objects
+    ice_handler_vtbl_ = new nr_ice_handler_vtbl();
+    ice_handler_vtbl_->select_pair = &IcePeer::select_pair;
+    ice_handler_vtbl_->stream_ready = &IcePeer::stream_ready;
+    ice_handler_vtbl_->stream_failed = &IcePeer::stream_failed;
+    ice_handler_vtbl_->ice_connected = &IcePeer::ice_connected;
+    ice_handler_vtbl_->msg_recvd = &IcePeer::msg_recvd;
+    ice_handler_vtbl_->ice_checking = &IcePeer::ice_checking;
+    ice_handler_vtbl_->ice_disconnected = &IcePeer::ice_disconnected;
+
+    ice_handler_ = new nr_ice_handler();
+    ice_handler_->vtbl = ice_handler_vtbl_;
+    ice_handler_->obj = this;
+
+    nr_ice_peer_ctx_create(ice_ctx_, ice_handler_,
+                           const_cast<char*>(name_.c_str()), &peer_ctx_);
+
+    nr_ice_add_media_stream(ice_ctx_, const_cast<char*>(name_.c_str()), "ufrag",
+                            "pass", 2, &ice_media_stream_);
+    EXPECT_EQ(2UL, GetStreamAttributes().size());
+
+    nr_ice_media_stream_initialize(ice_ctx_, ice_media_stream_);
+  }
+
+  virtual ~IcePeer() { Destroy(); }
+
+  void Destroy() {
+    test_utils_->SyncDispatchToSTS(WrapRunnable(this, &IcePeer::Destroy_s));
+  }
+
+  void Destroy_s() {
+    nr_ice_peer_ctx_destroy(&peer_ctx_);
+    delete ice_handler_;
+    delete ice_handler_vtbl_;
+    nr_ice_ctx_destroy(&ice_ctx_);
+  }
+
+  void Gather(bool default_route_only = false) {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &IcePeer::Gather_s, default_route_only));
+  }
+
+  void Gather_s(bool default_route_only = false) {
+    int r = nr_ice_gather(ice_ctx_, &IcePeer::gather_cb, this);
+    ASSERT_TRUE(r == 0 || r == R_WOULDBLOCK);
+  }
+
+  std::vector<std::string> GetStreamAttributes() {
+    std::vector<std::string> attributes;
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnableRet(&attributes, this, &IcePeer::GetStreamAttributes_s));
+    return attributes;
+  }
+
+  std::vector<std::string> GetStreamAttributes_s() {
+    char** attrs = nullptr;
+    int attrct;
+    std::vector<std::string> ret;
+
+    int r =
+        nr_ice_media_stream_get_attributes(ice_media_stream_, &attrs, &attrct);
+    EXPECT_EQ(0, r);
+
+    for (int i = 0; i < attrct; i++) {
+      ret.push_back(std::string(attrs[i]));
+      RFREE(attrs[i]);
+    }
+    RFREE(attrs);
+
+    return ret;
+  }
+
+  std::vector<std::string> GetGlobalAttributes() {
+    std::vector<std::string> attributes;
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnableRet(&attributes, this, &IcePeer::GetGlobalAttributes_s));
+    return attributes;
+  }
+
+  std::vector<std::string> GetGlobalAttributes_s() {
+    char** attrs = nullptr;
+    int attrct;
+    std::vector<std::string> ret;
+
+    nr_ice_get_global_attributes(ice_ctx_, &attrs, &attrct);
+
+    for (int i = 0; i < attrct; i++) {
+      ret.push_back(std::string(attrs[i]));
+      RFREE(attrs[i]);
+    }
+    RFREE(attrs);
+
+    return ret;
+  }
+
+  void ParseGlobalAttributes(std::vector<std::string> attrs) {
+    std::vector<char*> attrs_in;
+    attrs_in.reserve(attrs.size());
+    for (auto& attr : attrs) {
+      attrs_in.push_back(const_cast<char*>(attr.c_str()));
+    }
+
+    int r = nr_ice_peer_ctx_parse_global_attributes(
+        peer_ctx_, attrs_in.empty() ? nullptr : &attrs_in[0], attrs_in.size());
+    ASSERT_EQ(0, r);
+  }
+
+  void SetControlling(bool controlling) {
+    peer_ctx_->controlling = controlling ? 1 : 0;
+  }
+
+  void SetRemoteAttributes(std::vector<std::string> attributes) {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &IcePeer::SetRemoteAttributes_s, attributes));
+  }
+
+  void SetRemoteAttributes_s(std::vector<std::string> attributes) {
+    int r;
+
+    std::vector<char*> attrs;
+    attrs.reserve(attributes.size());
+    for (auto& attr : attributes) {
+      attrs.push_back(const_cast<char*>(attr.c_str()));
+    }
+
+    if (!attrs.empty()) {
+      r = nr_ice_peer_ctx_parse_stream_attributes(peer_ctx_, ice_media_stream_,
+                                                  &attrs[0], attrs.size());
+      ASSERT_EQ(0, r);
+    }
+  }
+
+  void StartChecks() {
+    test_utils_->SyncDispatchToSTS(WrapRunnable(this, &IcePeer::StartChecks_s));
+  }
+
+  void StartChecks_s() {
+    int r = nr_ice_peer_ctx_pair_candidates(peer_ctx_);
+    ASSERT_EQ(0, r);
+
+    r = nr_ice_peer_ctx_start_checks2(peer_ctx_, 1);
+    ASSERT_EQ(0, r);
+  }
+
+  // Handler callbacks
+  static int select_pair(void* obj, nr_ice_media_stream* stream,
+                         int component_id, nr_ice_cand_pair** potentials,
+                         int potential_ct) {
+    return 0;
+  }
+
+  static int stream_ready(void* obj, nr_ice_media_stream* stream) {
+    IcePeer* peer = static_cast<IcePeer*>(obj);
+    peer->stream_ready_ = true;
+    return 0;
+  }
+
+  static int stream_failed(void* obj, nr_ice_media_stream* stream) {
+    IcePeer* peer = static_cast<IcePeer*>(obj);
+    peer->stream_failed_ = true;
+    return 0;
+  }
+
+  static int ice_checking(void* obj, nr_ice_peer_ctx* pctx) {
+    IcePeer* peer = static_cast<IcePeer*>(obj);
+    peer->ice_checking_ = true;
+    return 0;
+  }
+
+  static int ice_connected(void* obj, nr_ice_peer_ctx* pctx) {
+    IcePeer* peer = static_cast<IcePeer*>(obj);
+    peer->ice_connected_ = true;
+    return 0;
+  }
+
+  static int ice_disconnected(void* obj, nr_ice_peer_ctx* pctx) {
+    IcePeer* peer = static_cast<IcePeer*>(obj);
+    peer->ice_disconnected_ = true;
+    return 0;
+  }
+
+  static int msg_recvd(void* obj, nr_ice_peer_ctx* pctx,
+                       nr_ice_media_stream* stream, int component_id,
+                       UCHAR* msg, int len) {
+    return 0;
+  }
+
+  static void gather_cb(NR_SOCKET s, int h, void* arg) {
+    IcePeer* peer = static_cast<IcePeer*>(arg);
+    peer->gather_cb_ = true;
+  }
+
+  std::string name_;
+
+  bool ice_checking_;
+  bool ice_connected_;
+  bool ice_disconnected_;
+  bool gather_cb_;
+  bool stream_ready_;
+  bool stream_failed_;
+
+  nr_ice_ctx* ice_ctx_;
+  nr_ice_handler* ice_handler_;
+  nr_ice_handler_vtbl* ice_handler_vtbl_;
+  nr_ice_media_stream* ice_media_stream_;
+  nr_ice_peer_ctx* peer_ctx_;
+  TestNat* nat_;
+  MtransportTestUtils* test_utils_;
+};
+
+class TestNrSocketIceUnitTest : public ::testing::Test {
+ public:
+  void SetUp() override {
+    NSS_NoDB_Init(nullptr);
+    NSS_SetDomesticPolicy();
+
+    test_utils_ = new MtransportTestUtils();
+    test_utils2_ = new MtransportTestUtils();
+
+    NrIceCtx::InitializeGlobals(NrIceCtx::GlobalConfig());
+  }
+
+  void TearDown() override {
+    delete test_utils_;
+    delete test_utils2_;
+  }
+
+  MtransportTestUtils* test_utils_;
+  MtransportTestUtils* test_utils2_;
+};
+
+TEST_F(TestNrSocketIceUnitTest, TestIcePeer) {
+  IcePeer peer("IcePeer", nullptr, NR_ICE_CTX_FLAGS_AGGRESSIVE_NOMINATION,
+               test_utils_);
+  ASSERT_NE(peer.ice_ctx_, nullptr);
+  ASSERT_NE(peer.peer_ctx_, nullptr);
+  ASSERT_NE(peer.ice_media_stream_, nullptr);
+  ASSERT_EQ(2UL, peer.GetStreamAttributes().size())
+      << "Should have ice-ufrag and ice-pwd";
+  peer.Gather();
+  ASSERT_LT(2UL, peer.GetStreamAttributes().size())
+      << "Should have ice-ufrag, ice-pwd, and at least one candidate.";
+}
+
+TEST_F(TestNrSocketIceUnitTest, TestIcePeersNoNAT) {
+  IcePeer peer("IcePeer", nullptr, NR_ICE_CTX_FLAGS_AGGRESSIVE_NOMINATION,
+               test_utils_);
+  IcePeer peer2("IcePeer2", nullptr, NR_ICE_CTX_FLAGS_AGGRESSIVE_NOMINATION,
+                test_utils2_);
+  peer.SetControlling(true);
+  peer2.SetControlling(false);
+
+  peer.Gather();
+  peer2.Gather();
+  std::vector<std::string> attrs = peer.GetGlobalAttributes();
+  peer2.ParseGlobalAttributes(attrs);
+  std::vector<std::string> attributes = peer.GetStreamAttributes();
+  peer2.SetRemoteAttributes(attributes);
+
+  attrs = peer2.GetGlobalAttributes();
+  peer.ParseGlobalAttributes(attrs);
+  attributes = peer2.GetStreamAttributes();
+  peer.SetRemoteAttributes(attributes);
+  peer2.StartChecks();
+  peer.StartChecks();
+
+  ASSERT_TRUE_WAIT(peer.ice_connected_, kDefaultTimeout);
+  ASSERT_TRUE_WAIT(peer2.ice_connected_, kDefaultTimeout);
+}
+
+TEST_F(TestNrSocketIceUnitTest, TestIcePeersPacketLoss) {
+  IcePeer peer("IcePeer", nullptr, NR_ICE_CTX_FLAGS_AGGRESSIVE_NOMINATION,
+               test_utils_);
+
+  RefPtr<TestNat> nat(new TestNat);
+  class NatDelegate : public TestNat::NatDelegate {
+   public:
+    NatDelegate() : messages(0) {}
+
+    int on_read(TestNat* nat, void* buf, size_t maxlen, size_t* len) override {
+      return 0;
+    }
+
+    int on_sendto(TestNat* nat, const void* msg, size_t len, int flags,
+                  const nr_transport_addr* to) override {
+      ++messages;
+      // 25% packet loss
+      if (messages % 4 == 0) {
+        return 1;
+      }
+      return 0;
+    }
+
+    int on_write(TestNat* nat, const void* msg, size_t len,
+                 size_t* written) override {
+      return 0;
+    }
+
+    int messages;
+  } delegate;
+  nat->nat_delegate_ = &delegate;
+
+  IcePeer peer2("IcePeer2", nat, NR_ICE_CTX_FLAGS_AGGRESSIVE_NOMINATION,
+                test_utils2_);
+  peer.SetControlling(true);
+  peer2.SetControlling(false);
+
+  peer.Gather();
+  peer2.Gather();
+  std::vector<std::string> attrs = peer.GetGlobalAttributes();
+  peer2.ParseGlobalAttributes(attrs);
+  std::vector<std::string> attributes = peer.GetStreamAttributes();
+  peer2.SetRemoteAttributes(attributes);
+
+  attrs = peer2.GetGlobalAttributes();
+  peer.ParseGlobalAttributes(attrs);
+  attributes = peer2.GetStreamAttributes();
+  peer.SetRemoteAttributes(attributes);
+  peer2.StartChecks();
+  peer.StartChecks();
+
+  ASSERT_TRUE_WAIT(peer.ice_connected_, kDefaultTimeout);
+  ASSERT_TRUE_WAIT(peer2.ice_connected_, kDefaultTimeout);
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/transport/test/test_nr_socket_unittest.cpp b/dom/media/webrtc/transport/test/test_nr_socket_unittest.cpp
new file mode 100644
index 0000000000..af2779accd
--- /dev/null
+++ b/dom/media/webrtc/transport/test/test_nr_socket_unittest.cpp
@@ -0,0 +1,800 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: bcampen@mozilla.com
+
+#include <cstddef>
+
+extern "C" {
+#include "r_errors.h"
+#include "async_wait.h"
+}
+
+#include "test_nr_socket.h"
+
+#include "nsCOMPtr.h"
+#include "nsNetCID.h"
+#include "nsServiceManagerUtils.h"
+#include "runnable_utils.h"
+
+#include <vector>
+
+#define GTEST_HAS_RTTI 0
+#include "gtest/gtest.h"
+#include "gtest_utils.h"
+
+#define DATA_BUF_SIZE 1024
+
+namespace mozilla {
+
+class TestNrSocketTest : public MtransportTest {
+ public:
+  TestNrSocketTest()
+      : MtransportTest(),
+        wait_done_for_main_(false),
+        sts_(),
+        public_addrs_(),
+        private_addrs_(),
+        nats_() {}
+
+  void SetUp() override {
+    MtransportTest::SetUp();
+
+    // Get the transport service as a dispatch target
+    nsresult rv;
+    sts_ = do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &rv);
+    EXPECT_TRUE(NS_SUCCEEDED(rv)) << "Failed to get STS: " << (int)rv;
+  }
+
+  void TearDown() override {
+    SyncDispatchToSTS(WrapRunnable(this, &TestNrSocketTest::TearDown_s));
+
+    MtransportTest::TearDown();
+  }
+
+  void TearDown_s() {
+    public_addrs_.clear();
+    private_addrs_.clear();
+    nats_.clear();
+    sts_ = nullptr;
+  }
+
+  RefPtr<TestNrSocket> CreateTestNrSocket_s(const char* ip_str, int proto,
+                                            TestNat* nat) {
+    // If no nat is supplied, we create a default NAT which is disabled. This
+    // is how we simulate a non-natted socket.
+    RefPtr<TestNrSocket> sock(new TestNrSocket(nat ? nat : new TestNat));
+    nr_transport_addr address;
+    nr_str_port_to_transport_addr(ip_str, 0, proto, &address);
+    int r = sock->create(&address);
+    if (r) {
+      return nullptr;
+    }
+    return sock;
+  }
+
+  void CreatePublicAddrs(size_t count, const char* ip_str = "127.0.0.1",
+                         int proto = IPPROTO_UDP) {
+    SyncDispatchToSTS(WrapRunnable(this, &TestNrSocketTest::CreatePublicAddrs_s,
+                                   count, ip_str, proto));
+  }
+
+  void CreatePublicAddrs_s(size_t count, const char* ip_str, int proto) {
+    while (count--) {
+      auto sock = CreateTestNrSocket_s(ip_str, proto, nullptr);
+      ASSERT_TRUE(sock)
+      << "Failed to create socket";
+      public_addrs_.push_back(sock);
+    }
+  }
+
+  RefPtr<TestNat> CreatePrivateAddrs(size_t size,
+                                     const char* ip_str = "127.0.0.1",
+                                     int proto = IPPROTO_UDP) {
+    RefPtr<TestNat> result;
+    SyncDispatchToSTS(WrapRunnableRet(&result, this,
+                                      &TestNrSocketTest::CreatePrivateAddrs_s,
+                                      size, ip_str, proto));
+    return result;
+  }
+
+  RefPtr<TestNat> CreatePrivateAddrs_s(size_t count, const char* ip_str,
+                                       int proto) {
+    RefPtr<TestNat> nat(new TestNat);
+    while (count--) {
+      auto sock = CreateTestNrSocket_s(ip_str, proto, nat);
+      if (!sock) {
+        EXPECT_TRUE(false) << "Failed to create socket";
+        break;
+      }
+      private_addrs_.push_back(sock);
+    }
+    nat->enabled_ = true;
+    nats_.push_back(nat);
+    return nat;
+  }
+
+  bool CheckConnectivityVia(
+      TestNrSocket* from, TestNrSocket* to, const nr_transport_addr& via,
+      nr_transport_addr* sender_external_address = nullptr) {
+    MOZ_ASSERT(from);
+
+    if (!WaitForWriteable(from)) {
+      return false;
+    }
+
+    int result = 0;
+    SyncDispatchToSTS(WrapRunnableRet(
+        &result, this, &TestNrSocketTest::SendData_s, from, via));
+    if (result) {
+      return false;
+    }
+
+    if (!WaitForReadable(to)) {
+      return false;
+    }
+
+    nr_transport_addr dummy_outparam;
+    if (!sender_external_address) {
+      sender_external_address = &dummy_outparam;
+    }
+
+    MOZ_ASSERT(to);
+    SyncDispatchToSTS(WrapRunnableRet(&result, this,
+                                      &TestNrSocketTest::RecvData_s, to,
+                                      sender_external_address));
+
+    return !result;
+  }
+
+  bool CheckConnectivity(TestNrSocket* from, TestNrSocket* to,
+                         nr_transport_addr* sender_external_address = nullptr) {
+    nr_transport_addr destination_address;
+    int r = GetAddress(to, &destination_address);
+    if (r) {
+      return false;
+    }
+
+    return CheckConnectivityVia(from, to, destination_address,
+                                sender_external_address);
+  }
+
+  bool CheckTcpConnectivity(TestNrSocket* from, TestNrSocket* to) {
+    NrSocketBase* accepted_sock;
+    if (!Connect(from, to, &accepted_sock)) {
+      std::cerr << "Connect failed" << std::endl;
+      return false;
+    }
+
+    // write on |from|, recv on |accepted_sock|
+    if (!WaitForWriteable(from)) {
+      std::cerr << __LINE__ << "WaitForWriteable (1) failed" << std::endl;
+      return false;
+    }
+
+    int r;
+    SyncDispatchToSTS(
+        WrapRunnableRet(&r, this, &TestNrSocketTest::SendDataTcp_s, from));
+    if (r) {
+      std::cerr << "SendDataTcp_s (1) failed" << std::endl;
+      return false;
+    }
+
+    if (!WaitForReadable(accepted_sock)) {
+      std::cerr << __LINE__ << "WaitForReadable (1) failed" << std::endl;
+      return false;
+    }
+
+    SyncDispatchToSTS(WrapRunnableRet(
+        &r, this, &TestNrSocketTest::RecvDataTcp_s, accepted_sock));
+    if (r) {
+      std::cerr << "RecvDataTcp_s (1) failed" << std::endl;
+      return false;
+    }
+
+    if (!WaitForWriteable(accepted_sock)) {
+      std::cerr << __LINE__ << "WaitForWriteable (2) failed" << std::endl;
+      return false;
+    }
+
+    SyncDispatchToSTS(WrapRunnableRet(
+        &r, this, &TestNrSocketTest::SendDataTcp_s, accepted_sock));
+    if (r) {
+      std::cerr << "SendDataTcp_s (2) failed" << std::endl;
+      return false;
+    }
+
+    if (!WaitForReadable(from)) {
+      std::cerr << __LINE__ << "WaitForReadable (2) failed" << std::endl;
+      return false;
+    }
+
+    SyncDispatchToSTS(
+        WrapRunnableRet(&r, this, &TestNrSocketTest::RecvDataTcp_s, from));
+    if (r) {
+      std::cerr << "RecvDataTcp_s (2) failed" << std::endl;
+      return false;
+    }
+
+    return true;
+  }
+
+  int GetAddress(TestNrSocket* sock, nr_transport_addr_* address) {
+    MOZ_ASSERT(sock);
+    MOZ_ASSERT(address);
+    int r;
+    SyncDispatchToSTS(WrapRunnableRet(&r, this, &TestNrSocketTest::GetAddress_s,
+                                      sock, address));
+    return r;
+  }
+
+  int GetAddress_s(TestNrSocket* sock, nr_transport_addr* address) {
+    return sock->getaddr(address);
+  }
+
+  int SendData_s(TestNrSocket* from, const nr_transport_addr& to) {
+    // It is up to caller to ensure that |from| is writeable.
+    const char buf[] = "foobajooba";
+    return from->sendto(buf, sizeof(buf), 0, &to);
+  }
+
+  int SendDataTcp_s(NrSocketBase* from) {
+    // It is up to caller to ensure that |from| is writeable.
+    const char buf[] = "foobajooba";
+    size_t written;
+    return from->write(buf, sizeof(buf), &written);
+  }
+
+  int RecvData_s(TestNrSocket* to, nr_transport_addr* from) {
+    // It is up to caller to ensure that |to| is readable
+    char buf[DATA_BUF_SIZE];
+    size_t len;
+    // Maybe check that data matches?
+    int r = to->recvfrom(buf, sizeof(buf), &len, 0, from);
+    if (!r && (len == 0)) {
+      r = R_INTERNAL;
+    }
+    return r;
+  }
+
+  int RecvDataTcp_s(NrSocketBase* to) {
+    // It is up to caller to ensure that |to| is readable
+    char buf[DATA_BUF_SIZE];
+    size_t len;
+    // Maybe check that data matches?
+    int r = to->read(buf, sizeof(buf), &len);
+    if (!r && (len == 0)) {
+      r = R_INTERNAL;
+    }
+    return r;
+  }
+
+  int Listen_s(TestNrSocket* to) {
+    // listen on |to|
+    int r = to->listen(1);
+    if (r) {
+      return r;
+    }
+    return 0;
+  }
+
+  int Connect_s(TestNrSocket* from, TestNrSocket* to) {
+    // connect on |from|
+    nr_transport_addr destination_address;
+    int r = to->getaddr(&destination_address);
+    if (r) {
+      return r;
+    }
+
+    r = from->connect(&destination_address);
+    if (r) {
+      return r;
+    }
+
+    return 0;
+  }
+
+  int Accept_s(TestNrSocket* to, NrSocketBase** accepted_sock) {
+    nr_socket* sock;
+    nr_transport_addr source_address;
+    int r = to->accept(&source_address, &sock);
+    if (r) {
+      return r;
+    }
+
+    *accepted_sock = reinterpret_cast<NrSocketBase*>(sock->obj);
+    return 0;
+  }
+
+  bool Connect(TestNrSocket* from, TestNrSocket* to,
+               NrSocketBase** accepted_sock) {
+    int r;
+    SyncDispatchToSTS(
+        WrapRunnableRet(&r, this, &TestNrSocketTest::Listen_s, to));
+    if (r) {
+      std::cerr << "Listen_s failed: " << r << std::endl;
+      return false;
+    }
+
+    SyncDispatchToSTS(
+        WrapRunnableRet(&r, this, &TestNrSocketTest::Connect_s, from, to));
+    if (r && r != R_WOULDBLOCK) {
+      std::cerr << "Connect_s failed: " << r << std::endl;
+      return false;
+    }
+
+    if (!WaitForReadable(to)) {
+      std::cerr << "WaitForReadable failed" << std::endl;
+      return false;
+    }
+
+    SyncDispatchToSTS(WrapRunnableRet(&r, this, &TestNrSocketTest::Accept_s, to,
+                                      accepted_sock));
+
+    if (r) {
+      std::cerr << "Accept_s failed: " << r << std::endl;
+      return false;
+    }
+    return true;
+  }
+
+  bool WaitForSocketState(NrSocketBase* sock, int state) {
+    MOZ_ASSERT(sock);
+    SyncDispatchToSTS(WrapRunnable(
+        this, &TestNrSocketTest::WaitForSocketState_s, sock, state));
+
+    bool res;
+    WAIT_(wait_done_for_main_, 500, res);
+    wait_done_for_main_ = false;
+
+    if (!res) {
+      SyncDispatchToSTS(
+          WrapRunnable(this, &TestNrSocketTest::CancelWait_s, sock, state));
+    }
+
+    return res;
+  }
+
+  void WaitForSocketState_s(NrSocketBase* sock, int state) {
+    NR_ASYNC_WAIT(sock, state, &WaitDone, this);
+  }
+
+  void CancelWait_s(NrSocketBase* sock, int state) { sock->cancel(state); }
+
+  bool WaitForReadable(NrSocketBase* sock) {
+    return WaitForSocketState(sock, NR_ASYNC_WAIT_READ);
+  }
+
+  bool WaitForWriteable(NrSocketBase* sock) {
+    return WaitForSocketState(sock, NR_ASYNC_WAIT_WRITE);
+  }
+
+  void SyncDispatchToSTS(nsIRunnable* runnable) {
+    NS_DispatchAndSpinEventLoopUntilComplete(
+        "TestNrSocketTest::SyncDispatchToSTS"_ns, sts_, do_AddRef(runnable));
+  }
+
+  static void WaitDone(void* sock, int how, void* test_fixture) {
+    TestNrSocketTest* test = static_cast<TestNrSocketTest*>(test_fixture);
+    test->wait_done_for_main_ = true;
+  }
+
+  // Simple busywait boolean for the test cases to spin on.
+  Atomic<bool> wait_done_for_main_;
+
+  nsCOMPtr<nsIEventTarget> sts_;
+  std::vector<RefPtr<TestNrSocket>> public_addrs_;
+  std::vector<RefPtr<TestNrSocket>> private_addrs_;
+  std::vector<RefPtr<TestNat>> nats_;
+};
+
+}  // namespace mozilla
+
+using mozilla::NrSocketBase;
+using mozilla::TestNat;
+using mozilla::TestNrSocketTest;
+
+TEST_F(TestNrSocketTest, UnsafePortRejectedUDP) {
+  nr_transport_addr address;
+  ASSERT_FALSE(nr_str_port_to_transport_addr("127.0.0.1",
+                                             // ssh
+                                             22, IPPROTO_UDP, &address));
+  ASSERT_TRUE(NrSocketBase::IsForbiddenAddress(&address));
+}
+
+TEST_F(TestNrSocketTest, UnsafePortRejectedTCP) {
+  nr_transport_addr address;
+  ASSERT_FALSE(nr_str_port_to_transport_addr("127.0.0.1",
+                                             // ssh
+                                             22, IPPROTO_TCP, &address));
+  ASSERT_TRUE(NrSocketBase::IsForbiddenAddress(&address));
+}
+
+TEST_F(TestNrSocketTest, SafePortAcceptedUDP) {
+  nr_transport_addr address;
+  ASSERT_FALSE(nr_str_port_to_transport_addr("127.0.0.1",
+                                             // stuns
+                                             5349, IPPROTO_UDP, &address));
+  ASSERT_FALSE(NrSocketBase::IsForbiddenAddress(&address));
+}
+
+TEST_F(TestNrSocketTest, SafePortAcceptedTCP) {
+  nr_transport_addr address;
+  ASSERT_FALSE(nr_str_port_to_transport_addr("127.0.0.1",
+                                             // turns
+                                             5349, IPPROTO_TCP, &address));
+  ASSERT_FALSE(NrSocketBase::IsForbiddenAddress(&address));
+}
+
+TEST_F(TestNrSocketTest, PublicConnectivity) {
+  CreatePublicAddrs(2);
+
+  ASSERT_TRUE(CheckConnectivity(public_addrs_[0], public_addrs_[1]));
+  ASSERT_TRUE(CheckConnectivity(public_addrs_[1], public_addrs_[0]));
+  ASSERT_TRUE(CheckConnectivity(public_addrs_[0], public_addrs_[0]));
+  ASSERT_TRUE(CheckConnectivity(public_addrs_[1], public_addrs_[1]));
+}
+
+TEST_F(TestNrSocketTest, PrivateConnectivity) {
+  RefPtr<TestNat> nat(CreatePrivateAddrs(2));
+  nat->filtering_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  nat->mapping_type_ = TestNat::ENDPOINT_INDEPENDENT;
+
+  ASSERT_TRUE(CheckConnectivity(private_addrs_[0], private_addrs_[1]));
+  ASSERT_TRUE(CheckConnectivity(private_addrs_[1], private_addrs_[0]));
+  ASSERT_TRUE(CheckConnectivity(private_addrs_[0], private_addrs_[0]));
+  ASSERT_TRUE(CheckConnectivity(private_addrs_[1], private_addrs_[1]));
+}
+
+TEST_F(TestNrSocketTest, NoConnectivityWithoutPinhole) {
+  RefPtr<TestNat> nat(CreatePrivateAddrs(1));
+  nat->filtering_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  nat->mapping_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  CreatePublicAddrs(1);
+
+  ASSERT_FALSE(CheckConnectivity(public_addrs_[0], private_addrs_[0]));
+}
+
+TEST_F(TestNrSocketTest, NoConnectivityBetweenSubnets) {
+  RefPtr<TestNat> nat1(CreatePrivateAddrs(1));
+  nat1->filtering_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  nat1->mapping_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  RefPtr<TestNat> nat2(CreatePrivateAddrs(1));
+  nat2->filtering_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  nat2->mapping_type_ = TestNat::ENDPOINT_INDEPENDENT;
+
+  ASSERT_FALSE(CheckConnectivity(private_addrs_[0], private_addrs_[1]));
+  ASSERT_FALSE(CheckConnectivity(private_addrs_[1], private_addrs_[0]));
+  ASSERT_TRUE(CheckConnectivity(private_addrs_[0], private_addrs_[0]));
+  ASSERT_TRUE(CheckConnectivity(private_addrs_[1], private_addrs_[1]));
+}
+
+TEST_F(TestNrSocketTest, FullConeAcceptIngress) {
+  RefPtr<TestNat> nat(CreatePrivateAddrs(1));
+  nat->filtering_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  nat->mapping_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  CreatePublicAddrs(2);
+
+  nr_transport_addr sender_external_address;
+  // Open pinhole to public IP 0
+  ASSERT_TRUE(CheckConnectivity(private_addrs_[0], public_addrs_[0],
+                                &sender_external_address));
+
+  // Verify that return traffic works
+  ASSERT_TRUE(CheckConnectivityVia(public_addrs_[0], private_addrs_[0],
+                                   sender_external_address));
+
+  // Verify that other public IP can use the pinhole
+  ASSERT_TRUE(CheckConnectivityVia(public_addrs_[1], private_addrs_[0],
+                                   sender_external_address));
+}
+
+TEST_F(TestNrSocketTest, FullConeOnePinhole) {
+  RefPtr<TestNat> nat(CreatePrivateAddrs(1));
+  nat->filtering_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  nat->mapping_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  CreatePublicAddrs(2);
+
+  nr_transport_addr sender_external_address;
+  // Open pinhole to public IP 0
+  ASSERT_TRUE(CheckConnectivity(private_addrs_[0], public_addrs_[0],
+                                &sender_external_address));
+
+  // Verify that return traffic works
+  ASSERT_TRUE(CheckConnectivityVia(public_addrs_[0], private_addrs_[0],
+                                   sender_external_address));
+
+  // Send traffic to other public IP, verify that it uses the same pinhole
+  nr_transport_addr sender_external_address2;
+  ASSERT_TRUE(CheckConnectivity(private_addrs_[0], public_addrs_[1],
+                                &sender_external_address2));
+  ASSERT_FALSE(nr_transport_addr_cmp(&sender_external_address,
+                                     &sender_external_address2,
+                                     NR_TRANSPORT_ADDR_CMP_MODE_ALL))
+  << "addr1: " << sender_external_address.as_string
+  << " addr2: " << sender_external_address2.as_string;
+}
+
+// OS 10.6 doesn't seem to allow us to open ports on 127.0.0.2, and while linux
+// does allow this, it has other behavior (see below) that prevents this test
+// from working.
+TEST_F(TestNrSocketTest, DISABLED_AddressRestrictedCone) {
+  RefPtr<TestNat> nat(CreatePrivateAddrs(1));
+  nat->filtering_type_ = TestNat::ADDRESS_DEPENDENT;
+  nat->mapping_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  CreatePublicAddrs(2, "127.0.0.1");
+  CreatePublicAddrs(1, "127.0.0.2");
+
+  nr_transport_addr sender_external_address;
+  // Open pinhole to public IP 0
+  ASSERT_TRUE(CheckConnectivity(private_addrs_[0], public_addrs_[0],
+                                &sender_external_address));
+
+  // Verify that return traffic works
+  ASSERT_TRUE(CheckConnectivityVia(public_addrs_[0], private_addrs_[0],
+                                   sender_external_address));
+
+  // Verify that another address on the same host can use the pinhole
+  ASSERT_TRUE(CheckConnectivityVia(public_addrs_[1], private_addrs_[0],
+                                   sender_external_address));
+
+  // Linux has a tendency to monkey around with source addresses, doing
+  // stuff like substituting 127.0.0.1 for packets sent by 127.0.0.2, and even
+  // going as far as substituting localhost for a packet sent from a real IP
+  // address when the destination is localhost. The only way to make this test
+  // work on linux is to have two real IP addresses.
+#ifndef __linux__
+  // Verify that an address on a different host can't use the pinhole
+  ASSERT_FALSE(CheckConnectivityVia(public_addrs_[2], private_addrs_[0],
+                                    sender_external_address));
+#endif
+
+  // Send traffic to other public IP, verify that it uses the same pinhole
+  nr_transport_addr sender_external_address2;
+  ASSERT_TRUE(CheckConnectivity(private_addrs_[0], public_addrs_[1],
+                                &sender_external_address2));
+  ASSERT_FALSE(nr_transport_addr_cmp(&sender_external_address,
+                                     &sender_external_address2,
+                                     NR_TRANSPORT_ADDR_CMP_MODE_ALL))
+  << "addr1: " << sender_external_address.as_string
+  << " addr2: " << sender_external_address2.as_string;
+
+  // Verify that the other public IP can now use the pinhole
+  ASSERT_TRUE(CheckConnectivityVia(public_addrs_[1], private_addrs_[0],
+                                   sender_external_address2));
+
+  // Send traffic to other public IP, verify that it uses the same pinhole
+  nr_transport_addr sender_external_address3;
+  ASSERT_TRUE(CheckConnectivity(private_addrs_[0], public_addrs_[2],
+                                &sender_external_address3));
+  ASSERT_FALSE(nr_transport_addr_cmp(&sender_external_address,
+                                     &sender_external_address3,
+                                     NR_TRANSPORT_ADDR_CMP_MODE_ALL))
+  << "addr1: " << sender_external_address.as_string
+  << " addr2: " << sender_external_address3.as_string;
+
+  // Verify that the other public IP can now use the pinhole
+  ASSERT_TRUE(CheckConnectivityVia(public_addrs_[2], private_addrs_[0],
+                                   sender_external_address3));
+}
+
+TEST_F(TestNrSocketTest, RestrictedCone) {
+  RefPtr<TestNat> nat(CreatePrivateAddrs(1));
+  nat->filtering_type_ = TestNat::PORT_DEPENDENT;
+  nat->mapping_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  CreatePublicAddrs(2);
+
+  nr_transport_addr sender_external_address;
+  // Open pinhole to public IP 0
+  ASSERT_TRUE(CheckConnectivity(private_addrs_[0], public_addrs_[0],
+                                &sender_external_address));
+
+  // Verify that return traffic works
+  ASSERT_TRUE(CheckConnectivityVia(public_addrs_[0], private_addrs_[0],
+                                   sender_external_address));
+
+  // Verify that other public IP cannot use the pinhole
+  ASSERT_FALSE(CheckConnectivityVia(public_addrs_[1], private_addrs_[0],
+                                    sender_external_address));
+
+  // Send traffic to other public IP, verify that it uses the same pinhole
+  nr_transport_addr sender_external_address2;
+  ASSERT_TRUE(CheckConnectivity(private_addrs_[0], public_addrs_[1],
+                                &sender_external_address2));
+  ASSERT_FALSE(nr_transport_addr_cmp(&sender_external_address,
+                                     &sender_external_address2,
+                                     NR_TRANSPORT_ADDR_CMP_MODE_ALL))
+  << "addr1: " << sender_external_address.as_string
+  << " addr2: " << sender_external_address2.as_string;
+
+  // Verify that the other public IP can now use the pinhole
+  ASSERT_TRUE(CheckConnectivityVia(public_addrs_[1], private_addrs_[0],
+                                   sender_external_address2));
+}
+
+TEST_F(TestNrSocketTest, PortDependentMappingFullCone) {
+  RefPtr<TestNat> nat(CreatePrivateAddrs(1));
+  nat->filtering_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  nat->mapping_type_ = TestNat::PORT_DEPENDENT;
+  CreatePublicAddrs(2);
+
+  nr_transport_addr sender_external_address0;
+  // Open pinhole to public IP 0
+  ASSERT_TRUE(CheckConnectivity(private_addrs_[0], public_addrs_[0],
+                                &sender_external_address0));
+
+  // Verify that return traffic works
+  ASSERT_TRUE(CheckConnectivityVia(public_addrs_[0], private_addrs_[0],
+                                   sender_external_address0));
+
+  // Verify that other public IP can use the pinhole
+  ASSERT_TRUE(CheckConnectivityVia(public_addrs_[1], private_addrs_[0],
+                                   sender_external_address0));
+
+  // Send traffic to other public IP, verify that it uses a different pinhole
+  nr_transport_addr sender_external_address1;
+  ASSERT_TRUE(CheckConnectivity(private_addrs_[0], public_addrs_[1],
+                                &sender_external_address1));
+  ASSERT_TRUE(nr_transport_addr_cmp(&sender_external_address0,
+                                    &sender_external_address1,
+                                    NR_TRANSPORT_ADDR_CMP_MODE_ALL))
+  << "addr1: " << sender_external_address0.as_string
+  << " addr2: " << sender_external_address1.as_string;
+
+  // Verify that return traffic works
+  ASSERT_TRUE(CheckConnectivityVia(public_addrs_[1], private_addrs_[0],
+                                   sender_external_address1));
+
+  // Verify that other public IP can use the original pinhole
+  ASSERT_TRUE(CheckConnectivityVia(public_addrs_[0], private_addrs_[0],
+                                   sender_external_address1));
+}
+
+TEST_F(TestNrSocketTest, Symmetric) {
+  RefPtr<TestNat> nat(CreatePrivateAddrs(1));
+  nat->filtering_type_ = TestNat::PORT_DEPENDENT;
+  nat->mapping_type_ = TestNat::PORT_DEPENDENT;
+  CreatePublicAddrs(2);
+
+  nr_transport_addr sender_external_address;
+  // Open pinhole to public IP 0
+  ASSERT_TRUE(CheckConnectivity(private_addrs_[0], public_addrs_[0],
+                                &sender_external_address));
+
+  // Verify that return traffic works
+  ASSERT_TRUE(CheckConnectivityVia(public_addrs_[0], private_addrs_[0],
+                                   sender_external_address));
+
+  // Verify that other public IP cannot use the pinhole
+  ASSERT_FALSE(CheckConnectivityVia(public_addrs_[1], private_addrs_[0],
+                                    sender_external_address));
+
+  // Send traffic to other public IP, verify that it uses a new pinhole
+  nr_transport_addr sender_external_address2;
+  ASSERT_TRUE(CheckConnectivity(private_addrs_[0], public_addrs_[1],
+                                &sender_external_address2));
+  ASSERT_TRUE(nr_transport_addr_cmp(&sender_external_address,
+                                    &sender_external_address2,
+                                    NR_TRANSPORT_ADDR_CMP_MODE_ALL));
+
+  // Verify that the other public IP can use the new pinhole
+  ASSERT_TRUE(CheckConnectivityVia(public_addrs_[1], private_addrs_[0],
+                                   sender_external_address2));
+}
+
+TEST_F(TestNrSocketTest, BlockUdp) {
+  RefPtr<TestNat> nat(CreatePrivateAddrs(2));
+  nat->block_udp_ = true;
+  CreatePublicAddrs(1);
+
+  nr_transport_addr sender_external_address;
+  ASSERT_FALSE(CheckConnectivity(private_addrs_[0], public_addrs_[0],
+                                 &sender_external_address));
+
+  // Make sure UDP behind the NAT still works
+  ASSERT_TRUE(CheckConnectivity(private_addrs_[0], private_addrs_[1]));
+  ASSERT_TRUE(CheckConnectivity(private_addrs_[1], private_addrs_[0]));
+}
+
+TEST_F(TestNrSocketTest, DenyHairpinning) {
+  RefPtr<TestNat> nat(CreatePrivateAddrs(2));
+  nat->filtering_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  nat->mapping_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  CreatePublicAddrs(1);
+
+  nr_transport_addr sender_external_address;
+  // Open pinhole to public IP 0
+  ASSERT_TRUE(CheckConnectivity(private_addrs_[0], public_addrs_[0],
+                                &sender_external_address));
+
+  // Verify that hairpinning is disallowed
+  ASSERT_FALSE(CheckConnectivityVia(private_addrs_[1], private_addrs_[0],
+                                    sender_external_address));
+}
+
+TEST_F(TestNrSocketTest, AllowHairpinning) {
+  RefPtr<TestNat> nat(CreatePrivateAddrs(2));
+  nat->filtering_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  nat->mapping_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  nat->mapping_timeout_ = 30000;
+  nat->allow_hairpinning_ = true;
+  CreatePublicAddrs(1);
+
+  nr_transport_addr sender_external_address;
+  // Open pinhole to public IP 0, obtain external address
+  ASSERT_TRUE(CheckConnectivity(private_addrs_[0], public_addrs_[0],
+                                &sender_external_address));
+
+  // Verify that hairpinning is allowed
+  ASSERT_TRUE(CheckConnectivityVia(private_addrs_[1], private_addrs_[0],
+                                   sender_external_address));
+}
+
+TEST_F(TestNrSocketTest, FullConeTimeout) {
+  RefPtr<TestNat> nat(CreatePrivateAddrs(1));
+  nat->filtering_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  nat->mapping_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  nat->mapping_timeout_ = 200;
+  CreatePublicAddrs(2);
+
+  nr_transport_addr sender_external_address;
+  // Open pinhole to public IP 0
+  ASSERT_TRUE(CheckConnectivity(private_addrs_[0], public_addrs_[0],
+                                &sender_external_address));
+
+  // Verify that return traffic works
+  ASSERT_TRUE(CheckConnectivityVia(public_addrs_[0], private_addrs_[0],
+                                   sender_external_address));
+
+  PR_Sleep(201);
+
+  // Verify that return traffic does not work
+  ASSERT_FALSE(CheckConnectivityVia(public_addrs_[0], private_addrs_[0],
+                                    sender_external_address));
+}
+
+TEST_F(TestNrSocketTest, PublicConnectivityTcp) {
+  CreatePublicAddrs(2, "127.0.0.1", IPPROTO_TCP);
+
+  ASSERT_TRUE(CheckTcpConnectivity(public_addrs_[0], public_addrs_[1]));
+}
+
+TEST_F(TestNrSocketTest, PrivateConnectivityTcp) {
+  RefPtr<TestNat> nat(CreatePrivateAddrs(2, "127.0.0.1", IPPROTO_TCP));
+  nat->filtering_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  nat->mapping_type_ = TestNat::ENDPOINT_INDEPENDENT;
+
+  ASSERT_TRUE(CheckTcpConnectivity(private_addrs_[0], private_addrs_[1]));
+}
+
+TEST_F(TestNrSocketTest, PrivateToPublicConnectivityTcp) {
+  RefPtr<TestNat> nat(CreatePrivateAddrs(1, "127.0.0.1", IPPROTO_TCP));
+  nat->filtering_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  nat->mapping_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  CreatePublicAddrs(1, "127.0.0.1", IPPROTO_TCP);
+
+  ASSERT_TRUE(CheckTcpConnectivity(private_addrs_[0], public_addrs_[0]));
+}
+
+TEST_F(TestNrSocketTest, NoConnectivityBetweenSubnetsTcp) {
+  RefPtr<TestNat> nat1(CreatePrivateAddrs(1, "127.0.0.1", IPPROTO_TCP));
+  nat1->filtering_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  nat1->mapping_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  RefPtr<TestNat> nat2(CreatePrivateAddrs(1, "127.0.0.1", IPPROTO_TCP));
+  nat2->filtering_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  nat2->mapping_type_ = TestNat::ENDPOINT_INDEPENDENT;
+
+  ASSERT_FALSE(CheckTcpConnectivity(private_addrs_[0], private_addrs_[1]));
+}
+
+TEST_F(TestNrSocketTest, NoConnectivityPublicToPrivateTcp) {
+  RefPtr<TestNat> nat(CreatePrivateAddrs(1, "127.0.0.1", IPPROTO_TCP));
+  nat->filtering_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  nat->mapping_type_ = TestNat::ENDPOINT_INDEPENDENT;
+  CreatePublicAddrs(1, "127.0.0.1", IPPROTO_TCP);
+
+  ASSERT_FALSE(CheckTcpConnectivity(public_addrs_[0], private_addrs_[0]));
+}
diff --git a/dom/media/webrtc/transport/test/transport_unittests.cpp b/dom/media/webrtc/transport/test/transport_unittests.cpp
new file mode 100644
index 0000000000..28f9359afb
--- /dev/null
+++ b/dom/media/webrtc/transport/test/transport_unittests.cpp
@@ -0,0 +1,1400 @@
+
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+#include <iostream>
+#include <string>
+#include <algorithm>
+#include <functional>
+
+#ifdef XP_MACOSX
+// ensure that Apple Security kit enum goes before "sslproto.h"
+#  include <CoreFoundation/CFAvailability.h>
+#  include <Security/CipherSuite.h>
+#endif
+
+#include "mozilla/UniquePtr.h"
+
+#include "sigslot.h"
+
+#include "logging.h"
+#include "ssl.h"
+#include "sslexp.h"
+#include "sslproto.h"
+
+#include "nsThreadUtils.h"
+
+#include "mediapacket.h"
+#include "dtlsidentity.h"
+#include "nricectx.h"
+#include "nricemediastream.h"
+#include "transportflow.h"
+#include "transportlayer.h"
+#include "transportlayerdtls.h"
+#include "transportlayerice.h"
+#include "transportlayerlog.h"
+#include "transportlayerloopback.h"
+
+#include "runnable_utils.h"
+
+#define GTEST_HAS_RTTI 0
+#include "gtest/gtest.h"
+#include "gtest_utils.h"
+
+using namespace mozilla;
+MOZ_MTLOG_MODULE("mtransport")
+
+const uint8_t kTlsChangeCipherSpecType = 0x14;
+const uint8_t kTlsHandshakeType = 0x16;
+
+const uint8_t kTlsHandshakeCertificate = 0x0b;
+const uint8_t kTlsHandshakeServerKeyExchange = 0x0c;
+
+const uint8_t kTlsFakeChangeCipherSpec[] = {
+    kTlsChangeCipherSpecType,  // Type
+    0xfe,
+    0xff,  // Version
+    0x00,
+    0x00,
+    0x00,
+    0x00,
+    0x00,
+    0x00,
+    0x00,
+    0x10,  // Fictitious sequence #
+    0x00,
+    0x01,  // Length
+    0x01   // Value
+};
+
+// Layer class which can't be initialized.
+class TransportLayerDummy : public TransportLayer {
+ public:
+  TransportLayerDummy(bool allow_init, bool* destroyed)
+      : allow_init_(allow_init), destroyed_(destroyed) {
+    *destroyed_ = false;
+  }
+
+  virtual ~TransportLayerDummy() { *destroyed_ = true; }
+
+  nsresult InitInternal() override {
+    return allow_init_ ? NS_OK : NS_ERROR_FAILURE;
+  }
+
+  TransportResult SendPacket(MediaPacket& packet) override {
+    MOZ_CRASH();  // Should never be called.
+    return 0;
+  }
+
+  TRANSPORT_LAYER_ID("lossy")
+
+ private:
+  bool allow_init_;
+  bool* destroyed_;
+};
+
+class Inspector {
+ public:
+  virtual ~Inspector() = default;
+
+  virtual void Inspect(TransportLayer* layer, const unsigned char* data,
+                       size_t len) = 0;
+};
+
+// Class to simulate various kinds of network lossage
+class TransportLayerLossy : public TransportLayer {
+ public:
+  TransportLayerLossy() : loss_mask_(0), packet_(0), inspector_(nullptr) {}
+  ~TransportLayerLossy() = default;
+
+  TransportResult SendPacket(MediaPacket& packet) override {
+    MOZ_MTLOG(ML_NOTICE, LAYER_INFO << "SendPacket(" << packet.len() << ")");
+
+    if (loss_mask_ & (1 << (packet_ % 32))) {
+      MOZ_MTLOG(ML_NOTICE, "Dropping packet");
+      ++packet_;
+      return packet.len();
+    }
+    if (inspector_) {
+      inspector_->Inspect(this, packet.data(), packet.len());
+    }
+
+    ++packet_;
+
+    return downward_->SendPacket(packet);
+  }
+
+  void SetLoss(uint32_t packet) { loss_mask_ |= (1 << (packet & 32)); }
+
+  void SetInspector(UniquePtr<Inspector> inspector) {
+    inspector_ = std::move(inspector);
+  }
+
+  void StateChange(TransportLayer* layer, State state) { TL_SET_STATE(state); }
+
+  void PacketReceived(TransportLayer* layer, MediaPacket& packet) {
+    SignalPacketReceived(this, packet);
+  }
+
+  TRANSPORT_LAYER_ID("lossy")
+
+ protected:
+  void WasInserted() override {
+    downward_->SignalPacketReceived.connect(
+        this, &TransportLayerLossy::PacketReceived);
+    downward_->SignalStateChange.connect(this,
+                                         &TransportLayerLossy::StateChange);
+
+    TL_SET_STATE(downward_->state());
+  }
+
+ private:
+  uint32_t loss_mask_;
+  uint32_t packet_;
+  UniquePtr<Inspector> inspector_;
+};
+
+// Process DTLS Records
+#define CHECK_LENGTH(expected)                \
+  do {                                        \
+    EXPECT_GE(remaining(), expected);         \
+    if (remaining() < expected) return false; \
+  } while (0)
+
+class TlsParser {
+ public:
+  TlsParser(const unsigned char* data, size_t len) : buffer_(), offset_(0) {
+    buffer_.Copy(data, len);
+  }
+
+  bool Read(unsigned char* val) {
+    if (remaining() < 1) {
+      return false;
+    }
+    *val = *ptr();
+    consume(1);
+    return true;
+  }
+
+  // Read an integral type of specified width.
+  bool Read(uint32_t* val, size_t len) {
+    if (len > sizeof(uint32_t)) return false;
+
+    *val = 0;
+
+    for (size_t i = 0; i < len; ++i) {
+      unsigned char tmp;
+
+      if (!Read(&tmp)) return false;
+
+      (*val) = ((*val) << 8) + tmp;
+    }
+
+    return true;
+  }
+
+  bool Read(unsigned char* val, size_t len) {
+    if (remaining() < len) {
+      return false;
+    }
+
+    if (val) {
+      memcpy(val, ptr(), len);
+    }
+    consume(len);
+
+    return true;
+  }
+
+ private:
+  size_t remaining() const { return buffer_.len() - offset_; }
+  const uint8_t* ptr() const { return buffer_.data() + offset_; }
+  void consume(size_t len) { offset_ += len; }
+
+  MediaPacket buffer_;
+  size_t offset_;
+};
+
+class DtlsRecordParser {
+ public:
+  DtlsRecordParser(const unsigned char* data, size_t len)
+      : buffer_(), offset_(0) {
+    buffer_.Copy(data, len);
+  }
+
+  bool NextRecord(uint8_t* ct, UniquePtr<MediaPacket>* buffer) {
+    if (!remaining()) return false;
+
+    CHECK_LENGTH(13U);
+    const uint8_t* ctp = reinterpret_cast<const uint8_t*>(ptr());
+    consume(11);  // ct + version + length
+
+    const uint16_t* tmp = reinterpret_cast<const uint16_t*>(ptr());
+    size_t length = ntohs(*tmp);
+    consume(2);
+
+    CHECK_LENGTH(length);
+    auto db = MakeUnique<MediaPacket>();
+    db->Copy(ptr(), length);
+    consume(length);
+
+    *ct = *ctp;
+    *buffer = std::move(db);
+
+    return true;
+  }
+
+ private:
+  size_t remaining() const { return buffer_.len() - offset_; }
+  const uint8_t* ptr() const { return buffer_.data() + offset_; }
+  void consume(size_t len) { offset_ += len; }
+
+  MediaPacket buffer_;
+  size_t offset_;
+};
+
+// Inspector that parses out DTLS records and passes
+// them on.
+class DtlsRecordInspector : public Inspector {
+ public:
+  virtual void Inspect(TransportLayer* layer, const unsigned char* data,
+                       size_t len) {
+    DtlsRecordParser parser(data, len);
+
+    uint8_t ct;
+    UniquePtr<MediaPacket> buf;
+    while (parser.NextRecord(&ct, &buf)) {
+      OnRecord(layer, ct, buf->data(), buf->len());
+    }
+  }
+
+  virtual void OnRecord(TransportLayer* layer, uint8_t content_type,
+                        const unsigned char* record, size_t len) = 0;
+};
+
+// Inspector that injects arbitrary packets based on
+// DTLS records of various types.
+class DtlsInspectorInjector : public DtlsRecordInspector {
+ public:
+  DtlsInspectorInjector(uint8_t packet_type, uint8_t handshake_type,
+                        const unsigned char* data, size_t len)
+      : packet_type_(packet_type), handshake_type_(handshake_type) {
+    packet_.Copy(data, len);
+  }
+
+  virtual void OnRecord(TransportLayer* layer, uint8_t content_type,
+                        const unsigned char* data, size_t len) {
+    // Only inject once.
+    if (!packet_.data()) {
+      return;
+    }
+
+    // Check that the first byte is as requested.
+    if (content_type != packet_type_) {
+      return;
+    }
+
+    if (handshake_type_ != 0xff) {
+      // Check that the packet is plausibly long enough.
+      if (len < 1) {
+        return;
+      }
+
+      // Check that the handshake type is as requested.
+      if (data[0] != handshake_type_) {
+        return;
+      }
+    }
+
+    layer->SendPacket(packet_);
+    packet_.Reset();
+  }
+
+ private:
+  uint8_t packet_type_;
+  uint8_t handshake_type_;
+  MediaPacket packet_;
+};
+
+// Make a copy of the first instance of a message.
+class DtlsInspectorRecordHandshakeMessage : public DtlsRecordInspector {
+ public:
+  explicit DtlsInspectorRecordHandshakeMessage(uint8_t handshake_type)
+      : handshake_type_(handshake_type), buffer_() {}
+
+  virtual void OnRecord(TransportLayer* layer, uint8_t content_type,
+                        const unsigned char* data, size_t len) {
+    // Only do this once.
+    if (buffer_.len()) {
+      return;
+    }
+
+    // Check that the first byte is as requested.
+    if (content_type != kTlsHandshakeType) {
+      return;
+    }
+
+    TlsParser parser(data, len);
+    unsigned char message_type;
+    // Read the handshake message type.
+    if (!parser.Read(&message_type)) {
+      return;
+    }
+    if (message_type != handshake_type_) {
+      return;
+    }
+
+    uint32_t length;
+    if (!parser.Read(&length, 3)) {
+      return;
+    }
+
+    uint32_t message_seq;
+    if (!parser.Read(&message_seq, 2)) {
+      return;
+    }
+
+    uint32_t fragment_offset;
+    if (!parser.Read(&fragment_offset, 3)) {
+      return;
+    }
+
+    uint32_t fragment_length;
+    if (!parser.Read(&fragment_length, 3)) {
+      return;
+    }
+
+    if ((fragment_offset != 0) || (fragment_length != length)) {
+      // This shouldn't happen because all current tests where we
+      // are using this code don't fragment.
+      return;
+    }
+
+    UniquePtr<uint8_t[]> buffer(new uint8_t[length]);
+    if (!parser.Read(buffer.get(), length)) {
+      return;
+    }
+    buffer_.Take(std::move(buffer), length);
+  }
+
+  const MediaPacket& buffer() { return buffer_; }
+
+ private:
+  uint8_t handshake_type_;
+  MediaPacket buffer_;
+};
+
+class TlsServerKeyExchangeECDHE {
+ public:
+  bool Parse(const unsigned char* data, size_t len) {
+    TlsParser parser(data, len);
+
+    uint8_t curve_type;
+    if (!parser.Read(&curve_type)) {
+      return false;
+    }
+
+    if (curve_type != 3) {  // named_curve
+      return false;
+    }
+
+    uint32_t named_curve;
+    if (!parser.Read(&named_curve, 2)) {
+      return false;
+    }
+
+    uint32_t point_length;
+    if (!parser.Read(&point_length, 1)) {
+      return false;
+    }
+
+    UniquePtr<uint8_t[]> key(new uint8_t[point_length]);
+    if (!parser.Read(key.get(), point_length)) {
+      return false;
+    }
+    public_key_.Take(std::move(key), point_length);
+
+    return true;
+  }
+
+  MediaPacket public_key_;
+};
+
+namespace {
+class TransportTestPeer : public sigslot::has_slots<> {
+ public:
+  TransportTestPeer(nsCOMPtr<nsIEventTarget> target, std::string name,
+                    MtransportTestUtils* utils)
+      : name_(name),
+        offerer_(name == "P1"),
+        target_(target),
+        received_packets_(0),
+        received_bytes_(0),
+        flow_(new TransportFlow(name)),
+        loopback_(new TransportLayerLoopback()),
+        logging_(new TransportLayerLogging()),
+        lossy_(new TransportLayerLossy()),
+        dtls_(new TransportLayerDtls()),
+        identity_(DtlsIdentity::Generate()),
+        ice_ctx_(),
+        streams_(),
+        peer_(nullptr),
+        gathering_complete_(false),
+        digest_("sha-1"),
+        enabled_cipersuites_(),
+        disabled_cipersuites_(),
+        test_utils_(utils) {
+    NrIceCtx::InitializeGlobals(NrIceCtx::GlobalConfig());
+    ice_ctx_ = NrIceCtx::Create(name);
+    std::vector<NrIceStunServer> stun_servers;
+    UniquePtr<NrIceStunServer> server(NrIceStunServer::Create(
+        std::string((char*)"stun.services.mozilla.com"), 3478));
+    stun_servers.push_back(*server);
+    EXPECT_TRUE(NS_SUCCEEDED(ice_ctx_->SetStunServers(stun_servers)));
+
+    dtls_->SetIdentity(identity_);
+    dtls_->SetRole(offerer_ ? TransportLayerDtls::SERVER
+                            : TransportLayerDtls::CLIENT);
+
+    nsresult res = identity_->ComputeFingerprint(&digest_);
+    EXPECT_TRUE(NS_SUCCEEDED(res));
+    EXPECT_EQ(20u, digest_.value_.size());
+  }
+
+  ~TransportTestPeer() {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &TransportTestPeer::DestroyFlow));
+  }
+
+  void DestroyFlow() {
+    disconnect_all();
+    if (flow_) {
+      loopback_->Disconnect();
+      flow_ = nullptr;
+    }
+    ice_ctx_->Destroy();
+    ice_ctx_ = nullptr;
+    streams_.clear();
+  }
+
+  void DisconnectDestroyFlow() {
+    test_utils_->SyncDispatchToSTS(NS_NewRunnableFunction(__func__, [this] {
+      loopback_->Disconnect();
+      disconnect_all();  // Disconnect from the signals;
+      flow_ = nullptr;
+    }));
+  }
+
+  void SetDtlsAllowAll() {
+    nsresult res = dtls_->SetVerificationAllowAll();
+    ASSERT_TRUE(NS_SUCCEEDED(res));
+  }
+
+  void SetAlpn(std::string str, bool withDefault, std::string extra = "") {
+    std::set<std::string> alpn;
+    alpn.insert(str);  // the one we want to select
+    if (!extra.empty()) {
+      alpn.insert(extra);
+    }
+    nsresult res = dtls_->SetAlpn(alpn, withDefault ? str : "");
+    ASSERT_EQ(NS_OK, res);
+  }
+
+  const std::string& GetAlpn() const { return dtls_->GetNegotiatedAlpn(); }
+
+  void SetDtlsPeer(TransportTestPeer* peer, int digests, unsigned int damage) {
+    unsigned int mask = 1;
+
+    for (int i = 0; i < digests; i++) {
+      DtlsDigest digest_to_set(peer->digest_);
+
+      if (damage & mask) digest_to_set.value_.data()[0]++;
+
+      nsresult res = dtls_->SetVerificationDigest(digest_to_set);
+
+      ASSERT_TRUE(NS_SUCCEEDED(res));
+
+      mask <<= 1;
+    }
+  }
+
+  void SetupSrtp() {
+    std::vector<uint16_t> srtp_ciphers =
+        TransportLayerDtls::GetDefaultSrtpCiphers();
+    SetSrtpCiphers(srtp_ciphers);
+  }
+
+  void SetSrtpCiphers(std::vector<uint16_t>& srtp_ciphers) {
+    ASSERT_TRUE(NS_SUCCEEDED(dtls_->SetSrtpCiphers(srtp_ciphers)));
+  }
+
+  void ConnectSocket_s(TransportTestPeer* peer) {
+    nsresult res;
+    res = loopback_->Init();
+    ASSERT_EQ((nsresult)NS_OK, res);
+
+    loopback_->Connect(peer->loopback_);
+    ASSERT_EQ((nsresult)NS_OK, loopback_->Init());
+    ASSERT_EQ((nsresult)NS_OK, logging_->Init());
+    ASSERT_EQ((nsresult)NS_OK, lossy_->Init());
+    ASSERT_EQ((nsresult)NS_OK, dtls_->Init());
+    dtls_->Chain(lossy_);
+    lossy_->Chain(logging_);
+    logging_->Chain(loopback_);
+
+    flow_->PushLayer(loopback_);
+    flow_->PushLayer(logging_);
+    flow_->PushLayer(lossy_);
+    flow_->PushLayer(dtls_);
+
+    if (dtls_->state() != TransportLayer::TS_ERROR) {
+      // Don't execute these blocks if DTLS didn't initialize.
+      TweakCiphers(dtls_->internal_fd());
+      if (post_setup_) {
+        post_setup_(dtls_->internal_fd());
+      }
+    }
+
+    dtls_->SignalPacketReceived.connect(this,
+                                        &TransportTestPeer::PacketReceived);
+  }
+
+  void TweakCiphers(PRFileDesc* fd) {
+    for (unsigned short& enabled_cipersuite : enabled_cipersuites_) {
+      SSL_CipherPrefSet(fd, enabled_cipersuite, PR_TRUE);
+    }
+    for (unsigned short& disabled_cipersuite : disabled_cipersuites_) {
+      SSL_CipherPrefSet(fd, disabled_cipersuite, PR_FALSE);
+    }
+  }
+
+  void ConnectSocket(TransportTestPeer* peer) {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &TransportTestPeer::ConnectSocket_s, peer));
+  }
+
+  nsresult InitIce_s() {
+    nsresult rv = ice_->Init();
+    NS_ENSURE_SUCCESS(rv, rv);
+    rv = dtls_->Init();
+    NS_ENSURE_SUCCESS(rv, rv);
+    dtls_->Chain(ice_);
+    flow_->PushLayer(ice_);
+    flow_->PushLayer(dtls_);
+    return NS_OK;
+  }
+
+  void InitIce() {
+    nsresult res;
+
+    // Attach our slots
+    ice_ctx_->SignalGatheringStateChange.connect(
+        this, &TransportTestPeer::GatheringStateChange);
+
+    char name[100];
+    snprintf(name, sizeof(name), "%s:stream%d", name_.c_str(),
+             (int)streams_.size());
+
+    // Create the media stream
+    RefPtr<NrIceMediaStream> stream = ice_ctx_->CreateStream(name, name, 1);
+
+    ASSERT_TRUE(stream != nullptr);
+    stream->SetIceCredentials("ufrag", "pass");
+    streams_.push_back(stream);
+
+    // Listen for candidates
+    stream->SignalCandidate.connect(this, &TransportTestPeer::GotCandidate);
+
+    // Create the transport layer
+    ice_ = new TransportLayerIce();
+    ice_->SetParameters(stream, 1);
+
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnableRet(&res, this, &TransportTestPeer::InitIce_s));
+
+    ASSERT_EQ((nsresult)NS_OK, res);
+
+    // Listen for media events
+    dtls_->SignalPacketReceived.connect(this,
+                                        &TransportTestPeer::PacketReceived);
+    dtls_->SignalStateChange.connect(this, &TransportTestPeer::StateChanged);
+
+    // Start gathering
+    test_utils_->SyncDispatchToSTS(WrapRunnableRet(
+        &res, ice_ctx_, &NrIceCtx::StartGathering, false, false));
+    ASSERT_TRUE(NS_SUCCEEDED(res));
+  }
+
+  void ConnectIce(TransportTestPeer* peer) {
+    peer_ = peer;
+
+    // If gathering is already complete, push the candidates over
+    if (gathering_complete_) GatheringComplete();
+  }
+
+  // New candidate
+  void GotCandidate(NrIceMediaStream* stream, const std::string& candidate,
+                    const std::string& ufrag, const std::string& mdns_addr,
+                    const std::string& actual_addr) {
+    std::cerr << "Got candidate " << candidate << " (ufrag=" << ufrag << ")"
+              << std::endl;
+  }
+
+  void GatheringStateChange(NrIceCtx* ctx, NrIceCtx::GatheringState state) {
+    (void)ctx;
+    if (state == NrIceCtx::ICE_CTX_GATHER_COMPLETE) {
+      GatheringComplete();
+    }
+  }
+
+  // Gathering complete, so send our candidates and start
+  // connecting on the other peer.
+  void GatheringComplete() {
+    nsresult res;
+
+    // Don't send to the other side
+    if (!peer_) {
+      gathering_complete_ = true;
+      return;
+    }
+
+    // First send attributes
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnableRet(&res, peer_->ice_ctx_, &NrIceCtx::ParseGlobalAttributes,
+                        ice_ctx_->GetGlobalAttributes()));
+    ASSERT_TRUE(NS_SUCCEEDED(res));
+
+    for (size_t i = 0; i < streams_.size(); ++i) {
+      test_utils_->SyncDispatchToSTS(WrapRunnableRet(
+          &res, peer_->streams_[i], &NrIceMediaStream::ConnectToPeer, "ufrag",
+          "pass", streams_[i]->GetAttributes()));
+
+      ASSERT_TRUE(NS_SUCCEEDED(res));
+    }
+
+    // Start checks on the other peer.
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnableRet(&res, peer_->ice_ctx_, &NrIceCtx::StartChecks));
+    ASSERT_TRUE(NS_SUCCEEDED(res));
+  }
+
+  // WrapRunnable/lambda and move semantics (MediaPacket is not copyable) don't
+  // get along yet, so we need a wrapper. Gross.
+  static TransportResult SendPacketWrapper(TransportLayer* layer,
+                                           MediaPacket* packet) {
+    return layer->SendPacket(*packet);
+  }
+
+  TransportResult SendPacket(MediaPacket& packet) {
+    TransportResult ret;
+
+    test_utils_->SyncDispatchToSTS(WrapRunnableNMRet(
+        &ret, &TransportTestPeer::SendPacketWrapper, dtls_, &packet));
+
+    return ret;
+  }
+
+  void StateChanged(TransportLayer* layer, TransportLayer::State state) {
+    if (state == TransportLayer::TS_OPEN) {
+      std::cerr << "Now connected" << std::endl;
+    }
+  }
+
+  void PacketReceived(TransportLayer* layer, MediaPacket& packet) {
+    std::cerr << "Received " << packet.len() << " bytes" << std::endl;
+    ++received_packets_;
+    received_bytes_ += packet.len();
+  }
+
+  void SetLoss(uint32_t loss) { lossy_->SetLoss(loss); }
+
+  void SetCombinePackets(bool combine) { loopback_->CombinePackets(combine); }
+
+  void SetInspector(UniquePtr<Inspector> inspector) {
+    lossy_->SetInspector(std::move(inspector));
+  }
+
+  void SetInspector(Inspector* in) {
+    UniquePtr<Inspector> inspector(in);
+
+    lossy_->SetInspector(std::move(inspector));
+  }
+
+  void SetCipherSuiteChanges(const std::vector<uint16_t>& enableThese,
+                             const std::vector<uint16_t>& disableThese) {
+    disabled_cipersuites_ = disableThese;
+    enabled_cipersuites_ = enableThese;
+  }
+
+  void SetPostSetup(const std::function<void(PRFileDesc*)>& setup) {
+    post_setup_ = std::move(setup);
+  }
+
+  TransportLayer::State state() {
+    TransportLayer::State tstate;
+
+    RUN_ON_THREAD(test_utils_->sts_target(),
+                  WrapRunnableRet(&tstate, dtls_, &TransportLayer::state));
+
+    return tstate;
+  }
+
+  bool connected() { return state() == TransportLayer::TS_OPEN; }
+
+  bool failed() { return state() == TransportLayer::TS_ERROR; }
+
+  size_t receivedPackets() { return received_packets_; }
+
+  size_t receivedBytes() { return received_bytes_; }
+
+  uint16_t cipherSuite() const {
+    nsresult rv;
+    uint16_t cipher;
+    RUN_ON_THREAD(
+        test_utils_->sts_target(),
+        WrapRunnableRet(&rv, dtls_, &TransportLayerDtls::GetCipherSuite,
+                        &cipher));
+
+    if (NS_FAILED(rv)) {
+      return TLS_NULL_WITH_NULL_NULL;  // i.e., not good
+    }
+    return cipher;
+  }
+
+  uint16_t srtpCipher() const {
+    nsresult rv;
+    uint16_t cipher;
+    RUN_ON_THREAD(test_utils_->sts_target(),
+                  WrapRunnableRet(&rv, dtls_,
+                                  &TransportLayerDtls::GetSrtpCipher, &cipher));
+    if (NS_FAILED(rv)) {
+      return 0;  // the SRTP equivalent of TLS_NULL_WITH_NULL_NULL
+    }
+    return cipher;
+  }
+
+ private:
+  std::string name_;
+  bool offerer_;
+  nsCOMPtr<nsIEventTarget> target_;
+  size_t received_packets_;
+  size_t received_bytes_;
+  RefPtr<TransportFlow> flow_;
+  TransportLayerLoopback* loopback_;
+  TransportLayerLogging* logging_;
+  TransportLayerLossy* lossy_;
+  TransportLayerDtls* dtls_;
+  TransportLayerIce* ice_;
+  RefPtr<DtlsIdentity> identity_;
+  RefPtr<NrIceCtx> ice_ctx_;
+  std::vector<RefPtr<NrIceMediaStream> > streams_;
+  TransportTestPeer* peer_;
+  bool gathering_complete_;
+  DtlsDigest digest_;
+  std::vector<uint16_t> enabled_cipersuites_;
+  std::vector<uint16_t> disabled_cipersuites_;
+  MtransportTestUtils* test_utils_;
+  std::function<void(PRFileDesc* fd)> post_setup_ = nullptr;
+};
+
+class TransportTest : public MtransportTest {
+ public:
+  TransportTest() {
+    fds_[0] = nullptr;
+    fds_[1] = nullptr;
+    p1_ = nullptr;
+    p2_ = nullptr;
+  }
+
+  void TearDown() override {
+    delete p1_;
+    delete p2_;
+
+    //    Can't detach these
+    //    PR_Close(fds_[0]);
+    //    PR_Close(fds_[1]);
+    MtransportTest::TearDown();
+  }
+
+  void DestroyPeerFlows() {
+    p1_->DisconnectDestroyFlow();
+    p2_->DisconnectDestroyFlow();
+  }
+
+  void SetUp() override {
+    MtransportTest::SetUp();
+
+    nsresult rv;
+    target_ = do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &rv);
+    ASSERT_TRUE(NS_SUCCEEDED(rv));
+
+    Reset();
+  }
+
+  void Reset() {
+    if (p1_) {
+      delete p1_;
+    }
+    if (p2_) {
+      delete p2_;
+    }
+    p1_ = new TransportTestPeer(target_, "P1", test_utils_);
+    p2_ = new TransportTestPeer(target_, "P2", test_utils_);
+  }
+
+  void SetupSrtp() {
+    p1_->SetupSrtp();
+    p2_->SetupSrtp();
+  }
+
+  void SetDtlsPeer(int digests = 1, unsigned int damage = 0) {
+    p1_->SetDtlsPeer(p2_, digests, damage);
+    p2_->SetDtlsPeer(p1_, digests, damage);
+  }
+
+  void SetDtlsAllowAll() {
+    p1_->SetDtlsAllowAll();
+    p2_->SetDtlsAllowAll();
+  }
+
+  void SetAlpn(std::string first, std::string second,
+               bool withDefaults = true) {
+    if (!first.empty()) {
+      p1_->SetAlpn(first, withDefaults, "bogus");
+    }
+    if (!second.empty()) {
+      p2_->SetAlpn(second, withDefaults);
+    }
+  }
+
+  void CheckAlpn(std::string first, std::string second) {
+    ASSERT_EQ(first, p1_->GetAlpn());
+    ASSERT_EQ(second, p2_->GetAlpn());
+  }
+
+  void ConnectSocket() {
+    ConnectSocketInternal();
+    ASSERT_TRUE_WAIT(p1_->connected(), 10000);
+    ASSERT_TRUE_WAIT(p2_->connected(), 10000);
+
+    ASSERT_EQ(p1_->cipherSuite(), p2_->cipherSuite());
+    ASSERT_EQ(p1_->srtpCipher(), p2_->srtpCipher());
+  }
+
+  void ConnectSocketExpectFail() {
+    ConnectSocketInternal();
+    ASSERT_TRUE_WAIT(p1_->failed(), 10000);
+    ASSERT_TRUE_WAIT(p2_->failed(), 10000);
+  }
+
+  void ConnectSocketExpectState(TransportLayer::State s1,
+                                TransportLayer::State s2) {
+    ConnectSocketInternal();
+    ASSERT_EQ_WAIT(s1, p1_->state(), 10000);
+    ASSERT_EQ_WAIT(s2, p2_->state(), 10000);
+  }
+
+  void ConnectIce() {
+    p1_->InitIce();
+    p2_->InitIce();
+    p1_->ConnectIce(p2_);
+    p2_->ConnectIce(p1_);
+    ASSERT_TRUE_WAIT(p1_->connected(), 10000);
+    ASSERT_TRUE_WAIT(p2_->connected(), 10000);
+  }
+
+  void TransferTest(size_t count, size_t bytes = 1024) {
+    unsigned char buf[bytes];
+
+    for (size_t i = 0; i < count; ++i) {
+      memset(buf, count & 0xff, sizeof(buf));
+      MediaPacket packet;
+      packet.Copy(buf, sizeof(buf));
+      TransportResult rv = p1_->SendPacket(packet);
+      ASSERT_TRUE(rv > 0);
+    }
+
+    std::cerr << "Received == " << p2_->receivedPackets() << " packets"
+              << std::endl;
+    ASSERT_TRUE_WAIT(count == p2_->receivedPackets(), 10000);
+    ASSERT_TRUE((count * sizeof(buf)) == p2_->receivedBytes());
+  }
+
+ protected:
+  void ConnectSocketInternal() {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(p1_, &TransportTestPeer::ConnectSocket, p2_));
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(p2_, &TransportTestPeer::ConnectSocket, p1_));
+  }
+
+  PRFileDesc* fds_[2];
+  TransportTestPeer* p1_;
+  TransportTestPeer* p2_;
+  nsCOMPtr<nsIEventTarget> target_;
+};
+
+TEST_F(TransportTest, TestNoDtlsVerificationSettings) {
+  ConnectSocketExpectFail();
+}
+
+static void DisableChaCha(TransportTestPeer* peer) {
+  // On ARM, ChaCha20Poly1305 might be preferred; disable it for the tests that
+  // want to check the cipher suite.  It doesn't matter which peer disables the
+  // suite, disabling on either side has the same effect.
+  std::vector<uint16_t> chachaSuites;
+  chachaSuites.push_back(TLS_ECDHE_ECDSA_WITH_CHACHA20_POLY1305_SHA256);
+  chachaSuites.push_back(TLS_ECDHE_RSA_WITH_CHACHA20_POLY1305_SHA256);
+  peer->SetCipherSuiteChanges(std::vector<uint16_t>(), chachaSuites);
+}
+
+TEST_F(TransportTest, TestConnect) {
+  SetDtlsPeer();
+  DisableChaCha(p1_);
+  ConnectSocket();
+
+  // check that we got the right suite
+  ASSERT_EQ(TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256, p1_->cipherSuite());
+
+  // no SRTP on this one
+  ASSERT_EQ(0, p1_->srtpCipher());
+}
+
+TEST_F(TransportTest, TestConnectSrtp) {
+  SetupSrtp();
+  SetDtlsPeer();
+  DisableChaCha(p2_);
+  ConnectSocket();
+
+  ASSERT_EQ(TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256, p1_->cipherSuite());
+
+  // SRTP is on with default value
+  ASSERT_EQ(kDtlsSrtpAeadAes128Gcm, p1_->srtpCipher());
+}
+
+TEST_F(TransportTest, TestConnectDestroyFlowsMainThread) {
+  SetDtlsPeer();
+  ConnectSocket();
+  DestroyPeerFlows();
+}
+
+TEST_F(TransportTest, TestConnectAllowAll) {
+  SetDtlsAllowAll();
+  ConnectSocket();
+}
+
+TEST_F(TransportTest, TestConnectAlpn) {
+  SetDtlsPeer();
+  SetAlpn("a", "a");
+  ConnectSocket();
+  CheckAlpn("a", "a");
+}
+
+TEST_F(TransportTest, TestConnectAlpnMismatch) {
+  SetDtlsPeer();
+  SetAlpn("something", "different");
+  ConnectSocketExpectFail();
+}
+
+TEST_F(TransportTest, TestConnectAlpnServerDefault) {
+  SetDtlsPeer();
+  SetAlpn("def", "");
+  // server allows default, client doesn't support
+  ConnectSocket();
+  CheckAlpn("def", "");
+}
+
+TEST_F(TransportTest, TestConnectAlpnClientDefault) {
+  SetDtlsPeer();
+  SetAlpn("", "clientdef");
+  // client allows default, but server will ignore the extension
+  ConnectSocket();
+  CheckAlpn("", "clientdef");
+}
+
+TEST_F(TransportTest, TestConnectClientNoAlpn) {
+  SetDtlsPeer();
+  // Here the server has ALPN, but no default is allowed.
+  // Reminder: p1 == server, p2 == client
+  SetAlpn("server-nodefault", "", false);
+  // The server doesn't see the extension, so negotiates without it.
+  // But then the server is forced to close when it discovers that ALPN wasn't
+  // negotiated; the client sees a close.
+  ConnectSocketExpectState(TransportLayer::TS_ERROR, TransportLayer::TS_CLOSED);
+}
+
+TEST_F(TransportTest, TestConnectServerNoAlpn) {
+  SetDtlsPeer();
+  SetAlpn("", "client-nodefault", false);
+  // The client aborts; the server doesn't realize this is a problem and just
+  // sees the close.
+  ConnectSocketExpectState(TransportLayer::TS_CLOSED, TransportLayer::TS_ERROR);
+}
+
+TEST_F(TransportTest, TestConnectNoDigest) {
+  SetDtlsPeer(0, 0);
+
+  ConnectSocketExpectFail();
+}
+
+TEST_F(TransportTest, TestConnectBadDigest) {
+  SetDtlsPeer(1, 1);
+
+  ConnectSocketExpectFail();
+}
+
+TEST_F(TransportTest, TestConnectTwoDigests) {
+  SetDtlsPeer(2, 0);
+
+  ConnectSocket();
+}
+
+TEST_F(TransportTest, TestConnectTwoDigestsFirstBad) {
+  SetDtlsPeer(2, 1);
+
+  ConnectSocket();
+}
+
+TEST_F(TransportTest, TestConnectTwoDigestsSecondBad) {
+  SetDtlsPeer(2, 2);
+
+  ConnectSocket();
+}
+
+TEST_F(TransportTest, TestConnectTwoDigestsBothBad) {
+  SetDtlsPeer(2, 3);
+
+  ConnectSocketExpectFail();
+}
+
+TEST_F(TransportTest, TestConnectInjectCCS) {
+  SetDtlsPeer();
+  p2_->SetInspector(MakeUnique<DtlsInspectorInjector>(
+      kTlsHandshakeType, kTlsHandshakeCertificate, kTlsFakeChangeCipherSpec,
+      sizeof(kTlsFakeChangeCipherSpec)));
+
+  ConnectSocket();
+}
+
+TEST_F(TransportTest, TestConnectVerifyNewECDHE) {
+  SetDtlsPeer();
+  DtlsInspectorRecordHandshakeMessage* i1 =
+      new DtlsInspectorRecordHandshakeMessage(kTlsHandshakeServerKeyExchange);
+  p1_->SetInspector(i1);
+  ConnectSocket();
+  TlsServerKeyExchangeECDHE dhe1;
+  ASSERT_TRUE(dhe1.Parse(i1->buffer().data(), i1->buffer().len()));
+
+  Reset();
+  SetDtlsPeer();
+  DtlsInspectorRecordHandshakeMessage* i2 =
+      new DtlsInspectorRecordHandshakeMessage(kTlsHandshakeServerKeyExchange);
+  p1_->SetInspector(i2);
+  ConnectSocket();
+  TlsServerKeyExchangeECDHE dhe2;
+  ASSERT_TRUE(dhe2.Parse(i2->buffer().data(), i2->buffer().len()));
+
+  // Now compare these two to see if they are the same.
+  ASSERT_FALSE((dhe1.public_key_.len() == dhe2.public_key_.len()) &&
+               (!memcmp(dhe1.public_key_.data(), dhe2.public_key_.data(),
+                        dhe1.public_key_.len())));
+}
+
+TEST_F(TransportTest, TestConnectVerifyReusedECDHE) {
+  auto set_reuse_ecdhe_key = [](PRFileDesc* fd) {
+    // TransportLayerDtls automatically sets this pref to false
+    // so set it back for test.
+    // This is pretty gross. Dig directly into the NSS FD. The problem
+    // is that we are testing a feature which TransaportLayerDtls doesn't
+    // expose.
+    SECStatus rv = SSL_OptionSet(fd, SSL_REUSE_SERVER_ECDHE_KEY, PR_TRUE);
+    ASSERT_EQ(SECSuccess, rv);
+  };
+
+  SetDtlsPeer();
+  DtlsInspectorRecordHandshakeMessage* i1 =
+      new DtlsInspectorRecordHandshakeMessage(kTlsHandshakeServerKeyExchange);
+  p1_->SetInspector(i1);
+  p1_->SetPostSetup(set_reuse_ecdhe_key);
+  ConnectSocket();
+  TlsServerKeyExchangeECDHE dhe1;
+  ASSERT_TRUE(dhe1.Parse(i1->buffer().data(), i1->buffer().len()));
+
+  Reset();
+  SetDtlsPeer();
+  DtlsInspectorRecordHandshakeMessage* i2 =
+      new DtlsInspectorRecordHandshakeMessage(kTlsHandshakeServerKeyExchange);
+
+  p1_->SetInspector(i2);
+  p1_->SetPostSetup(set_reuse_ecdhe_key);
+
+  ConnectSocket();
+  TlsServerKeyExchangeECDHE dhe2;
+  ASSERT_TRUE(dhe2.Parse(i2->buffer().data(), i2->buffer().len()));
+
+  // Now compare these two to see if they are the same.
+  ASSERT_EQ(dhe1.public_key_.len(), dhe2.public_key_.len());
+  ASSERT_TRUE(!memcmp(dhe1.public_key_.data(), dhe2.public_key_.data(),
+                      dhe1.public_key_.len()));
+}
+
+TEST_F(TransportTest, TestTransfer) {
+  SetDtlsPeer();
+  ConnectSocket();
+  TransferTest(1);
+}
+
+TEST_F(TransportTest, TestTransferMaxSize) {
+  SetDtlsPeer();
+  ConnectSocket();
+  /* transportlayerdtls uses a 9216 bytes buffer - as this test uses the
+   * loopback implementation it does not have to take into account the extra
+   * bytes added by the DTLS layer below. */
+  TransferTest(1, 9216);
+}
+
+TEST_F(TransportTest, TestTransferMultiple) {
+  SetDtlsPeer();
+  ConnectSocket();
+  TransferTest(3);
+}
+
+TEST_F(TransportTest, TestTransferCombinedPackets) {
+  SetDtlsPeer();
+  ConnectSocket();
+  p2_->SetCombinePackets(true);
+  TransferTest(3);
+}
+
+TEST_F(TransportTest, TestConnectLoseFirst) {
+  SetDtlsPeer();
+  p1_->SetLoss(0);
+  ConnectSocket();
+  TransferTest(1);
+}
+
+TEST_F(TransportTest, TestConnectIce) {
+  SetDtlsPeer();
+  ConnectIce();
+}
+
+TEST_F(TransportTest, TestTransferIceMaxSize) {
+  SetDtlsPeer();
+  ConnectIce();
+  /* nICEr and transportlayerdtls both use 9216 bytes buffers. But the DTLS
+   * layer add extra bytes to the packet, which size depends on chosen cipher
+   * etc. Sending more then 9216 bytes works, but on the receiving side the call
+   * to PR_recvfrom() will truncate any packet bigger then nICEr's buffer size
+   * of 9216 bytes, which then results in the DTLS layer discarding the packet.
+   * Therefore we leave some headroom (according to
+   * https://bugzilla.mozilla.org/show_bug.cgi?id=1214269#c29 256 bytes should
+   * be save choice) here for the DTLS bytes to make it safely into the
+   * receiving buffer in nICEr. */
+  TransferTest(1, 8960);
+}
+
+TEST_F(TransportTest, TestTransferIceMultiple) {
+  SetDtlsPeer();
+  ConnectIce();
+  TransferTest(3);
+}
+
+TEST_F(TransportTest, TestTransferIceCombinedPackets) {
+  SetDtlsPeer();
+  ConnectIce();
+  p2_->SetCombinePackets(true);
+  TransferTest(3);
+}
+
+// test the default configuration against a peer that supports only
+// one of the mandatory-to-implement suites, which should succeed
+static void ConfigureOneCipher(TransportTestPeer* peer, uint16_t suite) {
+  std::vector<uint16_t> justOne;
+  justOne.push_back(suite);
+  std::vector<uint16_t> everythingElse(
+      SSL_GetImplementedCiphers(),
+      SSL_GetImplementedCiphers() + SSL_GetNumImplementedCiphers());
+  std::remove(everythingElse.begin(), everythingElse.end(), suite);
+  peer->SetCipherSuiteChanges(justOne, everythingElse);
+}
+
+TEST_F(TransportTest, TestCipherMismatch) {
+  SetDtlsPeer();
+  ConfigureOneCipher(p1_, TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256);
+  ConfigureOneCipher(p2_, TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA);
+  ConnectSocketExpectFail();
+}
+
+TEST_F(TransportTest, TestCipherMandatoryOnlyGcm) {
+  SetDtlsPeer();
+  ConfigureOneCipher(p1_, TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256);
+  ConnectSocket();
+  ASSERT_EQ(TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256, p1_->cipherSuite());
+}
+
+TEST_F(TransportTest, TestCipherMandatoryOnlyCbc) {
+  SetDtlsPeer();
+  ConfigureOneCipher(p1_, TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA);
+  ConnectSocket();
+  ASSERT_EQ(TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA, p1_->cipherSuite());
+}
+
+TEST_F(TransportTest, TestSrtpMismatch) {
+  std::vector<uint16_t> setA;
+  setA.push_back(kDtlsSrtpAes128CmHmacSha1_80);
+  std::vector<uint16_t> setB;
+  setB.push_back(kDtlsSrtpAes128CmHmacSha1_32);
+
+  p1_->SetSrtpCiphers(setA);
+  p2_->SetSrtpCiphers(setB);
+  SetDtlsPeer();
+  ConnectSocketExpectFail();
+
+  ASSERT_EQ(0, p1_->srtpCipher());
+  ASSERT_EQ(0, p2_->srtpCipher());
+}
+
+static SECStatus NoopXtnHandler(PRFileDesc* fd, SSLHandshakeType message,
+                                const uint8_t* data, unsigned int len,
+                                SSLAlertDescription* alert, void* arg) {
+  return SECSuccess;
+}
+
+static PRBool WriteFixedXtn(PRFileDesc* fd, SSLHandshakeType message,
+                            uint8_t* data, unsigned int* len,
+                            unsigned int max_len, void* arg) {
+  // When we enable TLS 1.3, change ssl_hs_server_hello here to
+  // ssl_hs_encrypted_extensions.  At the same time, add a test that writes to
+  // ssl_hs_server_hello, which should fail.
+  if (message != ssl_hs_client_hello && message != ssl_hs_server_hello) {
+    return false;
+  }
+
+  auto v = reinterpret_cast<std::vector<uint8_t>*>(arg);
+  memcpy(data, &((*v)[0]), v->size());
+  *len = v->size();
+  return true;
+}
+
+// Note that |value| needs to be readable after this function returns.
+static void InstallBadSrtpExtensionWriter(TransportTestPeer* peer,
+                                          std::vector<uint8_t>* value) {
+  peer->SetPostSetup([value](PRFileDesc* fd) {
+    // Override the handler that is installed by the DTLS setup.
+    SECStatus rv = SSL_InstallExtensionHooks(
+        fd, ssl_use_srtp_xtn, WriteFixedXtn, value, NoopXtnHandler, nullptr);
+    ASSERT_EQ(SECSuccess, rv);
+  });
+}
+
+TEST_F(TransportTest, TestSrtpErrorServerSendsTwoSrtpCiphers) {
+  // Server (p1_) sends an extension with two values, and empty MKI.
+  std::vector<uint8_t> xtn = {0x04, 0x00, 0x01, 0x00, 0x02, 0x00};
+  InstallBadSrtpExtensionWriter(p1_, &xtn);
+  SetupSrtp();
+  SetDtlsPeer();
+  ConnectSocketExpectFail();
+}
+
+TEST_F(TransportTest, TestSrtpErrorServerSendsTwoMki) {
+  // Server (p1_) sends an MKI.
+  std::vector<uint8_t> xtn = {0x02, 0x00, 0x01, 0x01, 0x00};
+  InstallBadSrtpExtensionWriter(p1_, &xtn);
+  SetupSrtp();
+  SetDtlsPeer();
+  ConnectSocketExpectFail();
+}
+
+TEST_F(TransportTest, TestSrtpErrorServerSendsUnknownValue) {
+  std::vector<uint8_t> xtn = {0x02, 0x9a, 0xf1, 0x00};
+  InstallBadSrtpExtensionWriter(p1_, &xtn);
+  SetupSrtp();
+  SetDtlsPeer();
+  ConnectSocketExpectFail();
+}
+
+TEST_F(TransportTest, TestSrtpErrorServerSendsOverflow) {
+  std::vector<uint8_t> xtn = {0x32, 0x00, 0x01, 0x00};
+  InstallBadSrtpExtensionWriter(p1_, &xtn);
+  SetupSrtp();
+  SetDtlsPeer();
+  ConnectSocketExpectFail();
+}
+
+TEST_F(TransportTest, TestSrtpErrorServerSendsUnevenList) {
+  std::vector<uint8_t> xtn = {0x01, 0x00, 0x00};
+  InstallBadSrtpExtensionWriter(p1_, &xtn);
+  SetupSrtp();
+  SetDtlsPeer();
+  ConnectSocketExpectFail();
+}
+
+TEST_F(TransportTest, TestSrtpErrorClientSendsUnevenList) {
+  std::vector<uint8_t> xtn = {0x01, 0x00, 0x00};
+  InstallBadSrtpExtensionWriter(p2_, &xtn);
+  SetupSrtp();
+  SetDtlsPeer();
+  ConnectSocketExpectFail();
+}
+
+TEST_F(TransportTest, OnlyServerSendsSrtpXtn) {
+  p1_->SetupSrtp();
+  SetDtlsPeer();
+  // This should connect, but with no SRTP extension neogtiated.
+  // The client side might negotiate a data channel only.
+  ConnectSocket();
+  ASSERT_NE(TLS_NULL_WITH_NULL_NULL, p1_->cipherSuite());
+  ASSERT_EQ(0, p1_->srtpCipher());
+}
+
+TEST_F(TransportTest, OnlyClientSendsSrtpXtn) {
+  p2_->SetupSrtp();
+  SetDtlsPeer();
+  // This should connect, but with no SRTP extension neogtiated.
+  // The server side might negotiate a data channel only.
+  ConnectSocket();
+  ASSERT_NE(TLS_NULL_WITH_NULL_NULL, p1_->cipherSuite());
+  ASSERT_EQ(0, p1_->srtpCipher());
+}
+
+class TransportSrtpParameterTest
+    : public TransportTest,
+      public ::testing::WithParamInterface<uint16_t> {};
+
+INSTANTIATE_TEST_SUITE_P(
+    SrtpParamInit, TransportSrtpParameterTest,
+    ::testing::ValuesIn(TransportLayerDtls::GetDefaultSrtpCiphers()));
+
+TEST_P(TransportSrtpParameterTest, TestSrtpCiphersMismatchCombinations) {
+  uint16_t cipher = GetParam();
+  std::cerr << "Checking cipher: " << cipher << std::endl;
+
+  p1_->SetupSrtp();
+
+  std::vector<uint16_t> setB;
+  setB.push_back(cipher);
+
+  p2_->SetSrtpCiphers(setB);
+  SetDtlsPeer();
+  ConnectSocket();
+
+  ASSERT_EQ(cipher, p1_->srtpCipher());
+  ASSERT_EQ(cipher, p2_->srtpCipher());
+}
+
+// NSS doesn't support DHE suites on the server end.
+// This checks to see if we barf when that's the only option available.
+TEST_F(TransportTest, TestDheOnlyFails) {
+  SetDtlsPeer();
+
+  // p2_ is the client
+  // setting this on p1_ (the server) causes NSS to assert
+  ConfigureOneCipher(p2_, TLS_DHE_RSA_WITH_AES_128_CBC_SHA);
+  ConnectSocketExpectFail();
+}
+
+}  // end namespace
diff --git a/dom/media/webrtc/transport/test/turn_unittest.cpp b/dom/media/webrtc/transport/test/turn_unittest.cpp
new file mode 100644
index 0000000000..ae5d0386d9
--- /dev/null
+++ b/dom/media/webrtc/transport/test/turn_unittest.cpp
@@ -0,0 +1,432 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+// Some code copied from nICEr. License is:
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <stdlib.h>
+#include <iostream>
+
+#include "runnable_utils.h"
+
+#define GTEST_HAS_RTTI 0
+#include "gtest/gtest.h"
+#include "gtest_utils.h"
+
+#define USE_TURN
+
+// nICEr includes
+extern "C" {
+#include "nr_api.h"
+#include "transport_addr.h"
+#include "nr_crypto.h"
+#include "nr_socket.h"
+#include "nr_socket_local.h"
+#include "nr_socket_buffered_stun.h"
+#include "stun_client_ctx.h"
+#include "turn_client_ctx.h"
+}
+
+#include "nricectx.h"
+
+using namespace mozilla;
+
+static std::string kDummyTurnServer("192.0.2.1");  // From RFC 5737
+
+class TurnClient : public MtransportTest {
+ public:
+  TurnClient()
+      : MtransportTest(),
+        real_socket_(nullptr),
+        net_socket_(nullptr),
+        buffered_socket_(nullptr),
+        net_fd_(nullptr),
+        turn_ctx_(nullptr),
+        allocated_(false),
+        received_(0),
+        protocol_(IPPROTO_UDP) {}
+
+  ~TurnClient() = default;
+
+  static void SetUpTestCase() {
+    NrIceCtx::InitializeGlobals(NrIceCtx::GlobalConfig());
+  }
+
+  void SetTcp() { protocol_ = IPPROTO_TCP; }
+
+  void Init_s() {
+    int r;
+    nr_transport_addr addr;
+    r = nr_ip4_port_to_transport_addr(0, 0, protocol_, &addr);
+    ASSERT_EQ(0, r);
+
+    r = nr_socket_local_create(nullptr, &addr, &real_socket_);
+    ASSERT_EQ(0, r);
+
+    if (protocol_ == IPPROTO_TCP) {
+      int r = nr_socket_buffered_stun_create(
+          real_socket_, 100000, TURN_TCP_FRAMING, &buffered_socket_);
+      ASSERT_EQ(0, r);
+      net_socket_ = buffered_socket_;
+    } else {
+      net_socket_ = real_socket_;
+    }
+
+    r = nr_str_port_to_transport_addr(turn_server_.c_str(), 3478, protocol_,
+                                      &addr);
+    ASSERT_EQ(0, r);
+
+    std::vector<unsigned char> password_vec(turn_password_.begin(),
+                                            turn_password_.end());
+    Data password;
+    INIT_DATA(password, &password_vec[0], password_vec.size());
+    r = nr_turn_client_ctx_create("test", net_socket_, turn_user_.c_str(),
+                                  &password, &addr, nullptr, &turn_ctx_);
+    ASSERT_EQ(0, r);
+
+    r = nr_socket_getfd(net_socket_, &net_fd_);
+    ASSERT_EQ(0, r);
+
+    NR_ASYNC_WAIT(net_fd_, NR_ASYNC_WAIT_READ, socket_readable_cb, (void*)this);
+  }
+
+  void TearDown_s() {
+    nr_turn_client_ctx_destroy(&turn_ctx_);
+    if (net_fd_) {
+      NR_ASYNC_CANCEL(net_fd_, NR_ASYNC_WAIT_READ);
+    }
+
+    nr_socket_destroy(&buffered_socket_);
+  }
+
+  void TearDown() {
+    test_utils_->SyncDispatchToSTS(WrapRunnable(this, &TurnClient::TearDown_s));
+  }
+
+  void Allocate_s() {
+    Init_s();
+    ASSERT_TRUE(turn_ctx_);
+
+    int r = nr_turn_client_allocate(turn_ctx_, allocate_success_cb, this);
+    ASSERT_EQ(0, r);
+  }
+
+  void Allocate(bool expect_success = true) {
+    test_utils_->SyncDispatchToSTS(WrapRunnable(this, &TurnClient::Allocate_s));
+
+    if (expect_success) {
+      ASSERT_TRUE_WAIT(allocated_, 5000);
+    } else {
+      PR_Sleep(10000);
+      ASSERT_FALSE(allocated_);
+    }
+  }
+
+  void Allocated() {
+    if (turn_ctx_->state != NR_TURN_CLIENT_STATE_ALLOCATED) {
+      std::cerr << "Allocation failed" << std::endl;
+      return;
+    }
+    allocated_ = true;
+
+    int r;
+    nr_transport_addr addr;
+
+    r = nr_turn_client_get_relayed_address(turn_ctx_, &addr);
+    ASSERT_EQ(0, r);
+
+    relay_addr_ = addr.as_string;
+
+    std::cerr << "Allocation succeeded with addr=" << relay_addr_ << std::endl;
+  }
+
+  void Deallocate_s() {
+    ASSERT_TRUE(turn_ctx_);
+
+    std::cerr << "De-Allocating..." << std::endl;
+    int r = nr_turn_client_deallocate(turn_ctx_);
+    ASSERT_EQ(0, r);
+  }
+
+  void Deallocate() {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &TurnClient::Deallocate_s));
+  }
+
+  void RequestPermission_s(const std::string& target) {
+    nr_transport_addr addr;
+    int r;
+
+    // Expected pattern here is "IP4:127.0.0.1:3487"
+    ASSERT_EQ(0, target.compare(0, 4, "IP4:"));
+
+    size_t offset = target.rfind(':');
+    ASSERT_NE(std::string::npos, offset);
+
+    std::string host = target.substr(4, offset - 4);
+    std::string port = target.substr(offset + 1);
+
+    r = nr_str_port_to_transport_addr(host.c_str(), atoi(port.c_str()),
+                                      IPPROTO_UDP, &addr);
+    ASSERT_EQ(0, r);
+
+    r = nr_turn_client_ensure_perm(turn_ctx_, &addr);
+    ASSERT_EQ(0, r);
+  }
+
+  void RequestPermission(const std::string& target) {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &TurnClient::RequestPermission_s, target));
+  }
+
+  void Readable(NR_SOCKET s, int how, void* arg) {
+    // Re-arm
+    std::cerr << "Socket is readable" << std::endl;
+    NR_ASYNC_WAIT(s, how, socket_readable_cb, arg);
+
+    UCHAR buf[8192];
+    size_t len_s;
+    nr_transport_addr addr;
+
+    int r = nr_socket_recvfrom(net_socket_, buf, sizeof(buf), &len_s, 0, &addr);
+    if (r) {
+      std::cerr << "Error reading from socket" << std::endl;
+      return;
+    }
+
+    ASSERT_LT(len_s, (size_t)INT_MAX);
+    int len = (int)len_s;
+
+    if (nr_is_stun_response_message(buf, len)) {
+      std::cerr << "STUN response" << std::endl;
+      r = nr_turn_client_process_response(turn_ctx_, buf, len, &addr);
+
+      if (r && r != R_REJECTED && r != R_RETRY) {
+        std::cerr << "Error processing STUN: " << r << std::endl;
+      }
+    } else if (nr_is_stun_indication_message(buf, len)) {
+      std::cerr << "STUN indication" << std::endl;
+
+      /* Process the indication */
+      unsigned char data[NR_STUN_MAX_MESSAGE_SIZE];
+      size_t datal;
+      nr_transport_addr remote_addr;
+
+      r = nr_turn_client_parse_data_indication(
+          turn_ctx_, &addr, buf, len, data, &datal, sizeof(data), &remote_addr);
+      ASSERT_EQ(0, r);
+      std::cerr << "Received " << datal << " bytes from "
+                << remote_addr.as_string << std::endl;
+
+      received_ += datal;
+
+      for (size_t i = 0; i < datal; i++) {
+        ASSERT_EQ(i & 0xff, data[i]);
+      }
+    } else {
+      if (nr_is_stun_message(buf, len)) {
+        std::cerr << "STUN message of unexpected type" << std::endl;
+      } else {
+        std::cerr << "Not a STUN message" << std::endl;
+      }
+      return;
+    }
+  }
+
+  void SendTo_s(const std::string& target, int expect_return) {
+    nr_transport_addr addr;
+    int r;
+
+    // Expected pattern here is "IP4:127.0.0.1:3487"
+    ASSERT_EQ(0, target.compare(0, 4, "IP4:"));
+
+    size_t offset = target.rfind(':');
+    ASSERT_NE(std::string::npos, offset);
+
+    std::string host = target.substr(4, offset - 4);
+    std::string port = target.substr(offset + 1);
+
+    r = nr_str_port_to_transport_addr(host.c_str(), atoi(port.c_str()),
+                                      IPPROTO_UDP, &addr);
+    ASSERT_EQ(0, r);
+
+    unsigned char test[100];
+    for (size_t i = 0; i < sizeof(test); i++) {
+      test[i] = i & 0xff;
+    }
+
+    std::cerr << "Sending test message to " << target << " ..." << std::endl;
+
+    r = nr_turn_client_send_indication(turn_ctx_, test, sizeof(test), 0, &addr);
+    if (expect_return >= 0) {
+      ASSERT_EQ(expect_return, r);
+    }
+  }
+
+  void SendTo(const std::string& target, int expect_return = 0) {
+    test_utils_->SyncDispatchToSTS(
+        WrapRunnable(this, &TurnClient::SendTo_s, target, expect_return));
+  }
+
+  int received() const { return received_; }
+
+  static void socket_readable_cb(NR_SOCKET s, int how, void* arg) {
+    static_cast<TurnClient*>(arg)->Readable(s, how, arg);
+  }
+
+  static void allocate_success_cb(NR_SOCKET s, int how, void* arg) {
+    static_cast<TurnClient*>(arg)->Allocated();
+  }
+
+ protected:
+  std::string turn_server_;
+  nr_socket* real_socket_;
+  nr_socket* net_socket_;
+  nr_socket* buffered_socket_;
+  NR_SOCKET net_fd_;
+  nr_turn_client_ctx* turn_ctx_;
+  std::string relay_addr_;
+  bool allocated_;
+  int received_;
+  int protocol_;
+};
+
+TEST_F(TurnClient, Allocate) {
+  if (WarnIfTurnNotConfigured()) return;
+
+  Allocate();
+}
+
+TEST_F(TurnClient, AllocateTcp) {
+  if (WarnIfTurnNotConfigured()) return;
+
+  SetTcp();
+  Allocate();
+}
+
+TEST_F(TurnClient, AllocateAndHold) {
+  if (WarnIfTurnNotConfigured()) return;
+
+  Allocate();
+  PR_Sleep(20000);
+  ASSERT_TRUE(turn_ctx_->state == NR_TURN_CLIENT_STATE_ALLOCATED);
+}
+
+TEST_F(TurnClient, SendToSelf) {
+  if (WarnIfTurnNotConfigured()) return;
+
+  Allocate();
+  SendTo(relay_addr_);
+  ASSERT_TRUE_WAIT(received() == 100, 5000);
+  SendTo(relay_addr_);
+  ASSERT_TRUE_WAIT(received() == 200, 1000);
+}
+
+TEST_F(TurnClient, SendToSelfTcp) {
+  if (WarnIfTurnNotConfigured()) return;
+
+  SetTcp();
+  Allocate();
+  SendTo(relay_addr_);
+  ASSERT_TRUE_WAIT(received() == 100, 5000);
+  SendTo(relay_addr_);
+  ASSERT_TRUE_WAIT(received() == 200, 1000);
+}
+
+TEST_F(TurnClient, PermissionDenied) {
+  if (WarnIfTurnNotConfigured()) return;
+
+  Allocate();
+  RequestPermission(relay_addr_);
+  PR_Sleep(1000);
+
+  /* Fake a 403 response */
+  nr_turn_permission* perm;
+  perm = STAILQ_FIRST(&turn_ctx_->permissions);
+  ASSERT_TRUE(perm);
+  while (perm) {
+    perm->stun->last_error_code = 403;
+    std::cerr << "Set 403's on permission" << std::endl;
+    perm = STAILQ_NEXT(perm, entry);
+  }
+
+  SendTo(relay_addr_, R_NOT_PERMITTED);
+  ASSERT_TRUE(received() == 0);
+
+  // TODO: We should check if we can still send to a second destination, but
+  //      we would need a second TURN client as one client can only handle one
+  //      allocation (maybe as part of bug 1128128 ?).
+}
+
+TEST_F(TurnClient, DeallocateReceiveFailure) {
+  if (WarnIfTurnNotConfigured()) return;
+
+  Allocate();
+  SendTo(relay_addr_);
+  ASSERT_TRUE_WAIT(received() == 100, 5000);
+  Deallocate();
+  turn_ctx_->state = NR_TURN_CLIENT_STATE_ALLOCATED;
+  SendTo(relay_addr_);
+  PR_Sleep(1000);
+  ASSERT_TRUE(received() == 100);
+}
+
+TEST_F(TurnClient, DeallocateReceiveFailureTcp) {
+  if (WarnIfTurnNotConfigured()) return;
+
+  SetTcp();
+  Allocate();
+  SendTo(relay_addr_);
+  ASSERT_TRUE_WAIT(received() == 100, 5000);
+  Deallocate();
+  turn_ctx_->state = NR_TURN_CLIENT_STATE_ALLOCATED;
+  /* Either the connection got closed by the TURN server already, then the send
+   * is going to fail, which we simply ignore. Or the connection is still alive
+   * and we cand send the data, but it should not get forwarded to us. In either
+   * case we should not receive more data. */
+  SendTo(relay_addr_, -1);
+  PR_Sleep(1000);
+  ASSERT_TRUE(received() == 100);
+}
+
+TEST_F(TurnClient, AllocateDummyServer) {
+  if (WarnIfTurnNotConfigured()) return;
+
+  turn_server_ = kDummyTurnServer;
+  Allocate(false);
+}
diff --git a/dom/media/webrtc/transport/test/webrtcproxychannel_unittest.cpp b/dom/media/webrtc/transport/test/webrtcproxychannel_unittest.cpp
new file mode 100644
index 0000000000..5bfddc7a3f
--- /dev/null
+++ b/dom/media/webrtc/transport/test/webrtcproxychannel_unittest.cpp
@@ -0,0 +1,754 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include <algorithm>
+#include <mutex>
+
+#include "mozilla/net/WebrtcTCPSocket.h"
+#include "mozilla/net/WebrtcTCPSocketCallback.h"
+
+#include "nsISocketTransport.h"
+
+#define GTEST_HAS_RTTI 0
+#include "gtest/gtest.h"
+#include "gtest_utils.h"
+
+static const uint32_t kDefaultTestTimeout = 2000;
+static const char kReadData[] = "Hello, World!";
+static const size_t kReadDataLength = sizeof(kReadData) - 1;
+static const std::string kReadDataString =
+    std::string(kReadData, kReadDataLength);
+static int kDataLargeOuterLoopCount = 128;
+static int kDataLargeInnerLoopCount = 1024;
+
+namespace mozilla {
+
+using namespace net;
+using namespace testing;
+
+class WebrtcTCPSocketTestCallback;
+
+class FakeSocketTransportProvider : public nsISocketTransport {
+ public:
+  NS_DECL_THREADSAFE_ISUPPORTS
+
+  // nsISocketTransport
+  NS_IMETHOD GetHost(nsACString& aHost) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD GetPort(int32_t* aPort) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD GetScriptableOriginAttributes(
+      JSContext* cx, JS::MutableHandle<JS::Value> aOriginAttributes) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD SetScriptableOriginAttributes(
+      JSContext* cx, JS::Handle<JS::Value> aOriginAttributes) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  virtual nsresult GetOriginAttributes(
+      mozilla::OriginAttributes* _retval) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  virtual nsresult SetOriginAttributes(
+      const mozilla::OriginAttributes& aOriginAttrs) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD GetPeerAddr(mozilla::net::NetAddr* _retval) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD GetSelfAddr(mozilla::net::NetAddr* _retval) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD Bind(mozilla::net::NetAddr* aLocalAddr) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD GetScriptablePeerAddr(nsINetAddr** _retval) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD GetScriptableSelfAddr(nsINetAddr** _retval) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD GetTlsSocketControl(
+      nsITLSSocketControl** aTLSSocketControl) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD GetSecurityCallbacks(
+      nsIInterfaceRequestor** aSecurityCallbacks) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD SetSecurityCallbacks(
+      nsIInterfaceRequestor* aSecurityCallbacks) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD IsAlive(bool* _retval) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD GetTimeout(uint32_t aType, uint32_t* _retval) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD SetTimeout(uint32_t aType, uint32_t aValue) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD SetLinger(bool aPolarity, int16_t aTimeout) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD SetReuseAddrPort(bool reuseAddrPort) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD GetConnectionFlags(uint32_t* aConnectionFlags) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD SetConnectionFlags(uint32_t aConnectionFlags) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD SetIsPrivate(bool) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD GetTlsFlags(uint32_t* aTlsFlags) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD SetTlsFlags(uint32_t aTlsFlags) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD GetQoSBits(uint8_t* aQoSBits) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD SetQoSBits(uint8_t aQoSBits) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD GetRecvBufferSize(uint32_t* aRecvBufferSize) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD GetSendBufferSize(uint32_t* aSendBufferSize) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD GetKeepaliveEnabled(bool* aKeepaliveEnabled) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD SetKeepaliveEnabled(bool aKeepaliveEnabled) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD SetKeepaliveVals(int32_t keepaliveIdleTime,
+                              int32_t keepaliveRetryInterval) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD GetResetIPFamilyPreference(
+      bool* aResetIPFamilyPreference) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD GetEchConfigUsed(bool* aEchConfigUsed) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD SetEchConfig(const nsACString& aEchConfig) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD ResolvedByTRR(bool* _retval) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD GetEffectiveTRRMode(
+      nsIRequest::TRRMode* aEffectiveTRRMode) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD GetTrrSkipReason(nsITRRSkipReason::value* aSkipReason) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD GetRetryDnsIfPossible(bool* aRetryDns) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD GetStatus(nsresult* aStatus) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+
+  // nsITransport
+  NS_IMETHOD OpenInputStream(uint32_t aFlags, uint32_t aSegmentSize,
+                             uint32_t aSegmentCount,
+                             nsIInputStream** _retval) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD OpenOutputStream(uint32_t aFlags, uint32_t aSegmentSize,
+                              uint32_t aSegmentCount,
+                              nsIOutputStream** _retval) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+  NS_IMETHOD SetEventSink(nsITransportEventSink* aSink,
+                          nsIEventTarget* aEventTarget) override {
+    MOZ_ASSERT(false);
+    return NS_OK;
+  }
+
+  // fake except for these methods which are OK to call
+  // nsISocketTransport
+  NS_IMETHOD SetRecvBufferSize(uint32_t aRecvBufferSize) override {
+    return NS_OK;
+  }
+  NS_IMETHOD SetSendBufferSize(uint32_t aSendBufferSize) override {
+    return NS_OK;
+  }
+  // nsITransport
+  NS_IMETHOD Close(nsresult aReason) override { return NS_OK; }
+
+ protected:
+  virtual ~FakeSocketTransportProvider() = default;
+};
+
+NS_IMPL_ISUPPORTS(FakeSocketTransportProvider, nsISocketTransport, nsITransport)
+
+// Implements some common elements to WebrtcTCPSocketTestOutputStream and
+// WebrtcTCPSocketTestInputStream.
+class WebrtcTCPSocketTestStream {
+ public:
+  WebrtcTCPSocketTestStream();
+
+  void Fail() { mMustFail = true; }
+
+  size_t DataLength();
+  template <typename T>
+  void AppendElements(const T* aBuffer, size_t aLength);
+
+ protected:
+  virtual ~WebrtcTCPSocketTestStream() = default;
+
+  nsTArray<uint8_t> mData;
+  std::mutex mDataMutex;
+
+  bool mMustFail;
+};
+
+WebrtcTCPSocketTestStream::WebrtcTCPSocketTestStream() : mMustFail(false) {}
+
+template <typename T>
+void WebrtcTCPSocketTestStream::AppendElements(const T* aBuffer,
+                                               size_t aLength) {
+  std::lock_guard<std::mutex> guard(mDataMutex);
+  mData.AppendElements(aBuffer, aLength);
+}
+
+size_t WebrtcTCPSocketTestStream::DataLength() {
+  std::lock_guard<std::mutex> guard(mDataMutex);
+  return mData.Length();
+}
+
+class WebrtcTCPSocketTestInputStream : public nsIAsyncInputStream,
+                                       public WebrtcTCPSocketTestStream {
+ public:
+  NS_DECL_THREADSAFE_ISUPPORTS
+  NS_DECL_NSIASYNCINPUTSTREAM
+  NS_DECL_NSIINPUTSTREAM
+
+  WebrtcTCPSocketTestInputStream()
+      : mMaxReadSize(1024 * 1024), mAllowCallbacks(false) {}
+
+  void DoCallback();
+  void CallCallback(const nsCOMPtr<nsIInputStreamCallback>& aCallback);
+  void AllowCallbacks() { mAllowCallbacks = true; }
+
+  size_t mMaxReadSize;
+
+ protected:
+  virtual ~WebrtcTCPSocketTestInputStream() = default;
+
+ private:
+  nsCOMPtr<nsIInputStreamCallback> mCallback;
+  nsCOMPtr<nsIEventTarget> mCallbackTarget;
+
+  bool mAllowCallbacks;
+};
+
+NS_IMPL_ISUPPORTS(WebrtcTCPSocketTestInputStream, nsIAsyncInputStream,
+                  nsIInputStream)
+
+nsresult WebrtcTCPSocketTestInputStream::AsyncWait(
+    nsIInputStreamCallback* aCallback, uint32_t aFlags,
+    uint32_t aRequestedCount, nsIEventTarget* aEventTarget) {
+  MOZ_ASSERT(!aEventTarget, "no event target should be set");
+
+  mCallback = aCallback;
+  mCallbackTarget = NS_GetCurrentThread();
+
+  if (mAllowCallbacks && DataLength() > 0) {
+    DoCallback();
+  }
+
+  return NS_OK;
+}
+
+nsresult WebrtcTCPSocketTestInputStream::CloseWithStatus(nsresult aStatus) {
+  return Close();
+}
+
+nsresult WebrtcTCPSocketTestInputStream::Close() { return NS_OK; }
+
+nsresult WebrtcTCPSocketTestInputStream::Available(uint64_t* aAvailable) {
+  *aAvailable = DataLength();
+  return NS_OK;
+}
+
+nsresult WebrtcTCPSocketTestInputStream::StreamStatus() { return NS_OK; }
+
+nsresult WebrtcTCPSocketTestInputStream::Read(char* aBuffer, uint32_t aCount,
+                                              uint32_t* aRead) {
+  std::lock_guard<std::mutex> guard(mDataMutex);
+  if (mMustFail) {
+    return NS_ERROR_FAILURE;
+  }
+  *aRead = std::min({(size_t)aCount, mData.Length(), mMaxReadSize});
+  memcpy(aBuffer, mData.Elements(), *aRead);
+  mData.RemoveElementsAt(0, *aRead);
+  return *aRead > 0 ? NS_OK : NS_BASE_STREAM_WOULD_BLOCK;
+}
+
+nsresult WebrtcTCPSocketTestInputStream::ReadSegments(nsWriteSegmentFun aWriter,
+                                                      void* aClosure,
+                                                      uint32_t aCount,
+                                                      uint32_t* _retval) {
+  MOZ_ASSERT(false);
+  return NS_OK;
+}
+
+nsresult WebrtcTCPSocketTestInputStream::IsNonBlocking(bool* aIsNonBlocking) {
+  *aIsNonBlocking = true;
+  return NS_OK;
+}
+
+void WebrtcTCPSocketTestInputStream::CallCallback(
+    const nsCOMPtr<nsIInputStreamCallback>& aCallback) {
+  aCallback->OnInputStreamReady(this);
+}
+
+void WebrtcTCPSocketTestInputStream::DoCallback() {
+  if (mCallback) {
+    mCallbackTarget->Dispatch(
+        NewRunnableMethod<const nsCOMPtr<nsIInputStreamCallback>&>(
+            "WebrtcTCPSocketTestInputStream::DoCallback", this,
+            &WebrtcTCPSocketTestInputStream::CallCallback,
+            std::move(mCallback)));
+
+    mCallbackTarget = nullptr;
+  }
+}
+
+class WebrtcTCPSocketTestOutputStream : public nsIAsyncOutputStream,
+                                        public WebrtcTCPSocketTestStream {
+ public:
+  NS_DECL_THREADSAFE_ISUPPORTS
+  NS_DECL_NSIASYNCOUTPUTSTREAM
+  NS_DECL_NSIOUTPUTSTREAM
+
+  WebrtcTCPSocketTestOutputStream() : mMaxWriteSize(1024 * 1024) {}
+
+  void DoCallback();
+  void CallCallback(const nsCOMPtr<nsIOutputStreamCallback>& aCallback);
+
+  std::string DataString();
+
+  uint32_t mMaxWriteSize;
+
+ protected:
+  virtual ~WebrtcTCPSocketTestOutputStream() = default;
+
+ private:
+  nsCOMPtr<nsIOutputStreamCallback> mCallback;
+  nsCOMPtr<nsIEventTarget> mCallbackTarget;
+};
+
+NS_IMPL_ISUPPORTS(WebrtcTCPSocketTestOutputStream, nsIAsyncOutputStream,
+                  nsIOutputStream)
+
+nsresult WebrtcTCPSocketTestOutputStream::AsyncWait(
+    nsIOutputStreamCallback* aCallback, uint32_t aFlags,
+    uint32_t aRequestedCount, nsIEventTarget* aEventTarget) {
+  MOZ_ASSERT(!aEventTarget, "no event target should be set");
+
+  mCallback = aCallback;
+  mCallbackTarget = NS_GetCurrentThread();
+
+  return NS_OK;
+}
+
+nsresult WebrtcTCPSocketTestOutputStream::CloseWithStatus(nsresult aStatus) {
+  return Close();
+}
+
+nsresult WebrtcTCPSocketTestOutputStream::Close() { return NS_OK; }
+
+nsresult WebrtcTCPSocketTestOutputStream::Flush() { return NS_OK; }
+
+nsresult WebrtcTCPSocketTestOutputStream::StreamStatus() {
+  return mMustFail ? NS_ERROR_FAILURE : NS_OK;
+}
+
+nsresult WebrtcTCPSocketTestOutputStream::Write(const char* aBuffer,
+                                                uint32_t aCount,
+                                                uint32_t* aWrote) {
+  if (mMustFail) {
+    return NS_ERROR_FAILURE;
+  }
+  *aWrote = std::min(aCount, mMaxWriteSize);
+  AppendElements(aBuffer, *aWrote);
+  return NS_OK;
+}
+
+nsresult WebrtcTCPSocketTestOutputStream::WriteSegments(
+    nsReadSegmentFun aReader, void* aClosure, uint32_t aCount,
+    uint32_t* _retval) {
+  MOZ_ASSERT(false);
+  return NS_OK;
+}
+
+nsresult WebrtcTCPSocketTestOutputStream::WriteFrom(nsIInputStream* aFromStream,
+                                                    uint32_t aCount,
+                                                    uint32_t* _retval) {
+  MOZ_ASSERT(false);
+  return NS_OK;
+}
+
+nsresult WebrtcTCPSocketTestOutputStream::IsNonBlocking(bool* aIsNonBlocking) {
+  *aIsNonBlocking = true;
+  return NS_OK;
+}
+
+void WebrtcTCPSocketTestOutputStream::CallCallback(
+    const nsCOMPtr<nsIOutputStreamCallback>& aCallback) {
+  aCallback->OnOutputStreamReady(this);
+}
+
+void WebrtcTCPSocketTestOutputStream::DoCallback() {
+  if (mCallback) {
+    mCallbackTarget->Dispatch(
+        NewRunnableMethod<const nsCOMPtr<nsIOutputStreamCallback>&>(
+            "WebrtcTCPSocketTestOutputStream::CallCallback", this,
+            &WebrtcTCPSocketTestOutputStream::CallCallback,
+            std::move(mCallback)));
+
+    mCallbackTarget = nullptr;
+  }
+}
+
+std::string WebrtcTCPSocketTestOutputStream::DataString() {
+  std::lock_guard<std::mutex> guard(mDataMutex);
+  return std::string((char*)mData.Elements(), mData.Length());
+}
+
+// Fake as in not the real WebrtcTCPSocket but real enough
+class FakeWebrtcTCPSocket : public WebrtcTCPSocket {
+ public:
+  explicit FakeWebrtcTCPSocket(WebrtcTCPSocketCallback* aCallback)
+      : WebrtcTCPSocket(aCallback) {}
+
+ protected:
+  virtual ~FakeWebrtcTCPSocket() = default;
+
+  void InvokeOnClose(nsresult aReason) override;
+  void InvokeOnConnected() override;
+  void InvokeOnRead(nsTArray<uint8_t>&& aReadData) override;
+};
+
+void FakeWebrtcTCPSocket::InvokeOnClose(nsresult aReason) {
+  mProxyCallbacks->OnClose(aReason);
+}
+
+void FakeWebrtcTCPSocket::InvokeOnConnected() {
+  mProxyCallbacks->OnConnected("http"_ns);
+}
+
+void FakeWebrtcTCPSocket::InvokeOnRead(nsTArray<uint8_t>&& aReadData) {
+  mProxyCallbacks->OnRead(std::move(aReadData));
+}
+
+class WebrtcTCPSocketTest : public MtransportTest {
+ public:
+  WebrtcTCPSocketTest()
+      : MtransportTest(),
+        mSocketThread(nullptr),
+        mSocketTransport(nullptr),
+        mInputStream(nullptr),
+        mOutputStream(nullptr),
+        mChannel(nullptr),
+        mCallback(nullptr),
+        mOnCloseCalled(false),
+        mOnConnectedCalled(false) {}
+
+  // WebrtcTCPSocketCallback forwards from mCallback
+  void OnClose(nsresult aReason);
+  void OnConnected(const nsACString& aProxyType);
+  void OnRead(nsTArray<uint8_t>&& aReadData);
+
+  void SetUp() override;
+  void TearDown() override;
+
+  void DoTransportAvailable();
+
+  std::string ReadDataAsString();
+  std::string GetDataLarge();
+
+  nsCOMPtr<nsIEventTarget> mSocketThread;
+
+  nsCOMPtr<nsISocketTransport> mSocketTransport;
+  RefPtr<WebrtcTCPSocketTestInputStream> mInputStream;
+  RefPtr<WebrtcTCPSocketTestOutputStream> mOutputStream;
+  RefPtr<FakeWebrtcTCPSocket> mChannel;
+  RefPtr<WebrtcTCPSocketTestCallback> mCallback;
+
+  bool mOnCloseCalled;
+  bool mOnConnectedCalled;
+
+  size_t ReadDataLength();
+  template <typename T>
+  void AppendReadData(const T* aBuffer, size_t aLength);
+
+ private:
+  nsTArray<uint8_t> mReadData;
+  std::mutex mReadDataMutex;
+};
+
+class WebrtcTCPSocketTestCallback : public WebrtcTCPSocketCallback {
+ public:
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(WebrtcTCPSocketTestCallback, override)
+
+  explicit WebrtcTCPSocketTestCallback(WebrtcTCPSocketTest* aTest)
+      : mTest(aTest) {}
+
+  // WebrtcTCPSocketCallback
+  void OnClose(nsresult aReason) override;
+  void OnConnected(const nsACString& aProxyType) override;
+  void OnRead(nsTArray<uint8_t>&& aReadData) override;
+
+ protected:
+  virtual ~WebrtcTCPSocketTestCallback() = default;
+
+ private:
+  WebrtcTCPSocketTest* mTest;
+};
+
+void WebrtcTCPSocketTest::SetUp() {
+  nsresult rv;
+  // WebrtcTCPSocket's threading model is the same as mtransport
+  // all socket operations are done on the socket thread
+  // callbacks are invoked on the main thread
+  mSocketThread = do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &rv);
+  ASSERT_TRUE(NS_SUCCEEDED(rv));
+
+  mSocketTransport = new FakeSocketTransportProvider();
+  mInputStream = new WebrtcTCPSocketTestInputStream();
+  mOutputStream = new WebrtcTCPSocketTestOutputStream();
+  mCallback = new WebrtcTCPSocketTestCallback(this);
+  mChannel = new FakeWebrtcTCPSocket(mCallback.get());
+}
+
+void WebrtcTCPSocketTest::TearDown() {}
+
+// WebrtcTCPSocketCallback
+void WebrtcTCPSocketTest::OnRead(nsTArray<uint8_t>&& aReadData) {
+  AppendReadData(aReadData.Elements(), aReadData.Length());
+}
+
+void WebrtcTCPSocketTest::OnConnected(const nsACString& aProxyType) {
+  mOnConnectedCalled = true;
+}
+
+void WebrtcTCPSocketTest::OnClose(nsresult aReason) { mOnCloseCalled = true; }
+
+void WebrtcTCPSocketTest::DoTransportAvailable() {
+  if (!mSocketThread->IsOnCurrentThread()) {
+    mSocketThread->Dispatch(
+        NS_NewRunnableFunction("DoTransportAvailable", [this]() -> void {
+          nsresult rv;
+          rv = mChannel->OnTransportAvailable(mSocketTransport, mInputStream,
+                                              mOutputStream);
+          ASSERT_EQ(NS_OK, rv);
+        }));
+  } else {
+    // should always be called on the main thread
+    MOZ_ASSERT(0);
+  }
+}
+
+std::string WebrtcTCPSocketTest::ReadDataAsString() {
+  std::lock_guard<std::mutex> guard(mReadDataMutex);
+  return std::string((char*)mReadData.Elements(), mReadData.Length());
+}
+
+std::string WebrtcTCPSocketTest::GetDataLarge() {
+  std::string data;
+  for (int i = 0; i < kDataLargeOuterLoopCount * kDataLargeInnerLoopCount;
+       ++i) {
+    data += kReadData;
+  }
+  return data;
+}
+
+template <typename T>
+void WebrtcTCPSocketTest::AppendReadData(const T* aBuffer, size_t aLength) {
+  std::lock_guard<std::mutex> guard(mReadDataMutex);
+  mReadData.AppendElements(aBuffer, aLength);
+}
+
+size_t WebrtcTCPSocketTest::ReadDataLength() {
+  std::lock_guard<std::mutex> guard(mReadDataMutex);
+  return mReadData.Length();
+}
+
+void WebrtcTCPSocketTestCallback::OnClose(nsresult aReason) {
+  mTest->OnClose(aReason);
+}
+
+void WebrtcTCPSocketTestCallback::OnConnected(const nsACString& aProxyType) {
+  mTest->OnConnected(aProxyType);
+}
+
+void WebrtcTCPSocketTestCallback::OnRead(nsTArray<uint8_t>&& aReadData) {
+  mTest->OnRead(std::move(aReadData));
+}
+
+}  // namespace mozilla
+
+typedef mozilla::WebrtcTCPSocketTest WebrtcTCPSocketTest;
+
+TEST_F(WebrtcTCPSocketTest, SetUp) {}
+
+TEST_F(WebrtcTCPSocketTest, TransportAvailable) {
+  DoTransportAvailable();
+  ASSERT_TRUE_WAIT(mOnConnectedCalled, kDefaultTestTimeout);
+}
+
+TEST_F(WebrtcTCPSocketTest, Read) {
+  DoTransportAvailable();
+  ASSERT_TRUE_WAIT(mOnConnectedCalled, kDefaultTestTimeout);
+
+  mInputStream->AppendElements(kReadData, kReadDataLength);
+  mInputStream->DoCallback();
+
+  ASSERT_TRUE_WAIT(ReadDataAsString() == kReadDataString, kDefaultTestTimeout);
+}
+
+TEST_F(WebrtcTCPSocketTest, Write) {
+  DoTransportAvailable();
+  ASSERT_TRUE_WAIT(mOnConnectedCalled, kDefaultTestTimeout);
+
+  nsTArray<uint8_t> data;
+  data.AppendElements(kReadData, kReadDataLength);
+  mChannel->Write(std::move(data));
+
+  ASSERT_TRUE_WAIT(mChannel->CountUnwrittenBytes() == kReadDataLength,
+                   kDefaultTestTimeout);
+
+  mOutputStream->DoCallback();
+
+  ASSERT_TRUE_WAIT(mOutputStream->DataString() == kReadDataString,
+                   kDefaultTestTimeout);
+}
+
+TEST_F(WebrtcTCPSocketTest, ReadFail) {
+  DoTransportAvailable();
+  ASSERT_TRUE_WAIT(mOnConnectedCalled, kDefaultTestTimeout);
+
+  mInputStream->AppendElements(kReadData, kReadDataLength);
+  mInputStream->Fail();
+  mInputStream->DoCallback();
+
+  ASSERT_TRUE_WAIT(mOnCloseCalled, kDefaultTestTimeout);
+  ASSERT_EQ(0U, ReadDataLength());
+}
+
+TEST_F(WebrtcTCPSocketTest, WriteFail) {
+  DoTransportAvailable();
+  ASSERT_TRUE_WAIT(mOnConnectedCalled, kDefaultTestTimeout);
+
+  nsTArray<uint8_t> array;
+  array.AppendElements(kReadData, kReadDataLength);
+  mChannel->Write(std::move(array));
+
+  ASSERT_TRUE_WAIT(mChannel->CountUnwrittenBytes() == kReadDataLength,
+                   kDefaultTestTimeout);
+
+  mOutputStream->Fail();
+  mOutputStream->DoCallback();
+
+  ASSERT_TRUE_WAIT(mOnCloseCalled, kDefaultTestTimeout);
+  ASSERT_EQ(0U, mOutputStream->DataLength());
+}
+
+TEST_F(WebrtcTCPSocketTest, ReadLarge) {
+  DoTransportAvailable();
+  ASSERT_TRUE_WAIT(mOnConnectedCalled, kDefaultTestTimeout);
+
+  const std::string data = GetDataLarge();
+
+  mInputStream->AppendElements(data.c_str(), data.length());
+  // make sure reading loops more than once
+  mInputStream->mMaxReadSize = 3072;
+  mInputStream->AllowCallbacks();
+  mInputStream->DoCallback();
+
+  ASSERT_TRUE_WAIT(ReadDataAsString() == data, kDefaultTestTimeout);
+}
+
+TEST_F(WebrtcTCPSocketTest, WriteLarge) {
+  DoTransportAvailable();
+  ASSERT_TRUE_WAIT(mOnConnectedCalled, kDefaultTestTimeout);
+
+  const std::string data = GetDataLarge();
+
+  for (int i = 0; i < kDataLargeOuterLoopCount; ++i) {
+    nsTArray<uint8_t> array;
+    int chunkSize = kReadDataString.length() * kDataLargeInnerLoopCount;
+    int offset = i * chunkSize;
+    array.AppendElements(data.c_str() + offset, chunkSize);
+    mChannel->Write(std::move(array));
+  }
+
+  ASSERT_TRUE_WAIT(mChannel->CountUnwrittenBytes() == data.length(),
+                   kDefaultTestTimeout);
+
+  // make sure writing loops more than once per write request
+  mOutputStream->mMaxWriteSize = 1024;
+  mOutputStream->DoCallback();
+
+  ASSERT_TRUE_WAIT(mOutputStream->DataString() == data, kDefaultTestTimeout);
+}
diff --git a/dom/media/webrtc/transport/test_nr_socket.cpp b/dom/media/webrtc/transport/test_nr_socket.cpp
new file mode 100644
index 0000000000..064013a8b7
--- /dev/null
+++ b/dom/media/webrtc/transport/test_nr_socket.cpp
@@ -0,0 +1,1135 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+/*
+ */
+
+/*
+Based partially on original code from nICEr and nrappkit.
+
+nICEr copyright:
+
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+
+nrappkit copyright:
+
+   Copyright (C) 2001-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   ekr@rtfm.com  Thu Dec 20 20:14:49 2001
+*/
+
+// Original author: bcampen@mozilla.com [:bwc]
+
+extern "C" {
+#include "stun_msg.h"  // for NR_STUN_MAX_MESSAGE_SIZE
+#include "async_wait.h"
+#include "async_timer.h"
+#include "nr_socket.h"
+#include "stun.h"
+#include "transport_addr.h"
+}
+
+#include "mozilla/RefPtr.h"
+#include "test_nr_socket.h"
+
+namespace mozilla {
+
+static int test_nat_socket_create(void* obj, nr_transport_addr* addr,
+                                  nr_socket** sockp) {
+  RefPtr<NrSocketBase> sock = new TestNrSocket(static_cast<TestNat*>(obj));
+
+  int r, _status;
+
+  r = sock->create(addr);
+  if (r) ABORT(r);
+
+  r = nr_socket_create_int(static_cast<void*>(sock), sock->vtbl(), sockp);
+  if (r) ABORT(r);
+
+  _status = 0;
+
+  {
+    // We will release this reference in destroy(), not exactly the normal
+    // ownership model, but it is what it is.
+    NrSocketBase* dummy = sock.forget().take();
+    (void)dummy;
+  }
+
+abort:
+  return _status;
+}
+
+static int test_nat_socket_factory_destroy(void** obj) {
+  TestNat* nat = static_cast<TestNat*>(*obj);
+  *obj = nullptr;
+  nat->Release();
+  return 0;
+}
+
+static nr_socket_factory_vtbl test_nat_socket_factory_vtbl = {
+    test_nat_socket_create, test_nat_socket_factory_destroy};
+
+/* static */
+TestNat::NatBehavior TestNat::ToNatBehavior(const std::string& type) {
+  if (type.empty() || !type.compare("ENDPOINT_INDEPENDENT")) {
+    return TestNat::ENDPOINT_INDEPENDENT;
+  }
+  if (!type.compare("ADDRESS_DEPENDENT")) {
+    return TestNat::ADDRESS_DEPENDENT;
+  }
+  if (!type.compare("PORT_DEPENDENT")) {
+    return TestNat::PORT_DEPENDENT;
+  }
+
+  MOZ_ASSERT(false, "Invalid NAT behavior");
+  return TestNat::ENDPOINT_INDEPENDENT;
+}
+
+bool TestNat::has_port_mappings() const {
+  for (TestNrSocket* sock : sockets_) {
+    if (sock->has_port_mappings()) {
+      return true;
+    }
+  }
+  return false;
+}
+
+bool TestNat::is_my_external_tuple(const nr_transport_addr& addr) const {
+  for (TestNrSocket* sock : sockets_) {
+    if (sock->is_my_external_tuple(addr)) {
+      return true;
+    }
+  }
+
+  return false;
+}
+
+bool TestNat::is_an_internal_tuple(const nr_transport_addr& addr) const {
+  for (TestNrSocket* sock : sockets_) {
+    nr_transport_addr addr_behind_nat;
+    if (sock->getaddr(&addr_behind_nat)) {
+      MOZ_CRASH("TestNrSocket::getaddr failed!");
+    }
+
+    if (!nr_transport_addr_cmp(&addr, &addr_behind_nat,
+                               NR_TRANSPORT_ADDR_CMP_MODE_ALL)) {
+      return true;
+    }
+  }
+  return false;
+}
+
+int TestNat::create_socket_factory(nr_socket_factory** factorypp) {
+  int r = nr_socket_factory_create_int(this, &test_nat_socket_factory_vtbl,
+                                       factorypp);
+  if (!r) {
+    AddRef();
+  }
+  return r;
+}
+
+void TestNat::set_proxy_config(
+    std::shared_ptr<NrSocketProxyConfig> aProxyConfig) {
+  proxy_config_ = std::move(aProxyConfig);
+}
+
+TestNrSocket::TestNrSocket(TestNat* nat)
+    : nat_(nat), tls_(false), timer_handle_(nullptr) {
+  nat_->insert_socket(this);
+}
+
+TestNrSocket::~TestNrSocket() { nat_->erase_socket(this); }
+
+RefPtr<NrSocketBase> TestNrSocket::create_external_socket(
+    const nr_transport_addr& dest_addr) const {
+  MOZ_ASSERT(nat_->enabled_);
+  MOZ_ASSERT(!nat_->is_an_internal_tuple(dest_addr));
+
+  int r;
+  nr_transport_addr nat_external_addr;
+
+  // Open the socket on an arbitrary port, on the same address.
+  if ((r = nr_transport_addr_copy(&nat_external_addr,
+                                  &internal_socket_->my_addr()))) {
+    r_log(LOG_GENERIC, LOG_CRIT, "%s: Failure in nr_transport_addr_copy: %d",
+          __FUNCTION__, r);
+    return nullptr;
+  }
+
+  if ((r = nr_transport_addr_set_port(&nat_external_addr, 0))) {
+    r_log(LOG_GENERIC, LOG_CRIT,
+          "%s: Failure in nr_transport_addr_set_port: %d", __FUNCTION__, r);
+    return nullptr;
+  }
+
+  RefPtr<NrSocketBase> external_socket;
+  r = NrSocketBase::CreateSocket(&nat_external_addr, &external_socket,
+                                 nat_->proxy_config_);
+
+  if (r) {
+    r_log(LOG_GENERIC, LOG_CRIT, "%s: Failure in NrSocket::create: %d",
+          __FUNCTION__, r);
+    return nullptr;
+  }
+
+  return external_socket;
+}
+
+int TestNrSocket::create(nr_transport_addr* addr) {
+  tls_ = addr->tls;
+
+  r_log(LOG_GENERIC, LOG_DEBUG, "TestNrSocket %p create %s", this,
+        addr->as_string);
+  return NrSocketBase::CreateSocket(addr, &internal_socket_, nullptr);
+}
+
+int TestNrSocket::getaddr(nr_transport_addr* addrp) {
+  return internal_socket_->getaddr(addrp);
+}
+
+void TestNrSocket::close() {
+  r_log(LOG_GENERIC, LOG_DEBUG, "TestNrSocket %p %s closing", this,
+        internal_socket_->my_addr().as_string);
+  if (timer_handle_) {
+    NR_async_timer_cancel(timer_handle_);
+    timer_handle_ = nullptr;
+  }
+  internal_socket_->close();
+  for (RefPtr<PortMapping>& port_mapping : port_mappings_) {
+    port_mapping->external_socket_->close();
+  }
+}
+
+int TestNrSocket::listen(int backlog) {
+  MOZ_ASSERT(internal_socket_->my_addr().protocol == IPPROTO_TCP);
+  r_log(LOG_GENERIC, LOG_DEBUG, "TestNrSocket %p %s listening", this,
+        internal_socket_->my_addr().as_string);
+
+  return internal_socket_->listen(backlog);
+}
+
+int TestNrSocket::accept(nr_transport_addr* addrp, nr_socket** sockp) {
+  MOZ_ASSERT(internal_socket_->my_addr().protocol == IPPROTO_TCP);
+  int r = internal_socket_->accept(addrp, sockp);
+  if (r) {
+    return r;
+  }
+
+  if (nat_->enabled_ && !nat_->is_an_internal_tuple(*addrp)) {
+    nr_socket_destroy(sockp);
+    return R_IO_ERROR;
+  }
+
+  return 0;
+}
+
+void TestNrSocket::process_delayed_cb(NR_SOCKET s, int how, void* cb_arg) {
+  DeferredPacket* op = static_cast<DeferredPacket*>(cb_arg);
+  op->socket_->timer_handle_ = nullptr;
+  r_log(LOG_GENERIC, LOG_DEBUG, "TestNrSocket %s sending delayed STUN response",
+        op->internal_socket_->my_addr().as_string);
+  op->internal_socket_->sendto(op->buffer_.data(), op->buffer_.len(),
+                               op->flags_, &op->to_);
+
+  delete op;
+}
+
+int TestNrSocket::sendto(const void* msg, size_t len, int flags,
+                         const nr_transport_addr* to) {
+  MOZ_ASSERT(internal_socket_->my_addr().protocol != IPPROTO_TCP);
+  r_log(LOG_GENERIC, LOG_DEBUG, "TestNrSocket %p %s %s", this, __FUNCTION__,
+        to->as_string);
+
+  if (nat_->nat_delegate_ &&
+      nat_->nat_delegate_->on_sendto(nat_, msg, len, flags, to)) {
+    return nat_->error_code_for_drop_;
+  }
+
+  UCHAR* buf = static_cast<UCHAR*>(const_cast<void*>(msg));
+  if (nat_->block_stun_ && nr_is_stun_message(buf, len)) {
+    return nat_->error_code_for_drop_;
+  }
+
+  if (nr_is_stun_request_message(buf, len) &&
+      maybe_send_fake_response(buf, len, to)) {
+    return 0;
+  }
+
+  /* TODO: improve the functionality of this in bug 1253657 */
+  if (!nat_->enabled_ || nat_->is_an_internal_tuple(*to)) {
+    if (nat_->delay_stun_resp_ms_ && nr_is_stun_response_message(buf, len)) {
+      NR_ASYNC_TIMER_SET(
+          nat_->delay_stun_resp_ms_, process_delayed_cb,
+          new DeferredPacket(this, msg, len, flags, to, internal_socket_),
+          &timer_handle_);
+      return 0;
+    }
+    return internal_socket_->sendto(msg, len, flags, to);
+  }
+
+  destroy_stale_port_mappings();
+
+  if (to->protocol == IPPROTO_UDP && nat_->block_udp_) {
+    return nat_->error_code_for_drop_;
+  }
+
+  // Choose our port mapping based on our most selective criteria
+  PortMapping* port_mapping = get_port_mapping(
+      *to, std::max(nat_->filtering_type_, nat_->mapping_type_));
+
+  if (!port_mapping) {
+    // See if we have already made the external socket we need to use.
+    PortMapping* similar_port_mapping =
+        get_port_mapping(*to, nat_->mapping_type_);
+    RefPtr<NrSocketBase> external_socket;
+
+    if (similar_port_mapping) {
+      external_socket = similar_port_mapping->external_socket_;
+    } else {
+      external_socket = create_external_socket(*to);
+      if (!external_socket) {
+        MOZ_ASSERT(false);
+        return R_INTERNAL;
+      }
+    }
+
+    port_mapping = create_port_mapping(*to, external_socket);
+    port_mappings_.push_back(port_mapping);
+
+    if (poll_flags() & PR_POLL_READ) {
+      // Make sure the new port mapping is ready to receive traffic if the
+      // TestNrSocket is already waiting.
+      port_mapping->async_wait(NR_ASYNC_WAIT_READ, socket_readable_callback,
+                               this, (char*)__FUNCTION__, __LINE__);
+    }
+  }
+
+  // We probably don't want to propagate the flags, since this is a simulated
+  // external IP address.
+  return port_mapping->sendto(msg, len, *to);
+}
+
+int TestNrSocket::recvfrom(void* buf, size_t maxlen, size_t* len, int flags,
+                           nr_transport_addr* from) {
+  MOZ_ASSERT(internal_socket_->my_addr().protocol != IPPROTO_TCP);
+
+  if (!read_buffer_.empty()) {
+    UdpPacket& packet = read_buffer_.front();
+    *len = std::min(maxlen, packet.buffer_->len());
+    memcpy(buf, packet.buffer_->data(), *len);
+    nr_transport_addr_copy(from, &packet.remote_address_);
+    read_buffer_.pop_front();
+    return 0;
+  }
+
+  int r;
+  bool ingress_allowed = false;
+
+  if (readable_socket_) {
+    // If any of the external sockets got data, see if it will be passed through
+    r = readable_socket_->recvfrom(buf, maxlen, len, 0, from);
+    const nr_transport_addr to = readable_socket_->my_addr();
+    readable_socket_ = nullptr;
+    if (!r) {
+      PortMapping* port_mapping_used;
+      ingress_allowed = allow_ingress(to, *from, &port_mapping_used);
+      if (ingress_allowed) {
+        r_log(LOG_GENERIC, LOG_DEBUG, "TestNrSocket %s received from %s via %s",
+              internal_socket_->my_addr().as_string, from->as_string,
+              port_mapping_used->external_socket_->my_addr().as_string);
+        if (nat_->refresh_on_ingress_) {
+          port_mapping_used->last_used_ = PR_IntervalNow();
+        }
+      }
+    }
+  } else {
+    // If no external socket has data, see if there's any data that was sent
+    // directly to the TestNrSocket, and eat it if it isn't supposed to get
+    // through.
+    r = internal_socket_->recvfrom(buf, maxlen, len, flags, from);
+    if (!r) {
+      // We do not use allow_ingress() here because that only handles traffic
+      // landing on an external port.
+      ingress_allowed = (!nat_->enabled_ || nat_->is_an_internal_tuple(*from));
+      if (!ingress_allowed) {
+        r_log(LOG_GENERIC, LOG_INFO,
+              "TestNrSocket %s denying ingress from %s: "
+              "Not behind the same NAT",
+              internal_socket_->my_addr().as_string, from->as_string);
+      } else {
+        r_log(LOG_GENERIC, LOG_DEBUG, "TestNrSocket %s received from %s",
+              internal_socket_->my_addr().as_string, from->as_string);
+      }
+    }
+  }
+
+  // Kinda bad that we are forced to give the app a readable callback and then
+  // say "Oh, never mind...", but the alternative is to totally decouple the
+  // callbacks from STS and the callbacks the app sets. On the bright side, this
+  // speeds up unit tests where we are verifying that ingress is forbidden,
+  // since they'll get a readable callback and then an error, instead of having
+  // to wait for a timeout.
+  if (!ingress_allowed) {
+    *len = 0;
+    r = R_WOULDBLOCK;
+  }
+
+  return r;
+}
+
+bool TestNrSocket::allow_ingress(const nr_transport_addr& to,
+                                 const nr_transport_addr& from,
+                                 PortMapping** port_mapping_used) const {
+  // This is only called for traffic arriving at a port mapping
+  MOZ_ASSERT(nat_->enabled_);
+  MOZ_ASSERT(!nat_->is_an_internal_tuple(from));
+
+  // Find the port mapping (if any) that this packet landed on
+  for (PortMapping* port_mapping : port_mappings_) {
+    if (!nr_transport_addr_cmp(&to, &port_mapping->external_socket_->my_addr(),
+                               NR_TRANSPORT_ADDR_CMP_MODE_ALL)) {
+      *port_mapping_used = port_mapping;
+    }
+  }
+
+  if (NS_WARN_IF(!(*port_mapping_used))) {
+    MOZ_ASSERT(false);
+    r_log(LOG_GENERIC, LOG_INFO,
+          "TestNrSocket %s denying ingress from %s: "
+          "No port mapping for this local port! What?",
+          internal_socket_->my_addr().as_string, from.as_string);
+    return false;
+  }
+
+  if (!port_mapping_matches(**port_mapping_used, from, nat_->filtering_type_)) {
+    r_log(LOG_GENERIC, LOG_INFO,
+          "TestNrSocket %s denying ingress from %s: "
+          "Filtered (no port mapping for source)",
+          internal_socket_->my_addr().as_string, from.as_string);
+    return false;
+  }
+
+  if (is_port_mapping_stale(**port_mapping_used)) {
+    r_log(LOG_GENERIC, LOG_INFO,
+          "TestNrSocket %s denying ingress from %s: "
+          "Stale port mapping",
+          internal_socket_->my_addr().as_string, from.as_string);
+    return false;
+  }
+
+  if (!nat_->allow_hairpinning_ && nat_->is_my_external_tuple(from)) {
+    r_log(LOG_GENERIC, LOG_INFO,
+          "TestNrSocket %s denying ingress from %s: "
+          "Hairpinning disallowed",
+          internal_socket_->my_addr().as_string, from.as_string);
+    return false;
+  }
+
+  return true;
+}
+
+int TestNrSocket::connect(const nr_transport_addr* addr) {
+  r_log(LOG_GENERIC, LOG_DEBUG, "TestNrSocket %p %s connecting to %s", this,
+        internal_socket_->my_addr().as_string, addr->as_string);
+
+  if (connect_invoked_ || !port_mappings_.empty()) {
+    MOZ_CRASH("TestNrSocket::connect() called more than once!");
+    return R_INTERNAL;
+  }
+
+  if (maybe_get_redirect_targets(addr).isSome()) {
+    // If we are simulating STUN redirects for |addr|, we need to pretend that
+    // the TCP connection worked, since |addr| probably does not actually point
+    // at something that exists.
+    connect_fake_stun_address_.reset(new nr_transport_addr);
+    nr_transport_addr_copy(connect_fake_stun_address_.get(), addr);
+
+    // We dispatch this, otherwise nICEr can trip over its shoelaces
+    GetCurrentSerialEventTarget()->Dispatch(
+        NS_NewRunnableFunction("Async writeable callback for TestNrSocket",
+                               [this, self = RefPtr<TestNrSocket>(this)] {
+                                 if (poll_flags() & PR_POLL_WRITE) {
+                                   fire_callback(NR_ASYNC_WAIT_WRITE);
+                                 }
+                               }));
+
+    return R_WOULDBLOCK;
+  }
+
+  if (!nat_->enabled_ ||
+      addr->protocol == IPPROTO_UDP  // Horrible hack to allow default address
+                                     // discovery to work. Only works because
+                                     // we don't normally connect on UDP.
+      || nat_->is_an_internal_tuple(*addr)) {
+    // This will set connect_invoked_
+    return internal_socket_->connect(addr);
+  }
+
+  RefPtr<NrSocketBase> external_socket(create_external_socket(*addr));
+  if (!external_socket) {
+    return R_INTERNAL;
+  }
+
+  PortMapping* port_mapping = create_port_mapping(*addr, external_socket);
+  port_mappings_.push_back(port_mapping);
+  int r = port_mapping->external_socket_->connect(addr);
+  if (r && r != R_WOULDBLOCK) {
+    return r;
+  }
+
+  port_mapping->last_used_ = PR_IntervalNow();
+
+  if (poll_flags() & PR_POLL_READ) {
+    port_mapping->async_wait(NR_ASYNC_WAIT_READ,
+                             port_mapping_tcp_passthrough_callback, this,
+                             (char*)__FUNCTION__, __LINE__);
+  }
+
+  return r;
+}
+
+int TestNrSocket::write(const void* msg, size_t len, size_t* written) {
+  r_log(LOG_GENERIC, LOG_DEBUG, "TestNrSocket %p %s writing", this,
+        internal_socket_->my_addr().as_string);
+
+  UCHAR* buf = static_cast<UCHAR*>(const_cast<void*>(msg));
+
+  if (nat_->nat_delegate_ &&
+      nat_->nat_delegate_->on_write(nat_, msg, len, written)) {
+    return R_INTERNAL;
+  }
+
+  if (nat_->block_stun_ && nr_is_stun_message(buf, len)) {
+    // Should cause this socket to be abandoned
+    r_log(LOG_GENERIC, LOG_DEBUG,
+          "TestNrSocket %s dropping outgoing TCP "
+          "because it is configured to drop STUN",
+          my_addr().as_string);
+    return R_INTERNAL;
+  }
+
+  if (nr_is_stun_request_message(buf, len) && connect_fake_stun_address_ &&
+      maybe_send_fake_response(buf, len, connect_fake_stun_address_.get())) {
+    return 0;
+  }
+
+  if (nat_->block_tcp_ && !tls_) {
+    // Should cause this socket to be abandoned
+    r_log(LOG_GENERIC, LOG_DEBUG,
+          "TestNrSocket %s dropping outgoing TCP "
+          "because it is configured to drop TCP",
+          my_addr().as_string);
+    return R_INTERNAL;
+  }
+
+  if (nat_->block_tls_ && tls_) {
+    // Should cause this socket to be abandoned
+    r_log(LOG_GENERIC, LOG_DEBUG,
+          "TestNrSocket %s dropping outgoing TLS "
+          "because it is configured to drop TLS",
+          my_addr().as_string);
+    return R_INTERNAL;
+  }
+
+  if (port_mappings_.empty()) {
+    // The no-nat case, just pass call through.
+    r_log(LOG_GENERIC, LOG_DEBUG, "TestNrSocket %s writing",
+          my_addr().as_string);
+
+    return internal_socket_->write(msg, len, written);
+  }
+  destroy_stale_port_mappings();
+  if (port_mappings_.empty()) {
+    r_log(LOG_GENERIC, LOG_DEBUG,
+          "TestNrSocket %s dropping outgoing TCP "
+          "because the port mapping was stale",
+          my_addr().as_string);
+    return R_INTERNAL;
+  }
+  // This is TCP only
+  MOZ_ASSERT(port_mappings_.size() == 1);
+  r_log(LOG_GENERIC, LOG_DEBUG, "PortMapping %s -> %s writing",
+        port_mappings_.front()->external_socket_->my_addr().as_string,
+        port_mappings_.front()->remote_address_.as_string);
+  port_mappings_.front()->last_used_ = PR_IntervalNow();
+  return port_mappings_.front()->external_socket_->write(msg, len, written);
+}
+
+int TestNrSocket::read(void* buf, size_t maxlen, size_t* len) {
+  r_log(LOG_GENERIC, LOG_DEBUG, "TestNrSocket %p %s reading", this,
+        internal_socket_->my_addr().as_string);
+
+  if (!read_buffer_.empty()) {
+    r_log(LOG_GENERIC, LOG_DEBUG,
+          "TestNrSocket %p %s has stuff in read_buffer_", this,
+          internal_socket_->my_addr().as_string);
+    UdpPacket packet(std::move(read_buffer_.front()));
+    read_buffer_.pop_front();
+    *len = std::min(maxlen, packet.buffer_->len());
+    memcpy(buf, packet.buffer_->data(), *len);
+    if (*len != packet.buffer_->len()) {
+      // Put remaining bytes in new packet, at the front.
+      read_buffer_.emplace_front(packet.buffer_->data() + *len,
+                                 packet.buffer_->len() - *len,
+                                 packet.remote_address_);
+    }
+    return 0;
+  }
+
+  if (connect_fake_stun_address_) {
+    return R_WOULDBLOCK;
+  }
+
+  int r;
+
+  if (port_mappings_.empty()) {
+    r = internal_socket_->read(buf, maxlen, len);
+  } else {
+    MOZ_ASSERT(port_mappings_.size() == 1);
+    r = port_mappings_.front()->external_socket_->read(buf, maxlen, len);
+    if (!r && nat_->refresh_on_ingress_) {
+      port_mappings_.front()->last_used_ = PR_IntervalNow();
+    }
+  }
+
+  if (r) {
+    return r;
+  }
+
+  if (nat_->nat_delegate_ &&
+      nat_->nat_delegate_->on_read(nat_, buf, maxlen, len)) {
+    return R_INTERNAL;
+  }
+
+  if (nat_->block_tcp_ && !tls_) {
+    // Should cause this socket to be abandoned
+    return R_INTERNAL;
+  }
+
+  if (nat_->block_tls_ && tls_) {
+    // Should cause this socket to be abandoned
+    return R_INTERNAL;
+  }
+
+  UCHAR* cbuf = static_cast<UCHAR*>(const_cast<void*>(buf));
+  if (nat_->block_stun_ && nr_is_stun_message(cbuf, *len)) {
+    // Should cause this socket to be abandoned
+    return R_INTERNAL;
+  }
+
+  return r;
+}
+
+int TestNrSocket::async_wait(int how, NR_async_cb cb, void* cb_arg,
+                             char* function, int line) {
+  r_log(LOG_GENERIC, LOG_DEBUG, "TestNrSocket %s waiting for %s",
+        internal_socket_->my_addr().as_string,
+        how == NR_ASYNC_WAIT_READ ? "read" : "write");
+
+  int r;
+
+  if (how == NR_ASYNC_WAIT_READ) {
+    NrSocketBase::async_wait(how, cb, cb_arg, function, line);
+    if (!read_buffer_.empty()) {
+      fire_readable_callback();
+      return 0;
+    }
+
+    // Make sure we're waiting on the socket for the internal address
+    r = internal_socket_->async_wait(how, socket_readable_callback, this,
+                                     function, line);
+  } else {
+    if (connect_fake_stun_address_) {
+      // Fake TCP connection case; register the callback on this socket, not
+      // a real one.
+      return NrSocketBase::async_wait(how, cb, cb_arg, function, line);
+    }
+
+    // For write, just use the readiness of the internal socket, since we queue
+    // everything for the port mappings.
+    r = internal_socket_->async_wait(how, cb, cb_arg, function, line);
+  }
+
+  if (r) {
+    r_log(LOG_GENERIC, LOG_ERR,
+          "TestNrSocket %s failed to async_wait for "
+          "internal socket: %d\n",
+          internal_socket_->my_addr().as_string, r);
+    return r;
+  }
+
+  if (is_tcp_connection_behind_nat()) {
+    // Bypass all port-mapping related logic
+    return 0;
+  }
+
+  if (internal_socket_->my_addr().protocol == IPPROTO_TCP) {
+    // For a TCP connection through a simulated NAT, these signals are
+    // just passed through.
+    MOZ_ASSERT(port_mappings_.size() == 1);
+
+    return port_mappings_.front()->async_wait(
+        how, port_mapping_tcp_passthrough_callback, this, function, line);
+  }
+  if (how == NR_ASYNC_WAIT_READ) {
+    // For UDP port mappings, we decouple the writeable callbacks
+    for (PortMapping* port_mapping : port_mappings_) {
+      // Be ready to receive traffic on our port mappings
+      r = port_mapping->async_wait(how, socket_readable_callback, this,
+                                   function, line);
+      if (r) {
+        r_log(LOG_GENERIC, LOG_ERR,
+              "TestNrSocket %s failed to async_wait for "
+              "port mapping: %d\n",
+              internal_socket_->my_addr().as_string, r);
+        return r;
+      }
+    }
+  }
+
+  return 0;
+}
+
+void TestNrSocket::cancel_port_mapping_async_wait(int how) {
+  for (PortMapping* port_mapping : port_mappings_) {
+    port_mapping->cancel(how);
+  }
+}
+
+int TestNrSocket::cancel(int how) {
+  r_log(LOG_GENERIC, LOG_DEBUG, "TestNrSocket %s stop waiting for %s",
+        internal_socket_->my_addr().as_string,
+        how == NR_ASYNC_WAIT_READ ? "read" : "write");
+
+  if (connect_fake_stun_address_) {
+    return NrSocketBase::cancel(how);
+  }
+
+  // Writable callbacks are decoupled except for the TCP case
+  if (how == NR_ASYNC_WAIT_READ ||
+      internal_socket_->my_addr().protocol == IPPROTO_TCP) {
+    cancel_port_mapping_async_wait(how);
+  }
+
+  return internal_socket_->cancel(how);
+}
+
+bool TestNrSocket::has_port_mappings() const { return !port_mappings_.empty(); }
+
+bool TestNrSocket::is_my_external_tuple(const nr_transport_addr& addr) const {
+  for (PortMapping* port_mapping : port_mappings_) {
+    nr_transport_addr port_mapping_addr;
+    if (port_mapping->external_socket_->getaddr(&port_mapping_addr)) {
+      MOZ_CRASH("NrSocket::getaddr failed!");
+    }
+
+    if (!nr_transport_addr_cmp(&addr, &port_mapping_addr,
+                               NR_TRANSPORT_ADDR_CMP_MODE_ALL)) {
+      return true;
+    }
+  }
+  return false;
+}
+
+bool TestNrSocket::is_port_mapping_stale(
+    const PortMapping& port_mapping) const {
+  PRIntervalTime now = PR_IntervalNow();
+  PRIntervalTime elapsed_ticks = now - port_mapping.last_used_;
+  uint32_t idle_duration = PR_IntervalToMilliseconds(elapsed_ticks);
+  return idle_duration > nat_->mapping_timeout_;
+}
+
+void TestNrSocket::destroy_stale_port_mappings() {
+  for (auto i = port_mappings_.begin(); i != port_mappings_.end();) {
+    auto temp = i;
+    ++i;
+    if (is_port_mapping_stale(**temp)) {
+      r_log(LOG_GENERIC, LOG_INFO,
+            "TestNrSocket %s destroying port mapping %s -> %s",
+            internal_socket_->my_addr().as_string,
+            (*temp)->external_socket_->my_addr().as_string,
+            (*temp)->remote_address_.as_string);
+
+      port_mappings_.erase(temp);
+    }
+  }
+}
+
+void TestNrSocket::socket_readable_callback(void* real_sock_v, int how,
+                                            void* test_sock_v) {
+  TestNrSocket* test_socket = static_cast<TestNrSocket*>(test_sock_v);
+  NrSocketBase* real_socket = static_cast<NrSocketBase*>(real_sock_v);
+
+  test_socket->on_socket_readable(real_socket);
+}
+
+void TestNrSocket::on_socket_readable(NrSocketBase* real_socket) {
+  if (!readable_socket_ && (real_socket != internal_socket_)) {
+    readable_socket_ = real_socket;
+  }
+
+  fire_readable_callback();
+}
+
+void TestNrSocket::fire_readable_callback() {
+  MOZ_ASSERT(poll_flags() & PR_POLL_READ);
+  r_log(LOG_GENERIC, LOG_DEBUG, "TestNrSocket %p %s ready for read", this,
+        internal_socket_->my_addr().as_string);
+  fire_callback(NR_ASYNC_WAIT_READ);
+}
+
+void TestNrSocket::port_mapping_writeable_callback(void* ext_sock_v, int how,
+                                                   void* test_sock_v) {
+  TestNrSocket* test_socket = static_cast<TestNrSocket*>(test_sock_v);
+  NrSocketBase* external_socket = static_cast<NrSocketBase*>(ext_sock_v);
+
+  test_socket->write_to_port_mapping(external_socket);
+}
+
+void TestNrSocket::write_to_port_mapping(NrSocketBase* external_socket) {
+  MOZ_ASSERT(internal_socket_->my_addr().protocol != IPPROTO_TCP);
+
+  int r = 0;
+  for (PortMapping* port_mapping : port_mappings_) {
+    if (port_mapping->external_socket_ == external_socket) {
+      // If the send succeeds, or if there was nothing to send, we keep going
+      r = port_mapping->send_from_queue();
+      if (r) {
+        break;
+      }
+    }
+  }
+
+  if (r == R_WOULDBLOCK) {
+    // Re-register for writeable callbacks, since we still have stuff to send
+    NR_ASYNC_WAIT(external_socket, NR_ASYNC_WAIT_WRITE,
+                  &TestNrSocket::port_mapping_writeable_callback, this);
+  }
+}
+
+void TestNrSocket::port_mapping_tcp_passthrough_callback(void* ext_sock_v,
+                                                         int how,
+                                                         void* test_sock_v) {
+  TestNrSocket* test_socket = static_cast<TestNrSocket*>(test_sock_v);
+  r_log(LOG_GENERIC, LOG_DEBUG, "TestNrSocket %s firing %s callback",
+        test_socket->internal_socket_->my_addr().as_string,
+        how == NR_ASYNC_WAIT_READ ? "readable" : "writeable");
+
+  test_socket->internal_socket_->fire_callback(how);
+}
+
+bool TestNrSocket::is_tcp_connection_behind_nat() const {
+  return internal_socket_->my_addr().protocol == IPPROTO_TCP &&
+         port_mappings_.empty();
+}
+
+TestNrSocket::PortMapping* TestNrSocket::get_port_mapping(
+    const nr_transport_addr& remote_address,
+    TestNat::NatBehavior filter) const {
+  for (PortMapping* port_mapping : port_mappings_) {
+    if (port_mapping_matches(*port_mapping, remote_address, filter)) {
+      return port_mapping;
+    }
+  }
+  return nullptr;
+}
+
+/* static */
+bool TestNrSocket::port_mapping_matches(const PortMapping& port_mapping,
+                                        const nr_transport_addr& remote_addr,
+                                        TestNat::NatBehavior filter) {
+  int compare_flags;
+  switch (filter) {
+    case TestNat::ENDPOINT_INDEPENDENT:
+      compare_flags = NR_TRANSPORT_ADDR_CMP_MODE_PROTOCOL;
+      break;
+    case TestNat::ADDRESS_DEPENDENT:
+      compare_flags = NR_TRANSPORT_ADDR_CMP_MODE_ADDR;
+      break;
+    case TestNat::PORT_DEPENDENT:
+      compare_flags = NR_TRANSPORT_ADDR_CMP_MODE_ALL;
+      break;
+  }
+
+  return !nr_transport_addr_cmp(&remote_addr, &port_mapping.remote_address_,
+                                compare_flags);
+}
+
+TestNrSocket::PortMapping* TestNrSocket::create_port_mapping(
+    const nr_transport_addr& remote_address,
+    const RefPtr<NrSocketBase>& external_socket) const {
+  r_log(LOG_GENERIC, LOG_INFO, "TestNrSocket %s creating port mapping %s -> %s",
+        internal_socket_->my_addr().as_string,
+        external_socket->my_addr().as_string, remote_address.as_string);
+
+  return new PortMapping(remote_address, external_socket);
+}
+
+TestNrSocket::PortMapping::PortMapping(
+    const nr_transport_addr& remote_address,
+    const RefPtr<NrSocketBase>& external_socket)
+    : external_socket_(external_socket) {
+  nr_transport_addr_copy(&remote_address_, &remote_address);
+}
+
+int TestNrSocket::PortMapping::send_from_queue() {
+  MOZ_ASSERT(remote_address_.protocol != IPPROTO_TCP);
+  int r = 0;
+
+  while (!send_queue_.empty()) {
+    UdpPacket& packet = send_queue_.front();
+    r_log(LOG_GENERIC, LOG_DEBUG,
+          "PortMapping %s -> %s sending from queue to %s",
+          external_socket_->my_addr().as_string, remote_address_.as_string,
+          packet.remote_address_.as_string);
+
+    r = external_socket_->sendto(packet.buffer_->data(), packet.buffer_->len(),
+                                 0, &packet.remote_address_);
+
+    if (r) {
+      if (r != R_WOULDBLOCK) {
+        r_log(LOG_GENERIC, LOG_ERR, "%s: Fatal error %d, stop trying",
+              __FUNCTION__, r);
+        send_queue_.clear();
+      } else {
+        r_log(LOG_GENERIC, LOG_DEBUG, "Would block, will retry later");
+      }
+      break;
+    }
+
+    send_queue_.pop_front();
+  }
+
+  return r;
+}
+
+int TestNrSocket::PortMapping::sendto(const void* msg, size_t len,
+                                      const nr_transport_addr& to) {
+  MOZ_ASSERT(remote_address_.protocol != IPPROTO_TCP);
+  r_log(LOG_GENERIC, LOG_DEBUG, "PortMapping %s -> %s sending to %s",
+        external_socket_->my_addr().as_string, remote_address_.as_string,
+        to.as_string);
+
+  last_used_ = PR_IntervalNow();
+  int r = external_socket_->sendto(msg, len, 0, &to);
+
+  if (r == R_WOULDBLOCK) {
+    r_log(LOG_GENERIC, LOG_DEBUG, "Enqueueing UDP packet to %s", to.as_string);
+    send_queue_.emplace_back(msg, len, to);
+    return 0;
+  }
+  if (r) {
+    r_log(LOG_GENERIC, LOG_ERR, "Error: %d", r);
+  }
+
+  return r;
+}
+
+int TestNrSocket::PortMapping::async_wait(int how, NR_async_cb cb, void* cb_arg,
+                                          char* function, int line) {
+  r_log(LOG_GENERIC, LOG_DEBUG, "PortMapping %s -> %s waiting for %s",
+        external_socket_->my_addr().as_string, remote_address_.as_string,
+        how == NR_ASYNC_WAIT_READ ? "read" : "write");
+
+  return external_socket_->async_wait(how, cb, cb_arg, function, line);
+}
+
+int TestNrSocket::PortMapping::cancel(int how) {
+  r_log(LOG_GENERIC, LOG_DEBUG, "PortMapping %s -> %s stop waiting for %s",
+        external_socket_->my_addr().as_string, remote_address_.as_string,
+        how == NR_ASYNC_WAIT_READ ? "read" : "write");
+
+  return external_socket_->cancel(how);
+}
+
+class nr_stun_message_deleter {
+ public:
+  nr_stun_message_deleter() = default;
+  void operator()(nr_stun_message* msg) const { nr_stun_message_destroy(&msg); }
+};
+
+bool TestNrSocket::maybe_send_fake_response(const void* msg, size_t len,
+                                            const nr_transport_addr* to) {
+  Maybe<nsTArray<nsCString>> redirect_targets = maybe_get_redirect_targets(to);
+  if (!redirect_targets.isSome()) {
+    return false;
+  }
+
+  std::unique_ptr<nr_stun_message, nr_stun_message_deleter> request;
+  {
+    nr_stun_message* temp = nullptr;
+    if (NS_WARN_IF(nr_stun_message_create2(&temp, (unsigned char*)msg, len))) {
+      return false;
+    }
+    request.reset(temp);
+  }
+
+  if (NS_WARN_IF(nr_stun_decode_message(request.get(), nullptr, nullptr))) {
+    return false;
+  }
+
+  std::unique_ptr<nr_stun_message, nr_stun_message_deleter> response;
+  {
+    nr_stun_message* temp = nullptr;
+    if (nr_stun_message_create(&temp)) {
+      MOZ_CRASH("nr_stun_message_create failed!");
+    }
+    response.reset(temp);
+  }
+
+  nr_stun_form_error_response(request.get(), response.get(), 300,
+                              (char*)"Try alternate");
+
+  int port = 0;
+  if (nr_transport_addr_get_port(to, &port)) {
+    MOZ_CRASH();
+  }
+
+  for (const nsCString& address : *redirect_targets) {
+    r_log(LOG_GENERIC, LOG_DEBUG,
+          "TestNrSocket attempting to add alternate server %s", address.Data());
+    nr_transport_addr addr;
+    if (NS_WARN_IF(nr_str_port_to_transport_addr(address.Data(), port,
+                                                 IPPROTO_UDP, &addr))) {
+      continue;
+    }
+    if (nr_stun_message_add_alternate_server_attribute(response.get(), &addr)) {
+      MOZ_CRASH("nr_stun_message_add_alternate_server_attribute failed!");
+    }
+  }
+
+  if (nr_stun_encode_message(response.get())) {
+    MOZ_CRASH("nr_stun_encode_message failed!");
+  }
+
+  nr_transport_addr response_from;
+  if (nr_transport_addr_is_wildcard(to)) {
+    // |to| points to an FQDN, and nICEr is delegating DNS lookup to us; we
+    // aren't _actually_ going to do that though, so we select a bogus address
+    // for the response to come from. TEST-NET is a fairly reasonable thing to
+    // use for this.
+    int port = 0;
+    if (nr_transport_addr_get_port(to, &port)) {
+      MOZ_CRASH();
+    }
+    switch (to->ip_version) {
+      case NR_IPV4:
+        if (nr_str_port_to_transport_addr("198.51.100.1", port, to->protocol,
+                                          &response_from)) {
+          MOZ_CRASH();
+        }
+        break;
+      case NR_IPV6:
+        if (nr_str_port_to_transport_addr("::ffff:198.51.100.1", port,
+                                          to->protocol, &response_from)) {
+          MOZ_CRASH();
+        }
+        break;
+      default:
+        MOZ_CRASH();
+    }
+  } else {
+    nr_transport_addr_copy(&response_from, to);
+  }
+
+  read_buffer_.emplace_back(response->buffer, response->length, response_from);
+
+  // We dispatch this, otherwise nICEr can trip over its shoelaces
+  r_log(LOG_GENERIC, LOG_DEBUG,
+        "TestNrSocket %p scheduling callback for redirect response", this);
+  GetCurrentSerialEventTarget()->Dispatch(NS_NewRunnableFunction(
+      "Async readable callback for TestNrSocket",
+      [this, self = RefPtr<TestNrSocket>(this)] {
+        if (poll_flags() & PR_POLL_READ) {
+          fire_readable_callback();
+        } else {
+          r_log(LOG_GENERIC, LOG_DEBUG,
+                "TestNrSocket %p deferring callback for redirect response",
+                this);
+        }
+      }));
+
+  return true;
+}
+
+Maybe<nsTArray<nsCString>> TestNrSocket::maybe_get_redirect_targets(
+    const nr_transport_addr* to) const {
+  Maybe<nsTArray<nsCString>> result;
+
+  // 256 is overkill, but it hardly matters
+  char addrstring[256];
+  if (nr_transport_addr_get_addrstring(to, addrstring, 256)) {
+    MOZ_CRASH("nr_transport_addr_get_addrstring failed!");
+  }
+
+  r_log(LOG_GENERIC, LOG_DEBUG, "TestNrSocket checking redirect rules for %s",
+        addrstring);
+  auto it = nat_->stun_redirect_map_.find(nsCString(addrstring));
+  if (it != nat_->stun_redirect_map_.end()) {
+    result = Some(it->second);
+  }
+
+  return result;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/transport/test_nr_socket.h b/dom/media/webrtc/transport/test_nr_socket.h
new file mode 100644
index 0000000000..c6a796c063
--- /dev/null
+++ b/dom/media/webrtc/transport/test_nr_socket.h
@@ -0,0 +1,370 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+/*
+ */
+
+/*
+Based partially on original code from nICEr and nrappkit.
+
+nICEr copyright:
+
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+
+
+nrappkit copyright:
+
+   Copyright (C) 2001-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   ekr@rtfm.com  Thu Dec 20 20:14:49 2001
+*/
+
+// Original author: bcampen@mozilla.com [:bwc]
+
+#ifndef test_nr_socket__
+#define test_nr_socket__
+
+extern "C" {
+#include "transport_addr.h"
+}
+
+#include "nr_socket_prsock.h"
+
+extern "C" {
+#include "nr_socket.h"
+}
+
+#include <set>
+#include <map>
+#include <list>
+#include <string>
+
+#include "mozilla/UniquePtr.h"
+#include "prinrval.h"
+#include "mediapacket.h"
+
+namespace mozilla {
+
+class TestNrSocket;
+class NrSocketProxyConfig;
+
+/**
+ * A group of TestNrSockets that behave as if they were behind the same NAT.
+ * @note We deliberately avoid addref/release of TestNrSocket here to avoid
+ *       masking lifetime errors elsewhere.
+ */
+class TestNat {
+ public:
+  /**
+   * This allows TestNat traffic to be passively inspected.
+   * If a non-zero (error) value is returned, the packet will be dropped,
+   * allowing for tests to extend how packet manipulation is done by
+   * TestNat with having to modify TestNat itself.
+   */
+  class NatDelegate {
+   public:
+    virtual int on_read(TestNat* nat, void* buf, size_t maxlen,
+                        size_t* len) = 0;
+    virtual int on_sendto(TestNat* nat, const void* msg, size_t len, int flags,
+                          const nr_transport_addr* to) = 0;
+    virtual int on_write(TestNat* nat, const void* msg, size_t len,
+                         size_t* written) = 0;
+  };
+
+  typedef enum {
+    /** For mapping, one port is used for all destinations.
+     *  For filtering, allow any external address/port. */
+    ENDPOINT_INDEPENDENT,
+
+    /** For mapping, one port for each destination address (for any port).
+     *  For filtering, allow incoming traffic from addresses that outgoing
+     *  traffic has been sent to. */
+    ADDRESS_DEPENDENT,
+
+    /** For mapping, one port for each destination address/port.
+     *  For filtering, allow incoming traffic only from addresses/ports that
+     *  outgoing traffic has been sent to. */
+    PORT_DEPENDENT,
+  } NatBehavior;
+
+  TestNat()
+      : enabled_(false),
+        filtering_type_(ENDPOINT_INDEPENDENT),
+        mapping_type_(ENDPOINT_INDEPENDENT),
+        mapping_timeout_(30000),
+        allow_hairpinning_(false),
+        refresh_on_ingress_(false),
+        block_udp_(false),
+        block_stun_(false),
+        block_tcp_(false),
+        block_tls_(false),
+        error_code_for_drop_(0),
+        delay_stun_resp_ms_(0),
+        nat_delegate_(nullptr),
+        sockets_() {}
+
+  bool has_port_mappings() const;
+
+  // Helps determine whether we're hairpinning
+  bool is_my_external_tuple(const nr_transport_addr& addr) const;
+  bool is_an_internal_tuple(const nr_transport_addr& addr) const;
+
+  int create_socket_factory(nr_socket_factory** factorypp);
+
+  void insert_socket(TestNrSocket* socket) { sockets_.insert(socket); }
+
+  void erase_socket(TestNrSocket* socket) { sockets_.erase(socket); }
+
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(TestNat);
+
+  static NatBehavior ToNatBehavior(const std::string& type);
+
+  void set_proxy_config(std::shared_ptr<NrSocketProxyConfig> aProxyConfig);
+
+  bool enabled_;
+  TestNat::NatBehavior filtering_type_;
+  TestNat::NatBehavior mapping_type_;
+  uint32_t mapping_timeout_;
+  bool allow_hairpinning_;
+  bool refresh_on_ingress_;
+  bool block_udp_;
+  bool block_stun_;
+  bool block_tcp_;
+  bool block_tls_;
+  bool error_code_for_drop_;
+  /* Note: this can only delay a single response so far (bug 1253657) */
+  uint32_t delay_stun_resp_ms_;
+
+  // When we see an outgoing STUN request with a destination address or
+  // destination FQDN that matches a key in this map, we respond with a STUN/300
+  // with a list of ALTERNATE-SERVER fields based on the value in this map.
+  std::map<nsCString, CopyableTArray<nsCString>> stun_redirect_map_;
+
+  NatDelegate* nat_delegate_;
+  std::shared_ptr<NrSocketProxyConfig> proxy_config_;
+
+ private:
+  std::set<TestNrSocket*> sockets_;
+
+  ~TestNat() = default;
+};
+
+/**
+ * Subclass of NrSocketBase that can simulate things like being behind a NAT,
+ * packet loss, latency, packet rewriting, etc. Also exposes some stuff that
+ * assists in diagnostics.
+ * This is accomplished by wrapping an "internal" socket (that handles traffic
+ * behind the NAT), and a collection of "external" sockets (that handle traffic
+ * into/out of the NAT)
+ */
+class TestNrSocket : public NrSocketBase {
+ public:
+  explicit TestNrSocket(TestNat* nat);
+
+  bool has_port_mappings() const;
+  bool is_my_external_tuple(const nr_transport_addr& addr) const;
+
+  // Overrides of NrSocketBase
+  int create(nr_transport_addr* addr) override;
+  int sendto(const void* msg, size_t len, int flags,
+             const nr_transport_addr* to) override;
+  int recvfrom(void* buf, size_t maxlen, size_t* len, int flags,
+               nr_transport_addr* from) override;
+  int getaddr(nr_transport_addr* addrp) override;
+  void close() override;
+  int connect(const nr_transport_addr* addr) override;
+  int write(const void* msg, size_t len, size_t* written) override;
+  int read(void* buf, size_t maxlen, size_t* len) override;
+
+  int listen(int backlog) override;
+  int accept(nr_transport_addr* addrp, nr_socket** sockp) override;
+  int async_wait(int how, NR_async_cb cb, void* cb_arg, char* function,
+                 int line) override;
+  int cancel(int how) override;
+
+  // Need override since this is virtual in NrSocketBase
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(TestNrSocket, override)
+
+ private:
+  virtual ~TestNrSocket();
+
+  class UdpPacket {
+   public:
+    UdpPacket(const void* msg, size_t len, const nr_transport_addr& addr)
+        : buffer_(new MediaPacket) {
+      buffer_->Copy(static_cast<const uint8_t*>(msg), len);
+      nr_transport_addr_copy(&remote_address_, &addr);
+    }
+
+    UdpPacket(UdpPacket&& aOrig) = default;
+
+    ~UdpPacket() = default;
+
+    nr_transport_addr remote_address_;
+    UniquePtr<MediaPacket> buffer_;
+  };
+
+  class PortMapping {
+   public:
+    PortMapping(const nr_transport_addr& remote_address,
+                const RefPtr<NrSocketBase>& external_socket);
+
+    int sendto(const void* msg, size_t len, const nr_transport_addr& to);
+    int async_wait(int how, NR_async_cb cb, void* cb_arg, char* function,
+                   int line);
+    int cancel(int how);
+    int send_from_queue();
+    NS_INLINE_DECL_THREADSAFE_REFCOUNTING(PortMapping);
+
+    PRIntervalTime last_used_;
+    RefPtr<NrSocketBase> external_socket_;
+    // For non-symmetric, most of the data here doesn't matter
+    nr_transport_addr remote_address_;
+
+   private:
+    ~PortMapping() { external_socket_->close(); }
+
+    // If external_socket_ returns E_WOULDBLOCK, we don't want to propagate
+    // that to the code using the TestNrSocket. We can also perhaps use this
+    // to help simulate things like latency.
+    std::list<UdpPacket> send_queue_;
+  };
+
+  struct DeferredPacket {
+    DeferredPacket(TestNrSocket* sock, const void* data, size_t len, int flags,
+                   const nr_transport_addr* addr,
+                   RefPtr<NrSocketBase> internal_socket)
+        : socket_(sock),
+          buffer_(),
+          flags_(flags),
+          internal_socket_(internal_socket) {
+      buffer_.Copy(reinterpret_cast<const uint8_t*>(data), len);
+      nr_transport_addr_copy(&to_, addr);
+    }
+
+    TestNrSocket* socket_;
+    MediaPacket buffer_;
+    int flags_;
+    nr_transport_addr to_;
+    RefPtr<NrSocketBase> internal_socket_;
+  };
+
+  bool is_port_mapping_stale(const PortMapping& port_mapping) const;
+  bool allow_ingress(const nr_transport_addr& to, const nr_transport_addr& from,
+                     PortMapping** port_mapping_used) const;
+  void destroy_stale_port_mappings();
+
+  static void socket_readable_callback(void* real_sock_v, int how,
+                                       void* test_sock_v);
+  void on_socket_readable(NrSocketBase* external_or_internal_socket);
+  void fire_readable_callback();
+
+  static void port_mapping_tcp_passthrough_callback(void* ext_sock_v, int how,
+                                                    void* test_sock_v);
+  void cancel_port_mapping_async_wait(int how);
+
+  static void port_mapping_writeable_callback(void* ext_sock_v, int how,
+                                              void* test_sock_v);
+  void write_to_port_mapping(NrSocketBase* external_socket);
+  bool is_tcp_connection_behind_nat() const;
+
+  PortMapping* get_port_mapping(const nr_transport_addr& remote_addr,
+                                TestNat::NatBehavior filter) const;
+  static bool port_mapping_matches(const PortMapping& port_mapping,
+                                   const nr_transport_addr& remote_addr,
+                                   TestNat::NatBehavior filter);
+  PortMapping* create_port_mapping(
+      const nr_transport_addr& remote_addr,
+      const RefPtr<NrSocketBase>& external_socket) const;
+  RefPtr<NrSocketBase> create_external_socket(
+      const nr_transport_addr& remote_addr) const;
+
+  static void process_delayed_cb(NR_SOCKET s, int how, void* cb_arg);
+
+  bool maybe_send_fake_response(const void* msg, size_t len,
+                                const nr_transport_addr* to);
+  Maybe<nsTArray<nsCString>> maybe_get_redirect_targets(
+      const nr_transport_addr* to) const;
+
+  RefPtr<NrSocketBase> readable_socket_;
+  // The socket for the "internal" address; used to talk to stuff behind the
+  // same nat.
+  RefPtr<NrSocketBase> internal_socket_;
+  RefPtr<TestNat> nat_;
+  bool tls_;
+  // Since our comparison logic is different depending on what kind of NAT
+  // we simulate, and the STL does not make it very easy to switch out the
+  // comparison function at runtime, and these lists are going to be very
+  // small anyway, we just brute-force it.
+  std::list<RefPtr<PortMapping>> port_mappings_;
+
+  void* timer_handle_;
+
+  // Just used for fake stun responses right now. Not _necessarily_ just UDP
+  // stuff, UdpPacket just has what we need to make this work for UDP.
+  std::list<UdpPacket> read_buffer_;
+  std::unique_ptr<nr_transport_addr> connect_fake_stun_address_;
+};
+
+}  // namespace mozilla
+
+#endif  // test_nr_socket__
diff --git a/dom/media/webrtc/transport/third_party/moz.build b/dom/media/webrtc/transport/third_party/moz.build
new file mode 100644
index 0000000000..852ce4ac4b
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/moz.build
@@ -0,0 +1,39 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+include('/build/gyp.mozbuild')
+
+GYP_DIRS += [
+    'nICEr',
+    'nrappkit',
+]
+
+# Set gyp vars that webrtc needs when building under various analysis tools.
+# Primarily this prevents webrtc from setting NVALGRIND and breaking builds.
+gyp_vars_copy = gyp_vars.copy()
+if CONFIG['MOZ_VALGRIND']:
+    gyp_vars_copy.update(build_for_tool="memcheck")
+elif CONFIG['MOZ_ASAN']:
+    gyp_vars_copy.update(build_for_tool="asan")
+elif CONFIG['MOZ_TSAN']:
+    gyp_vars_copy.update(build_for_tool="tsan")
+
+# These files cannot be built in unified mode because of name clashes on RCSSTRING
+
+GYP_DIRS['nICEr'].input = 'nICEr/nicer.gyp'
+GYP_DIRS['nICEr'].variables = gyp_vars_copy
+GYP_DIRS['nICEr'].sandbox_vars['FINAL_LIBRARY'] = 'xul'
+GYP_DIRS['nrappkit'].input = 'nrappkit/nrappkit.gyp'
+GYP_DIRS['nrappkit'].variables = gyp_vars_copy
+GYP_DIRS['nrappkit'].sandbox_vars['FINAL_LIBRARY'] = 'xul'
+
+
+if CONFIG["CC_TYPE"] == "gcc":
+    GYP_DIRS['nICEr'].sandbox_vars['CFLAGS'] = ['-Wno-stringop-overflow']
+    GYP_DIRS['nrappkit'].sandbox_vars['CFLAGS'] = ['-Wno-stringop-overflow']
+    if int(CONFIG["CC_VERSION"].split(".")[0]) >= 8:
+        GYP_DIRS['nICEr'].sandbox_vars['CFLAGS'] += ['-Wno-stringop-truncation']
+        GYP_DIRS['nrappkit'].sandbox_vars['CFLAGS'] += ['-Wno-stringop-truncation']
diff --git a/dom/media/webrtc/transport/third_party/nICEr/COPYRIGHT b/dom/media/webrtc/transport/third_party/nICEr/COPYRIGHT
new file mode 100644
index 0000000000..2005fbd104
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/COPYRIGHT
@@ -0,0 +1,36 @@
+Portions of this software are subject to the following copyrights:
+
+   Copyright (C) 2007, Adobe Systems Inc.
+   Copyright (C) 2007-2008, Network Resonance, Inc.
+
+Each source file bears an individual copyright notice.
+
+The following license applies to this distribution as a whole.
+
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
diff --git a/dom/media/webrtc/transport/third_party/nICEr/README b/dom/media/webrtc/transport/third_party/nICEr/README
new file mode 100644
index 0000000000..390203dec5
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/README
@@ -0,0 +1,74 @@
+                             nICEr 1.0
+
+PREREQUISITES:
+-------------
+You must first obtain and build the following packages:
+
+* nrappkit
+  - http://nrappkit.sourceforge.net/
+
+* OpenSSL-0.9.8g
+  - http://www.openssl.org/source/openssl-0.9.8g.tar.gz
+
+
+For best results, the "ice-<version>" directory should be at
+the same level as the "nrappkit" and "openssl-0.9.8g"
+directories.  I.e., the directory structure should look like:
+
+   nrappkit/
+   ice-<version>/
+   openssl/
+          include/
+          lib/VC/
+
+
+BUILDING ON UNIX:
+----------------
+Once the prerequisite packages are built, 'cd' to the
+relevant build directory, one of:
+
+   ice-<version>/make/darwin
+   ice-<version>/make/linux-fedora
+   ice-<version>/make/ubuntu
+
+and simply do a "make".
+
+
+BUILDING ON WINDOWS:
+-------------------
+The Visual C++ project files are configured to expect the
+directory structure described above.
+ 
+Note: Binary Windows builds of OpenSSL can be found at:
+   http://www.slproweb.com/products/Win32OpenSSL.html
+
+Once the prerequisite packages are built, open the VC++ 9.0
+solution file: ICE/make/win32/ice.sln and build the solution.
+Note: Since the VC++ project/solution files are version 9.0,
+Visual Studio 2008 is required.
+
+
+STATUS:
+------
+The ICE code has been tested on the following platforms:
+-- Fedora Core 4 (Intel 32-bit)
+-- Fedora Core 6 (Intel 32-bit)
+-- Ubuntu 6.10 
+-- MacOSX 10.4.9
+-- Windows Vista (Home Premium)
+-- Windows XP Pro
+-- Windows 2000 SP4
+
+
+KNOWN ISSUES:
+------------
+-- TURN SET-ACTIVE-DESTINATION mode not yet supported.
+
+-- Problems may exist with the TURN client implementation; the TURN code
+   has received minimal testing due to the unavailability of a real
+   TURN server to test against.
+
+-- The ICE-Lite implementation is not complete.
+
+-- The new "impatient" timeout has not yet been thoroughly tested.
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/moz.yaml b/dom/media/webrtc/transport/third_party/nICEr/moz.yaml
new file mode 100644
index 0000000000..873769f9c2
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/moz.yaml
@@ -0,0 +1,117 @@
+schema: 1
+
+bugzilla:
+  product: Core
+  component: "WebRTC: Networking"
+
+origin:
+  name: nICEr
+  description: FIXME
+
+  url: https://github.com/resiprocate/nICEr/
+
+  release: b14598f34d12373069693a1f0535fe354a3f1fd5
+  revision: b14598f34d12373069693a1f0535fe354a3f1fd5
+
+  license: BSD-3-Clause
+  license-file: COPYRIGHT
+
+vendoring:
+  url: https://github.com/resiprocate/nICEr/
+  source-hosting: github
+  tracking: commit
+
+  exclude:
+    - api.txt
+    - src/
+
+  keep:
+    - nicer.gyp
+    # Crypto
+    - src/crypto/nr_crypto.c
+    - src/crypto/nr_crypto.h
+    #./src/crypto/nr_crypto_openssl.c
+    #./src/crypto/nr_crypto_openssl.h
+
+    # ICE
+    - src/ice/ice_candidate.c
+    - src/ice/ice_candidate.h
+    - src/ice/ice_candidate_pair.c
+    - src/ice/ice_candidate_pair.h
+    - src/ice/ice_codeword.h
+    - src/ice/ice_component.c
+    - src/ice/ice_component.h
+    - src/ice/ice_ctx.c
+    - src/ice/ice_ctx.h
+    - src/ice/ice_handler.h
+    - src/ice/ice_media_stream.c
+    - src/ice/ice_media_stream.h
+    - src/ice/ice_parser.c
+    - src/ice/ice_peer_ctx.c
+    - src/ice/ice_peer_ctx.h
+    - src/ice/ice_reg.h
+    - src/ice/ice_socket.c
+    - src/ice/ice_socket.h
+
+    # Net
+    - src/net/nr_socket.c
+    - src/net/nr_socket.h
+    #./src/net/nr_socket_local.c
+    - src/net/nr_socket_local.h
+    - src/net/transport_addr.c
+    - src/net/transport_addr.h
+    - src/net/transport_addr_reg.c
+    - src/net/transport_addr_reg.h
+    - src/net/nr_interface_prioritizer.c
+    - src/net/nr_interface_prioritizer.h
+    - src/net/nr_resolver.h
+    - src/net/nr_resolver.c
+    - src/net/nr_socket_multi_tcp.h
+    - src/net/nr_socket_multi_tcp.c
+    - src/net/nr_socket_wrapper.h
+    - src/net/nr_socket_wrapper.c
+    - src/net/local_addr.h
+    - src/net/local_addr.c
+
+    # STUN
+    - src/stun/addrs.c
+    - src/stun/addrs.h
+    - src/stun/addrs-bsd.c
+    - src/stun/addrs-bsd.h
+    - src/stun/addrs-netlink.c
+    - src/stun/addrs-netlink.h
+    - src/stun/addrs-win32.c
+    - src/stun/addrs-win32.h
+    - src/stun/nr_socket_buffered_stun.h
+    - src/stun/nr_socket_buffered_stun.c
+    - src/stun/nr_socket_turn.c
+    - src/stun/nr_socket_turn.h
+    - src/stun/stun.h
+    - src/stun/stun_build.c
+    - src/stun/stun_build.h
+    - src/stun/stun_client_ctx.c
+    - src/stun/stun_client_ctx.h
+    - src/stun/stun_codec.c
+    - src/stun/stun_codec.h
+    - src/stun/stun_hint.c
+    - src/stun/stun_hint.h
+    - src/stun/stun_msg.c
+    - src/stun/stun_msg.h
+    - src/stun/stun_proc.c
+    - src/stun/stun_proc.h
+    - src/stun/stun_reg.h
+    - src/stun/stun_server_ctx.c
+    - src/stun/stun_server_ctx.h
+    - src/stun/stun_util.c
+    - src/stun/stun_util.h
+    - src/stun/turn_client_ctx.c
+    - src/stun/turn_client_ctx.h
+
+    # Util
+    - src/util/cb_args.c
+    - src/util/cb_args.h
+    - src/util/ice_util.c
+    - src/util/ice_util.h
+
+  patches:
+    - non-unified-build.patch
diff --git a/dom/media/webrtc/transport/third_party/nICEr/nicer.gyp b/dom/media/webrtc/transport/third_party/nICEr/nicer.gyp
new file mode 100644
index 0000000000..488b9b229c
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/nicer.gyp
@@ -0,0 +1,276 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+#
+# nrappkit.gyp
+#
+#
+{
+  'variables' : {
+    'have_ethtool_cmd_speed_hi%': 1
+  },
+  'targets' : [
+      {
+          'target_name' : 'nicer',
+          'type' : 'static_library',
+
+          'include_dirs' : [
+              ## EXTERNAL
+              # nrappkit
+              '../nrappkit/src/event',
+              '../nrappkit/src/log',
+              '../nrappkit/src/plugin',
+              '../nrappkit/src/registry',
+              '../nrappkit/src/share',
+              '../nrappkit/src/stats',
+              '../nrappkit/src/util',
+              '../nrappkit/src/util/libekr',
+              '../nrappkit/src/port/generic/include',
+
+              # INTERNAL
+              "./src/crypto",
+              "./src/ice",
+              "./src/net",
+              "./src/stun",
+              "./src/util",
+          ],
+
+          'sources' : [
+                # Crypto
+                "./src/crypto/nr_crypto.c",
+                "./src/crypto/nr_crypto.h",
+                #"./src/crypto/nr_crypto_openssl.c",
+                #"./src/crypto/nr_crypto_openssl.h",
+
+                # ICE
+                "./src/ice/ice_candidate.c",
+                "./src/ice/ice_candidate.h",
+                "./src/ice/ice_candidate_pair.c",
+                "./src/ice/ice_candidate_pair.h",
+                "./src/ice/ice_codeword.h",
+                "./src/ice/ice_component.c",
+                "./src/ice/ice_component.h",
+                "./src/ice/ice_ctx.c",
+                "./src/ice/ice_ctx.h",
+                "./src/ice/ice_handler.h",
+                "./src/ice/ice_media_stream.c",
+                "./src/ice/ice_media_stream.h",
+                "./src/ice/ice_parser.c",
+                "./src/ice/ice_peer_ctx.c",
+                "./src/ice/ice_peer_ctx.h",
+                "./src/ice/ice_reg.h",
+                "./src/ice/ice_socket.c",
+                "./src/ice/ice_socket.h",
+
+                # Net
+                "./src/net/nr_resolver.c",
+                "./src/net/nr_resolver.h",
+                "./src/net/nr_socket_wrapper.c",
+                "./src/net/nr_socket_wrapper.h",
+                "./src/net/nr_socket.c",
+                "./src/net/nr_socket.h",
+                #"./src/net/nr_socket_local.c",
+                "./src/net/nr_socket_local.h",
+                "./src/net/nr_socket_multi_tcp.c",
+                "./src/net/nr_socket_multi_tcp.h",
+                "./src/net/transport_addr.c",
+                "./src/net/transport_addr.h",
+                "./src/net/transport_addr_reg.c",
+                "./src/net/transport_addr_reg.h",
+                "./src/net/local_addr.c",
+                "./src/net/local_addr.h",
+                "./src/net/nr_interface_prioritizer.c",
+                "./src/net/nr_interface_prioritizer.h",
+
+                # STUN
+                "./src/stun/addrs.c",
+                "./src/stun/addrs.h",
+                "./src/stun/addrs-bsd.c",
+                "./src/stun/addrs-bsd.h",
+                "./src/stun/addrs-netlink.c",
+                "./src/stun/addrs-netlink.h",
+                "./src/stun/addrs-win32.c",
+                "./src/stun/addrs-win32.h",
+                "./src/stun/nr_socket_turn.c",
+                "./src/stun/nr_socket_turn.h",
+                "./src/stun/nr_socket_buffered_stun.c",
+                "./src/stun/nr_socket_buffered_stun.h",
+                "./src/stun/stun.h",
+                "./src/stun/stun_build.c",
+                "./src/stun/stun_build.h",
+                "./src/stun/stun_client_ctx.c",
+                "./src/stun/stun_client_ctx.h",
+                "./src/stun/stun_codec.c",
+                "./src/stun/stun_codec.h",
+                "./src/stun/stun_hint.c",
+                "./src/stun/stun_hint.h",
+                "./src/stun/stun_msg.c",
+                "./src/stun/stun_msg.h",
+                "./src/stun/stun_proc.c",
+                "./src/stun/stun_proc.h",
+                "./src/stun/stun_reg.h",
+                "./src/stun/stun_server_ctx.c",
+                "./src/stun/stun_server_ctx.h",
+                "./src/stun/stun_util.c",
+                "./src/stun/stun_util.h",
+                "./src/stun/turn_client_ctx.c",
+                "./src/stun/turn_client_ctx.h",
+
+                # Util
+                "./src/util/cb_args.c",
+                "./src/util/cb_args.h",
+                "./src/util/ice_util.c",
+                "./src/util/ice_util.h",
+
+
+          ],
+
+          'defines' : [
+              'SANITY_CHECKS',
+              'USE_TURN',
+              'USE_ICE',
+              'USE_RFC_3489_BACKWARDS_COMPATIBLE',
+              'USE_STUND_0_96',
+              'USE_STUN_PEDANTIC',
+              'USE_TURN',
+              'NR_SOCKET_IS_VOID_PTR',
+              'restrict=',
+              'R_PLATFORM_INT_TYPES=<stdint.h>',
+              'R_DEFINED_INT2=int16_t',
+              'R_DEFINED_UINT2=uint16_t',
+              'R_DEFINED_INT4=int32_t',
+              'R_DEFINED_UINT4=uint32_t',
+              'R_DEFINED_INT8=int64_t',
+              'R_DEFINED_UINT8=uint64_t',
+          ],
+
+          'conditions' : [
+              ## Mac and BSDs
+              [ 'OS == "mac" or OS == "ios"', {
+                'defines' : [
+                    'DARWIN',
+                ],
+              }],
+              [ 'os_bsd == 1', {
+                'defines' : [
+                    'BSD',
+                ],
+              }],
+              [ 'OS == "mac" or OS == "ios" or os_bsd == 1', {
+                'cflags_mozilla': [
+                    '-Wall',
+                    '-Wno-parentheses',
+                    '-Wno-strict-prototypes',
+                    '-Wmissing-prototypes',
+                    '-Wno-format',
+                    '-Wno-format-security',
+                 ],
+                 'defines' : [
+                     'HAVE_LIBM=1',
+                     'HAVE_STRDUP=1',
+                     'HAVE_STRLCPY=1',
+                     'HAVE_SYS_TIME_H=1',
+                     'HAVE_VFPRINTF=1',
+                     'NEW_STDIO'
+                     'RETSIGTYPE=void',
+                     'TIME_WITH_SYS_TIME_H=1',
+                     '__UNUSED__=__attribute__((unused))',
+                 ],
+
+                 'include_dirs': [
+                     '../nrappkit/src/port/darwin/include'
+                 ],
+
+                 'sources': [
+                 ],
+              }],
+
+              ## Win
+              [ 'OS == "win"', {
+                'defines' : [
+                    'WIN32',
+                    '_WINSOCK_DEPRECATED_NO_WARNINGS',
+                    'USE_ICE',
+                    'USE_TURN',
+                    'USE_RFC_3489_BACKWARDS_COMPATIBLE',
+                    'USE_STUND_0_96',
+                    'USE_STUN_PEDANTIC',
+                    '_CRT_SECURE_NO_WARNINGS',
+                    '__UNUSED__=',
+                    'HAVE_STRDUP',
+                    'NO_REG_RPC'
+                ],
+
+                 'include_dirs': [
+                     '../nrappkit/src/port/win32/include'
+                 ],
+              }],
+
+              # Windows, clang-cl build
+              [ 'clang_cl == 1', {
+                'cflags_mozilla': [
+                    '-Xclang',
+                    '-Wall',
+                    '-Xclang',
+                    '-Wno-parentheses',
+                    '-Wno-pointer-sign',
+                    '-Wno-strict-prototypes',
+                    '-Xclang',
+                    '-Wno-unused-function',
+                    '-Wmissing-prototypes',
+                    '-Wno-format',
+                    '-Wno-format-security',
+                 ],
+              }],
+
+              ## Linux/Android
+              [ '(OS == "linux") or (OS == "android")', {
+                'cflags_mozilla': [
+                    '-Wall',
+                    '-Wno-parentheses',
+                    '-Wno-strict-prototypes',
+                    '-Wmissing-prototypes',
+                    '-Wno-format',
+                    '-Wno-format-security',
+                 ],
+                 'defines' : [
+                     'LINUX',
+                     'HAVE_LIBM=1',
+                     'HAVE_STRDUP=1',
+                     'HAVE_STRLCPY=1',
+                     'HAVE_SYS_TIME_H=1',
+                     'HAVE_VFPRINTF=1',
+                     'NEW_STDIO'
+                     'RETSIGTYPE=void',
+                     'TIME_WITH_SYS_TIME_H=1',
+                     '__UNUSED__=__attribute__((unused))',
+                 ],
+
+                 'include_dirs': [
+                     '../nrappkit/src/port/linux/include'
+                 ],
+
+                 'sources': [
+                 ],
+             }],
+             ['have_ethtool_cmd_speed_hi==0', {
+               'defines': [
+                  "DONT_HAVE_ETHTOOL_SPEED_HI",
+               ]
+             }],
+        # libFuzzer instrumentation is not compatible with TSan.
+        # See also the comment in build/moz.configure/toolchain.configure.
+        ['(libfuzzer == 1) and (tsan == 0) and (libfuzzer_fuzzer_no_link_flag == 1)', {
+          'cflags_mozilla': [
+            '-fsanitize=fuzzer-no-link'
+         ],
+        }],
+        ['(libfuzzer == 1) and (tsan == 0) and (libfuzzer_fuzzer_no_link_flag == 0)', {
+          'cflags_mozilla': [
+            '-fsanitize-coverage=trace-pc-guard,trace-cmp'
+         ],
+        }],
+          ],
+      }]
+}
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/non-unified-build.patch b/dom/media/webrtc/transport/third_party/nICEr/non-unified-build.patch
new file mode 100644
index 0000000000..41f206e67a
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/non-unified-build.patch
@@ -0,0 +1,40 @@
+diff --git a/src/net/nr_socket_wrapper.c b/src/net/nr_socket_wrapper.c
+index 0c9ec5674407d..4ad59527c12ec 100644
+--- a/src/net/nr_socket_wrapper.c
++++ b/src/net/nr_socket_wrapper.c
+@@ -36,6 +36,8 @@ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ #include <nr_api.h>
+ #include "nr_socket_wrapper.h"
+ 
++#include <assert.h>
++
+ int nr_socket_wrapper_factory_create_int(void *obj, nr_socket_wrapper_factory_vtbl *vtbl,
+                                          nr_socket_wrapper_factory **wrapperp)
+ {
+diff --git a/src/net/transport_addr.h b/nICEr/src/net/transport_addr.h
+index c75c70a1d94fa..1783d2e4506a8 100644
+--- a/src/net/transport_addr.h
++++ b/src/net/transport_addr.h
+@@ -38,7 +38,22 @@ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ #include <stdbool.h>
+ #include <sys/types.h>
+ #ifdef WIN32
++
++// FIXME: This is dangerous, but exactly the pattern used in
++// nrappkit/src/port/win32/include/csi_platform.h
++// Not good because INT8 are typedefed to different values in
++// <winsock2.h> and <r_types.h>.
++// {
++
++#define UINT8 UBLAH_IGNORE_ME_PLEASE
++#define INT8 BLAH_IGNORE_ME_PLEASE
+ #include <winsock2.h>
++#undef UINT8
++#undef INT8
++#include <r_types.h>
++
++// }
++
+ #include <ws2tcpip.h>
+ #else
+ #include <sys/socket.h>
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/crypto/nr_crypto.c b/dom/media/webrtc/transport/third_party/nICEr/src/crypto/nr_crypto.c
new file mode 100644
index 0000000000..4c430b0e8b
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/crypto/nr_crypto.c
@@ -0,0 +1,67 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <nr_api.h>
+#include "nr_crypto.h"
+
+static int nr_ice_crypto_dummy_random_bytes(UCHAR *buf, size_t len)
+  {
+    fprintf(stderr,"Need to define crypto API implementation\n");
+
+    exit(1);
+  }
+
+static int nr_ice_crypto_dummy_hmac_sha1(UCHAR *key, size_t key_l, UCHAR *buf, size_t buf_l, UCHAR digest[20])
+  {
+    fprintf(stderr,"Need to define crypto API implementation\n");
+
+    exit(1);
+  }
+
+static int nr_ice_crypto_dummy_md5(UCHAR *buf, size_t buf_l, UCHAR digest[16])
+  {
+    fprintf(stderr,"Need to define crypto API implementation\n");
+
+    exit(1);
+  }
+
+static nr_ice_crypto_vtbl nr_ice_crypto_dummy_vtbl= {
+  nr_ice_crypto_dummy_random_bytes,
+  nr_ice_crypto_dummy_hmac_sha1,
+  nr_ice_crypto_dummy_md5
+};
+
+
+
+nr_ice_crypto_vtbl *nr_crypto_vtbl=&nr_ice_crypto_dummy_vtbl;
+
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/crypto/nr_crypto.h b/dom/media/webrtc/transport/third_party/nICEr/src/crypto/nr_crypto.h
new file mode 100644
index 0000000000..5fbbd8f097
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/crypto/nr_crypto.h
@@ -0,0 +1,52 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _nr_crypto_h
+#define _nr_crypto_h
+
+
+typedef struct nr_ice_crypto_vtbl_ {
+  int (*random_bytes)(UCHAR *buf, size_t len);
+  int (*hmac_sha1)(UCHAR *key, size_t key_l, UCHAR *buf, size_t buf_l, UCHAR digest[20]);
+  int (*md5)(UCHAR *buf, size_t buf_l, UCHAR digest[16]);
+} nr_ice_crypto_vtbl;
+
+extern nr_ice_crypto_vtbl *nr_crypto_vtbl;
+
+#define nr_crypto_random_bytes(a,b) nr_crypto_vtbl->random_bytes(a,b)
+#define nr_crypto_hmac_sha1(a,b,c,d,e) nr_crypto_vtbl->hmac_sha1(a,b,c,d,e)
+#define nr_crypto_md5(a,b,c) nr_crypto_vtbl->md5(a,b,c)
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_candidate.c b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_candidate.c
new file mode 100644
index 0000000000..b7cf2c9a99
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_candidate.c
@@ -0,0 +1,1052 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <csi_platform.h>
+#include <assert.h>
+#include <stdio.h>
+#include <string.h>
+#include <sys/queue.h>
+#include <sys/types.h>
+#ifdef WIN32
+#include <winsock2.h>
+#else
+#include <sys/socket.h>
+#include <netinet/in.h>
+#include <arpa/inet.h>
+#endif
+#include "nr_api.h"
+#include "registry.h"
+#include "nr_socket.h"
+#include "async_timer.h"
+
+#include "stun_client_ctx.h"
+#include "stun_server_ctx.h"
+#include "turn_client_ctx.h"
+#include "ice_ctx.h"
+#include "ice_candidate.h"
+#include "ice_codeword.h"
+#include "ice_reg.h"
+#include "ice_util.h"
+#include "nr_socket_turn.h"
+#include "nr_socket.h"
+#include "nr_socket_multi_tcp.h"
+
+static int next_automatic_preference = 127;
+
+static int nr_ice_candidate_initialize2(nr_ice_candidate *cand);
+static int nr_ice_get_foundation(nr_ice_ctx *ctx,nr_ice_candidate *cand);
+static int nr_ice_srvrflx_start_stun(nr_ice_candidate *cand);
+static void nr_ice_srvrflx_stun_finished_cb(NR_SOCKET sock, int how, void *cb_arg);
+#ifdef USE_TURN
+static int nr_ice_start_relay_turn(nr_ice_candidate *cand);
+static void nr_ice_turn_allocated_cb(NR_SOCKET sock, int how, void *cb_arg);
+static int nr_ice_candidate_resolved_cb(void *cb_arg, nr_transport_addr *addr);
+#endif /* USE_TURN */
+
+void nr_ice_candidate_compute_codeword(nr_ice_candidate *cand)
+  {
+    char as_string[1024];
+
+    snprintf(as_string,
+             sizeof(as_string),
+             "%s(%s)",
+             cand->addr.as_string,
+             cand->label);
+
+    nr_ice_compute_codeword(as_string,strlen(as_string),cand->codeword);
+  }
+
+char *nr_ice_candidate_type_names[]={0,"host","srflx","prflx","relay",0};
+char *nr_ice_candidate_tcp_type_names[]={0,"active","passive","so",0};
+
+static const char *nr_ctype_name(nr_ice_candidate_type ctype) {
+  assert(ctype<CTYPE_MAX && ctype>0);
+  if (ctype <= 0 || ctype >= CTYPE_MAX) {
+    return "ERROR";
+  }
+  return nr_ice_candidate_type_names[ctype];
+}
+
+static const char *nr_tcp_type_name(nr_socket_tcp_type tcp_type) {
+  assert(tcp_type<TCP_TYPE_MAX && tcp_type>0);
+  if (tcp_type <= 0 || tcp_type >= TCP_TYPE_MAX) {
+    return "ERROR";
+  }
+  return nr_ice_candidate_tcp_type_names[tcp_type];
+}
+
+static int nr_ice_candidate_format_stun_label(char *label, size_t size, nr_ice_candidate *cand)
+  {
+  *label = 0;
+  snprintf(label, size, "%s(%s|%s)", nr_ctype_name(cand->type),
+           cand->base.as_string, cand->stun_server->addr.as_string);
+
+  return (0);
+  }
+
+int nr_ice_candidate_create(nr_ice_ctx *ctx,nr_ice_component *comp,nr_ice_socket *isock, nr_socket *osock, nr_ice_candidate_type ctype, nr_socket_tcp_type tcp_type, nr_ice_stun_server *stun_server, UCHAR component_id, nr_ice_candidate **candp)
+  {
+    assert(!(comp->stream->flags & NR_ICE_CTX_FLAGS_RELAY_ONLY) || ctype == RELAYED);
+    nr_ice_candidate *cand=0;
+    nr_ice_candidate *tmp=0;
+    int r,_status;
+    char label[512];
+
+    if(!(cand=RCALLOC(sizeof(nr_ice_candidate))))
+      ABORT(R_NO_MEMORY);
+    cand->state=NR_ICE_CAND_STATE_CREATED;
+    cand->ctx=ctx;
+    cand->isock=isock;
+    cand->osock=osock;
+    cand->type=ctype;
+    cand->tcp_type=tcp_type;
+    cand->stun_server=stun_server;
+    cand->component_id=component_id;
+    cand->component=comp;
+    cand->stream=comp->stream;
+
+    /* Extract the addr as the base */
+    if(r=nr_socket_getaddr(cand->isock->sock,&cand->base))
+      ABORT(r);
+
+    switch(ctype) {
+      case HOST:
+        snprintf(label, sizeof(label), "host(%s)", cand->base.as_string);
+        break;
+
+      case SERVER_REFLEXIVE:
+        if(r=nr_ice_candidate_format_stun_label(label, sizeof(label), cand))
+          ABORT(r);
+        break;
+
+      case RELAYED:
+        if(r=nr_ice_candidate_format_stun_label(label, sizeof(label), cand))
+          ABORT(r);
+        break;
+
+      case PEER_REFLEXIVE:
+        snprintf(label, sizeof(label), "prflx");
+        break;
+
+      default:
+        assert(0); /* Can't happen */
+        ABORT(R_BAD_ARGS);
+    }
+
+    if (tcp_type) {
+      const char* ttype = nr_tcp_type_name(tcp_type);
+      const int tlen = strlen(ttype)+1; /* plus space */
+      const size_t llen=strlen(label);
+      if (snprintf(label+llen, sizeof(label)-llen, " %s", ttype) != tlen) {
+        r_log(LOG_ICE,LOG_ERR,"ICE(%s): truncated tcp type added to buffer",
+          ctx->label);
+      }
+    }
+
+    if(!(cand->label=r_strdup(label)))
+      ABORT(R_NO_MEMORY);
+
+    if(r=nr_ice_get_foundation(ctx,cand))
+      ABORT(r);
+    if(r=nr_ice_candidate_compute_priority(cand))
+      ABORT(r);
+
+    TAILQ_FOREACH(tmp,&isock->candidates,entry_sock){
+      if(cand->priority==tmp->priority){
+        r_log(LOG_ICE,LOG_ERR,"ICE(%s): duplicate priority %u candidate %s and candidate %s",
+          ctx->label,cand->priority,cand->label,tmp->label);
+      }
+    }
+
+    if(ctype==RELAYED)
+      cand->u.relayed.turn_sock=osock;
+
+
+    /* Add the candidate to the isock list*/
+    TAILQ_INSERT_TAIL(&isock->candidates,cand,entry_sock);
+
+    nr_ice_candidate_compute_codeword(cand);
+
+    r_log(LOG_ICE,LOG_DEBUG,"ICE(%s)/CAND(%s): created candidate %s with type %s",
+      ctx->label,cand->codeword,cand->label,nr_ctype_name(ctype));
+
+    *candp=cand;
+
+    _status=0;
+  abort:
+    if (_status){
+      r_log(LOG_ICE,LOG_ERR,"ICE(%s): Failed to create candidate of type %s", ctx->label,nr_ctype_name(ctype));
+      nr_ice_candidate_destroy(&cand);
+    }
+    return(_status);
+  }
+
+
+/* Create a peer reflexive candidate */
+int nr_ice_peer_peer_rflx_candidate_create(nr_ice_ctx *ctx,char *label, nr_ice_component *comp,nr_transport_addr *addr, nr_ice_candidate **candp)
+  {
+    nr_ice_candidate *cand=0;
+    nr_ice_candidate_type ctype=PEER_REFLEXIVE;
+    int r,_status;
+
+    if(!(cand=RCALLOC(sizeof(nr_ice_candidate))))
+      ABORT(R_NO_MEMORY);
+    if(!(cand->label=r_strdup(label)))
+      ABORT(R_NO_MEMORY);
+
+    cand->state=NR_ICE_CAND_STATE_INITIALIZED;
+    cand->ctx=ctx;
+    cand->type=ctype;
+    cand->component_id=comp->component_id;
+    cand->component=comp;
+    cand->stream=comp->stream;
+
+
+    r_log(LOG_ICE,LOG_DEBUG,"ICE(%s)/CAND(%s): creating candidate with type %s",
+      ctx->label,label,nr_ctype_name(ctype));
+
+    if(r=nr_transport_addr_copy(&cand->base,addr))
+      ABORT(r);
+    if(r=nr_transport_addr_copy(&cand->addr,addr))
+      ABORT(r);
+    /* Bogus foundation */
+    if(!(cand->foundation=r_strdup(cand->addr.as_string)))
+      ABORT(R_NO_MEMORY);
+
+    nr_ice_candidate_compute_codeword(cand);
+
+    *candp=cand;
+
+    _status=0;
+  abort:
+    if (_status){
+      nr_ice_candidate_destroy(&cand);
+    }
+    return(_status);
+  }
+
+static void nr_ice_candidate_mark_done(nr_ice_candidate *cand, int state)
+  {
+    if (!cand) {
+      assert(0);
+      return;
+    }
+
+    /* If this is a relay candidate, there's likely to be a srflx that is
+     * piggybacking on it. Make sure it is marked done too. */
+    if ((cand->type == RELAYED) && cand->u.relayed.srvflx_candidate) {
+      nr_ice_candidate *srflx=cand->u.relayed.srvflx_candidate;
+      if (state == NR_ICE_CAND_STATE_INITIALIZED &&
+          nr_turn_client_get_mapped_address(cand->u.relayed.turn,
+                                            &srflx->addr)) {
+        r_log(LOG_ICE, LOG_WARNING, "ICE(%s)/CAND(%s): Failed to get mapped address from TURN allocate response, srflx failed.", cand->ctx->label, cand->label);
+        nr_ice_candidate_mark_done(srflx, NR_ICE_CAND_STATE_FAILED);
+      } else {
+        nr_ice_candidate_mark_done(srflx, state);
+      }
+    }
+
+    NR_async_cb done_cb=cand->done_cb;
+    cand->done_cb=0;
+    cand->state=state;
+    /* This might destroy cand! */
+    if (done_cb) {
+      done_cb(0,0,cand->cb_arg);
+    }
+  }
+
+int nr_ice_candidate_destroy(nr_ice_candidate **candp)
+  {
+    nr_ice_candidate *cand=0;
+
+    if(!candp || !*candp)
+      return(0);
+
+    cand=*candp;
+
+    nr_ice_candidate_stop_gathering(cand);
+
+    switch(cand->type){
+      case HOST:
+        break;
+#ifdef USE_TURN
+      case RELAYED:
+        // record stats back to the ice ctx on destruction
+        if (cand->u.relayed.turn) {
+          nr_accumulate_count(&(cand->ctx->stats.turn_401s), cand->u.relayed.turn->cnt_401s);
+          nr_accumulate_count(&(cand->ctx->stats.turn_403s), cand->u.relayed.turn->cnt_403s);
+          nr_accumulate_count(&(cand->ctx->stats.turn_438s), cand->u.relayed.turn->cnt_438s);
+
+          nr_turn_stun_ctx* stun_ctx;
+          stun_ctx = STAILQ_FIRST(&cand->u.relayed.turn->stun_ctxs);
+          while (stun_ctx) {
+            nr_accumulate_count(&(cand->ctx->stats.stun_retransmits), stun_ctx->stun->retransmit_ct);
+
+            stun_ctx = STAILQ_NEXT(stun_ctx, entry);
+          }
+        }
+        if (cand->u.relayed.turn_handle)
+          nr_ice_socket_deregister(cand->isock, cand->u.relayed.turn_handle);
+        if (cand->u.relayed.srvflx_candidate)
+          cand->u.relayed.srvflx_candidate->u.srvrflx.relay_candidate=0;
+        nr_turn_client_ctx_destroy(&cand->u.relayed.turn);
+        nr_socket_destroy(&cand->u.relayed.turn_sock);
+        break;
+#endif /* USE_TURN */
+      case SERVER_REFLEXIVE:
+        if (cand->u.srvrflx.stun_handle)
+          nr_ice_socket_deregister(cand->isock, cand->u.srvrflx.stun_handle);
+        if (cand->u.srvrflx.relay_candidate)
+          cand->u.srvrflx.relay_candidate->u.relayed.srvflx_candidate=0;
+        nr_stun_client_ctx_destroy(&cand->u.srvrflx.stun);
+        break;
+      default:
+        break;
+    }
+
+    RFREE(cand->mdns_addr);
+    RFREE(cand->foundation);
+    RFREE(cand->label);
+    RFREE(cand);
+
+    return(0);
+  }
+
+void nr_ice_candidate_stop_gathering(nr_ice_candidate *cand)
+  {
+    if (cand->state == NR_ICE_CAND_STATE_INITIALIZING) {
+      /* Make sure the ICE ctx isn't still waiting around for this candidate
+       * to init. */
+      nr_ice_candidate_mark_done(cand, NR_ICE_CAND_STATE_FAILED);
+    }
+
+    NR_async_timer_cancel(cand->delay_timer);
+    cand->delay_timer=0;
+    NR_async_timer_cancel(cand->ready_cb_timer);
+    cand->ready_cb_timer=0;
+
+    if(cand->resolver_handle){
+      nr_resolver_cancel(cand->ctx->resolver,cand->resolver_handle);
+      cand->resolver_handle=0;
+    }
+  }
+
+/* This algorithm is not super-fast, but I don't think we need a hash
+   table just yet and it produces a small foundation string */
+static int nr_ice_get_foundation(nr_ice_ctx *ctx,nr_ice_candidate *cand)
+  {
+    nr_ice_foundation *foundation;
+    int i=0;
+    char fnd[20];
+    int _status;
+
+    foundation=STAILQ_FIRST(&ctx->foundations);
+    while(foundation){
+      if(nr_transport_addr_cmp(&cand->base,&foundation->addr,NR_TRANSPORT_ADDR_CMP_MODE_ADDR))
+        goto next;
+      // cast necessary because there is no guarantee that enum is signed.
+      // foundation->type should probably match nr_ice_candidate_type
+      if((int)cand->type != foundation->type)
+        goto next;
+      if(cand->type == SERVER_REFLEXIVE || cand->type == RELAYED) {
+        if(nr_transport_addr_cmp(&cand->stun_server->addr, &foundation->stun_server_addr, NR_TRANSPORT_ADDR_CMP_MODE_ADDR))
+          goto next;
+      }
+
+      snprintf(fnd,sizeof(fnd),"%d",i);
+      if(!(cand->foundation=r_strdup(fnd)))
+        ABORT(R_NO_MEMORY);
+      return(0);
+
+    next:
+      foundation=STAILQ_NEXT(foundation,entry);
+      i++;
+    }
+
+    if(!(foundation=RCALLOC(sizeof(nr_ice_foundation))))
+      ABORT(R_NO_MEMORY);
+    nr_transport_addr_copy(&foundation->addr,&cand->base);
+    foundation->type=cand->type;
+    if(cand->type == SERVER_REFLEXIVE || cand->type == RELAYED) {
+      nr_transport_addr_copy(&foundation->stun_server_addr, &cand->stun_server->addr);
+    }
+    STAILQ_INSERT_TAIL(&ctx->foundations,foundation,entry);
+
+    snprintf(fnd,sizeof(fnd),"%d",i);
+    if(!(cand->foundation=r_strdup(fnd)))
+      ABORT(R_NO_MEMORY);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_ice_candidate_compute_priority(nr_ice_candidate *cand)
+  {
+    UCHAR type_preference;
+    UCHAR interface_preference;
+    UCHAR stun_priority;
+    UCHAR direction_priority=0;
+    int r,_status;
+
+    if (cand->base.protocol != IPPROTO_UDP && cand->base.protocol != IPPROTO_TCP){
+      r_log(LOG_ICE,LOG_ERR,"Unknown protocol type %u",
+            (unsigned int)cand->base.protocol);
+      ABORT(R_INTERNAL);
+    }
+
+    switch(cand->type){
+      case HOST:
+        if(cand->base.protocol == IPPROTO_UDP) {
+          if(r=NR_reg_get_uchar(NR_ICE_REG_PREF_TYPE_HOST,&type_preference))
+            ABORT(r);
+        } else if(cand->base.protocol == IPPROTO_TCP) {
+          if(r=NR_reg_get_uchar(NR_ICE_REG_PREF_TYPE_HOST_TCP,&type_preference))
+            ABORT(r);
+        }
+        stun_priority=0;
+        break;
+      case RELAYED:
+        if(cand->base.protocol == IPPROTO_UDP) {
+          if(r=NR_reg_get_uchar(NR_ICE_REG_PREF_TYPE_RELAYED,&type_preference))
+            ABORT(r);
+        } else if(cand->base.protocol == IPPROTO_TCP) {
+          if(r=NR_reg_get_uchar(NR_ICE_REG_PREF_TYPE_RELAYED_TCP,&type_preference))
+            ABORT(r);
+        }
+        stun_priority=31-cand->stun_server->id;
+        break;
+      case SERVER_REFLEXIVE:
+        if(cand->base.protocol == IPPROTO_UDP) {
+          if(r=NR_reg_get_uchar(NR_ICE_REG_PREF_TYPE_SRV_RFLX,&type_preference))
+            ABORT(r);
+        } else if(cand->base.protocol == IPPROTO_TCP) {
+          if(r=NR_reg_get_uchar(NR_ICE_REG_PREF_TYPE_SRV_RFLX_TCP,&type_preference))
+            ABORT(r);
+        }
+        stun_priority=31-cand->stun_server->id;
+        break;
+      case PEER_REFLEXIVE:
+        if(cand->base.protocol == IPPROTO_UDP) {
+          if(r=NR_reg_get_uchar(NR_ICE_REG_PREF_TYPE_PEER_RFLX,&type_preference))
+            ABORT(r);
+        } else if(cand->base.protocol == IPPROTO_TCP) {
+          if(r=NR_reg_get_uchar(NR_ICE_REG_PREF_TYPE_PEER_RFLX_TCP,&type_preference))
+            ABORT(r);
+        }
+        stun_priority=0;
+        break;
+      default:
+        ABORT(R_INTERNAL);
+    }
+
+    if(cand->base.protocol == IPPROTO_TCP){
+      switch (cand->tcp_type) {
+        case TCP_TYPE_ACTIVE:
+          if (cand->type == HOST)
+            direction_priority=6;
+          else
+            direction_priority=4;
+          break;
+        case  TCP_TYPE_PASSIVE:
+          if (cand->type == HOST)
+            direction_priority=4;
+          else
+            direction_priority=2;
+          break;
+        case  TCP_TYPE_SO:
+          if (cand->type == HOST)
+            direction_priority=2;
+          else
+            direction_priority=6;
+          break;
+        case  TCP_TYPE_NONE:
+          break;
+        case TCP_TYPE_MAX:
+        default:
+          assert(0);
+          ABORT(R_INTERNAL);
+      }
+    }
+
+    if(type_preference > 126)
+      r_log(LOG_ICE,LOG_ERR,"Illegal type preference %d",type_preference);
+
+    if(!cand->ctx->interface_prioritizer) {
+      /* Prioritizer is not set, read from registry */
+      if(r=NR_reg_get2_uchar(NR_ICE_REG_PREF_INTERFACE_PRFX,cand->base.ifname,
+        &interface_preference)) {
+        if (r==R_NOT_FOUND) {
+          if (next_automatic_preference == 1) {
+            r_log(LOG_ICE,LOG_ERR,"Out of preference values. Can't assign one for interface %s",cand->base.ifname);
+            ABORT(R_NOT_FOUND);
+          }
+          r_log(LOG_ICE,LOG_DEBUG,"Automatically assigning preference for interface %s->%d",cand->base.ifname,
+            next_automatic_preference);
+          if (r=NR_reg_set2_uchar(NR_ICE_REG_PREF_INTERFACE_PRFX,cand->base.ifname,next_automatic_preference)){
+            ABORT(r);
+          }
+          interface_preference=next_automatic_preference << 1;
+          next_automatic_preference--;
+          if (cand->base.ip_version == NR_IPV6) {
+            /* Prefer IPV6 over IPV4 on the same interface. */
+            interface_preference += 1;
+          }
+        }
+        else {
+          ABORT(r);
+        }
+      }
+    }
+    else {
+      char key_of_interface[MAXIFNAME + 41];
+      nr_transport_addr addr;
+
+      if(r=nr_socket_getaddr(cand->isock->sock, &addr))
+        ABORT(r);
+
+      if(r=nr_transport_addr_fmt_ifname_addr_string(&addr,key_of_interface,
+         sizeof(key_of_interface))) {
+        ABORT(r);
+      }
+      if(r=nr_interface_prioritizer_get_priority(cand->ctx->interface_prioritizer,
+         key_of_interface,&interface_preference)) {
+        ABORT(r);
+      }
+    }
+
+    assert(stun_priority < 32);
+    assert(direction_priority < 8);
+
+    cand->priority=
+      (type_preference << 24) |
+      (interface_preference << 16) |
+      (direction_priority << 13) |
+      (stun_priority << 8) |
+      (256 - cand->component_id);
+
+    /* S 4.1.2 */
+    assert(cand->priority>=1&&cand->priority<=2147483647);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+static void nr_ice_candidate_fire_ready_cb(NR_SOCKET s, int how, void *cb_arg)
+  {
+    nr_ice_candidate *cand = cb_arg;
+
+    cand->ready_cb_timer = 0;
+    cand->ready_cb(0, 0, cand->ready_cb_arg);
+  }
+
+static int nr_ice_candidate_use_nr_resolver(nr_transport_addr *addr)
+  {
+    if(addr->fqdn[0] == 0) {
+      return 0;
+    }
+
+    char use = 0;
+    if(addr->protocol == IPPROTO_UDP) {
+      NR_reg_get_char(NR_ICE_REG_USE_NR_RESOLVER_FOR_UDP, &use);
+    } else if(addr->protocol == IPPROTO_TCP) {
+      NR_reg_get_char(NR_ICE_REG_USE_NR_RESOLVER_FOR_TCP, &use);
+    } else {
+      assert(0);
+    }
+
+    return use;
+  }
+
+int nr_ice_candidate_initialize(nr_ice_candidate *cand, NR_async_cb ready_cb, void *cb_arg)
+  {
+    int r,_status;
+    int protocol=NR_RESOLVE_PROTOCOL_STUN;
+    cand->done_cb=ready_cb;
+    cand->cb_arg=cb_arg;
+    cand->state=NR_ICE_CAND_STATE_INITIALIZING;
+
+    switch(cand->type){
+      case HOST:
+        if(r=nr_socket_getaddr(cand->isock->sock,&cand->addr))
+          ABORT(r);
+        cand->osock=cand->isock->sock;
+        // Post this so that it doesn't happen in-line
+        cand->ready_cb = ready_cb;
+        cand->ready_cb_arg = cb_arg;
+        NR_ASYNC_TIMER_SET(0, nr_ice_candidate_fire_ready_cb, (void *)cand, &cand->ready_cb_timer);
+        break;
+#ifdef USE_TURN
+      case RELAYED:
+        protocol=NR_RESOLVE_PROTOCOL_TURN;
+        /* Fall through */
+#endif
+      case SERVER_REFLEXIVE:
+        if (nr_transport_addr_cmp(&cand->base, &cand->stun_server->addr,
+                                  NR_TRANSPORT_ADDR_CMP_MODE_PROTOCOL)) {
+          r_log(LOG_ICE, LOG_INFO,
+                "ICE-CANDIDATE(%s): Skipping srflx/relayed candidate because "
+                "of IP version/transport mis-match with STUN/TURN server "
+                "(%u/%s - %u/%s).",
+                cand->label, cand->base.ip_version,
+                cand->base.protocol == IPPROTO_UDP ? "UDP" : "TCP",
+                cand->stun_server->addr.ip_version,
+                cand->stun_server->addr.protocol == IPPROTO_UDP ? "UDP"
+                                                                : "TCP");
+          ABORT(R_NOT_FOUND); /* Same error code when DNS lookup fails */
+        }
+
+        if(nr_ice_candidate_use_nr_resolver(&cand->stun_server->addr)) {
+          r_log(
+              LOG_ICE, LOG_DEBUG,
+              "ICE-CANDIDATE(%s): Starting DNS resolution (%u/%s - %u/%s).",
+              cand->label, cand->base.ip_version,
+              cand->base.protocol == IPPROTO_UDP ? "UDP" : "TCP",
+              cand->stun_server->addr.ip_version,
+              cand->stun_server->addr.protocol == IPPROTO_UDP ? "UDP" : "TCP");
+          nr_resolver_resource resource;
+          int port;
+          resource.domain_name = cand->stun_server->addr.fqdn;
+          if (r = nr_transport_addr_get_port(&cand->stun_server->addr, &port)) {
+            ABORT(r);
+          }
+          resource.port = (uint16_t)port;
+          resource.stun_turn=protocol;
+          resource.transport_protocol = cand->stun_server->addr.protocol;
+
+          switch (cand->base.ip_version) {
+            case NR_IPV4:
+              resource.address_family=AF_INET;
+              break;
+            case NR_IPV6:
+              resource.address_family=AF_INET6;
+              break;
+            default:
+              assert(0);
+              ABORT(R_BAD_ARGS);
+          }
+
+          /* Try to resolve */
+          if(!cand->ctx->resolver) {
+            r_log(LOG_ICE, LOG_ERR, "ICE-CANDIDATE(%s): Can't use DNS names without a resolver", cand->label);
+            ABORT(R_BAD_ARGS);
+          }
+
+          if(r=nr_resolver_resolve(cand->ctx->resolver,
+                                   &resource,
+                                   nr_ice_candidate_resolved_cb,
+                                   (void *)cand,
+                                   &cand->resolver_handle)){
+            r_log(LOG_ICE,LOG_ERR,"ICE-CANDIDATE(%s): Could not invoke DNS resolver",cand->label);
+            ABORT(r);
+          }
+        } else {
+          /* No nr_resolver for this, just copy the address and finish init */
+          if (r = nr_transport_addr_copy(&cand->stun_server_addr,
+                                         &cand->stun_server->addr)) {
+            r_log(LOG_ICE,LOG_ERR,"ICE-CANDIDATE(%s): Could not copy STUN server addr", cand->label);
+            ABORT(r);
+          }
+
+          if(r=nr_ice_candidate_initialize2(cand))
+            ABORT(r);
+        }
+        break;
+      default:
+        ABORT(R_INTERNAL);
+    }
+
+    nr_ice_candidate_compute_codeword(cand);
+
+    _status=0;
+  abort:
+    if(_status && _status!=R_WOULDBLOCK)
+      nr_ice_candidate_mark_done(cand, NR_ICE_CAND_STATE_FAILED);
+    return(_status);
+  }
+
+
+static int nr_ice_candidate_resolved_cb(void *cb_arg, nr_transport_addr *addr)
+  {
+    nr_ice_candidate *cand=cb_arg;
+    int r,_status;
+
+    cand->resolver_handle=0;
+
+    if(addr){
+      r_log(LOG_ICE,LOG_DEBUG,"ICE(%s): resolved candidate %s. addr=%s",
+            cand->ctx->label,cand->label,addr->as_string);
+    }
+    else {
+      r_log(LOG_ICE,LOG_WARNING,"ICE(%s): failed to resolve candidate %s.",
+            cand->ctx->label,cand->label);
+      ABORT(R_NOT_FOUND);
+    }
+
+    if (nr_transport_addr_check_compatibility(addr, &cand->base)) {
+      r_log(LOG_ICE,LOG_WARNING,"ICE(%s): Skipping STUN server because of link local mis-match for candidate %s",cand->ctx->label,cand->label);
+      ABORT(R_NOT_FOUND);
+    }
+
+    /* Copy the address */
+    if(r=nr_transport_addr_copy(&cand->stun_server_addr,addr))
+      ABORT(r);
+
+    if (cand->tcp_type == TCP_TYPE_PASSIVE || cand->tcp_type == TCP_TYPE_SO){
+      if (r=nr_socket_multi_tcp_stun_server_connect(cand->osock, addr))
+        ABORT(r);
+    }
+
+    /* Now start initializing */
+    if(r=nr_ice_candidate_initialize2(cand))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    if(_status && _status!=R_WOULDBLOCK) {
+      nr_ice_candidate_mark_done(cand, NR_ICE_CAND_STATE_FAILED);
+    }
+    return(_status);
+  }
+
+static int nr_ice_candidate_initialize2(nr_ice_candidate *cand)
+  {
+    int r,_status;
+
+    switch(cand->type){
+      case HOST:
+        assert(0); /* Can't happen */
+        ABORT(R_INTERNAL);
+        break;
+#ifdef USE_TURN
+      case RELAYED:
+        if(r=nr_ice_start_relay_turn(cand))
+          ABORT(r);
+        ABORT(R_WOULDBLOCK);
+        break;
+#endif /* USE_TURN */
+      case SERVER_REFLEXIVE:
+        /* Need to start stun */
+        if(r=nr_ice_srvrflx_start_stun(cand))
+          ABORT(r);
+        cand->osock=cand->isock->sock;
+        ABORT(R_WOULDBLOCK);
+        break;
+      default:
+        ABORT(R_INTERNAL);
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+static void nr_ice_srvrflx_start_stun_timer_cb(NR_SOCKET s, int how, void *cb_arg)
+  {
+    nr_ice_candidate *cand=cb_arg;
+    int r,_status;
+
+    cand->delay_timer=0;
+
+/* TODO: if the response is a BINDING-ERROR-RESPONSE, then restart
+ * TODO: using NR_STUN_CLIENT_MODE_BINDING_REQUEST because the
+ * TODO: server may not have understood the 0.96-style request */
+    if(r=nr_stun_client_start(cand->u.srvrflx.stun, NR_STUN_CLIENT_MODE_BINDING_REQUEST_NO_AUTH, nr_ice_srvrflx_stun_finished_cb, cand))
+      ABORT(r);
+
+    if(r=nr_ice_ctx_remember_id(cand->ctx, cand->u.srvrflx.stun->request))
+      ABORT(r);
+
+    if(r=nr_ice_socket_register_stun_client(cand->isock,cand->u.srvrflx.stun,&cand->u.srvrflx.stun_handle))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    if (_status && (cand->u.srvrflx.stun->state==NR_STUN_CLIENT_STATE_RUNNING)) {
+      nr_stun_client_failed(cand->u.srvrflx.stun);
+    }
+    return;
+  }
+
+static int nr_ice_srvrflx_start_stun(nr_ice_candidate *cand)
+  {
+    int r,_status;
+
+    assert(!cand->delay_timer);
+    if(r=nr_stun_client_ctx_create(cand->label, cand->isock->sock,
+      &cand->stun_server_addr, cand->stream->ctx->gather_rto,
+      &cand->u.srvrflx.stun))
+      ABORT(r);
+
+    NR_ASYNC_TIMER_SET(cand->stream->ctx->stun_delay,nr_ice_srvrflx_start_stun_timer_cb,cand,&cand->delay_timer);
+    cand->stream->ctx->stun_delay += cand->stream->ctx->Ta;
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+#ifdef USE_TURN
+static void nr_ice_start_relay_turn_timer_cb(NR_SOCKET s, int how, void *cb_arg)
+  {
+    nr_ice_candidate *cand=cb_arg;
+    int r,_status;
+
+    cand->delay_timer=0;
+
+    if(r=nr_turn_client_allocate(cand->u.relayed.turn, nr_ice_turn_allocated_cb, cb_arg))
+      ABORT(r);
+
+    if(r=nr_ice_socket_register_turn_client(cand->isock, cand->u.relayed.turn,
+                                            cand->osock, &cand->u.relayed.turn_handle))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    if(_status && (cand->u.relayed.turn->state==NR_TURN_CLIENT_STATE_ALLOCATING)){
+      nr_turn_client_failed(cand->u.relayed.turn);
+    }
+    return;
+  }
+
+static int nr_ice_start_relay_turn(nr_ice_candidate *cand)
+  {
+    int r,_status;
+    assert(!cand->delay_timer);
+    if(r=nr_turn_client_ctx_create(cand->label, cand->isock->sock,
+                                   cand->u.relayed.server->username,
+                                   cand->u.relayed.server->password,
+                                   &cand->stun_server_addr,
+                                   cand->component->ctx,
+                                   &cand->u.relayed.turn))
+      ABORT(r);
+
+    if(r=nr_socket_turn_set_ctx(cand->osock, cand->u.relayed.turn))
+      ABORT(r);
+
+    NR_ASYNC_TIMER_SET(cand->stream->ctx->stun_delay,nr_ice_start_relay_turn_timer_cb,cand,&cand->delay_timer);
+    cand->stream->ctx->stun_delay += cand->stream->ctx->Ta;
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+#endif /* USE_TURN */
+
+static void nr_ice_srvrflx_stun_finished_cb(NR_SOCKET sock, int how, void *cb_arg)
+  {
+    int _status;
+    nr_ice_candidate *cand=cb_arg;
+
+    r_log(LOG_ICE,LOG_DEBUG,"ICE(%s)/CAND(%s): %s",cand->ctx->label,cand->label,__FUNCTION__);
+
+    /* Deregister to suppress duplicates */
+    if(cand->u.srvrflx.stun_handle){ /* This test because we might have failed before CB registered */
+      nr_ice_socket_deregister(cand->isock,cand->u.srvrflx.stun_handle);
+      cand->u.srvrflx.stun_handle=0;
+    }
+
+    switch(cand->u.srvrflx.stun->state){
+      /* OK, we should have a mapped address */
+      case NR_STUN_CLIENT_STATE_DONE:
+        /* Copy the address */
+        nr_transport_addr_copy(&cand->addr, &cand->u.srvrflx.stun->results.stun_binding_response.mapped_addr);
+        cand->addr.protocol=cand->base.protocol;
+        nr_transport_addr_fmt_addr_string(&cand->addr);
+        nr_stun_client_ctx_destroy(&cand->u.srvrflx.stun);
+        nr_ice_candidate_mark_done(cand, NR_ICE_CAND_STATE_INITIALIZED);
+        cand=0;
+        break;
+
+      /* This failed, so go to the next STUN server if there is one */
+      case NR_STUN_CLIENT_STATE_FAILED:
+        ABORT(R_NOT_FOUND);
+        break;
+      default:
+        ABORT(R_INTERNAL);
+    }
+    _status = 0;
+  abort:
+    if(_status){
+      nr_ice_candidate_mark_done(cand, NR_ICE_CAND_STATE_FAILED);
+    }
+  }
+
+#ifdef USE_TURN
+static void nr_ice_turn_allocated_cb(NR_SOCKET s, int how, void *cb_arg)
+  {
+    int r,_status;
+    nr_ice_candidate *cand=cb_arg;
+    nr_turn_client_ctx *turn=cand->u.relayed.turn;
+    char *label;
+    nr_transport_addr relay_addr;
+
+    switch(turn->state){
+      /* OK, we should have a mapped address */
+      case NR_TURN_CLIENT_STATE_ALLOCATED:
+        if (r=nr_turn_client_get_relayed_address(turn, &relay_addr))
+          ABORT(r);
+
+        if(r=nr_concat_strings(&label,"turn-relay(",cand->base.as_string,"|",
+                               relay_addr.as_string,")",NULL))
+          ABORT(r);
+
+        r_log(LOG_ICE,LOG_DEBUG,"TURN-CLIENT(%s)/CAND(%s): Switching from TURN to RELAY (%s)",cand->u.relayed.turn->label,cand->label,label);
+
+        /* Copy the relayed address into the candidate addr and
+           into the candidate base. Note that we need to keep the
+           ifname in the base. */
+        if (r=nr_transport_addr_copy(&cand->addr, &relay_addr))
+          ABORT(r);
+        if (r=nr_transport_addr_copy_keep_ifname(&cand->base, &relay_addr))  /* Need to keep interface for priority calculation */
+          ABORT(r);
+
+        r_log(LOG_ICE,LOG_DEBUG,"ICE(%s)/CAND(%s): new relay base=%s addr=%s", cand->ctx->label, cand->label, cand->base.as_string, cand->addr.as_string);
+
+        RFREE(cand->label);
+        cand->label=label;
+        nr_ice_candidate_mark_done(cand, NR_ICE_CAND_STATE_INITIALIZED);
+        cand = 0;
+
+        break;
+
+    case NR_TURN_CLIENT_STATE_FAILED:
+    case NR_TURN_CLIENT_STATE_CANCELLED:
+      r_log(NR_LOG_TURN, LOG_WARNING,
+            "ICE-CANDIDATE(%s): nr_turn_allocated_cb called with state %d",
+            cand->label, turn->state);
+      /* This failed, so go to the next TURN server if there is one */
+      ABORT(R_NOT_FOUND);
+      break;
+    default:
+      assert(0); /* should never happen */
+      ABORT(R_INTERNAL);
+    }
+
+    _status=0;
+  abort:
+    if(_status){
+      if (cand) {
+        r_log(NR_LOG_TURN, LOG_WARNING,
+              "ICE-CANDIDATE(%s): nr_turn_allocated_cb failed", cand->label);
+        nr_ice_candidate_mark_done(cand, NR_ICE_CAND_STATE_FAILED);
+      }
+    }
+  }
+#endif /* USE_TURN */
+
+/* Format the candidate attribute as per ICE S 15.1 */
+int nr_ice_format_candidate_attribute(nr_ice_candidate *cand, char *attr, int maxlen, int obfuscate_srflx_addr)
+  {
+    int r,_status;
+    char addr[64];
+    int port;
+    int len;
+    nr_transport_addr *raddr;
+
+    assert(!strcmp(nr_ice_candidate_type_names[HOST], "host"));
+    assert(!strcmp(nr_ice_candidate_type_names[RELAYED], "relay"));
+
+    if (cand->mdns_addr) {
+      /* mdns_addr is NSID_LENGTH which is 39, - 2 for removing the "{" and "}"
+         + 6 for ".local" for a total of 43. */
+      strncpy(addr, cand->mdns_addr, sizeof(addr) - 1);
+    } else {
+      if(r=nr_transport_addr_get_addrstring(&cand->addr,addr,sizeof(addr)))
+        ABORT(r);
+    }
+    if(r=nr_transport_addr_get_port(&cand->addr,&port))
+      ABORT(r);
+    /* https://tools.ietf.org/html/rfc6544#section-4.5 */
+    if (cand->base.protocol==IPPROTO_TCP && cand->tcp_type==TCP_TYPE_ACTIVE)
+      port=9;
+    snprintf(attr,maxlen,"candidate:%s %d %s %u %s %d typ %s",
+      cand->foundation, cand->component_id, cand->addr.protocol==IPPROTO_UDP?"UDP":"TCP",cand->priority, addr, port,
+      nr_ctype_name(cand->type));
+
+    len=strlen(attr); attr+=len; maxlen-=len;
+
+    /* raddr, rport */
+    raddr = (cand->stream->flags &
+             (NR_ICE_CTX_FLAGS_RELAY_ONLY |
+              NR_ICE_CTX_FLAGS_DISABLE_HOST_CANDIDATES)) ?
+      &cand->addr : &cand->base;
+
+    switch(cand->type){
+      case HOST:
+        break;
+      case SERVER_REFLEXIVE:
+        if (obfuscate_srflx_addr) {
+          snprintf(attr,maxlen," raddr 0.0.0.0 rport 0");
+        } else {
+          if(r=nr_transport_addr_get_addrstring(raddr,addr,sizeof(addr)))
+            ABORT(r);
+          if(r=nr_transport_addr_get_port(raddr,&port))
+            ABORT(r);
+          snprintf(attr,maxlen," raddr %s rport %d",addr,port);
+        }
+        break;
+      case PEER_REFLEXIVE:
+        if(r=nr_transport_addr_get_addrstring(raddr,addr,sizeof(addr)))
+          ABORT(r);
+        if(r=nr_transport_addr_get_port(raddr,&port))
+          ABORT(r);
+        snprintf(attr,maxlen," raddr %s rport %d",addr,port);
+        break;
+      case RELAYED:
+        // comes from XorMappedAddress via AllocateResponse
+        if(r=nr_transport_addr_get_addrstring(raddr,addr,sizeof(addr)))
+          ABORT(r);
+        if(r=nr_transport_addr_get_port(raddr,&port))
+          ABORT(r);
+
+        snprintf(attr,maxlen," raddr %s rport %d",addr,port);
+        break;
+      default:
+        assert(0);
+        ABORT(R_INTERNAL);
+        break;
+    }
+
+    if (cand->base.protocol==IPPROTO_TCP && cand->tcp_type){
+      len=strlen(attr);
+      attr+=len;
+      maxlen-=len;
+      snprintf(attr,maxlen," tcptype %s",nr_tcp_type_name(cand->tcp_type));
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_candidate.h b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_candidate.h
new file mode 100644
index 0000000000..40e6545e37
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_candidate.h
@@ -0,0 +1,124 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _ice_candidate_h
+#define _ice_candidate_h
+#ifdef __cplusplus
+using namespace std;
+extern "C" {
+#endif /* __cplusplus */
+
+typedef enum {HOST=1, SERVER_REFLEXIVE, PEER_REFLEXIVE, RELAYED, CTYPE_MAX} nr_ice_candidate_type;
+
+struct nr_ice_candidate_ {
+  char *label;
+  char codeword[5];
+  int state;
+  int trickled;
+#define NR_ICE_CAND_STATE_CREATED          1
+#define NR_ICE_CAND_STATE_INITIALIZING     2
+#define NR_ICE_CAND_STATE_INITIALIZED      3
+#define NR_ICE_CAND_STATE_FAILED           4
+#define NR_ICE_CAND_PEER_CANDIDATE_UNPAIRED 9
+#define NR_ICE_CAND_PEER_CANDIDATE_PAIRED   10
+  struct nr_ice_ctx_ *ctx;
+  nr_ice_socket *isock;               /* The socket to read from
+                                         (it contains all other candidates
+                                         on this socket) */
+  nr_socket *osock;                   /* The socket to write to */
+  nr_ice_media_stream *stream;        /* The media stream this is associated with */
+  nr_ice_component *component;        /* The component this is associated with */
+  nr_ice_candidate_type type;         /* The type of the candidate (S 4.1.1) */
+  nr_socket_tcp_type tcp_type;
+  UCHAR component_id;                 /* The component id (S 4.1.2.1) */
+  nr_transport_addr addr;             /* The advertised address;
+                                         JDR calls this the candidate */
+  nr_transport_addr base;             /* The base address (S 2.1)*/
+  char *mdns_addr;                    /* MDNS address, if any */
+  char *foundation;                   /* Foundation for the candidate (S 4) */
+  UINT4 priority;                     /* The priority value (S 5.4 */
+  nr_ice_stun_server *stun_server;
+  nr_transport_addr stun_server_addr; /* Resolved STUN server address */
+  void *delay_timer;
+  void *resolver_handle;
+
+  /* Holding data for STUN and TURN */
+  union {
+    struct {
+      nr_stun_client_ctx *stun;
+      void *stun_handle;
+      /* If this is a srflx that is piggybacking on a relay candidate, this is
+       * a back pointer to that relay candidate. */
+      nr_ice_candidate *relay_candidate;
+    } srvrflx;
+    struct {
+      nr_turn_client_ctx *turn;
+      nr_ice_turn_server *server;
+      nr_ice_candidate *srvflx_candidate;
+      nr_socket *turn_sock;
+      void *turn_handle;
+    } relayed;
+  } u;
+
+  NR_async_cb done_cb;
+  void *cb_arg;
+
+  NR_async_cb ready_cb;
+  void *ready_cb_arg;
+  void *ready_cb_timer;
+
+  TAILQ_ENTRY(nr_ice_candidate_) entry_sock;
+  TAILQ_ENTRY(nr_ice_candidate_) entry_comp;
+};
+
+extern char *nr_ice_candidate_type_names[];
+extern char *nr_ice_candidate_tcp_type_names[];
+
+
+int nr_ice_candidate_create(struct nr_ice_ctx_ *ctx,nr_ice_component *component, nr_ice_socket *isock, nr_socket *osock, nr_ice_candidate_type ctype, nr_socket_tcp_type tcp_type, nr_ice_stun_server *stun_server, UCHAR component_id, nr_ice_candidate **candp);
+int nr_ice_candidate_initialize(nr_ice_candidate *cand, NR_async_cb ready_cb, void *cb_arg);
+void nr_ice_candidate_compute_codeword(nr_ice_candidate *cand);
+int nr_ice_candidate_process_stun(nr_ice_candidate *cand, UCHAR *msg, int len, nr_transport_addr *faddr);
+int nr_ice_candidate_destroy(nr_ice_candidate **candp);
+void nr_ice_candidate_stop_gathering(nr_ice_candidate *cand);
+int nr_ice_format_candidate_attribute(nr_ice_candidate *cand, char *attr, int maxlen, int obfuscate_srflx_addr);
+int nr_ice_peer_candidate_from_attribute(nr_ice_ctx *ctx,char *attr,nr_ice_media_stream *stream,nr_ice_candidate **candp);
+int nr_ice_peer_peer_rflx_candidate_create(nr_ice_ctx *ctx,char *label, nr_ice_component *comp,nr_transport_addr *addr, nr_ice_candidate **candp);
+int nr_ice_candidate_compute_priority(nr_ice_candidate *cand);
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_candidate_pair.c b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_candidate_pair.c
new file mode 100644
index 0000000000..6a7233963d
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_candidate_pair.c
@@ -0,0 +1,689 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <assert.h>
+#include <string.h>
+#include <nr_api.h>
+#include "async_timer.h"
+#include "ice_ctx.h"
+#include "ice_util.h"
+#include "ice_codeword.h"
+#include "stun.h"
+
+static char *nr_ice_cand_pair_states[]={"UNKNOWN","FROZEN","WAITING","IN_PROGRESS","FAILED","SUCCEEDED","CANCELLED"};
+
+static void nr_ice_candidate_pair_restart_stun_role_change_cb(NR_SOCKET s, int how, void *cb_arg);
+static void nr_ice_candidate_pair_compute_codeword(nr_ice_cand_pair *pair,
+  nr_ice_candidate *lcand, nr_ice_candidate *rcand);
+
+static void nr_ice_candidate_pair_set_priority(nr_ice_cand_pair *pair)
+  {
+    /* Priority computation S 5.7.2 */
+    UINT8 controlling_priority, controlled_priority;
+    if(pair->pctx->controlling)
+    {
+      controlling_priority=pair->local->priority;
+      controlled_priority=pair->remote->priority;
+    }
+    else{
+      controlling_priority=pair->remote->priority;
+      controlled_priority=pair->local->priority;
+    }
+    pair->priority=(MIN(controlling_priority, controlled_priority))<<32 |
+      (MAX(controlling_priority, controlled_priority))<<1 |
+      (controlled_priority > controlling_priority?0:1);
+  }
+
+int nr_ice_candidate_pair_create(nr_ice_peer_ctx *pctx, nr_ice_candidate *lcand,nr_ice_candidate *rcand,nr_ice_cand_pair **pairp)
+  {
+    nr_ice_cand_pair *pair=0;
+    int r,_status;
+    UINT4 RTO;
+    nr_ice_candidate tmpcand;
+    UINT8 t_priority;
+
+    if(!(pair=RCALLOC(sizeof(nr_ice_cand_pair))))
+      ABORT(R_NO_MEMORY);
+
+    pair->pctx=pctx;
+
+    nr_ice_candidate_pair_compute_codeword(pair,lcand,rcand);
+
+    if(r=nr_concat_strings(&pair->as_string,pair->codeword,"|",lcand->addr.as_string,"|",
+        rcand->addr.as_string,"(",lcand->label,"|",rcand->label,")", NULL))
+      ABORT(r);
+
+    nr_ice_candidate_pair_set_state(pctx,pair,NR_ICE_PAIR_STATE_FROZEN);
+    pair->local=lcand;
+    pair->remote=rcand;
+
+    nr_ice_candidate_pair_set_priority(pair);
+
+    /*
+       TODO(bcampen@mozilla.com): Would be nice to log why this candidate was
+       created (initial pair generation, triggered check, and new trickle
+       candidate seem to be the possibilities here).
+    */
+    r_log(LOG_ICE,LOG_INFO,"ICE(%s)/CAND-PAIR(%s): Pairing candidate %s (%x):%s (%x) priority=%llu (%llx)",pctx->ctx->label,pair->codeword,lcand->addr.as_string,lcand->priority,rcand->addr.as_string,rcand->priority,pair->priority,pair->priority);
+
+    /* Foundation */
+    if(r=nr_concat_strings(&pair->foundation,lcand->foundation,"|",
+      rcand->foundation,NULL))
+      ABORT(r);
+
+    /* Compute the RTO per S 16 */
+    RTO = MAX(100, (pctx->ctx->Ta * pctx->waiting_pairs));
+
+    /* Make a bogus candidate to compute a theoretical peer reflexive
+     * priority per S 7.1.1.1 */
+    memcpy(&tmpcand, lcand, sizeof(tmpcand));
+    tmpcand.type = PEER_REFLEXIVE;
+    if (r=nr_ice_candidate_compute_priority(&tmpcand))
+      ABORT(r);
+    t_priority = tmpcand.priority;
+
+    /* Our sending context */
+    if(r=nr_stun_client_ctx_create(pair->as_string,
+      lcand->osock,
+      &rcand->addr,RTO,&pair->stun_client))
+      ABORT(r);
+    if(!(pair->stun_client->params.ice_binding_request.username=r_strdup(rcand->stream->l2r_user)))
+      ABORT(R_NO_MEMORY);
+    if(r=r_data_copy(&pair->stun_client->params.ice_binding_request.password,
+      &rcand->stream->l2r_pass))
+      ABORT(r);
+    /* TODO(ekr@rtfm.com): Do we need to frob this when we change role. Bug 890667 */
+    pair->stun_client->params.ice_binding_request.control = pctx->controlling?
+      NR_ICE_CONTROLLING:NR_ICE_CONTROLLED;
+    pair->stun_client->params.ice_binding_request.priority=t_priority;
+
+    pair->stun_client->params.ice_binding_request.tiebreaker=pctx->tiebreaker;
+
+    *pairp=pair;
+
+    _status=0;
+  abort:
+    if(_status){
+      nr_ice_candidate_pair_destroy(&pair);
+    }
+    return(_status);
+  }
+
+int nr_ice_candidate_pair_destroy(nr_ice_cand_pair **pairp)
+  {
+    nr_ice_cand_pair *pair;
+
+    if(!pairp || !*pairp)
+      return(0);
+
+    pair=*pairp;
+    *pairp=0;
+
+    // record stats back to the ice ctx on destruction
+    if (pair->stun_client) {
+      nr_accumulate_count(&(pair->local->ctx->stats.stun_retransmits), pair->stun_client->retransmit_ct);
+    }
+
+    RFREE(pair->as_string);
+    RFREE(pair->foundation);
+    nr_ice_socket_deregister(pair->local->isock,pair->stun_client_handle);
+    if (pair->stun_client) {
+      RFREE(pair->stun_client->params.ice_binding_request.username);
+      RFREE(pair->stun_client->params.ice_binding_request.password.data);
+      nr_stun_client_ctx_destroy(&pair->stun_client);
+    }
+
+    NR_async_timer_cancel(pair->stun_cb_timer);
+    NR_async_timer_cancel(pair->restart_role_change_cb_timer);
+    NR_async_timer_cancel(pair->restart_nominated_cb_timer);
+
+    RFREE(pair);
+    return(0);
+  }
+
+int nr_ice_candidate_pair_unfreeze(nr_ice_peer_ctx *pctx, nr_ice_cand_pair *pair)
+  {
+    assert(pair->state==NR_ICE_PAIR_STATE_FROZEN);
+
+    nr_ice_candidate_pair_set_state(pctx,pair,NR_ICE_PAIR_STATE_WAITING);
+
+    return(0);
+  }
+
+static void nr_ice_candidate_pair_stun_cb(NR_SOCKET s, int how, void *cb_arg)
+  {
+    int r,_status;
+    nr_ice_cand_pair *pair=cb_arg;
+    nr_ice_cand_pair *actual_pair=0;
+    nr_ice_candidate *cand=0;
+    nr_stun_message *sres;
+    nr_transport_addr *request_src;
+    nr_transport_addr *request_dst;
+    nr_transport_addr *response_src;
+    nr_transport_addr response_dst;
+    nr_stun_message_attribute *attr;
+
+    pair->stun_cb_timer=0;
+
+    r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s)/STREAM(%s)/CAND-PAIR(%s): STUN cb on pair addr = %s",
+      pair->pctx->label,pair->local->stream->label,pair->codeword,pair->as_string);
+
+    /* This ordinarily shouldn't happen, but can if we're
+       doing the second check to confirm nomination.
+       Just bail out */
+    if(pair->state==NR_ICE_PAIR_STATE_SUCCEEDED)
+      goto done;
+
+    switch(pair->stun_client->state){
+      case NR_STUN_CLIENT_STATE_FAILED:
+        sres=pair->stun_client->response;
+        if(sres && nr_stun_message_has_attribute(sres,NR_STUN_ATTR_ERROR_CODE,&attr)&&attr->u.error_code.number==487){
+
+          /*
+           * Flip the controlling bit; subsequent 487s for other pairs will be
+           * ignored, since we abandon their STUN transactions.
+           */
+          nr_ice_peer_ctx_switch_controlling_role(pair->pctx);
+
+          return;
+        }
+        /* Fall through */
+      case NR_STUN_CLIENT_STATE_TIMED_OUT:
+        nr_ice_candidate_pair_set_state(pair->pctx,pair,NR_ICE_PAIR_STATE_FAILED);
+        break;
+      case NR_STUN_CLIENT_STATE_DONE:
+        /* make sure the addresses match up S 7.1.2.2 */
+        response_src=&pair->stun_client->peer_addr;
+        request_dst=&pair->remote->addr;
+        if (nr_transport_addr_cmp(response_src,request_dst,NR_TRANSPORT_ADDR_CMP_MODE_ALL)){
+          r_log(LOG_ICE,LOG_WARNING,"ICE-PEER(%s)/CAND-PAIR(%s): Peer address mismatch %s != %s",pair->pctx->label,pair->codeword,response_src->as_string,request_dst->as_string);
+          nr_ice_candidate_pair_set_state(pair->pctx,pair,NR_ICE_PAIR_STATE_FAILED);
+          break;
+        }
+        request_src=&pair->stun_client->my_addr;
+        nr_socket_getaddr(pair->local->osock,&response_dst);
+        if (nr_transport_addr_cmp(request_src,&response_dst,NR_TRANSPORT_ADDR_CMP_MODE_ALL)){
+          r_log(LOG_ICE,LOG_WARNING,"ICE-PEER(%s)/CAND-PAIR(%s): Local address mismatch %s != %s",pair->pctx->label,pair->codeword,request_src->as_string,response_dst.as_string);
+          nr_ice_candidate_pair_set_state(pair->pctx,pair,NR_ICE_PAIR_STATE_FAILED);
+          break;
+        }
+
+        if(strlen(pair->stun_client->results.ice_binding_response.mapped_addr.as_string)==0){
+          /* we're using the mapped_addr returned by the server to lookup our
+           * candidate, but if the server fails to do that we can't perform
+           * the lookup -- this may be a BUG because if we've gotten here
+           * then the transaction ID check succeeded, and perhaps we should
+           * just assume that it's the server we're talking to and that our
+           * peer is ok, but I'm not sure how that'll interact with the
+           * peer reflexive logic below */
+          r_log(LOG_ICE,LOG_WARNING,"ICE-PEER(%s)/CAND-PAIR(%s): server failed to return mapped address on pair %s", pair->pctx->label,pair->codeword,pair->as_string);
+          nr_ice_candidate_pair_set_state(pair->pctx,pair,NR_ICE_PAIR_STATE_FAILED);
+          break;
+        }
+        else if(!nr_transport_addr_cmp(&pair->local->addr,&pair->stun_client->results.ice_binding_response.mapped_addr,NR_TRANSPORT_ADDR_CMP_MODE_ALL)){
+          nr_ice_candidate_pair_set_state(pair->pctx,pair,NR_ICE_PAIR_STATE_SUCCEEDED);
+        }
+        else if(pair->stun_client->state == NR_STUN_CLIENT_STATE_DONE) {
+          /* OK, this didn't correspond to a pair on the check list, but
+             it probably matches one of our candidates */
+
+          cand=TAILQ_FIRST(&pair->local->component->candidates);
+          while(cand){
+            if(!nr_transport_addr_cmp(&cand->addr,&pair->stun_client->results.ice_binding_response.mapped_addr,NR_TRANSPORT_ADDR_CMP_MODE_ALL)) {
+              r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s): found pre-existing local candidate of type %d for mapped address %s", pair->pctx->label,cand->type,cand->addr.as_string);
+              assert(cand->type != HOST);
+              break;
+            }
+
+            cand=TAILQ_NEXT(cand,entry_comp);
+          }
+
+          if(!cand) {
+            /* OK, nothing found, must be a new peer reflexive */
+            if (pair->local->stream->flags & NR_ICE_CTX_FLAGS_RELAY_ONLY) {
+              /* Any STUN response with a reflexive address in it is unwanted
+                 when we'll send on relay only. Bail since cand is used below. */
+              goto done;
+            }
+            if(r=nr_ice_candidate_create(pair->pctx->ctx,
+              pair->local->component,pair->local->isock,pair->local->osock,
+              PEER_REFLEXIVE,pair->local->tcp_type,0,pair->local->component->component_id,&cand))
+              ABORT(r);
+            if(r=nr_transport_addr_copy(&cand->addr,&pair->stun_client->results.ice_binding_response.mapped_addr))
+              ABORT(r);
+            cand->state=NR_ICE_CAND_STATE_INITIALIZED;
+            TAILQ_INSERT_TAIL(&pair->local->component->candidates,cand,entry_comp);
+          } else {
+            /* Check if we have a pair for this candidate already. */
+            if(r=nr_ice_media_stream_find_pair(pair->remote->stream, cand, pair->remote, &actual_pair)) {
+              r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s): no pair exists for %s and %s", pair->pctx->label,cand->addr.as_string, pair->remote->addr.as_string);
+            }
+          }
+
+          if(!actual_pair) {
+            if(r=nr_ice_candidate_pair_create(pair->pctx,cand,pair->remote, &actual_pair))
+              ABORT(r);
+
+            if(r=nr_ice_component_insert_pair(actual_pair->remote->component,actual_pair))
+              ABORT(r);
+
+            /* If the original pair was nominated, make us nominated too. */
+            if(pair->peer_nominated)
+              actual_pair->peer_nominated=1;
+
+            /* Now mark the orig pair failed */
+            nr_ice_candidate_pair_set_state(pair->pctx,pair,NR_ICE_PAIR_STATE_FAILED);
+          }
+
+          assert(actual_pair);
+          nr_ice_candidate_pair_set_state(actual_pair->pctx,actual_pair,NR_ICE_PAIR_STATE_SUCCEEDED);
+          pair=actual_pair;
+
+        }
+
+        /* Should we set nominated? */
+        if(pair->pctx->controlling){
+          if(pair->pctx->ctx->flags & NR_ICE_CTX_FLAGS_AGGRESSIVE_NOMINATION)
+            pair->nominated=1;
+        }
+        else{
+          if(pair->peer_nominated)
+            pair->nominated=1;
+        }
+
+
+        /* increment the number of valid pairs in the component */
+        /* We don't bother to maintain a separate valid list */
+        pair->remote->component->valid_pairs++;
+
+        /* S 7.1.2.2: unfreeze other pairs with the same foundation*/
+        if(r=nr_ice_media_stream_unfreeze_pairs_foundation(pair->remote->stream,pair->foundation))
+          ABORT(r);
+
+        /* Deal with this pair being nominated */
+        if(pair->nominated){
+          if(r=nr_ice_component_nominated_pair(pair->remote->component, pair))
+            ABORT(r);
+        }
+
+        break;
+      default:
+        ABORT(R_INTERNAL);
+    }
+
+    /* If we're controlling but in regular mode, ask the handler
+       if he wants to nominate something and stop... */
+    if(pair->pctx->controlling && !(pair->pctx->ctx->flags & NR_ICE_CTX_FLAGS_AGGRESSIVE_NOMINATION)){
+
+      if(r=nr_ice_component_select_pair(pair->pctx,pair->remote->component)){
+        if(r!=R_NOT_FOUND)
+          ABORT(r);
+      }
+    }
+
+  done:
+    _status=0;
+  abort:
+    if (_status) {
+      // cb doesn't return anything, but we should probably log that we aborted
+      // This also quiets the unused variable warnings.
+      r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s)/STREAM(%s)/CAND-PAIR(%s): STUN cb pair addr = %s abort with status: %d",
+        pair->pctx->label,pair->local->stream->label,pair->codeword,pair->as_string, _status);
+    }
+    return;
+  }
+
+static void nr_ice_candidate_pair_restart(nr_ice_peer_ctx *pctx, nr_ice_cand_pair *pair)
+  {
+    int r,_status;
+    UINT4 mode;
+
+    nr_ice_candidate_pair_set_state(pctx,pair,NR_ICE_PAIR_STATE_IN_PROGRESS);
+
+    /* Start STUN */
+    if(pair->pctx->controlling && (pair->pctx->ctx->flags & NR_ICE_CTX_FLAGS_AGGRESSIVE_NOMINATION))
+      mode=NR_ICE_CLIENT_MODE_USE_CANDIDATE;
+    else
+      mode=NR_ICE_CLIENT_MODE_BINDING_REQUEST;
+
+    nr_stun_client_reset(pair->stun_client);
+
+    if(r=nr_stun_client_start(pair->stun_client,mode,nr_ice_candidate_pair_stun_cb,pair))
+      ABORT(r);
+
+    if ((r=nr_ice_ctx_remember_id(pair->pctx->ctx, pair->stun_client->request))) {
+      /* ignore if this fails (which it shouldn't) because it's only an
+       * optimization and the cleanup routines are not going to do the right
+       * thing if this fails */
+      assert(0);
+    }
+
+    _status=0;
+  abort:
+    if(_status){
+      /* Don't fire the CB, but schedule it to fire ASAP */
+      assert(!pair->stun_cb_timer);
+      NR_ASYNC_TIMER_SET(0,nr_ice_candidate_pair_stun_cb,pair, &pair->stun_cb_timer);
+      _status=0;
+    }
+  }
+
+int nr_ice_candidate_pair_start(nr_ice_peer_ctx *pctx, nr_ice_cand_pair *pair)
+  {
+    int r,_status;
+
+    /* Register the stun ctx for when responses come in*/
+    if(r=nr_ice_socket_register_stun_client(pair->local->isock,pair->stun_client,&pair->stun_client_handle))
+      ABORT(r);
+
+    nr_ice_candidate_pair_restart(pctx, pair);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+static int nr_ice_candidate_copy_for_triggered_check(nr_ice_cand_pair *pair)
+  {
+    int r,_status;
+    nr_ice_cand_pair *copy;
+
+    if(r=nr_ice_candidate_pair_create(pair->pctx, pair->local, pair->remote, &copy))
+      ABORT(r);
+
+    /* Preserve nomination status */
+    copy->peer_nominated= pair->peer_nominated;
+    copy->nominated = pair->nominated;
+
+    r_log(LOG_ICE,LOG_INFO,"CAND-PAIR(%s): Adding pair to check list and trigger check queue: %s",pair->codeword,pair->as_string);
+    nr_ice_candidate_pair_insert(&pair->remote->stream->check_list,copy);
+    nr_ice_candidate_pair_trigger_check_append(&pair->remote->stream->trigger_check_queue,copy);
+
+    copy->triggered = 1;
+    nr_ice_candidate_pair_set_state(copy->pctx,copy,NR_ICE_PAIR_STATE_WAITING);
+
+    _status=0;
+  abort:
+    return(_status);
+}
+
+int nr_ice_candidate_pair_do_triggered_check(nr_ice_peer_ctx *pctx, nr_ice_cand_pair *pair)
+  {
+    int r,_status;
+
+    if(pair->state==NR_ICE_PAIR_STATE_CANCELLED) {
+      r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s)/CAND_PAIR(%s): Ignoring matching but canceled pair",pctx->label,pair->codeword);
+      return(0);
+    } else if(pair->state==NR_ICE_PAIR_STATE_SUCCEEDED) {
+      r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s)/CAND_PAIR(%s): No new trigger check for succeeded pair",pctx->label,pair->codeword);
+      return(0);
+    } else if (pair->local->stream->obsolete) {
+      r_log(LOG_ICE, LOG_DEBUG,
+            "ICE-PEER(%s)/CAND_PAIR(%s): No new trigger check for pair with "
+            "obsolete stream",
+            pctx->label, pair->codeword);
+      return (0);
+    }
+
+    /* Do not run this logic more than once on a given pair */
+    if(!pair->triggered){
+      r_log(LOG_ICE,LOG_INFO,"ICE-PEER(%s)/CAND-PAIR(%s): triggered check on %s",pctx->label,pair->codeword,pair->as_string);
+
+      pair->triggered=1;
+
+      switch(pair->state){
+        case NR_ICE_PAIR_STATE_FAILED:
+          /* OK, there was a pair, it's just invalid: According to Section
+           * 7.2.1.4, we need to resurrect it */
+          r_log(LOG_ICE,LOG_INFO,"ICE-PEER(%s)/CAND-PAIR(%s): received STUN check on failed pair, resurrecting: %s",pctx->label,pair->codeword,pair->as_string);
+          /* fall through */
+        case NR_ICE_PAIR_STATE_FROZEN:
+          nr_ice_candidate_pair_set_state(pctx,pair,NR_ICE_PAIR_STATE_WAITING);
+          /* fall through even further */
+        case NR_ICE_PAIR_STATE_WAITING:
+          /* Append it additionally to the trigger check queue */
+          r_log(LOG_ICE,LOG_INFO,"ICE-PEER(%s)/CAND-PAIR(%s): Inserting pair to trigger check queue: %s",pctx->label,pair->codeword,pair->as_string);
+          nr_ice_candidate_pair_trigger_check_append(&pair->remote->stream->trigger_check_queue,pair);
+          break;
+        case NR_ICE_PAIR_STATE_IN_PROGRESS:
+          /* Instead of trying to maintain two stun contexts on the same pair,
+           * and handling heterogenous responses and error conditions, we instead
+           * create a second pair that is identical except that it has the
+           * |triggered| bit set. We also cancel the original pair, but it can
+           * still succeed on its own in the special waiting state. */
+          if(r=nr_ice_candidate_copy_for_triggered_check(pair))
+            ABORT(r);
+          nr_ice_candidate_pair_cancel(pair->pctx,pair,1);
+          break;
+        default:
+          /* all states are handled - a new/unknown state should not
+           * automatically enter the start_checks() below */
+          assert(0);
+          break;
+      }
+
+      /* Ensure that the timers are running to start checks on the topmost entry
+       * of the triggered check queue. */
+      if(r=nr_ice_media_stream_start_checks(pair->pctx,pair->remote->stream))
+        ABORT(r);
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+void nr_ice_candidate_pair_cancel(nr_ice_peer_ctx *pctx,nr_ice_cand_pair *pair, int move_to_wait_state)
+  {
+    if(pair->state != NR_ICE_PAIR_STATE_FAILED){
+      /* If it's already running we need to terminate the stun */
+      if(pair->state==NR_ICE_PAIR_STATE_IN_PROGRESS){
+        if(move_to_wait_state) {
+          nr_stun_client_wait(pair->stun_client);
+        } else {
+          nr_stun_client_cancel(pair->stun_client);
+        }
+      }
+      nr_ice_candidate_pair_set_state(pctx,pair,NR_ICE_PAIR_STATE_CANCELLED);
+    }
+  }
+
+int nr_ice_candidate_pair_select(nr_ice_cand_pair *pair)
+  {
+    int r,_status;
+
+    if(!pair){
+      r_log(LOG_ICE,LOG_ERR,"ICE-PAIR: No pair chosen");
+      ABORT(R_BAD_ARGS);
+    }
+
+    if(pair->state!=NR_ICE_PAIR_STATE_SUCCEEDED){
+      r_log(LOG_ICE,LOG_ERR,"ICE-PEER(%s)/CAND-PAIR(%s): tried to install non-succeeded pair, ignoring: %s",pair->pctx->label,pair->codeword,pair->as_string);
+    }
+    else{
+      /* Ok, they chose one */
+      /* 1. Send a new request with nominated. Do it as a scheduled
+            event to avoid reentrancy issues. Only do this if it hasn't
+            happened already (though this shouldn't happen.)
+      */
+      if(!pair->restart_nominated_cb_timer)
+        NR_ASYNC_TIMER_SET(0,nr_ice_candidate_pair_restart_stun_nominated_cb,pair,&pair->restart_nominated_cb_timer);
+
+      /* 2. Tell ourselves this pair is ready */
+      if(r=nr_ice_component_nominated_pair(pair->remote->component, pair))
+        ABORT(r);
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+ }
+
+void nr_ice_candidate_pair_set_state(nr_ice_peer_ctx *pctx, nr_ice_cand_pair *pair, int state)
+  {
+    r_log(LOG_ICE,LOG_INFO,"ICE-PEER(%s)/CAND-PAIR(%s): setting pair to state %s: %s",
+      pctx->label,pair->codeword,nr_ice_cand_pair_states[state],pair->as_string);
+
+    /* NOTE: This function used to reference pctx->state instead of
+       pair->state and the assignment to pair->state was at the top
+       of this function. Because pctx->state was never changed, this seems to have
+       been a typo. The natural logic is "if the state changed
+       decrement the counter" so this implies we should be checking
+       the pair state rather than the pctx->state.
+
+       This didn't cause big problems because waiting_pairs was only
+       used for pacing, so the pacing just was kind of broken.
+
+       This note is here as a reminder until we do more testing
+       and make sure that in fact this was a typo.
+    */
+    if(pair->state!=NR_ICE_PAIR_STATE_WAITING){
+      if(state==NR_ICE_PAIR_STATE_WAITING)
+        pctx->waiting_pairs++;
+    }
+    else{
+      if(state!=NR_ICE_PAIR_STATE_WAITING)
+        pctx->waiting_pairs--;
+
+      assert(pctx->waiting_pairs>=0);
+    }
+    pair->state=state;
+
+
+    if(pair->state==NR_ICE_PAIR_STATE_FAILED ||
+       pair->state==NR_ICE_PAIR_STATE_CANCELLED){
+      nr_ice_component_failed_pair(pair->remote->component, pair);
+    }
+  }
+
+void nr_ice_candidate_pair_dump_state(nr_ice_cand_pair *pair, int log_level)
+  {
+    r_log(LOG_ICE,log_level,"CAND-PAIR(%s): pair %s: state=%s, priority=0x%llx\n",pair->codeword,pair->as_string,nr_ice_cand_pair_states[pair->state],pair->priority);
+  }
+
+
+int nr_ice_candidate_pair_trigger_check_append(nr_ice_cand_pair_head *head,nr_ice_cand_pair *pair)
+  {
+    if(pair->triggered_check_queue_entry.tqe_next ||
+       pair->triggered_check_queue_entry.tqe_prev)
+      return(0);
+
+    TAILQ_INSERT_TAIL(head,pair,triggered_check_queue_entry);
+
+    return(0);
+  }
+
+void nr_ice_candidate_pair_insert(nr_ice_cand_pair_head *head,nr_ice_cand_pair *pair)
+  {
+    nr_ice_cand_pair *c1;
+
+    c1=TAILQ_FIRST(head);
+    while(c1){
+      if(c1->priority < pair->priority){
+        TAILQ_INSERT_BEFORE(c1,pair,check_queue_entry);
+        break;
+      }
+
+      c1=TAILQ_NEXT(c1,check_queue_entry);
+    }
+    if(!c1) TAILQ_INSERT_TAIL(head,pair,check_queue_entry);
+  }
+
+void nr_ice_candidate_pair_restart_stun_nominated_cb(NR_SOCKET s, int how, void *cb_arg)
+  {
+    nr_ice_cand_pair *pair=cb_arg;
+    int r,_status;
+
+    pair->restart_nominated_cb_timer=0;
+
+    r_log(LOG_ICE,LOG_INFO,"ICE-PEER(%s)/STREAM(%s)/CAND-PAIR(%s)/COMP(%d): Restarting pair as nominated: %s",pair->pctx->label,pair->local->stream->label,pair->codeword,pair->remote->component->component_id,pair->as_string);
+
+    nr_stun_client_reset(pair->stun_client);
+
+    if(r=nr_stun_client_start(pair->stun_client,NR_ICE_CLIENT_MODE_USE_CANDIDATE,nr_ice_candidate_pair_stun_cb,pair))
+      ABORT(r);
+
+    if(r=nr_ice_ctx_remember_id(pair->pctx->ctx, pair->stun_client->request))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    if (_status) {
+      // cb doesn't return anything, but we should probably log that we aborted
+      // This also quiets the unused variable warnings.
+      r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s)/STREAM(%s)/CAND-PAIR(%s)/COMP(%d): STUN nominated cb pair as nominated: %s abort with status: %d",
+        pair->pctx->label,pair->local->stream->label,pair->codeword,pair->remote->component->component_id,pair->as_string, _status);
+    }
+    return;
+  }
+
+static void nr_ice_candidate_pair_restart_stun_role_change_cb(NR_SOCKET s, int how, void *cb_arg)
+ {
+    nr_ice_cand_pair *pair=cb_arg;
+
+    pair->restart_role_change_cb_timer=0;
+
+    r_log(LOG_ICE,LOG_INFO,"ICE-PEER(%s)/STREAM(%s)/CAND-PAIR(%s):COMP(%d): Restarting pair as %s: %s",pair->pctx->label,pair->local->stream->label,pair->codeword,pair->remote->component->component_id,pair->pctx->controlling ? "CONTROLLING" : "CONTROLLED",pair->as_string);
+
+    nr_ice_candidate_pair_restart(pair->pctx, pair);
+  }
+
+void nr_ice_candidate_pair_role_change(nr_ice_cand_pair *pair)
+  {
+    pair->stun_client->params.ice_binding_request.control = pair->pctx->controlling ? NR_ICE_CONTROLLING : NR_ICE_CONTROLLED;
+    nr_ice_candidate_pair_set_priority(pair);
+
+    if(pair->state == NR_ICE_PAIR_STATE_IN_PROGRESS) {
+      /* We could try only restarting in-progress pairs when they receive their
+       * 487, but this ends up being simpler, because any extra 487 are dropped.
+       */
+      if(!pair->restart_role_change_cb_timer)
+        NR_ASYNC_TIMER_SET(0,nr_ice_candidate_pair_restart_stun_role_change_cb,pair,&pair->restart_role_change_cb_timer);
+    }
+  }
+
+static void nr_ice_candidate_pair_compute_codeword(nr_ice_cand_pair *pair,
+  nr_ice_candidate *lcand, nr_ice_candidate *rcand)
+  {
+    char as_string[2048];
+
+    snprintf(as_string,
+             sizeof(as_string),
+             "%s|%s(%s|%s)",
+             lcand->addr.as_string,
+             rcand->addr.as_string,
+             lcand->label,
+             rcand->label);
+
+    nr_ice_compute_codeword(as_string,strlen(as_string),pair->codeword);
+  }
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_candidate_pair.h b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_candidate_pair.h
new file mode 100644
index 0000000000..49da1f7802
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_candidate_pair.h
@@ -0,0 +1,101 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _ice_candidate_pair_h
+#define _ice_candidate_pair_h
+#ifdef __cplusplus
+using namespace std;
+extern "C" {
+#endif /* __cplusplus */
+
+
+struct nr_ice_cand_pair_ {
+  nr_ice_peer_ctx *pctx;
+  char codeword[5];
+  char *as_string;
+  int state;                          /* The current state (S 5.7.3) */
+#define NR_ICE_PAIR_STATE_FROZEN           1
+#define NR_ICE_PAIR_STATE_WAITING          2
+#define NR_ICE_PAIR_STATE_IN_PROGRESS      3
+#define NR_ICE_PAIR_STATE_FAILED           4
+#define NR_ICE_PAIR_STATE_SUCCEEDED        5
+#define NR_ICE_PAIR_STATE_CANCELLED        6
+
+  UCHAR peer_nominated;               /* The peer sent USE-CANDIDATE
+                                         on this check */
+  UCHAR nominated;                    /* Is this nominated or not */
+
+  UCHAR triggered;            /* Ignore further trigger check requests */
+
+  UINT8 priority;                  /* The priority for this pair */
+  nr_ice_candidate *local;            /* The local candidate */
+  nr_ice_candidate *remote;           /* The remote candidate */
+  char *foundation;                   /* The combined foundations */
+
+  // for RTCIceCandidatePairStats
+  UINT8 bytes_sent;
+  UINT8 bytes_recvd;
+  struct timeval last_sent;
+  struct timeval last_recvd;
+
+  nr_stun_client_ctx *stun_client;    /* STUN context when acting as a client */
+  void *stun_client_handle;
+
+  void *stun_cb_timer;
+  void *restart_role_change_cb_timer;
+  void *restart_nominated_cb_timer;
+
+  TAILQ_ENTRY(nr_ice_cand_pair_) check_queue_entry;   /* the check list */
+  TAILQ_ENTRY(nr_ice_cand_pair_) triggered_check_queue_entry;  /* the trigger check queue */
+};
+
+int nr_ice_candidate_pair_create(nr_ice_peer_ctx *pctx, nr_ice_candidate *lcand,nr_ice_candidate *rcand,nr_ice_cand_pair **pairp);
+int nr_ice_candidate_pair_unfreeze(nr_ice_peer_ctx *pctx, nr_ice_cand_pair *pair);
+int nr_ice_candidate_pair_start(nr_ice_peer_ctx *pctx, nr_ice_cand_pair *pair);
+void nr_ice_candidate_pair_set_state(nr_ice_peer_ctx *pctx, nr_ice_cand_pair *pair, int state);
+void nr_ice_candidate_pair_dump_state(nr_ice_cand_pair *pair, int log_level);
+void nr_ice_candidate_pair_cancel(nr_ice_peer_ctx *pctx,nr_ice_cand_pair *pair, int move_to_wait_state);
+int nr_ice_candidate_pair_select(nr_ice_cand_pair *pair);
+int nr_ice_candidate_pair_do_triggered_check(nr_ice_peer_ctx *pctx, nr_ice_cand_pair *pair);
+void nr_ice_candidate_pair_insert(nr_ice_cand_pair_head *head,nr_ice_cand_pair *pair);
+int nr_ice_candidate_pair_trigger_check_append(nr_ice_cand_pair_head *head,nr_ice_cand_pair *pair);
+void nr_ice_candidate_pair_restart_stun_nominated_cb(NR_SOCKET s, int how, void *cb_arg);
+int nr_ice_candidate_pair_destroy(nr_ice_cand_pair **pairp);
+void nr_ice_candidate_pair_role_change(nr_ice_cand_pair *pair);
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_codeword.h b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_codeword.h
new file mode 100644
index 0000000000..76fce0b69b
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_codeword.h
@@ -0,0 +1,41 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _ice_codeword_h
+#define _ice_codeword_h
+
+void nr_ice_compute_codeword(char *buf, int len,char *codeword);
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_component.c b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_component.c
new file mode 100644
index 0000000000..584a9466bc
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_component.c
@@ -0,0 +1,1786 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <string.h>
+#include <assert.h>
+#include <nr_api.h>
+#include <registry.h>
+#include <async_timer.h>
+#include "ice_ctx.h"
+#include "ice_codeword.h"
+#include "stun.h"
+#include "nr_socket_local.h"
+#include "nr_socket_turn.h"
+#include "nr_socket_wrapper.h"
+#include "nr_socket_buffered_stun.h"
+#include "nr_socket_multi_tcp.h"
+#include "ice_reg.h"
+#include "nr_crypto.h"
+#include "r_time.h"
+
+static void nr_ice_component_refresh_consent_cb(NR_SOCKET s, int how, void *cb_arg);
+static int nr_ice_component_stun_server_default_cb(void *cb_arg,nr_stun_server_ctx *stun_ctx,nr_socket *sock, nr_stun_server_request *req, int *dont_free, int *error);
+static int nr_ice_pre_answer_request_destroy(nr_ice_pre_answer_request **parp);
+int nr_ice_component_can_candidate_addr_pair(nr_transport_addr *local, nr_transport_addr *remote);
+int nr_ice_component_can_candidate_tcptype_pair(nr_socket_tcp_type left, nr_socket_tcp_type right);
+void nr_ice_component_consent_calc_consent_timer(nr_ice_component *comp);
+void nr_ice_component_consent_schedule_consent_timer(nr_ice_component *comp);
+int nr_ice_component_refresh_consent(nr_stun_client_ctx *ctx, NR_async_cb finished_cb, void *cb_arg);
+int nr_ice_component_setup_consent(nr_ice_component *comp);
+int nr_ice_pre_answer_enqueue(nr_ice_component *comp, nr_socket *sock, nr_stun_server_request *req, int *dont_free);
+
+/* This function takes ownership of the contents of req (but not req itself) */
+static int nr_ice_pre_answer_request_create(nr_transport_addr *dst, nr_stun_server_request *req, nr_ice_pre_answer_request **parp)
+  {
+    int r, _status;
+    nr_ice_pre_answer_request *par = 0;
+    nr_stun_message_attribute *attr;
+
+    if (!(par = RCALLOC(sizeof(nr_ice_pre_answer_request))))
+      ABORT(R_NO_MEMORY);
+
+    par->req = *req; /* Struct assignment */
+    memset(req, 0, sizeof(*req)); /* Zero contents to avoid confusion */
+
+    if (r=nr_transport_addr_copy(&par->local_addr, dst))
+      ABORT(r);
+    if (!nr_stun_message_has_attribute(par->req.request, NR_STUN_ATTR_USERNAME, &attr))
+      ABORT(R_INTERNAL);
+    if (!(par->username = r_strdup(attr->u.username)))
+      ABORT(R_NO_MEMORY);
+
+    *parp=par;
+    _status=0;
+ abort:
+    if (_status) {
+      /* Erase the request so we don't free it */
+      memset(&par->req, 0, sizeof(nr_stun_server_request));
+      nr_ice_pre_answer_request_destroy(&par);
+    }
+
+    return(_status);
+  }
+
+static int nr_ice_pre_answer_request_destroy(nr_ice_pre_answer_request **parp)
+  {
+    nr_ice_pre_answer_request *par;
+
+    if (!parp || !*parp)
+      return(0);
+
+    par = *parp;
+    *parp = 0;
+
+    nr_stun_message_destroy(&par->req.request);
+    nr_stun_message_destroy(&par->req.response);
+
+    RFREE(par->username);
+    RFREE(par);
+
+    return(0);
+  }
+
+int nr_ice_component_create(nr_ice_media_stream *stream, int component_id, nr_ice_component **componentp)
+  {
+    int _status;
+    nr_ice_component *comp=0;
+
+    if(!(comp=RCALLOC(sizeof(nr_ice_component))))
+      ABORT(R_NO_MEMORY);
+
+    comp->state=NR_ICE_COMPONENT_UNPAIRED;
+    comp->component_id=component_id;
+    comp->stream=stream;
+    comp->ctx=stream->ctx;
+
+    STAILQ_INIT(&comp->sockets);
+    TAILQ_INIT(&comp->candidates);
+    STAILQ_INIT(&comp->pre_answer_reqs);
+
+    STAILQ_INSERT_TAIL(&stream->components,comp,entry);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_ice_component_destroy(nr_ice_component **componentp)
+  {
+    nr_ice_component *component;
+    nr_ice_socket *s1,*s2;
+    nr_ice_candidate *c1,*c2;
+    nr_ice_pre_answer_request *r1,*r2;
+
+    if(!componentp || !*componentp)
+      return(0);
+
+    component=*componentp;
+    *componentp=0;
+
+    nr_ice_component_consent_destroy(component);
+
+    /* Detach ourselves from the sockets */
+    if (component->local_component){
+      nr_ice_socket *isock=STAILQ_FIRST(&component->local_component->sockets);
+      while(isock){
+        nr_stun_server_remove_client(isock->stun_server, component);
+        isock=STAILQ_NEXT(isock, entry);
+      }
+    }
+
+    /* candidates MUST be destroyed before the sockets so that
+       they can deregister */
+    TAILQ_FOREACH_SAFE(c1, &component->candidates, entry_comp, c2){
+      TAILQ_REMOVE(&component->candidates,c1,entry_comp);
+      nr_ice_candidate_destroy(&c1);
+    }
+
+    STAILQ_FOREACH_SAFE(s1, &component->sockets, entry, s2){
+      STAILQ_REMOVE(&component->sockets,s1,nr_ice_socket_,entry);
+      nr_ice_socket_destroy(&s1);
+    }
+
+    STAILQ_FOREACH_SAFE(r1, &component->pre_answer_reqs, entry, r2){
+      STAILQ_REMOVE(&component->pre_answer_reqs,r1,nr_ice_pre_answer_request_, entry);
+      nr_ice_pre_answer_request_destroy(&r1);
+    }
+
+    RFREE(component);
+    return(0);
+  }
+
+static int nr_ice_component_create_stun_server_ctx(nr_ice_component *component, nr_ice_socket *isock, nr_transport_addr *addr, char *lufrag, Data *pwd)
+  {
+    char label[256];
+    int r,_status;
+
+    /* Create a STUN server context for this socket */
+    snprintf(label, sizeof(label), "server(%s)", addr->as_string);
+    if(r=nr_stun_server_ctx_create(label,&isock->stun_server))
+      ABORT(r);
+
+   /* Add the default STUN credentials so that we can respond before
+      we hear about the peer.*/
+    if(r=nr_stun_server_add_default_client(isock->stun_server, lufrag, pwd, nr_ice_component_stun_server_default_cb, component))
+      ABORT(r);
+
+    /* Do this last; if this function fails, we should not be taking a reference to isock */
+    if(r=nr_ice_socket_register_stun_server(isock,isock->stun_server,&isock->stun_server_handle))
+      ABORT(r);
+
+    _status = 0;
+ abort:
+    return(_status);
+  }
+
+static int nr_ice_component_initialize_udp(struct nr_ice_ctx_ *ctx,nr_ice_component *component, nr_local_addr *addrs, int addr_ct, char *lufrag, Data *pwd)
+  {
+    nr_socket *sock;
+    nr_ice_socket *isock=0;
+    nr_ice_candidate *cand=0;
+    int i;
+    int j;
+    int r,_status;
+
+    if(ctx->flags & NR_ICE_CTX_FLAGS_ONLY_PROXY) {
+      /* No UDP support if we must use a proxy */
+      return 0;
+    }
+
+    /* Now one ice_socket for each address */
+    for(i=0;i<addr_ct;i++){
+      char suppress;
+
+      if(r=NR_reg_get2_char(NR_ICE_REG_SUPPRESS_INTERFACE_PRFX,addrs[i].addr.ifname,&suppress)){
+        if(r!=R_NOT_FOUND)
+          ABORT(r);
+      }
+      else{
+        if(suppress)
+          continue;
+      }
+      r_log(LOG_ICE,LOG_DEBUG,"ICE-STREAM(%s): host address %s",component->stream->label,addrs[i].addr.as_string);
+      if((r=nr_socket_factory_create_socket(ctx->socket_factory,&addrs[i].addr,&sock))){
+        r_log(LOG_ICE,LOG_WARNING,"ICE-STREAM(%s): couldn't create socket for address %s",component->stream->label,addrs[i].addr.as_string);
+        continue;
+      }
+
+      if(r=nr_ice_socket_create(ctx,component,sock,NR_ICE_SOCKET_TYPE_DGRAM,&isock))
+        ABORT(r);
+
+      /* Create a STUN server context for this socket */
+      if ((r=nr_ice_component_create_stun_server_ctx(component,isock,&addrs[i].addr,lufrag,pwd)))
+        ABORT(r);
+
+      /* Make sure we don't leak this. Failures might result in it being
+       * unused, but we hand off references to this in enough places below
+       * that unwinding it all becomes impractical. */
+      STAILQ_INSERT_TAIL(&component->sockets,isock,entry);
+
+      if (!(component->stream->flags & NR_ICE_CTX_FLAGS_RELAY_ONLY)) {
+        /* Create one host candidate */
+        if(r=nr_ice_candidate_create(ctx,component,isock,sock,HOST,0,0,
+          component->component_id,&cand))
+          ABORT(r);
+
+        TAILQ_INSERT_TAIL(&component->candidates,cand,entry_comp);
+        component->candidate_ct++;
+        cand=0;
+
+        /* And a srvrflx candidate for each STUN server */
+        for(j=0;j<component->stream->stun_server_ct;j++){
+          r_log(LOG_ICE,LOG_DEBUG,"ICE-STREAM(%s): Checking STUN server %s %s", component->stream->label, component->stream->stun_servers[j].addr.fqdn, component->stream->stun_servers[j].addr.as_string);
+          /* Skip non-UDP */
+          if (component->stream->stun_servers[j].addr.protocol != IPPROTO_UDP) continue;
+
+          if (nr_transport_addr_check_compatibility(
+                  &addrs[i].addr, &component->stream->stun_servers[j].addr)) {
+            r_log(LOG_ICE,LOG_INFO,"ICE-STREAM(%s): Skipping STUN server because of address type mis-match",component->stream->label);
+            continue;
+          }
+
+          /* Ensure id is set (nr_ice_ctx_set_stun_servers does not) */
+          component->stream->stun_servers[j].id = j;
+          if(r=nr_ice_candidate_create(ctx,component,
+            isock,sock,SERVER_REFLEXIVE,0,
+            &component->stream->stun_servers[j],component->component_id,&cand))
+            ABORT(r);
+          TAILQ_INSERT_TAIL(&component->candidates,cand,entry_comp);
+          component->candidate_ct++;
+          cand=0;
+        }
+      }
+      else{
+        r_log(LOG_ICE,LOG_WARNING,"ICE-STREAM(%s): relay only option results in no host candidate for %s",component->stream->label,addrs[i].addr.as_string);
+      }
+
+#ifdef USE_TURN
+      if ((component->stream->flags & NR_ICE_CTX_FLAGS_RELAY_ONLY) &&
+          (component->stream->turn_server_ct == 0)) {
+        r_log(LOG_ICE,LOG_ERR,"ICE-STREAM(%s): relay only option is set without any TURN server configured",component->stream->label);
+      }
+      /* And both a srvrflx and relayed candidate for each TURN server (unless
+         we're in relay-only mode, in which case just the relayed one) */
+      for(j=0;j<component->stream->turn_server_ct;j++){
+        nr_socket *turn_sock;
+        nr_ice_candidate *srvflx_cand=0;
+
+        r_log(LOG_ICE,LOG_DEBUG,"ICE-STREAM(%s): Checking TURN server %s %s", component->stream->label, component->stream->turn_servers[j].turn_server.addr.fqdn, component->stream->turn_servers[j].turn_server.addr.as_string);
+
+        /* Skip non-UDP */
+        if (component->stream->turn_servers[j].turn_server.addr.protocol != IPPROTO_UDP)
+          continue;
+
+        if (nr_transport_addr_check_compatibility(
+                &addrs[i].addr, &component->stream->turn_servers[j].turn_server.addr)) {
+          r_log(LOG_ICE,LOG_INFO,"ICE-STREAM(%s): Skipping TURN server because of address type mis-match",component->stream->label);
+          continue;
+        }
+
+        if (!(component->stream->flags & NR_ICE_CTX_FLAGS_RELAY_ONLY)) {
+          /* Ensure id is set with a unique value */
+          component->stream->turn_servers[j].turn_server.id = j + component->stream->stun_server_ct;
+          /* srvrflx */
+          if(r=nr_ice_candidate_create(ctx,component,
+            isock,sock,SERVER_REFLEXIVE,0,
+            &component->stream->turn_servers[j].turn_server,component->component_id,&cand))
+            ABORT(r);
+          cand->state=NR_ICE_CAND_STATE_INITIALIZING; /* Don't start */
+          cand->done_cb=nr_ice_gather_finished_cb;
+          cand->cb_arg=cand;
+
+          TAILQ_INSERT_TAIL(&component->candidates,cand,entry_comp);
+          component->candidate_ct++;
+          srvflx_cand=cand;
+          cand=0;
+        }
+        /* relayed*/
+        if(r=nr_socket_turn_create(&turn_sock))
+          ABORT(r);
+        if(r=nr_ice_candidate_create(ctx,component,
+          isock,turn_sock,RELAYED,0,
+          &component->stream->turn_servers[j].turn_server,component->component_id,&cand))
+           ABORT(r);
+        if (srvflx_cand) {
+          cand->u.relayed.srvflx_candidate=srvflx_cand;
+          srvflx_cand->u.srvrflx.relay_candidate=cand;
+        }
+        cand->u.relayed.server=&component->stream->turn_servers[j];
+        TAILQ_INSERT_TAIL(&component->candidates,cand,entry_comp);
+        component->candidate_ct++;
+
+        cand=0;
+      }
+#endif /* USE_TURN */
+    }
+
+    _status = 0;
+ abort:
+    return(_status);
+  }
+
+static int nr_ice_component_get_port_from_ephemeral_range(uint16_t *port)
+  {
+    int _status, r;
+    void *buf = port;
+    if(r=nr_crypto_random_bytes(buf, 2))
+      ABORT(r);
+    *port|=49152; /* make it fit into IANA ephemeral port range >= 49152 */
+    _status=0;
+abort:
+    return(_status);
+  }
+
+static int nr_ice_component_create_tcp_host_candidate(struct nr_ice_ctx_ *ctx,
+  nr_ice_component *component, nr_transport_addr *interface_addr, nr_socket_tcp_type tcp_type,
+  int backlog, int so_sock_ct, char *lufrag, Data *pwd, nr_ice_socket **isock)
+  {
+    int r,_status;
+    nr_ice_candidate *cand=0;
+    int tries=3;
+    nr_ice_socket *isock_tmp=0;
+    nr_socket *nrsock=0;
+    nr_transport_addr addr;
+    uint16_t local_port;
+
+    if ((r=nr_transport_addr_copy(&addr,interface_addr)))
+      ABORT(r);
+    addr.protocol=IPPROTO_TCP;
+
+    do{
+      if (!tries--)
+        ABORT(r);
+
+      if((r=nr_ice_component_get_port_from_ephemeral_range(&local_port)))
+        ABORT(r);
+
+      if ((r=nr_transport_addr_set_port(&addr, local_port)))
+        ABORT(r);
+
+      if((r=nr_transport_addr_fmt_addr_string(&addr)))
+        ABORT(r);
+
+      /* It would be better to stop trying if there is error other than
+         port already used, but it'd require significant work to support this. */
+      r=nr_socket_multi_tcp_create(ctx,component,&addr,tcp_type,so_sock_ct,NR_STUN_MAX_MESSAGE_SIZE,&nrsock);
+
+    } while(r);
+
+    if((tcp_type == TCP_TYPE_PASSIVE) && (r=nr_socket_listen(nrsock,backlog)))
+      ABORT(r);
+
+    if((r=nr_ice_socket_create(ctx,component,nrsock,NR_ICE_SOCKET_TYPE_STREAM_TCP,&isock_tmp)))
+      ABORT(r);
+
+    /* nr_ice_socket took ownership of nrsock */
+    nrsock=NULL;
+
+    /* Create a STUN server context for this socket */
+    if ((r=nr_ice_component_create_stun_server_ctx(component,isock_tmp,&addr,lufrag,pwd)))
+      ABORT(r);
+
+    if((r=nr_ice_candidate_create(ctx,component,isock_tmp,isock_tmp->sock,HOST,tcp_type,0,
+      component->component_id,&cand)))
+      ABORT(r);
+
+    if (isock)
+      *isock=isock_tmp;
+
+    TAILQ_INSERT_TAIL(&component->candidates,cand,entry_comp);
+    component->candidate_ct++;
+
+    STAILQ_INSERT_TAIL(&component->sockets,isock_tmp,entry);
+
+    _status=0;
+abort:
+    if (_status) {
+      nr_ice_socket_destroy(&isock_tmp);
+      nr_socket_destroy(&nrsock);
+    }
+    return(_status);
+  }
+
+static int nr_ice_component_initialize_tcp(struct nr_ice_ctx_ *ctx,nr_ice_component *component, nr_local_addr *addrs, int addr_ct, char *lufrag, Data *pwd)
+  {
+    nr_ice_candidate *cand=0;
+    int i;
+    int j;
+    int r,_status;
+    int so_sock_ct=0;
+    int backlog=10;
+    char ice_tcp_disabled=1;
+
+    r_log(LOG_ICE,LOG_DEBUG,"nr_ice_component_initialize_tcp");
+
+    if(r=NR_reg_get_int4(NR_ICE_REG_ICE_TCP_SO_SOCK_COUNT,&so_sock_ct)){
+      if(r!=R_NOT_FOUND)
+        ABORT(r);
+    }
+
+    if(r=NR_reg_get_int4(NR_ICE_REG_ICE_TCP_LISTEN_BACKLOG,&backlog)){
+      if(r!=R_NOT_FOUND)
+        ABORT(r);
+    }
+
+    if ((r=NR_reg_get_char(NR_ICE_REG_ICE_TCP_DISABLE, &ice_tcp_disabled))) {
+      if (r != R_NOT_FOUND)
+        ABORT(r);
+    }
+    if ((component->stream->flags & NR_ICE_CTX_FLAGS_RELAY_ONLY) ||
+        (component->stream->flags & NR_ICE_CTX_FLAGS_ONLY_PROXY)) {
+      r_log(LOG_ICE,LOG_WARNING,"ICE-STREAM(%s): relay/proxy only option results in ICE TCP being disabled",component->stream->label);
+      ice_tcp_disabled = 1;
+    }
+
+    for(i=0;i<addr_ct;i++){
+      char suppress;
+      nr_ice_socket *isock_psv=0;
+      nr_ice_socket *isock_so=0;
+
+      if(r=NR_reg_get2_char(NR_ICE_REG_SUPPRESS_INTERFACE_PRFX,addrs[i].addr.ifname,&suppress)){
+        if(r!=R_NOT_FOUND)
+          ABORT(r);
+      }
+      else if(suppress) {
+          continue;
+      }
+
+      if (!ice_tcp_disabled) {
+        /* passive host candidate */
+        if ((r=nr_ice_component_create_tcp_host_candidate(ctx, component, &addrs[i].addr,
+          TCP_TYPE_PASSIVE, backlog, 0, lufrag, pwd, &isock_psv))) {
+          r_log(LOG_ICE,LOG_WARNING,"ICE-STREAM(%s): failed to create passive TCP host candidate: %d",component->stream->label,r);
+        }
+
+        /* active host candidate */
+        if ((r=nr_ice_component_create_tcp_host_candidate(ctx, component, &addrs[i].addr,
+          TCP_TYPE_ACTIVE, 0, 0, lufrag, pwd, NULL))) {
+          r_log(LOG_ICE,LOG_WARNING,"ICE-STREAM(%s): failed to create active TCP host candidate: %d",component->stream->label,r);
+        }
+
+        /* simultaneous-open host candidate */
+        if (so_sock_ct) {
+          if ((r=nr_ice_component_create_tcp_host_candidate(ctx, component, &addrs[i].addr,
+            TCP_TYPE_SO, 0, so_sock_ct, lufrag, pwd, &isock_so))) {
+            r_log(LOG_ICE,LOG_WARNING,"ICE-STREAM(%s): failed to create simultanous open TCP host candidate: %d",component->stream->label,r);
+          }
+        }
+
+        /* And srvrflx candidates for each STUN server */
+        for(j=0;j<component->stream->stun_server_ct;j++){
+          if (component->stream->stun_servers[j].addr.protocol != IPPROTO_TCP) continue;
+
+          if (isock_psv) {
+            if(r=nr_ice_candidate_create(ctx,component,
+              isock_psv,isock_psv->sock,SERVER_REFLEXIVE,TCP_TYPE_PASSIVE,
+              &component->stream->stun_servers[j],component->component_id,&cand))
+              ABORT(r);
+            TAILQ_INSERT_TAIL(&component->candidates,cand,entry_comp);
+            component->candidate_ct++;
+            cand=0;
+          }
+
+          if (isock_so) {
+            if(r=nr_ice_candidate_create(ctx,component,
+              isock_so,isock_so->sock,SERVER_REFLEXIVE,TCP_TYPE_SO,
+              &component->stream->stun_servers[j],component->component_id,&cand))
+              ABORT(r);
+            TAILQ_INSERT_TAIL(&component->candidates,cand,entry_comp);
+            component->candidate_ct++;
+            cand=0;
+          }
+        }
+      }
+
+#ifdef USE_TURN
+      /* Create a new relayed candidate for each addr/TURN server pair */
+      for(j=0;j<component->stream->turn_server_ct;j++){
+        nr_transport_addr addr;
+        nr_socket *local_sock;
+        nr_socket *buffered_sock;
+        nr_socket *turn_sock;
+        nr_ice_socket *turn_isock;
+
+        r_log(LOG_ICE,LOG_DEBUG,"ICE-STREAM(%s): Checking TURN server %s %s", component->stream->label, component->stream->turn_servers[j].turn_server.addr.fqdn, component->stream->turn_servers[j].turn_server.addr.as_string);
+
+        /* Skip non-TCP */
+        if (component->stream->turn_servers[j].turn_server.addr.protocol != IPPROTO_TCP)
+          continue;
+
+        /* Create relay candidate */
+        if ((r=nr_transport_addr_copy(&addr, &addrs[i].addr)))
+          ABORT(r);
+        addr.protocol = IPPROTO_TCP;
+
+        if (nr_transport_addr_check_compatibility(
+                &addr, &component->stream->turn_servers[j].turn_server.addr)) {
+          r_log(LOG_ICE,LOG_INFO,"ICE-STREAM(%s): Skipping TURN server because of address type mis-match",component->stream->label);
+          continue;
+        }
+
+        if (!ice_tcp_disabled) {
+          /* Use TURN server to get srflx candidates */
+          if (isock_psv) {
+            if(r=nr_ice_candidate_create(ctx,component,
+              isock_psv,isock_psv->sock,SERVER_REFLEXIVE,TCP_TYPE_PASSIVE,
+              &component->stream->turn_servers[j].turn_server,component->component_id,&cand))
+              ABORT(r);
+            TAILQ_INSERT_TAIL(&component->candidates,cand,entry_comp);
+            component->candidate_ct++;
+            cand=0;
+          }
+
+          if (isock_so) {
+            if(r=nr_ice_candidate_create(ctx,component,
+              isock_so,isock_so->sock,SERVER_REFLEXIVE,TCP_TYPE_SO,
+              &component->stream->turn_servers[j].turn_server,component->component_id,&cand))
+              ABORT(r);
+            TAILQ_INSERT_TAIL(&component->candidates,cand,entry_comp);
+            component->candidate_ct++;
+            cand=0;
+          }
+        }
+
+        if (component->stream->turn_servers[j].turn_server.addr.fqdn[0] != 0) {
+          /* If we're going to use TLS, make sure that's recorded */
+          addr.tls = component->stream->turn_servers[j].turn_server.addr.tls;
+        }
+
+        if ((r=nr_transport_addr_fmt_addr_string(&addr)))
+          ABORT(r);
+
+        r_log(LOG_ICE, LOG_DEBUG,
+              "ICE-STREAM(%s): Creating socket for address %s (turn server %s)",
+              component->stream->label, addr.as_string,
+              component->stream->turn_servers[j].turn_server.addr.as_string);
+
+        /* Create a local socket */
+        if((r=nr_socket_factory_create_socket(ctx->socket_factory,&addr,&local_sock))){
+          r_log(LOG_ICE,LOG_DEBUG,"ICE-STREAM(%s): couldn't create socket for address %s",component->stream->label,addr.as_string);
+          continue;
+        }
+
+        r_log(LOG_ICE,LOG_DEBUG,"nr_ice_component_initialize_tcp creating TURN TCP wrappers");
+
+        /* The TCP buffered socket */
+        if((r=nr_socket_buffered_stun_create(local_sock, NR_STUN_MAX_MESSAGE_SIZE, TURN_TCP_FRAMING, &buffered_sock)))
+          ABORT(r);
+
+        /* The TURN socket */
+        if(r=nr_socket_turn_create(&turn_sock))
+          ABORT(r);
+
+        /* Create an ICE socket */
+        if((r=nr_ice_socket_create(ctx, component, buffered_sock, NR_ICE_SOCKET_TYPE_STREAM_TURN, &turn_isock)))
+          ABORT(r);
+
+
+        /* Create a STUN server context for this socket */
+        if ((r=nr_ice_component_create_stun_server_ctx(component,turn_isock,&addr,lufrag,pwd)))
+          ABORT(r);
+
+        /* Make sure we don't leak this. Failures might result in it being
+         * unused, but we hand off references to this in enough places below
+         * that unwinding it all becomes impractical. */
+        STAILQ_INSERT_TAIL(&component->sockets,turn_isock,entry);
+
+        /* Attach ourselves to it */
+        if(r=nr_ice_candidate_create(ctx,component,
+          turn_isock,turn_sock,RELAYED,TCP_TYPE_NONE,
+          &component->stream->turn_servers[j].turn_server,component->component_id,&cand))
+          ABORT(r);
+        cand->u.relayed.srvflx_candidate=NULL;
+        cand->u.relayed.server=&component->stream->turn_servers[j];
+        TAILQ_INSERT_TAIL(&component->candidates,cand,entry_comp);
+        component->candidate_ct++;
+        cand=0;
+      }
+#endif /* USE_TURN */
+    }
+
+    _status = 0;
+ abort:
+    return(_status);
+  }
+
+
+/* Make all the candidates we can make at the beginning */
+int nr_ice_component_initialize(struct nr_ice_ctx_ *ctx,nr_ice_component *component)
+  {
+    int r,_status;
+    nr_local_addr *addrs=ctx->local_addrs;
+    int addr_ct=ctx->local_addr_ct;
+    char *lufrag;
+    char *lpwd;
+    Data pwd;
+    nr_ice_candidate *cand;
+
+    if (component->candidate_ct) {
+      r_log(LOG_ICE,LOG_DEBUG,"ICE(%s): component with id %d already has candidates, probably restarting gathering because of a new stream",ctx->label,component->component_id);
+      return(0);
+    }
+
+    r_log(LOG_ICE,LOG_DEBUG,"ICE(%s): initializing component with id %d",ctx->label,component->component_id);
+
+    if(addr_ct==0){
+      r_log(LOG_ICE,LOG_ERR,"ICE(%s): no local addresses available",ctx->label);
+      ABORT(R_NOT_FOUND);
+    }
+
+    /* Note: we need to recompute these because
+       we have not yet computed the values in the peer media stream.*/
+    lufrag=component->stream->ufrag;
+    assert(lufrag);
+    if (!lufrag)
+      ABORT(R_INTERNAL);
+    lpwd=component->stream->pwd;
+    assert(lpwd);
+    if (!lpwd)
+      ABORT(R_INTERNAL);
+    INIT_DATA(pwd, (UCHAR *)lpwd, strlen(lpwd));
+
+    /* Initialize the UDP candidates */
+    if (r=nr_ice_component_initialize_udp(ctx, component, addrs, addr_ct, lufrag, &pwd))
+      r_log(LOG_ICE,LOG_INFO,"ICE(%s): failed to create UDP candidates with error %d",ctx->label,r);
+    /* And the TCP candidates */
+    if (r=nr_ice_component_initialize_tcp(ctx, component, addrs, addr_ct, lufrag, &pwd))
+      r_log(LOG_ICE,LOG_INFO,"ICE(%s): failed to create TCP candidates with error %d",ctx->label,r);
+
+    /* count the candidates that will be initialized */
+    cand=TAILQ_FIRST(&component->candidates);
+    if(!cand){
+      r_log(LOG_ICE,LOG_ERR,"ICE(%s): couldn't create any valid candidates",ctx->label);
+      ABORT(R_NOT_FOUND);
+    }
+
+    while(cand){
+      ctx->uninitialized_candidates++;
+      cand=TAILQ_NEXT(cand,entry_comp);
+    }
+
+    /* Now initialize all the candidates */
+    cand=TAILQ_FIRST(&component->candidates);
+    while(cand){
+      if(cand->state!=NR_ICE_CAND_STATE_INITIALIZING){
+        nr_ice_candidate_initialize(cand,nr_ice_gather_finished_cb,cand);
+      }
+      cand=TAILQ_NEXT(cand,entry_comp);
+    }
+    _status=0;
+ abort:
+    return(_status);
+  }
+
+void nr_ice_component_stop_gathering(nr_ice_component *component)
+  {
+    nr_ice_candidate *c1,*c2;
+    TAILQ_FOREACH_SAFE(c1, &component->candidates, entry_comp, c2){
+      nr_ice_candidate_stop_gathering(c1);
+    }
+  }
+
+int nr_ice_component_is_done_gathering(nr_ice_component *comp)
+  {
+    nr_ice_candidate *cand=TAILQ_FIRST(&comp->candidates);
+    while(cand){
+      if(cand->state != NR_ICE_CAND_STATE_INITIALIZED &&
+         cand->state != NR_ICE_CAND_STATE_FAILED){
+        return 0;
+      }
+      cand=TAILQ_NEXT(cand,entry_comp);
+    }
+    return 1;
+  }
+
+
+static int nr_ice_any_peer_paired(nr_ice_candidate* cand) {
+  nr_ice_peer_ctx* pctx=STAILQ_FIRST(&cand->ctx->peers);
+  while(pctx && pctx->state == NR_ICE_PEER_STATE_UNPAIRED){
+    /* Is it worth actually looking through the check lists? Probably not. */
+    pctx=STAILQ_NEXT(pctx,entry);
+  }
+  return pctx != NULL;
+}
+
+/*
+  Compare this newly initialized candidate against the other initialized
+  candidates and discard the lower-priority one if they are redundant.
+
+   This algorithm combined with the other algorithms, favors
+   host > srflx > relay
+ */
+int nr_ice_component_maybe_prune_candidate(nr_ice_ctx *ctx, nr_ice_component *comp, nr_ice_candidate *c1, int *was_pruned)
+  {
+    nr_ice_candidate *c2, *tmp = NULL;
+
+    *was_pruned = 0;
+    c2 = TAILQ_FIRST(&comp->candidates);
+    while(c2){
+      if((c1 != c2) &&
+         (c2->state == NR_ICE_CAND_STATE_INITIALIZED) &&
+         !nr_transport_addr_cmp(&c1->base,&c2->base,NR_TRANSPORT_ADDR_CMP_MODE_ALL) &&
+         !nr_transport_addr_cmp(&c1->addr,&c2->addr,NR_TRANSPORT_ADDR_CMP_MODE_ALL)){
+
+        if((c1->type == c2->type) ||
+           (!(ctx->flags & NR_ICE_CTX_FLAGS_DISABLE_HOST_CANDIDATES) &&
+            !(ctx->flags & NR_ICE_CTX_FLAGS_OBFUSCATE_HOST_ADDRESSES) &&
+            ((c1->type==HOST && c2->type == SERVER_REFLEXIVE) ||
+             (c2->type==HOST && c1->type == SERVER_REFLEXIVE)))){
+
+          /*
+             These are redundant. Remove the lower pri one, or if pairing has
+             already occurred, remove the newest one.
+
+             Since this algorithmis run whenever a new candidate
+             is initialized, there should at most one duplicate.
+           */
+          if ((c1->priority <= c2->priority) || nr_ice_any_peer_paired(c2)) {
+            tmp = c1;
+            *was_pruned = 1;
+          }
+          else {
+            tmp = c2;
+          }
+          break;
+        }
+      }
+
+      c2=TAILQ_NEXT(c2,entry_comp);
+    }
+
+    if (tmp) {
+      r_log(LOG_ICE,LOG_DEBUG,"ICE(%s)/CAND(%s): Removing redundant candidate",
+            ctx->label,tmp->label);
+
+      TAILQ_REMOVE(&comp->candidates,tmp,entry_comp);
+      comp->candidate_ct--;
+      TAILQ_REMOVE(&tmp->isock->candidates,tmp,entry_sock);
+
+      nr_ice_candidate_destroy(&tmp);
+    }
+
+    return 0;
+  }
+
+static int nr_ice_component_pair_matches_check(nr_ice_component *comp, nr_ice_cand_pair *pair, nr_transport_addr *local_addr, nr_stun_server_request *req)
+  {
+    if(pair->remote->component->component_id!=comp->component_id)
+      return(0);
+
+    if(nr_transport_addr_cmp(&pair->local->base,local_addr,NR_TRANSPORT_ADDR_CMP_MODE_ALL))
+      return(0);
+
+    if(nr_transport_addr_cmp(&pair->remote->addr,&req->src_addr,NR_TRANSPORT_ADDR_CMP_MODE_ALL))
+      return(0);
+
+    return(1);
+  }
+
+static int nr_ice_component_handle_triggered_check(nr_ice_component *comp, nr_ice_cand_pair *pair, nr_stun_server_request *req, int *error)
+  {
+    nr_stun_message *sreq=req->request;
+    int r=0,_status;
+
+    if(nr_stun_message_has_attribute(sreq,NR_STUN_ATTR_USE_CANDIDATE,0)){
+      if(comp->stream->pctx->controlling){
+        r_log(LOG_ICE,LOG_WARNING,"ICE-PEER(%s)/CAND_PAIR(%s): Peer sent USE-CANDIDATE but is controlled",comp->stream->pctx->label, pair->codeword);
+      }
+      else{
+        /* If this is the first time we've noticed this is nominated...*/
+        pair->peer_nominated=1;
+
+        if(pair->state==NR_ICE_PAIR_STATE_SUCCEEDED && !pair->nominated){
+          pair->nominated=1;
+
+          if(r=nr_ice_component_nominated_pair(pair->remote->component, pair)) {
+            *error=(r==R_NO_MEMORY)?500:400;
+            ABORT(r);
+          }
+        }
+      }
+    }
+
+    /* Note: the RFC says to trigger first and then nominate. But in that case
+     * the canceled trigger pair would get nominated and the cloned trigger pair
+     * would not get the nomination status cloned with it.*/
+    if(r=nr_ice_candidate_pair_do_triggered_check(comp->stream->pctx,pair)) {
+      *error=(r==R_NO_MEMORY)?500:400;
+      ABORT(r);
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+/* Section 7.2.1 */
+static int nr_ice_component_process_incoming_check(nr_ice_component *comp, nr_transport_addr *local_addr, nr_stun_server_request *req, int *error)
+  {
+    nr_ice_cand_pair *pair;
+    nr_ice_candidate *pcand=0;
+    nr_stun_message *sreq=req->request;
+    nr_stun_message_attribute *attr;
+    int r=0,_status;
+    int found_valid=0;
+
+    r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s)/STREAM(%s)/COMP(%d): received request from %s",comp->stream->pctx->label,comp->stream->label,comp->component_id,req->src_addr.as_string);
+
+    if (comp->state == NR_ICE_COMPONENT_DISABLED)
+      ABORT(R_REJECTED);
+
+    /* Check for role conficts (7.2.1.1) */
+    if(comp->stream->pctx->controlling){
+      if(nr_stun_message_has_attribute(sreq,NR_STUN_ATTR_ICE_CONTROLLING,&attr)){
+        /* OK, there is a conflict. Who's right? */
+        r_log(LOG_ICE,LOG_INFO,"ICE-PEER(%s): role conflict, both controlling",comp->stream->pctx->label);
+
+        if(attr->u.ice_controlling > comp->stream->pctx->tiebreaker){
+          /* Update the peer ctx. This will propagate to all candidate pairs
+             in the context. */
+          nr_ice_peer_ctx_switch_controlling_role(comp->stream->pctx);
+        }
+        else {
+          /* We are: throw an error */
+          r_log(LOG_ICE,LOG_WARNING,"ICE-PEER(%s): returning 487 role conflict",comp->stream->pctx->label);
+
+          *error=487;
+          ABORT(R_REJECTED);
+        }
+      }
+    }
+    else{
+      if(nr_stun_message_has_attribute(sreq,NR_STUN_ATTR_ICE_CONTROLLED,&attr)){
+        /* OK, there is a conflict. Who's right? */
+        r_log(LOG_ICE,LOG_INFO,"ICE-PEER(%s): role conflict, both controlled",comp->stream->pctx->label);
+
+        if(attr->u.ice_controlled < comp->stream->pctx->tiebreaker){
+          /* Update the peer ctx. This will propagate to all candidate pairs
+             in the context. */
+          nr_ice_peer_ctx_switch_controlling_role(comp->stream->pctx);
+        }
+        else {
+          /* We are: throw an error */
+          r_log(LOG_ICE,LOG_WARNING,"ICE-PEER(%s): returning 487 role conflict",comp->stream->pctx->label);
+
+          *error=487;
+          ABORT(R_REJECTED);
+        }
+      }
+    }
+
+    r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s): This STUN request appears to map to local addr %s",comp->stream->pctx->label,local_addr->as_string);
+
+    pair=TAILQ_FIRST(&comp->stream->check_list);
+    while(pair){
+      /* Since triggered checks create duplicate pairs (in this implementation)
+       * we are willing to handle multiple matches here. */
+      if(nr_ice_component_pair_matches_check(comp, pair, local_addr, req)){
+        r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s)/CAND_PAIR(%s): Found a matching pair for received check: %s",comp->stream->pctx->label,pair->codeword,pair->as_string);
+        if(r=nr_ice_component_handle_triggered_check(comp, pair, req, error))
+          ABORT(r);
+        ++found_valid;
+      }
+      pair=TAILQ_NEXT(pair,check_queue_entry);
+    }
+
+    if(!found_valid){
+      /* There were no matching pairs, so we need to create a new peer
+       * reflexive candidate pair. */
+
+      if(!nr_stun_message_has_attribute(sreq,NR_STUN_ATTR_PRIORITY,&attr)){
+        r_log(LOG_ICE,LOG_WARNING,"ICE-PEER(%s): Rejecting stun request without priority",comp->stream->pctx->label);
+        *error=400;
+        ABORT(R_BAD_DATA);
+      }
+
+      /* Find our local component candidate */
+      nr_ice_candidate *cand;
+
+      r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s): no matching pair",comp->stream->pctx->label);
+      cand=TAILQ_FIRST(&comp->local_component->candidates);
+      while(cand){
+        if(!nr_transport_addr_cmp(&cand->addr,local_addr,NR_TRANSPORT_ADDR_CMP_MODE_ALL))
+          break;
+
+        cand=TAILQ_NEXT(cand,entry_comp);
+      }
+
+      /* Well, this really shouldn't happen, but it's an error from the
+         other side, so we just throw an error and keep going */
+      if(!cand){
+        r_log(LOG_ICE,LOG_WARNING,"ICE-PEER(%s): stun request to unknown local address %s, discarding",comp->stream->pctx->label,local_addr->as_string);
+
+        *error=400;
+        ABORT(R_NOT_FOUND);
+      }
+
+      /* Now make a peer reflexive (remote) candidate */
+      if(r=nr_ice_peer_peer_rflx_candidate_create(comp->stream->pctx->ctx,"prflx",comp,&req->src_addr,&pcand)) {
+        *error=(r==R_NO_MEMORY)?500:400;
+        ABORT(r);
+      }
+      pcand->priority=attr->u.priority;
+      pcand->state=NR_ICE_CAND_PEER_CANDIDATE_PAIRED;
+
+      /* Finally, create the candidate pair, insert into the check list, and
+       * apply the incoming check to it. */
+      if(r=nr_ice_candidate_pair_create(comp->stream->pctx,cand,pcand,
+           &pair)) {
+        *error=(r==R_NO_MEMORY)?500:400;
+        ABORT(r);
+      }
+
+      nr_ice_candidate_pair_set_state(pair->pctx,pair,NR_ICE_PAIR_STATE_FROZEN);
+      if(r=nr_ice_component_insert_pair(comp,pair)) {
+        *error=(r==R_NO_MEMORY)?500:400;
+        ABORT(r);
+      }
+
+      /* Do this last, since any call to ABORT will destroy pcand */
+      TAILQ_INSERT_TAIL(&comp->candidates,pcand,entry_comp);
+      pcand=0;
+
+      /* Finally start the trigger check if needed */
+      if(r=nr_ice_component_handle_triggered_check(comp, pair, req, error))
+        ABORT(r);
+    }
+
+    _status=0;
+  abort:
+    if(_status){
+      nr_ice_candidate_destroy(&pcand);
+      assert(*error != 0);
+      if(r!=R_NO_MEMORY) assert(*error != 500);
+    }
+    return(_status);
+  }
+
+static int nr_ice_component_stun_server_cb(void *cb_arg,nr_stun_server_ctx *stun_ctx,nr_socket *sock, nr_stun_server_request *req, int *dont_free, int *error)
+  {
+    nr_ice_component *pcomp=cb_arg;
+    nr_transport_addr local_addr;
+    int r,_status;
+
+    if(pcomp->state==NR_ICE_COMPONENT_FAILED) {
+      *error=400;
+      ABORT(R_REJECTED);
+    }
+
+    if (pcomp->local_component->stream->obsolete) {
+      /* Don't do any triggered check stuff in thiis case. */
+      return 0;
+    }
+
+    /* Find the candidate pair that this maps to */
+    if(r=nr_socket_getaddr(sock,&local_addr)) {
+      *error=500;
+      ABORT(r);
+    }
+
+    if (r=nr_ice_component_process_incoming_check(pcomp, &local_addr, req, error))
+      ABORT(r);
+
+    _status=0;
+ abort:
+    return(_status);
+  }
+
+int nr_ice_component_service_pre_answer_requests(nr_ice_peer_ctx *pctx, nr_ice_component *pcomp, char *username, int *serviced)
+  {
+    nr_ice_pre_answer_request *r1,*r2;
+    nr_ice_component *comp = pcomp->local_component;
+    int r,_status;
+
+    if (serviced)
+      *serviced = 0;
+
+    r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s)/STREAM(%s)/COMP(%d): looking for pre-answer requests",pctx->label,comp->stream->label,comp->component_id);
+
+    STAILQ_FOREACH_SAFE(r1, &comp->pre_answer_reqs, entry, r2) {
+      if (!strcmp(r1->username, username)) {
+        int error = 0;
+
+        r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s)/STREAM(%s)/COMP(%d): found pre-answer request",pctx->label,comp->stream->label,comp->component_id);
+        r = nr_ice_component_process_incoming_check(pcomp, &r1->local_addr, &r1->req, &error);
+        if (r) {
+          r_log(LOG_ICE,LOG_INFO,"ICE-PEER(%s)/STREAM(%s)/COMP(%d): error processing pre-answer request. Would have returned %d",pctx->label,comp->stream->label,comp->component_id, error);
+        }
+        (*serviced)++;
+        STAILQ_REMOVE(&comp->pre_answer_reqs,r1,nr_ice_pre_answer_request_, entry);
+        nr_ice_pre_answer_request_destroy(&r1);
+      }
+    }
+
+    _status=0;
+     return(_status);
+  }
+
+int nr_ice_component_can_candidate_tcptype_pair(nr_socket_tcp_type left, nr_socket_tcp_type right)
+  {
+    if (left && !right)
+      return(0);
+    if (!left && right)
+      return(0);
+    if (left == TCP_TYPE_ACTIVE && right != TCP_TYPE_PASSIVE)
+      return(0);
+    if (left == TCP_TYPE_SO && right != TCP_TYPE_SO)
+      return(0);
+    if (left == TCP_TYPE_PASSIVE)
+      return(0);
+
+    return(1);
+  }
+
+/* filter out pairings which won't work. */
+int nr_ice_component_can_candidate_addr_pair(nr_transport_addr *local, nr_transport_addr *remote)
+  {
+    if(local->ip_version != remote->ip_version)
+      return(0);
+    if(local->protocol != remote->protocol)
+      return(0);
+    if(nr_transport_addr_is_link_local(local) !=
+       nr_transport_addr_is_link_local(remote))
+      return(0);
+    /* This prevents our ice_unittest (or broken clients) from pairing a
+     * loopback with a host candidate. */
+    if(nr_transport_addr_is_loopback(local) !=
+       nr_transport_addr_is_loopback(remote))
+      return(0);
+
+    return(1);
+  }
+
+int nr_ice_component_pair_candidate(nr_ice_peer_ctx *pctx, nr_ice_component *pcomp, nr_ice_candidate *lcand, int pair_all_remote)
+  {
+    int r, _status;
+    nr_ice_candidate *pcand;
+    nr_ice_cand_pair *pair=0;
+    char codeword[5];
+
+    nr_ice_compute_codeword(lcand->label,strlen(lcand->label),codeword);
+    r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s)/CAND(%s): Pairing local candidate %s",pctx->label,codeword,lcand->label);
+
+    switch(lcand->type){
+      case HOST:
+        break;
+      case SERVER_REFLEXIVE:
+      case PEER_REFLEXIVE:
+        /* Don't actually pair these candidates */
+        goto done;
+        break;
+      case RELAYED:
+        break;
+      default:
+        assert(0);
+        ABORT(R_INTERNAL);
+        break;
+    }
+
+    TAILQ_FOREACH(pcand, &pcomp->candidates, entry_comp){
+      if(!nr_ice_component_can_candidate_addr_pair(&lcand->addr, &pcand->addr))
+        continue;
+      if(!nr_ice_component_can_candidate_tcptype_pair(lcand->tcp_type, pcand->tcp_type))
+        continue;
+
+      /* https://tools.ietf.org/html/draft-ietf-rtcweb-mdns-ice-candidates-03#section-3.3.2 */
+      if(lcand->type == RELAYED && pcand->mdns_addr && strlen(pcand->mdns_addr)) {
+        continue;
+      }
+
+      /*
+        Two modes, depending on |pair_all_remote|
+
+        1. Pair remote candidates which have not been paired
+           (used in initial pairing or in processing the other side's
+           trickle candidates).
+        2. Pair any remote candidate (used when processing our own
+           trickle candidates).
+      */
+      if (pair_all_remote || (pcand->state == NR_ICE_CAND_PEER_CANDIDATE_UNPAIRED)) {
+        if (pair_all_remote) {
+          /* When a remote candidate arrives after the start of checking, but
+           * before the gathering of local candidates, it can be in UNPAIRED */
+          pcand->state = NR_ICE_CAND_PEER_CANDIDATE_PAIRED;
+        }
+
+        nr_ice_compute_codeword(pcand->label,strlen(pcand->label),codeword);
+        r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s)/CAND(%s): Pairing with peer candidate %s", pctx->label, codeword, pcand->label);
+
+        if(r=nr_ice_candidate_pair_create(pctx,lcand,pcand,&pair))
+          ABORT(r);
+
+        if(r=nr_ice_component_insert_pair(pcomp, pair))
+          ABORT(r);
+      }
+    }
+
+   done:
+    _status = 0;
+   abort:
+    return(_status);
+  }
+
+int nr_ice_component_pair_candidates(nr_ice_peer_ctx *pctx, nr_ice_component *lcomp,nr_ice_component *pcomp)
+  {
+    nr_ice_candidate *lcand, *pcand;
+    nr_ice_socket *isock;
+    int r,_status;
+
+    r_log(LOG_ICE,LOG_DEBUG,"Pairing candidates======");
+
+    /* Create the candidate pairs */
+    lcand=TAILQ_FIRST(&lcomp->candidates);
+
+    if (!lcand) {
+      /* No local candidates, initialized or not! */
+      ABORT(R_FAILED);
+    }
+
+    while(lcand){
+      if (lcand->state == NR_ICE_CAND_STATE_INITIALIZED) {
+        if ((r = nr_ice_component_pair_candidate(pctx, pcomp, lcand, 0)))
+          ABORT(r);
+      }
+
+      lcand=TAILQ_NEXT(lcand,entry_comp);
+    }
+
+    /* Mark all peer candidates as paired */
+    pcand=TAILQ_FIRST(&pcomp->candidates);
+    while(pcand){
+      pcand->state = NR_ICE_CAND_PEER_CANDIDATE_PAIRED;
+
+      pcand=TAILQ_NEXT(pcand,entry_comp);
+
+    }
+
+    /* Now register the STUN server callback for this component.
+       Note that this is a per-component CB so we only need to
+       do this once.
+    */
+    if (pcomp->state != NR_ICE_COMPONENT_RUNNING) {
+      isock=STAILQ_FIRST(&lcomp->sockets);
+      while(isock){
+        if(r=nr_stun_server_add_client(isock->stun_server,pctx->label,
+          pcomp->stream->r2l_user,&pcomp->stream->r2l_pass,nr_ice_component_stun_server_cb,pcomp)) {
+            ABORT(r);
+        }
+        isock=STAILQ_NEXT(isock,entry);
+      }
+    }
+
+    pcomp->state = NR_ICE_COMPONENT_RUNNING;
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_ice_pre_answer_enqueue(nr_ice_component *comp, nr_socket *sock, nr_stun_server_request *req, int *dont_free)
+  {
+    int r = 0;
+    int _status;
+    nr_ice_pre_answer_request *r1, *r2;
+    nr_transport_addr dst_addr;
+    nr_ice_pre_answer_request *par = 0;
+
+    if (r=nr_socket_getaddr(sock, &dst_addr))
+      ABORT(r);
+
+    STAILQ_FOREACH_SAFE(r1, &comp->pre_answer_reqs, entry, r2) {
+      if (!nr_transport_addr_cmp(&r1->local_addr, &dst_addr,
+                                 NR_TRANSPORT_ADDR_CMP_MODE_ALL) &&
+          !nr_transport_addr_cmp(&r1->req.src_addr, &req->src_addr,
+                                 NR_TRANSPORT_ADDR_CMP_MODE_ALL)) {
+        return(0);
+      }
+    }
+
+    if (r=nr_ice_pre_answer_request_create(&dst_addr, req, &par))
+      ABORT(r);
+
+    r_log(LOG_ICE,LOG_DEBUG, "ICE(%s)/STREAM(%s)/COMP(%d): Enqueuing STUN request pre-answer from %s",
+          comp->ctx->label, comp->stream->label, comp->component_id,
+          req->src_addr.as_string);
+
+    *dont_free = 1;
+    STAILQ_INSERT_TAIL(&comp->pre_answer_reqs, par, entry);
+
+    _status=0;
+abort:
+    return(_status);
+  }
+
+/* Fires when we have an incoming candidate that doesn't correspond to an existing
+   remote peer. This is either pre-answer or just spurious. Store it in the
+   component for use when we see the actual answer, at which point we need
+   to do the procedures from S 7.2.1 in nr_ice_component_stun_server_cb.
+ */
+static int nr_ice_component_stun_server_default_cb(void *cb_arg,nr_stun_server_ctx *stun_ctx,nr_socket *sock, nr_stun_server_request *req, int *dont_free, int *error)
+  {
+    int r, _status;
+    nr_ice_component *comp = (nr_ice_component *)cb_arg;
+
+    r_log(LOG_ICE,LOG_DEBUG,"ICE(%s)/STREAM(%s)/COMP(%d): Received STUN request pre-answer from %s",
+          comp->ctx->label, comp->stream->label, comp->component_id,
+          req->src_addr.as_string);
+
+    if (r=nr_ice_pre_answer_enqueue(comp, sock, req, dont_free)) {
+      r_log(LOG_ICE,LOG_ERR,"ICE(%s)/STREAM(%s)/COMP(%d): Failed (%d) to enque pre-answer request from %s",
+          comp->ctx->label, comp->stream->label, comp->component_id, r,
+          req->src_addr.as_string);
+      ABORT(r);
+    }
+
+    _status=0;
+ abort:
+    return(_status);
+  }
+
+#define NR_ICE_CONSENT_TIMER_DEFAULT 5000
+#define NR_ICE_CONSENT_TIMEOUT_DEFAULT 30000
+
+static void nr_ice_component_consent_failed(nr_ice_component *comp)
+  {
+    if (!comp->can_send) {
+      return;
+    }
+
+    r_log(LOG_ICE,LOG_INFO,"ICE(%s)/STREAM(%s)/COMP(%d): Consent refresh failed",
+          comp->ctx->label, comp->stream->label, comp->component_id);
+    comp->can_send = 0;
+
+    if (comp->consent_timeout) {
+      NR_async_timer_cancel(comp->consent_timeout);
+      comp->consent_timeout = 0;
+    }
+    if (comp->consent_timer) {
+      NR_async_timer_cancel(comp->consent_timer);
+      comp->consent_timer = 0;
+    }
+    /* We are turning the consent failure into a ICE component failure to
+     * alert the browser via ICE connection state change about this event. */
+    nr_ice_media_stream_component_failed(comp->stream, comp);
+  }
+
+static void nr_ice_component_consent_timeout_cb(NR_SOCKET s, int how, void *cb_arg)
+  {
+    nr_ice_component *comp=cb_arg;
+
+    comp->consent_timeout = 0;
+
+    r_log(LOG_ICE,LOG_WARNING,"ICE(%s)/STREAM(%s)/COMP(%d): Consent refresh final time out",
+          comp->ctx->label, comp->stream->label, comp->component_id);
+    nr_ice_component_consent_failed(comp);
+  }
+
+
+void nr_ice_component_disconnected(nr_ice_component *comp)
+  {
+    if (!comp->can_send) {
+      return;
+    }
+
+    if (comp->disconnected) {
+      return;
+    }
+
+    r_log(LOG_ICE,LOG_WARNING,"ICE(%s)/STREAM(%s)/COMP(%d): component disconnected",
+          comp->ctx->label, comp->stream->label, comp->component_id);
+    comp->disconnected = 1;
+
+    /* a single disconnected component disconnects the stream */
+    nr_ice_media_stream_set_disconnected(comp->stream, NR_ICE_MEDIA_STREAM_DISCONNECTED);
+  }
+
+static void nr_ice_component_consent_refreshed(nr_ice_component *comp)
+  {
+    uint16_t tval;
+
+    if (!comp->can_send) {
+      return;
+    }
+
+    gettimeofday(&comp->consent_last_seen, 0);
+    r_log(LOG_ICE,LOG_DEBUG,"ICE(%s)/STREAM(%s)/COMP(%d): consent_last_seen is now %lu",
+        comp->ctx->label, comp->stream->label, comp->component_id,
+        comp->consent_last_seen.tv_sec);
+
+    comp->disconnected = 0;
+
+    nr_ice_media_stream_check_if_connected(comp->stream);
+
+    if (comp->consent_timeout)
+      NR_async_timer_cancel(comp->consent_timeout);
+
+    tval = NR_ICE_CONSENT_TIMEOUT_DEFAULT;
+    if (comp->ctx->test_timer_divider)
+      tval = tval / comp->ctx->test_timer_divider;
+
+    NR_ASYNC_TIMER_SET(tval, nr_ice_component_consent_timeout_cb, comp,
+                       &comp->consent_timeout);
+  }
+
+static void nr_ice_component_refresh_consent_cb(NR_SOCKET s, int how, void *cb_arg)
+  {
+    nr_ice_component *comp=cb_arg;
+
+    switch (comp->consent_ctx->state) {
+      case NR_STUN_CLIENT_STATE_FAILED:
+        if (comp->consent_ctx->error_code == 403) {
+          r_log(LOG_ICE, LOG_INFO, "ICE(%s)/STREAM(%s)/COMP(%d): Consent revoked by peer",
+                comp->ctx->label, comp->stream->label, comp->component_id);
+          nr_ice_component_consent_failed(comp);
+        }
+        break;
+      case NR_STUN_CLIENT_STATE_DONE:
+        r_log(LOG_ICE, LOG_INFO, "ICE(%s)/STREAM(%s)/COMP(%d): Consent refreshed",
+              comp->ctx->label, comp->stream->label, comp->component_id);
+        nr_ice_component_consent_refreshed(comp);
+        break;
+      case NR_STUN_CLIENT_STATE_TIMED_OUT:
+        r_log(LOG_ICE, LOG_INFO, "ICE(%s)/STREAM(%s)/COMP(%d): A single consent refresh request timed out",
+              comp->ctx->label, comp->stream->label, comp->component_id);
+        nr_ice_component_disconnected(comp);
+        break;
+      default:
+        break;
+    }
+  }
+
+int nr_ice_component_refresh_consent(nr_stun_client_ctx *ctx, NR_async_cb finished_cb, void *cb_arg)
+  {
+    int r,_status;
+
+    nr_stun_client_reset(ctx);
+
+    if (r=nr_stun_client_start(ctx, NR_ICE_CLIENT_MODE_BINDING_REQUEST, finished_cb, cb_arg))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+void nr_ice_component_consent_calc_consent_timer(nr_ice_component *comp)
+  {
+    uint16_t trange, trand, tval;
+
+    trange = NR_ICE_CONSENT_TIMER_DEFAULT * 20 / 100;
+    tval = NR_ICE_CONSENT_TIMER_DEFAULT - trange;
+    if (!nr_crypto_random_bytes((UCHAR*)&trand, sizeof(trand)))
+      tval += (trand % (trange * 2));
+
+    if (comp->ctx->test_timer_divider)
+      tval = tval / comp->ctx->test_timer_divider;
+
+    /* The timeout of the transaction is the maximum time until we send the
+     * next consent request. */
+    comp->consent_ctx->maximum_transmits_timeout_ms = tval;
+  }
+
+static void nr_ice_component_consent_timer_cb(NR_SOCKET s, int how, void *cb_arg)
+  {
+    nr_ice_component *comp=cb_arg;
+    int r;
+
+    if (!comp->consent_ctx) {
+      return;
+    }
+
+    if (comp->consent_timer) {
+      NR_async_timer_cancel(comp->consent_timer);
+    }
+    comp->consent_timer = 0;
+
+    comp->consent_ctx->params.ice_binding_request.username =
+      comp->stream->l2r_user;
+    comp->consent_ctx->params.ice_binding_request.password =
+      comp->stream->l2r_pass;
+    comp->consent_ctx->params.ice_binding_request.control =
+      comp->stream->pctx->controlling?
+      NR_ICE_CONTROLLING:NR_ICE_CONTROLLED;
+    comp->consent_ctx->params.ice_binding_request.tiebreaker =
+      comp->stream->pctx->tiebreaker;
+    comp->consent_ctx->params.ice_binding_request.priority =
+      comp->active->local->priority;
+
+    nr_ice_component_consent_calc_consent_timer(comp);
+
+    if (r=nr_ice_component_refresh_consent(comp->consent_ctx,
+                                           nr_ice_component_refresh_consent_cb,
+                                           comp)) {
+      r_log(LOG_ICE,LOG_ERR,"ICE(%s)/STREAM(%s)/COMP(%d): Refresh consent failed with %d",
+            comp->ctx->label, comp->stream->label, comp->component_id, r);
+    }
+
+    nr_ice_component_consent_schedule_consent_timer(comp);
+
+  }
+
+void nr_ice_component_consent_schedule_consent_timer(nr_ice_component *comp)
+  {
+    if (!comp->can_send) {
+      return;
+    }
+
+    NR_ASYNC_TIMER_SET(comp->consent_ctx->maximum_transmits_timeout_ms,
+                       nr_ice_component_consent_timer_cb, comp,
+                       &comp->consent_timer);
+  }
+
+void nr_ice_component_refresh_consent_now(nr_ice_component *comp)
+  {
+    nr_ice_component_consent_timer_cb(0, 0, comp);
+  }
+
+void nr_ice_component_consent_destroy(nr_ice_component *comp)
+  {
+    if (comp->consent_timer) {
+      NR_async_timer_cancel(comp->consent_timer);
+      comp->consent_timer = 0;
+    }
+    if (comp->consent_timeout) {
+      NR_async_timer_cancel(comp->consent_timeout);
+      comp->consent_timeout = 0;
+    }
+    if (comp->consent_handle) {
+      nr_ice_socket_deregister(comp->active->local->isock,
+                               comp->consent_handle);
+      comp->consent_handle = 0;
+    }
+    if (comp->consent_ctx) {
+      nr_stun_client_ctx_destroy(&comp->consent_ctx);
+      comp->consent_ctx = 0;
+    }
+  }
+
+int nr_ice_component_setup_consent(nr_ice_component *comp)
+  {
+    int r,_status;
+
+    r_log(LOG_ICE,LOG_DEBUG,"ICE(%s)/STREAM(%s)/COMP(%d): Setting up refresh consent",
+          comp->ctx->label, comp->stream->label, comp->component_id);
+
+    nr_ice_component_consent_destroy(comp);
+
+    if (r=nr_stun_client_ctx_create("consent", comp->active->local->osock,
+                                    &comp->active->remote->addr, 0,
+                                    &comp->consent_ctx))
+      ABORT(r);
+    /* Consent request get send only once. */
+    comp->consent_ctx->maximum_transmits = 1;
+
+    if (r=nr_ice_socket_register_stun_client(comp->active->local->isock,
+            comp->consent_ctx, &comp->consent_handle))
+      ABORT(r);
+
+    comp->can_send = 1;
+    comp->disconnected = 0;
+    nr_ice_component_consent_refreshed(comp);
+
+    nr_ice_component_consent_calc_consent_timer(comp);
+    nr_ice_component_consent_schedule_consent_timer(comp);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_ice_component_nominated_pair(nr_ice_component *comp, nr_ice_cand_pair *pair)
+  {
+    int r,_status;
+    nr_ice_cand_pair *p2;
+
+    /* Are we changing what the nominated pair is? */
+    if(comp->nominated){
+      if(comp->nominated->priority >= pair->priority)
+        return(0);
+      r_log(LOG_ICE,LOG_INFO,"ICE-PEER(%s)/STREAM(%s)/COMP(%d)/CAND-PAIR(%s): replacing pair %s with CAND-PAIR(%s)",comp->stream->pctx->label,comp->stream->label,comp->component_id,comp->nominated->codeword,comp->nominated->as_string,pair->codeword);
+      /* As consent doesn't hold a reference to its isock this needs to happen
+       * before making the new pair the active one. */
+      nr_ice_component_consent_destroy(comp);
+    }
+
+    /* Set the new nominated pair */
+    r_log(LOG_ICE,LOG_INFO,"ICE-PEER(%s)/STREAM(%s)/COMP(%d)/CAND-PAIR(%s): nominated pair is %s",comp->stream->pctx->label,comp->stream->label,comp->component_id,pair->codeword,pair->as_string);
+    comp->state=NR_ICE_COMPONENT_NOMINATED;
+    comp->nominated=pair;
+    comp->active=pair;
+
+    r_log(LOG_ICE,LOG_INFO,"ICE-PEER(%s)/STREAM(%s)/COMP(%d)/CAND-PAIR(%s): cancelling all pairs but %s",comp->stream->pctx->label,comp->stream->label,comp->component_id,pair->codeword,pair->as_string);
+
+    /* Cancel checks in WAITING and FROZEN per ICE S 8.1.2 */
+    p2=TAILQ_FIRST(&comp->stream->trigger_check_queue);
+    while(p2){
+      if((p2 != pair) &&
+         (p2->remote->component->component_id == comp->component_id)) {
+        assert(p2->state == NR_ICE_PAIR_STATE_WAITING ||
+               p2->state == NR_ICE_PAIR_STATE_CANCELLED);
+        r_log(LOG_ICE,LOG_INFO,"ICE-PEER(%s)/STREAM(%s)/COMP(%d)/CAND-PAIR(%s): cancelling FROZEN/WAITING pair %s in trigger check queue because CAND-PAIR(%s) was nominated.",comp->stream->pctx->label,comp->stream->label,comp->component_id,p2->codeword,p2->as_string,pair->codeword);
+
+        nr_ice_candidate_pair_cancel(pair->pctx,p2,0);
+      }
+
+      p2=TAILQ_NEXT(p2,triggered_check_queue_entry);
+    }
+    p2=TAILQ_FIRST(&comp->stream->check_list);
+    while(p2){
+      if((p2 != pair) &&
+         (p2->remote->component->component_id == comp->component_id) &&
+         ((p2->state == NR_ICE_PAIR_STATE_FROZEN) ||
+          (p2->state == NR_ICE_PAIR_STATE_WAITING))) {
+        r_log(LOG_ICE,LOG_INFO,"ICE-PEER(%s)/STREAM(%s)/COMP(%d)/CAND-PAIR(%s): cancelling FROZEN/WAITING pair %s because CAND-PAIR(%s) was nominated.",comp->stream->pctx->label,comp->stream->label,comp->component_id,p2->codeword,p2->as_string,pair->codeword);
+
+        nr_ice_candidate_pair_cancel(pair->pctx,p2,0);
+      }
+
+      p2=TAILQ_NEXT(p2,check_queue_entry);
+    }
+    r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s)/STREAM(%s)/COMP(%d): cancelling done",comp->stream->pctx->label,comp->stream->label,comp->component_id);
+
+    if(r=nr_ice_component_setup_consent(comp))
+      ABORT(r);
+
+    nr_ice_media_stream_component_nominated(comp->stream,comp);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+static int nr_ice_component_have_all_pairs_failed(nr_ice_component *comp)
+  {
+    nr_ice_cand_pair *p2;
+
+    p2=TAILQ_FIRST(&comp->stream->check_list);
+    while(p2){
+      if(comp->component_id==p2->local->component_id){
+        switch(p2->state){
+        case NR_ICE_PAIR_STATE_FROZEN:
+        case NR_ICE_PAIR_STATE_WAITING:
+        case NR_ICE_PAIR_STATE_IN_PROGRESS:
+        case NR_ICE_PAIR_STATE_SUCCEEDED:
+            return(0);
+        case NR_ICE_PAIR_STATE_FAILED:
+        case NR_ICE_PAIR_STATE_CANCELLED:
+            /* states that will never be recovered from */
+            break;
+        default:
+            assert(0);
+            break;
+        }
+      }
+
+      p2=TAILQ_NEXT(p2,check_queue_entry);
+    }
+
+    return(1);
+  }
+
+void nr_ice_component_failed_pair(nr_ice_component *comp, nr_ice_cand_pair *pair)
+  {
+    nr_ice_component_check_if_failed(comp);
+  }
+
+void nr_ice_component_check_if_failed(nr_ice_component *comp)
+  {
+    if (comp->state == NR_ICE_COMPONENT_RUNNING) {
+      /* Don't do anything to streams that aren't currently running */
+      r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s)/STREAM(%s)/COMP(%d): Checking whether component needs to be marked failed.",comp->stream->pctx->label,comp->stream->label,comp->component_id);
+
+      if (!comp->stream->pctx->trickle_grace_period_timer &&
+          nr_ice_component_have_all_pairs_failed(comp)) {
+        r_log(LOG_ICE,LOG_INFO,"ICE-PEER(%s)/STREAM(%s)/COMP(%d): All pairs are failed, and grace period has elapsed. Marking component as failed.",comp->stream->pctx->label,comp->stream->label,comp->component_id);
+        nr_ice_media_stream_component_failed(comp->stream,comp);
+      }
+    }
+  }
+
+int nr_ice_component_select_pair(nr_ice_peer_ctx *pctx, nr_ice_component *comp)
+  {
+    nr_ice_cand_pair **pairs=0;
+    int ct=0;
+    nr_ice_cand_pair *pair;
+    int r,_status;
+
+    /* Size the array */
+    pair=TAILQ_FIRST(&comp->stream->check_list);
+    while(pair){
+      if (comp->component_id == pair->local->component_id)
+          ct++;
+
+      pair=TAILQ_NEXT(pair,check_queue_entry);
+    }
+
+    /* Make and fill the array */
+    if(!(pairs=RCALLOC(sizeof(nr_ice_cand_pair *)*ct)))
+      ABORT(R_NO_MEMORY);
+
+    ct=0;
+    pair=TAILQ_FIRST(&comp->stream->check_list);
+    while(pair){
+      if (comp->component_id == pair->local->component_id)
+          pairs[ct++]=pair;
+
+      pair=TAILQ_NEXT(pair,check_queue_entry);
+    }
+
+    if (pctx->handler) {
+      if(r=pctx->handler->vtbl->select_pair(pctx->handler->obj,
+        comp->stream,comp->component_id,pairs,ct))
+        ABORT(r);
+    }
+
+    _status=0;
+  abort:
+    RFREE(pairs);
+    return(_status);
+  }
+
+
+/* Close the underlying sockets for everything but the nominated candidate */
+int nr_ice_component_finalize(nr_ice_component *lcomp, nr_ice_component *rcomp)
+  {
+    nr_ice_socket *isock=0;
+    nr_ice_socket *s1,*s2;
+
+    if(rcomp->state==NR_ICE_COMPONENT_NOMINATED){
+      assert(rcomp->active == rcomp->nominated);
+      isock=rcomp->nominated->local->isock;
+    }
+
+    STAILQ_FOREACH_SAFE(s1, &lcomp->sockets, entry, s2){
+      if(s1!=isock){
+        STAILQ_REMOVE(&lcomp->sockets,s1,nr_ice_socket_,entry);
+        nr_ice_socket_destroy(&s1);
+      }
+    }
+
+    return(0);
+  }
+
+
+int nr_ice_component_insert_pair(nr_ice_component *pcomp, nr_ice_cand_pair *pair)
+  {
+    int _status;
+
+    /* Pairs for peer reflexive are marked SUCCEEDED immediately */
+    if (pair->state != NR_ICE_PAIR_STATE_FROZEN &&
+        pair->state != NR_ICE_PAIR_STATE_SUCCEEDED){
+      assert(0);
+      ABORT(R_BAD_ARGS);
+    }
+
+    /* We do not throw an error after this, because we've inserted the pair. */
+    nr_ice_candidate_pair_insert(&pair->remote->stream->check_list,pair);
+
+    /* Make sure the check timer is running, if the stream was previously
+     * started. We will not start streams just because a pair was created,
+     * unless it is the first pair to be created across all streams. */
+    r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s)/CAND-PAIR(%s): Ensure that check timer is running for new pair %s.",pair->remote->stream->pctx->label, pair->codeword, pair->as_string);
+
+    if(pair->remote->stream->ice_state == NR_ICE_MEDIA_STREAM_CHECKS_ACTIVE ||
+       (pair->remote->stream->ice_state == NR_ICE_MEDIA_STREAM_CHECKS_FROZEN &&
+        !pair->remote->stream->pctx->checks_started)){
+      if(nr_ice_media_stream_start_checks(pair->remote->stream->pctx, pair->remote->stream)) {
+        r_log(LOG_ICE,LOG_WARNING,"ICE-PEER(%s)/CAND-PAIR(%s): Could not restart checks for new pair %s.",pair->remote->stream->pctx->label, pair->codeword, pair->as_string);
+      }
+    }
+
+    _status=0;
+  abort:
+    if (_status) {
+      nr_ice_candidate_pair_destroy(&pair);
+    }
+    return(_status);
+  }
+
+int nr_ice_component_get_default_candidate(nr_ice_component *comp, nr_ice_candidate **candp, int ip_version)
+  {
+    int _status;
+    nr_ice_candidate *cand;
+    nr_ice_candidate *best_cand = NULL;
+
+    /* We have the component. Now find the "best" candidate, making
+       use of the fact that more "reliable" candidate types have
+       higher numbers. So, we sort by type and then priority within
+       type
+    */
+    cand=TAILQ_FIRST(&comp->candidates);
+    while(cand){
+      if (!nr_ice_ctx_hide_candidate(comp->ctx, cand) &&
+          cand->addr.ip_version == ip_version) {
+        if (!best_cand) {
+          best_cand = cand;
+        }
+        else if (best_cand->type < cand->type) {
+          best_cand = cand;
+        } else if (best_cand->type == cand->type &&
+                   best_cand->priority < cand->priority) {
+          best_cand = cand;
+        }
+      }
+
+      cand=TAILQ_NEXT(cand,entry_comp);
+    }
+
+    /* No candidates */
+    if (!best_cand)
+      ABORT(R_NOT_FOUND);
+
+    *candp = best_cand;
+
+    _status=0;
+  abort:
+    return(_status);
+
+  }
+
+
+void nr_ice_component_dump_state(nr_ice_component *comp, int log_level)
+  {
+    nr_ice_candidate *cand;
+
+    if (comp->local_component) {
+      r_log(LOG_ICE,log_level,"ICE(%s)/ICE-STREAM(%s): Remote component %d in state %d - dumping candidates",comp->ctx->label,comp->stream->label,comp->component_id,comp->state);
+    } else {
+      r_log(LOG_ICE,log_level,"ICE(%s)/ICE-STREAM(%s): Local component %d - dumping candidates",comp->ctx->label,comp->stream->label,comp->component_id);
+    }
+
+    cand=TAILQ_FIRST(&comp->candidates);
+    while(cand){
+      r_log(LOG_ICE,log_level,"ICE(%s)/ICE-STREAM(%s)/CAND(%s): %s",comp->ctx->label,comp->stream->label,cand->codeword,cand->label);
+      cand=TAILQ_NEXT(cand,entry_comp);
+    }
+  }
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_component.h b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_component.h
new file mode 100644
index 0000000000..0b12a68d58
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_component.h
@@ -0,0 +1,111 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _ice_component_h
+#define _ice_component_h
+#ifdef __cplusplus
+using namespace std;
+extern "C" {
+#endif /* __cplusplus */
+
+typedef struct nr_ice_pre_answer_request_ {
+  nr_stun_server_request req;
+  char *username;
+  nr_transport_addr local_addr;
+
+  STAILQ_ENTRY(nr_ice_pre_answer_request_) entry;
+} nr_ice_pre_answer_request;
+
+typedef STAILQ_HEAD(nr_ice_pre_answer_request_head_, nr_ice_pre_answer_request_) nr_ice_pre_answer_request_head;
+
+struct nr_ice_component_ {
+  int state;
+#define NR_ICE_COMPONENT_UNPAIRED           0
+#define NR_ICE_COMPONENT_RUNNING            1
+#define NR_ICE_COMPONENT_NOMINATED          2
+#define NR_ICE_COMPONENT_FAILED             3
+#define NR_ICE_COMPONENT_DISABLED           4
+  struct nr_ice_ctx_ *ctx;
+  struct nr_ice_media_stream_ *stream;
+  nr_ice_component *local_component;
+
+  int component_id;
+  nr_ice_socket_head sockets;
+  nr_ice_candidate_head candidates;
+  int candidate_ct;
+  nr_ice_pre_answer_request_head pre_answer_reqs;
+
+  int valid_pairs;
+  struct nr_ice_cand_pair_ *nominated; /* Highest priority nomninated pair */
+  struct nr_ice_cand_pair_ *active;
+
+  nr_stun_client_ctx *consent_ctx;
+  void *consent_timer;
+  void *consent_timeout;
+  void *consent_handle;
+  int can_send;
+  int disconnected;
+  struct timeval consent_last_seen;
+
+  STAILQ_ENTRY(nr_ice_component_)entry;
+};
+
+typedef STAILQ_HEAD(nr_ice_component_head_,nr_ice_component_) nr_ice_component_head;
+
+int nr_ice_component_create(struct nr_ice_media_stream_ *stream, int component_id, nr_ice_component **componentp);
+int nr_ice_component_destroy(nr_ice_component **componentp);
+int nr_ice_component_initialize(struct nr_ice_ctx_ *ctx,nr_ice_component *component);
+void nr_ice_component_stop_gathering(nr_ice_component *component);
+int nr_ice_component_is_done_gathering(nr_ice_component *comp);
+int nr_ice_component_maybe_prune_candidate(nr_ice_ctx *ctx, nr_ice_component *comp, nr_ice_candidate *c1, int *was_pruned);
+int nr_ice_component_pair_candidate(nr_ice_peer_ctx *pctx, nr_ice_component *pcomp, nr_ice_candidate *lcand, int pair_all_remote);
+int nr_ice_component_pair_candidates(nr_ice_peer_ctx *pctx, nr_ice_component *lcomp, nr_ice_component *pcomp);
+int nr_ice_component_service_pre_answer_requests(nr_ice_peer_ctx *pctx, nr_ice_component *pcomp, char *username, int *serviced);
+int nr_ice_component_nominated_pair(nr_ice_component *comp, nr_ice_cand_pair *pair);
+void nr_ice_component_failed_pair(nr_ice_component *comp, nr_ice_cand_pair *pair);
+void nr_ice_component_check_if_failed(nr_ice_component *comp);
+int nr_ice_component_select_pair(nr_ice_peer_ctx *pctx, nr_ice_component *comp);
+int nr_ice_component_set_failed(nr_ice_component *comp);
+int nr_ice_component_finalize(nr_ice_component *lcomp, nr_ice_component *rcomp);
+int nr_ice_component_insert_pair(nr_ice_component *pcomp, nr_ice_cand_pair *pair);
+int nr_ice_component_get_default_candidate(nr_ice_component *comp, nr_ice_candidate **candp, int ip_version);
+void nr_ice_component_consent_destroy(nr_ice_component *comp);
+void nr_ice_component_refresh_consent_now(nr_ice_component *comp);
+void nr_ice_component_disconnected(nr_ice_component *comp);
+void nr_ice_component_dump_state(nr_ice_component *comp, int log_level);
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+#endif
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_ctx.c b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_ctx.c
new file mode 100644
index 0000000000..0d498845a4
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_ctx.c
@@ -0,0 +1,1125 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <csi_platform.h>
+#include <assert.h>
+#include <sys/types.h>
+#ifdef WIN32
+#include <winsock2.h>
+#else
+#include <sys/socket.h>
+#include <netinet/in.h>
+#include <arpa/inet.h>
+#endif
+#include <sys/queue.h>
+#include <string.h>
+#include <nr_api.h>
+#include <registry.h>
+#include "stun.h"
+#include "ice_ctx.h"
+#include "ice_reg.h"
+#include "nr_crypto.h"
+#include "async_timer.h"
+#include "util.h"
+#include "nr_socket_local.h"
+
+#define ICE_UFRAG_LEN 8
+#define ICE_PWD_LEN 32
+
+int LOG_ICE = 0;
+
+static int nr_ice_random_string(char *str, int len);
+static int nr_ice_fetch_stun_servers(int ct, nr_ice_stun_server **out);
+#ifdef USE_TURN
+static int nr_ice_fetch_turn_servers(int ct, nr_ice_turn_server **out);
+#endif /* USE_TURN */
+static int nr_ice_ctx_pair_new_trickle_candidates(nr_ice_ctx *ctx, nr_ice_candidate *cand);
+static int no_op(void **obj) {
+  return 0;
+}
+
+static nr_socket_factory_vtbl default_socket_factory_vtbl = {
+  nr_socket_local_create,
+  no_op
+};
+
+int nr_ice_fetch_stun_servers(int ct, nr_ice_stun_server **out)
+  {
+    int r,_status;
+    nr_ice_stun_server *servers = 0;
+    int i;
+    NR_registry child;
+    char *addr=0;
+    UINT2 port;
+    in_addr_t addr_int;
+
+    if(!(servers=RCALLOC(sizeof(nr_ice_stun_server)*ct)))
+      ABORT(R_NO_MEMORY);
+
+    for(i=0;i<ct;i++){
+      if(r=NR_reg_get_child_registry(NR_ICE_REG_STUN_SRV_PRFX,i,child))
+        ABORT(r);
+      /* Assume we have a v4 addr for now */
+      if(r=NR_reg_alloc2_string(child,"addr",&addr))
+        ABORT(r);
+      addr_int=inet_addr(addr);
+      if(addr_int==INADDR_NONE){
+        r_log(LOG_ICE,LOG_ERR,"Invalid address %s;",addr);
+        ABORT(R_BAD_ARGS);
+      }
+      if(r=NR_reg_get2_uint2(child,"port",&port)) {
+        if (r != R_NOT_FOUND)
+          ABORT(r);
+        port = 3478;
+      }
+      if (r = nr_ip4_port_to_transport_addr(ntohl(addr_int), port, IPPROTO_UDP,
+                                            &servers[i].addr))
+        ABORT(r);
+      RFREE(addr);
+      addr=0;
+    }
+
+    *out = servers;
+
+    _status=0;
+  abort:
+    RFREE(addr);
+    if (_status) RFREE(servers);
+    return(_status);
+  }
+
+int nr_ice_ctx_set_stun_servers(nr_ice_ctx *ctx,nr_ice_stun_server *servers,int ct)
+  {
+    int _status;
+
+    if(ctx->stun_servers_cfg){
+      RFREE(ctx->stun_servers_cfg);
+      ctx->stun_servers_cfg=NULL;
+      ctx->stun_server_ct_cfg=0;
+    }
+
+    if (ct) {
+      if(!(ctx->stun_servers_cfg=RCALLOC(sizeof(nr_ice_stun_server)*ct)))
+        ABORT(R_NO_MEMORY);
+
+      memcpy(ctx->stun_servers_cfg,servers,sizeof(nr_ice_stun_server)*ct);
+      ctx->stun_server_ct_cfg = ct;
+    }
+
+    _status=0;
+ abort:
+    return(_status);
+  }
+
+int nr_ice_ctx_set_turn_servers(nr_ice_ctx *ctx,nr_ice_turn_server *servers,int ct)
+  {
+    int _status;
+
+    if(ctx->turn_servers_cfg){
+      for (int i = 0; i < ctx->turn_server_ct_cfg; i++) {
+        RFREE(ctx->turn_servers_cfg[i].username);
+        r_data_destroy(&ctx->turn_servers_cfg[i].password);
+      }
+      RFREE(ctx->turn_servers_cfg);
+      ctx->turn_servers_cfg=NULL;
+      ctx->turn_server_ct_cfg=0;
+    }
+
+    if(ct) {
+      if(!(ctx->turn_servers_cfg=RCALLOC(sizeof(nr_ice_turn_server)*ct)))
+        ABORT(R_NO_MEMORY);
+
+      memcpy(ctx->turn_servers_cfg,servers,sizeof(nr_ice_turn_server)*ct);
+      ctx->turn_server_ct_cfg = ct;
+    }
+
+    _status=0;
+ abort:
+    return(_status);
+  }
+
+int nr_ice_ctx_copy_turn_servers(nr_ice_ctx *ctx, nr_ice_turn_server *servers, int ct)
+  {
+    int _status, i, r;
+
+    if (r = nr_ice_ctx_set_turn_servers(ctx, servers, ct)) {
+      ABORT(r);
+    }
+
+    // make copies of the username and password so they aren't freed twice
+    for (i = 0; i < ct; ++i) {
+      if (!(ctx->turn_servers_cfg[i].username = r_strdup(servers[i].username))) {
+        ABORT(R_NO_MEMORY);
+      }
+      if (r = r_data_create(&ctx->turn_servers_cfg[i].password,
+                            servers[i].password->data,
+                            servers[i].password->len)) {
+        ABORT(r);
+      }
+    }
+
+    _status=0;
+   abort:
+    return(_status);
+  }
+
+static int nr_ice_ctx_set_local_addrs(nr_ice_ctx *ctx,nr_local_addr *addrs,int ct)
+  {
+    int _status,i,r;
+
+    if(ctx->local_addrs) {
+      RFREE(ctx->local_addrs);
+      ctx->local_addr_ct=0;
+      ctx->local_addrs=0;
+    }
+
+    if (ct) {
+      if(!(ctx->local_addrs=RCALLOC(sizeof(nr_local_addr)*ct)))
+        ABORT(R_NO_MEMORY);
+
+      for (i=0;i<ct;++i) {
+        if (r=nr_local_addr_copy(ctx->local_addrs+i,addrs+i)) {
+          ABORT(r);
+        }
+      }
+      ctx->local_addr_ct = ct;
+    }
+
+    _status=0;
+   abort:
+    return(_status);
+  }
+
+int nr_ice_ctx_set_resolver(nr_ice_ctx *ctx, nr_resolver *resolver)
+  {
+    int _status;
+
+    if (ctx->resolver) {
+      ABORT(R_ALREADY);
+    }
+
+    ctx->resolver = resolver;
+
+    _status=0;
+   abort:
+    return(_status);
+  }
+
+int nr_ice_ctx_set_interface_prioritizer(nr_ice_ctx *ctx, nr_interface_prioritizer *ip)
+  {
+    int _status;
+
+    if (ctx->interface_prioritizer) {
+      ABORT(R_ALREADY);
+    }
+
+    ctx->interface_prioritizer = ip;
+
+    _status=0;
+   abort:
+    return(_status);
+  }
+
+void nr_ice_ctx_set_socket_factory(nr_ice_ctx *ctx, nr_socket_factory *factory)
+  {
+    nr_socket_factory_destroy(&ctx->socket_factory);
+    ctx->socket_factory = factory;
+  }
+
+#ifdef USE_TURN
+int nr_ice_fetch_turn_servers(int ct, nr_ice_turn_server **out)
+  {
+    int r,_status;
+    nr_ice_turn_server *servers = 0;
+    int i;
+    NR_registry child;
+    char *addr=0;
+    UINT2 port;
+    in_addr_t addr_int;
+    Data data={0};
+
+    if(!(servers=RCALLOC(sizeof(nr_ice_turn_server)*ct)))
+      ABORT(R_NO_MEMORY);
+
+    for(i=0;i<ct;i++){
+      if(r=NR_reg_get_child_registry(NR_ICE_REG_TURN_SRV_PRFX,i,child))
+        ABORT(r);
+      /* Assume we have a v4 addr for now */
+      if(r=NR_reg_alloc2_string(child,"addr",&addr))
+        ABORT(r);
+      addr_int=inet_addr(addr);
+      if(addr_int==INADDR_NONE){
+        r_log(LOG_ICE,LOG_ERR,"Invalid address %s",addr);
+        ABORT(R_BAD_ARGS);
+      }
+      if(r=NR_reg_get2_uint2(child,"port",&port)) {
+        if (r != R_NOT_FOUND)
+          ABORT(r);
+        port = 3478;
+      }
+      if (r = nr_ip4_port_to_transport_addr(ntohl(addr_int), port, IPPROTO_UDP,
+                                            &servers[i].turn_server.addr))
+        ABORT(r);
+
+
+      if(r=NR_reg_alloc2_string(child,NR_ICE_REG_TURN_SRV_USERNAME,&servers[i].username)){
+        if(r!=R_NOT_FOUND)
+          ABORT(r);
+      }
+
+      if(r=NR_reg_alloc2_data(child,NR_ICE_REG_TURN_SRV_PASSWORD,&data)){
+        if(r!=R_NOT_FOUND)
+          ABORT(r);
+      }
+      else {
+        servers[i].password=RCALLOC(sizeof(*servers[i].password));
+        if(!servers[i].password)
+          ABORT(R_NO_MEMORY);
+        servers[i].password->data = data.data;
+        servers[i].password->len = data.len;
+        data.data=0;
+      }
+
+      RFREE(addr);
+      addr=0;
+    }
+
+    *out = servers;
+
+    _status=0;
+  abort:
+    RFREE(data.data);
+    RFREE(addr);
+    if (_status) RFREE(servers);
+    return(_status);
+  }
+#endif /* USE_TURN */
+
+#define MAXADDRS 100 /* Ridiculously high */
+int nr_ice_ctx_create(char *label, UINT4 flags, nr_ice_ctx **ctxp)
+  {
+    nr_ice_ctx *ctx=0;
+    int r,_status;
+
+    if(r=r_log_register("ice", &LOG_ICE))
+      ABORT(r);
+
+    if(!(ctx=RCALLOC(sizeof(nr_ice_ctx))))
+      ABORT(R_NO_MEMORY);
+
+    ctx->flags=flags;
+
+    if(!(ctx->label=r_strdup(label)))
+      ABORT(R_NO_MEMORY);
+
+    /* Get the STUN servers */
+    if(r=NR_reg_get_child_count(NR_ICE_REG_STUN_SRV_PRFX,
+      (unsigned int *)&ctx->stun_server_ct_cfg)||ctx->stun_server_ct_cfg==0) {
+      r_log(LOG_ICE,LOG_DEBUG,"ICE(%s): No STUN servers specified in nICEr registry", ctx->label);
+      ctx->stun_server_ct_cfg=0;
+    }
+
+    /* 31 is the max for our priority algorithm */
+    if(ctx->stun_server_ct_cfg>31){
+      r_log(LOG_ICE,LOG_WARNING,"ICE(%s): Too many STUN servers specified: max=31", ctx->label);
+      ctx->stun_server_ct_cfg=31;
+    }
+
+    if(ctx->stun_server_ct_cfg>0){
+      if(r=nr_ice_fetch_stun_servers(ctx->stun_server_ct_cfg,&ctx->stun_servers_cfg)){
+        r_log(LOG_ICE,LOG_ERR,"ICE(%s): Couldn't load STUN servers from registry", ctx->label);
+        ctx->stun_server_ct_cfg=0;
+        ABORT(r);
+      }
+    }
+
+#ifdef USE_TURN
+    /* Get the TURN servers */
+    if(r=NR_reg_get_child_count(NR_ICE_REG_TURN_SRV_PRFX,
+      (unsigned int *)&ctx->turn_server_ct_cfg)||ctx->turn_server_ct_cfg==0) {
+      r_log(LOG_ICE,LOG_DEBUG,"ICE(%s): No TURN servers specified in nICEr registry", ctx->label);
+      ctx->turn_server_ct_cfg=0;
+    }
+#else
+    ctx->turn_server_ct_cfg=0;
+#endif /* USE_TURN */
+
+    ctx->local_addrs=0;
+    ctx->local_addr_ct=0;
+
+    /* 31 is the max for our priority algorithm */
+    if((ctx->stun_server_ct_cfg+ctx->turn_server_ct_cfg)>31){
+      r_log(LOG_ICE,LOG_WARNING,"ICE(%s): Too many STUN/TURN servers specified: max=31", ctx->label);
+      ctx->turn_server_ct_cfg=31-ctx->stun_server_ct_cfg;
+    }
+
+#ifdef USE_TURN
+    if(ctx->turn_server_ct_cfg>0){
+      if(r=nr_ice_fetch_turn_servers(ctx->turn_server_ct_cfg,&ctx->turn_servers_cfg)){
+        ctx->turn_server_ct_cfg=0;
+        r_log(LOG_ICE,LOG_ERR,"ICE(%s): Couldn't load TURN servers from registry", ctx->label);
+        ABORT(r);
+      }
+    }
+#endif /* USE_TURN */
+
+
+    ctx->Ta = 20;
+
+    ctx->test_timer_divider = 0;
+
+    if (r=nr_socket_factory_create_int(NULL, &default_socket_factory_vtbl, &ctx->socket_factory))
+      ABORT(r);
+
+    if ((r=NR_reg_get_string((char *)NR_ICE_REG_PREF_FORCE_INTERFACE_NAME, ctx->force_net_interface, sizeof(ctx->force_net_interface)))) {
+      if (r == R_NOT_FOUND) {
+        ctx->force_net_interface[0] = 0;
+      } else {
+        ABORT(r);
+      }
+    }
+
+    ctx->target_for_default_local_address_lookup=0;
+
+    STAILQ_INIT(&ctx->streams);
+    STAILQ_INIT(&ctx->sockets);
+    STAILQ_INIT(&ctx->foundations);
+    STAILQ_INIT(&ctx->peers);
+    STAILQ_INIT(&ctx->ids);
+
+    *ctxp=ctx;
+
+    _status=0;
+  abort:
+    if (_status && ctx) nr_ice_ctx_destroy(&ctx);
+
+    return(_status);
+  }
+
+  void nr_ice_ctx_add_flags(nr_ice_ctx* ctx, UINT4 flags) {
+    ctx->flags |= flags;
+  }
+
+  void nr_ice_ctx_remove_flags(nr_ice_ctx* ctx, UINT4 flags) {
+    ctx->flags &= ~flags;
+  }
+
+  void nr_ice_ctx_destroy(nr_ice_ctx** ctxp) {
+    if (!ctxp || !*ctxp) return;
+
+    nr_ice_ctx* ctx = *ctxp;
+    nr_ice_foundation *f1,*f2;
+    nr_ice_media_stream *s1,*s2;
+    int i;
+    nr_ice_stun_id *id1,*id2;
+
+    ctx->done_cb = 0;
+    ctx->trickle_cb = 0;
+
+    STAILQ_FOREACH_SAFE(s1, &ctx->streams, entry, s2){
+      STAILQ_REMOVE(&ctx->streams,s1,nr_ice_media_stream_,entry);
+      nr_ice_media_stream_destroy(&s1);
+    }
+
+    RFREE(ctx->label);
+
+    RFREE(ctx->stun_servers_cfg);
+
+    RFREE(ctx->local_addrs);
+
+    RFREE(ctx->target_for_default_local_address_lookup);
+
+    for (i = 0; i < ctx->turn_server_ct_cfg; i++) {
+        RFREE(ctx->turn_servers_cfg[i].username);
+        r_data_destroy(&ctx->turn_servers_cfg[i].password);
+    }
+    RFREE(ctx->turn_servers_cfg);
+
+    f1=STAILQ_FIRST(&ctx->foundations);
+    while(f1){
+      f2=STAILQ_NEXT(f1,entry);
+      RFREE(f1);
+      f1=f2;
+    }
+
+    STAILQ_FOREACH_SAFE(id1, &ctx->ids, entry, id2){
+      STAILQ_REMOVE(&ctx->ids,id1,nr_ice_stun_id_,entry);
+      RFREE(id1);
+    }
+
+    nr_resolver_destroy(&ctx->resolver);
+    nr_interface_prioritizer_destroy(&ctx->interface_prioritizer);
+    nr_socket_factory_destroy(&ctx->socket_factory);
+
+    RFREE(ctx);
+
+    *ctxp=0;
+  }
+
+void nr_ice_gather_finished_cb(NR_SOCKET s, int h, void *cb_arg)
+  {
+    int r;
+    nr_ice_candidate *cand=cb_arg;
+    nr_ice_ctx *ctx;
+    nr_ice_media_stream *stream;
+    int component_id;
+
+    assert(cb_arg);
+    if (!cb_arg)
+      return;
+    ctx = cand->ctx;
+    stream = cand->stream;
+    component_id = cand->component_id;
+
+    ctx->uninitialized_candidates--;
+    if (cand->state == NR_ICE_CAND_STATE_FAILED) {
+      r_log(LOG_ICE, LOG_WARNING,
+            "ICE(%s)/CAND(%s): failed to initialize, %d remaining", ctx->label,
+            cand->label, ctx->uninitialized_candidates);
+    } else {
+      r_log(LOG_ICE, LOG_DEBUG, "ICE(%s)/CAND(%s): initialized, %d remaining",
+            ctx->label, cand->label, ctx->uninitialized_candidates);
+    }
+
+    /* Avoid the need for yet another initialization function */
+    if (cand->state == NR_ICE_CAND_STATE_INITIALIZING && cand->type == HOST)
+      cand->state = NR_ICE_CAND_STATE_INITIALIZED;
+
+    if (cand->state == NR_ICE_CAND_STATE_INITIALIZED) {
+      int was_pruned = 0;
+
+      if (r=nr_ice_component_maybe_prune_candidate(ctx, cand->component,
+                                                   cand, &was_pruned)) {
+          r_log(LOG_ICE, LOG_NOTICE, "ICE(%s): Problem pruning candidates",ctx->label);
+      }
+
+      if (was_pruned) {
+        cand = NULL;
+      }
+
+      /* If we are initialized, the candidate wasn't pruned,
+         and we have a trickle ICE callback fire the callback */
+      if (ctx->trickle_cb && cand &&
+          !nr_ice_ctx_hide_candidate(ctx, cand)) {
+        ctx->trickle_cb(ctx->trickle_cb_arg, ctx, cand->stream, cand->component_id, cand);
+
+        if (nr_ice_ctx_pair_new_trickle_candidates(ctx, cand)) {
+          r_log(LOG_ICE,LOG_ERR, "ICE(%s): All could not pair new trickle candidate",ctx->label);
+          /* But continue */
+        }
+      }
+    }
+
+    if (nr_ice_media_stream_is_done_gathering(stream) &&
+        ctx->trickle_cb) {
+      ctx->trickle_cb(ctx->trickle_cb_arg, ctx, stream, component_id, NULL);
+    }
+
+    if(ctx->uninitialized_candidates==0){
+      r_log(LOG_ICE, LOG_INFO, "ICE(%s): All candidates initialized",
+            ctx->label);
+      if (ctx->done_cb) {
+        ctx->done_cb(0,0,ctx->cb_arg);
+      }
+      else {
+        r_log(LOG_ICE, LOG_INFO,
+              "ICE(%s): No done_cb. We were probably destroyed.", ctx->label);
+      }
+    }
+    else {
+      r_log(LOG_ICE,LOG_DEBUG,"ICE(%s): Waiting for %d candidates to be initialized",ctx->label, ctx->uninitialized_candidates);
+    }
+  }
+
+static int nr_ice_ctx_pair_new_trickle_candidates(nr_ice_ctx *ctx, nr_ice_candidate *cand)
+  {
+    int r,_status;
+    nr_ice_peer_ctx *pctx;
+
+    pctx=STAILQ_FIRST(&ctx->peers);
+    while(pctx){
+      if (pctx->state == NR_ICE_PEER_STATE_PAIRED) {
+        r = nr_ice_peer_ctx_pair_new_trickle_candidate(ctx, pctx, cand);
+        if (r)
+          ABORT(r);
+      }
+
+      pctx=STAILQ_NEXT(pctx,entry);
+    }
+
+    _status=0;
+ abort:
+    return(_status);
+  }
+
+/* Get the default address by creating a UDP socket, binding it to a wildcard
+   address, and connecting it to the remote IP. Because this is UDP, no packets
+   are sent. This lets us query the local address assigned to the socket by the
+   kernel.
+
+   If the context's remote address is NULL, then the application wasn't loaded
+   over the network, and we can fall back on connecting to a known public
+   address (namely Google's):
+
+   IPv4: 8.8.8.8
+   IPv6: 2001:4860:4860::8888
+*/
+static int nr_ice_get_default_address(nr_ice_ctx *ctx, int ip_version, nr_transport_addr* addrp)
+  {
+    int r,_status;
+    nr_transport_addr addr, known_remote_addr;
+    nr_transport_addr *remote_addr=ctx->target_for_default_local_address_lookup;
+    nr_socket *sock=0;
+
+    switch(ip_version) {
+      case NR_IPV4:
+        if ((r=nr_str_port_to_transport_addr("0.0.0.0", 0, IPPROTO_UDP, &addr)))
+          ABORT(r);
+        if (!remote_addr || nr_transport_addr_is_loopback(remote_addr)) {
+          if ((r=nr_str_port_to_transport_addr("8.8.8.8", 53, IPPROTO_UDP, &known_remote_addr)))
+            ABORT(r);
+          remote_addr=&known_remote_addr;
+        }
+        break;
+      case NR_IPV6:
+        if ((r=nr_str_port_to_transport_addr("::0", 0, IPPROTO_UDP, &addr)))
+          ABORT(r);
+        if (!remote_addr || nr_transport_addr_is_loopback(remote_addr)) {
+          if ((r=nr_str_port_to_transport_addr("2001:4860:4860::8888", 53, IPPROTO_UDP, &known_remote_addr)))
+            ABORT(r);
+          remote_addr=&known_remote_addr;
+        }
+        break;
+      default:
+        assert(0);
+        ABORT(R_INTERNAL);
+    }
+
+    if ((r=nr_socket_factory_create_socket(ctx->socket_factory, &addr, &sock)))
+      ABORT(r);
+    if ((r=nr_socket_connect(sock, remote_addr)))
+      ABORT(r);
+    if ((r=nr_socket_getaddr(sock, addrp)))
+      ABORT(r);
+
+    r_log(LOG_GENERIC, LOG_DEBUG, "Default address: %s", addrp->as_string);
+
+    _status=0;
+  abort:
+    nr_socket_destroy(&sock);
+    return(_status);
+  }
+
+static int nr_ice_get_default_local_address(nr_ice_ctx *ctx, int ip_version, nr_local_addr* addrs, int addr_ct, nr_local_addr *addrp)
+  {
+    int r,_status;
+    nr_transport_addr default_addr;
+    int i;
+
+    if ((r=nr_ice_get_default_address(ctx, ip_version, &default_addr)))
+        ABORT(r);
+
+    for (i=0; i < addr_ct; ++i) {
+      // if default addr is found in local addrs, copy the more fully
+      // complete local addr to the output arg.  Don't need to worry
+      // about comparing ports here.
+      if (!nr_transport_addr_cmp(&default_addr, &addrs[i].addr,
+                                 NR_TRANSPORT_ADDR_CMP_MODE_ADDR)) {
+        if ((r=nr_local_addr_copy(addrp, &addrs[i])))
+          ABORT(r);
+        break;
+      }
+    }
+
+    // if default addr is not in local addrs, just copy the transport addr
+    // to output arg.
+    if (i == addr_ct) {
+      if ((r=nr_transport_addr_copy(&addrp->addr, &default_addr)))
+        ABORT(r);
+      (void)strlcpy(addrp->addr.ifname, "default route", sizeof(addrp->addr.ifname));
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+/* if handed a IPv4 default_local_addr, looks for IPv6 address on same interface
+   if handed a IPv6 default_local_addr, looks for IPv4 address on same interface
+*/
+static int nr_ice_get_assoc_interface_address(nr_local_addr* default_local_addr,
+                                              nr_local_addr* local_addrs, int addr_ct,
+                                              nr_local_addr* assoc_addrp)
+  {
+    int r, _status;
+    int i, ip_version;
+
+    if (!default_local_addr || !local_addrs || !addr_ct) {
+      ABORT(R_BAD_ARGS);
+    }
+
+    /* set _status to R_EOD in case we don't find an associated address */
+    _status = R_EOD;
+
+    /* look for IPv6 if we have IPv4, look for IPv4 if we have IPv6 */
+    ip_version = (NR_IPV4 == default_local_addr->addr.ip_version?NR_IPV6:NR_IPV4);
+
+    for (i=0; i<addr_ct; ++i) {
+      /* if we find the ip_version we're looking for on the matching interface,
+         copy it to assoc_addrp.
+      */
+      if (local_addrs[i].addr.ip_version == ip_version &&
+          !strcmp(local_addrs[i].addr.ifname, default_local_addr->addr.ifname)) {
+        if (r=nr_local_addr_copy(assoc_addrp, &local_addrs[i])) {
+          ABORT(r);
+        }
+        _status = 0;
+        break;
+      }
+    }
+
+  abort:
+    return(_status);
+  }
+
+int nr_ice_set_local_addresses(nr_ice_ctx *ctx,
+                               nr_local_addr* stun_addrs, int stun_addr_ct)
+  {
+    int r,_status;
+    nr_local_addr local_addrs[MAXADDRS];
+    nr_local_addr *addrs = 0;
+    int i,addr_ct;
+    nr_local_addr default_addrs[2];
+    int default_addr_ct = 0;
+
+    if (!stun_addrs || !stun_addr_ct) {
+      r_log(LOG_ICE,LOG_ERR,"ICE(%s): no stun addrs provided",ctx->label);
+      ABORT(R_BAD_ARGS);
+    }
+
+    addr_ct = MIN(stun_addr_ct, MAXADDRS);
+    r_log(LOG_ICE, LOG_DEBUG, "ICE(%s): copy %d pre-fetched stun addrs", ctx->label, addr_ct);
+    for (i=0; i<addr_ct; ++i) {
+      if (r=nr_local_addr_copy(&local_addrs[i], &stun_addrs[i])) {
+        ABORT(r);
+      }
+    }
+
+    // removes duplicates and, based on prefs, loopback and link_local addrs
+    if (r=nr_stun_filter_local_addresses(local_addrs, &addr_ct)) {
+      ABORT(r);
+    }
+
+    if (ctx->force_net_interface[0] && addr_ct) {
+      /* Limit us to only addresses on a single interface */
+      int force_addr_ct = 0;
+      for(i=0;i<addr_ct;i++){
+        if (!strcmp(local_addrs[i].addr.ifname, ctx->force_net_interface)) {
+          // copy it down in the array, if needed
+          if (i != force_addr_ct) {
+            if (r=nr_local_addr_copy(&local_addrs[force_addr_ct], &local_addrs[i])) {
+              ABORT(r);
+            }
+          }
+          force_addr_ct++;
+        }
+      }
+      addr_ct = force_addr_ct;
+    }
+
+    r_log(LOG_ICE, LOG_DEBUG,
+          "ICE(%s): use only default local addresses: %s\n",
+          ctx->label,
+          (char*)(ctx->flags & NR_ICE_CTX_FLAGS_ONLY_DEFAULT_ADDRS?"yes":"no"));
+    if ((!addr_ct) || (ctx->flags & NR_ICE_CTX_FLAGS_ONLY_DEFAULT_ADDRS)) {
+      if (ctx->target_for_default_local_address_lookup) {
+        /* Get just the default IPv4 or IPv6 addr */
+        if(!nr_ice_get_default_local_address(
+               ctx, ctx->target_for_default_local_address_lookup->ip_version,
+               local_addrs, addr_ct, &default_addrs[default_addr_ct])) {
+          nr_local_addr *new_addr = &default_addrs[default_addr_ct];
+
+          ++default_addr_ct;
+
+          /* If we have a default target address, check for an associated
+             address on the same interface.  For example, if the default
+             target address is IPv6, this will find an associated IPv4
+             address on the same interface.
+             This makes ICE w/ dual stacks work better - Bug 1609124.
+          */
+          if(!nr_ice_get_assoc_interface_address(
+              new_addr, local_addrs, addr_ct,
+              &default_addrs[default_addr_ct])) {
+            ++default_addr_ct;
+          }
+        }
+      } else {
+        /* Get just the default IPv4 and IPv6 addrs */
+        if(!nr_ice_get_default_local_address(ctx, NR_IPV4, local_addrs, addr_ct,
+                                             &default_addrs[default_addr_ct])) {
+          ++default_addr_ct;
+        }
+        if(!nr_ice_get_default_local_address(ctx, NR_IPV6, local_addrs, addr_ct,
+                                             &default_addrs[default_addr_ct])) {
+          ++default_addr_ct;
+        }
+      }
+      if (!default_addr_ct) {
+        r_log(LOG_ICE,LOG_ERR,"ICE(%s): failed to find default addresses",ctx->label);
+        ABORT(R_FAILED);
+      }
+      addrs = default_addrs;
+      addr_ct = default_addr_ct;
+    }
+    else {
+      addrs = local_addrs;
+    }
+
+    /* Sort interfaces by preference */
+    if(ctx->interface_prioritizer) {
+      for(i=0;i<addr_ct;i++){
+        if((r=nr_interface_prioritizer_add_interface(ctx->interface_prioritizer,addrs+i)) && (r!=R_ALREADY)) {
+          r_log(LOG_ICE,LOG_ERR,"ICE(%s): unable to add interface ",ctx->label);
+          ABORT(r);
+        }
+      }
+      if(r=nr_interface_prioritizer_sort_preference(ctx->interface_prioritizer)) {
+        r_log(LOG_ICE,LOG_ERR,"ICE(%s): unable to sort interface by preference",ctx->label);
+        ABORT(r);
+      }
+    }
+
+    if (r=nr_ice_ctx_set_local_addrs(ctx,addrs,addr_ct)) {
+      ABORT(r);
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_ice_set_target_for_default_local_address_lookup(nr_ice_ctx *ctx, const char *target_ip, UINT2 target_port)
+  {
+    int r,_status;
+
+    if (ctx->target_for_default_local_address_lookup) {
+      RFREE(ctx->target_for_default_local_address_lookup);
+      ctx->target_for_default_local_address_lookup=0;
+    }
+
+    if (!(ctx->target_for_default_local_address_lookup=RCALLOC(sizeof(nr_transport_addr))))
+      ABORT(R_NO_MEMORY);
+
+    if ((r=nr_str_port_to_transport_addr(target_ip, target_port, IPPROTO_UDP, ctx->target_for_default_local_address_lookup))) {
+      RFREE(ctx->target_for_default_local_address_lookup);
+      ctx->target_for_default_local_address_lookup=0;
+      ABORT(r);
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_ice_gather(nr_ice_ctx *ctx, NR_async_cb done_cb, void *cb_arg)
+  {
+    int r,_status;
+    nr_ice_media_stream *stream;
+    nr_local_addr stun_addrs[MAXADDRS];
+    int stun_addr_ct;
+
+    if (!ctx->local_addrs) {
+      if((r=nr_stun_find_local_addresses(stun_addrs,MAXADDRS,&stun_addr_ct))) {
+        ABORT(r);
+      }
+      if((r=nr_ice_set_local_addresses(ctx,stun_addrs,stun_addr_ct))) {
+        ABORT(r);
+      }
+    }
+
+    if(STAILQ_EMPTY(&ctx->streams)) {
+      r_log(LOG_ICE,LOG_ERR,"ICE(%s): Missing streams to initialize",ctx->label);
+      ABORT(R_BAD_ARGS);
+    }
+
+    r_log(LOG_ICE,LOG_DEBUG,"ICE(%s): Initializing candidates",ctx->label);
+    ctx->done_cb=done_cb;
+    ctx->cb_arg=cb_arg;
+
+    /* Initialize all the media stream/component pairs */
+    stream=STAILQ_FIRST(&ctx->streams);
+    while(stream){
+      if(!stream->obsolete) {
+        if(r=nr_ice_media_stream_initialize(ctx,stream)) {
+          r_log(LOG_ICE,LOG_ERR,"ICE(%s): Failed to initialize a stream; this might not be an unrecoverable error, if this stream will be marked obsolete soon due to an ICE restart.",ctx->label);
+        }
+      }
+
+      stream=STAILQ_NEXT(stream,entry);
+    }
+
+    if(ctx->uninitialized_candidates)
+      ABORT(R_WOULDBLOCK);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_ice_add_media_stream(nr_ice_ctx *ctx,const char *label,const char *ufrag,const char *pwd,int components, nr_ice_media_stream **streamp)
+  {
+    int r,_status;
+
+    if(r=nr_ice_media_stream_create(ctx,label,ufrag,pwd,components,streamp))
+      ABORT(r);
+
+    STAILQ_INSERT_TAIL(&ctx->streams,*streamp,entry);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_ice_remove_media_stream(nr_ice_ctx *ctx,nr_ice_media_stream **streamp)
+  {
+    int r,_status;
+    nr_ice_peer_ctx *pctx;
+    nr_ice_media_stream *peer_stream;
+
+    pctx=STAILQ_FIRST(&ctx->peers);
+    while(pctx){
+      if(!nr_ice_peer_ctx_find_pstream(pctx, *streamp, &peer_stream)) {
+        if(r=nr_ice_peer_ctx_remove_pstream(pctx, &peer_stream)) {
+          ABORT(r);
+        }
+      }
+
+      pctx=STAILQ_NEXT(pctx,entry);
+    }
+
+    STAILQ_REMOVE(&ctx->streams,*streamp,nr_ice_media_stream_,entry);
+    if(r=nr_ice_media_stream_destroy(streamp)) {
+      ABORT(r);
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_ice_get_global_attributes(nr_ice_ctx *ctx,char ***attrsp, int *attrctp)
+  {
+    *attrctp=0;
+    *attrsp=0;
+    return(0);
+  }
+
+static int nr_ice_random_string(char *str, int len)
+  {
+    unsigned char bytes[100];
+    size_t needed;
+    int r,_status;
+
+    if(len%2) ABORT(R_BAD_ARGS);
+    needed=len/2;
+
+    if(needed>sizeof(bytes)) ABORT(R_BAD_ARGS);
+
+    if(r=nr_crypto_random_bytes(bytes,needed))
+      ABORT(r);
+
+    if(r=nr_bin2hex(bytes,needed,(unsigned char *)str))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+/* This is incredibly annoying: we now have a datagram but we don't
+   know which peer it's from, and we need to be able to tell the
+   API user. So, offer it to each peer and if one bites, assume
+   the others don't want it
+*/
+int nr_ice_ctx_deliver_packet(nr_ice_ctx *ctx, nr_ice_component *comp, nr_transport_addr *source_addr, UCHAR *data, int len)
+  {
+    nr_ice_peer_ctx *pctx;
+    int r;
+
+    pctx=STAILQ_FIRST(&ctx->peers);
+    while(pctx){
+      r=nr_ice_peer_ctx_deliver_packet_maybe(pctx, comp, source_addr, data, len);
+      if(!r)
+        break;
+
+      pctx=STAILQ_NEXT(pctx,entry);
+    }
+
+    if(!pctx)
+      r_log(LOG_ICE,LOG_WARNING,"ICE(%s): Packet received from %s which doesn't match any known peer",ctx->label,source_addr->as_string);
+
+    return(0);
+  }
+
+int nr_ice_ctx_is_known_id(nr_ice_ctx *ctx, UCHAR id[12])
+  {
+    nr_ice_stun_id *xid;
+
+    xid=STAILQ_FIRST(&ctx->ids);
+    while(xid){
+      if (!memcmp(xid->id, id, 12))
+          return 1;
+
+      xid=STAILQ_NEXT(xid,entry);
+    }
+
+    return 0;
+  }
+
+int nr_ice_ctx_remember_id(nr_ice_ctx *ctx, nr_stun_message *msg)
+{
+    int _status;
+    nr_ice_stun_id *xid;
+
+    xid = RCALLOC(sizeof(*xid));
+    if (!xid)
+        ABORT(R_NO_MEMORY);
+
+    assert(sizeof(xid->id) == sizeof(msg->header.id));
+#if __STDC_VERSION__ >= 201112L
+    _Static_assert(sizeof(xid->id) == sizeof(msg->header.id),"Message ID Size Mismatch");
+#endif
+    memcpy(xid->id, &msg->header.id, sizeof(xid->id));
+
+    STAILQ_INSERT_TAIL(&ctx->ids,xid,entry);
+
+    _status=0;
+  abort:
+    return(_status);
+}
+
+
+/* Clean up some of the resources (mostly file descriptors) used
+   by candidates we didn't choose. Note that this still leaves
+   a fair amount of non-system stuff floating around. This gets
+   cleaned up when you destroy the ICE ctx */
+int nr_ice_ctx_finalize(nr_ice_ctx *ctx, nr_ice_peer_ctx *pctx)
+  {
+    nr_ice_media_stream *lstr,*rstr;
+
+    r_log(LOG_ICE,LOG_DEBUG,"Finalizing ICE ctx %s, peer=%s",ctx->label,pctx->label);
+    /*
+       First find the peer stream, if any
+    */
+    lstr=STAILQ_FIRST(&ctx->streams);
+    while(lstr){
+      rstr=STAILQ_FIRST(&pctx->peer_streams);
+
+      while(rstr){
+        if(rstr->local_stream==lstr)
+          break;
+
+        rstr=STAILQ_NEXT(rstr,entry);
+      }
+
+      nr_ice_media_stream_finalize(lstr,rstr);
+
+      lstr=STAILQ_NEXT(lstr,entry);
+    }
+
+    return(0);
+  }
+
+
+int nr_ice_ctx_set_trickle_cb(nr_ice_ctx *ctx, nr_ice_trickle_candidate_cb cb, void *cb_arg)
+{
+  ctx->trickle_cb = cb;
+  ctx->trickle_cb_arg = cb_arg;
+
+  return 0;
+}
+
+int nr_ice_ctx_hide_candidate(nr_ice_ctx *ctx, nr_ice_candidate *cand)
+  {
+    if (cand->state != NR_ICE_CAND_STATE_INITIALIZED) {
+      return 1;
+    }
+
+    if (ctx->flags & NR_ICE_CTX_FLAGS_DISABLE_HOST_CANDIDATES) {
+      if (cand->type == HOST)
+        return 1;
+    }
+
+    if (cand->stream->obsolete) {
+      return 1;
+    }
+
+    return 0;
+  }
+
+int nr_ice_get_new_ice_ufrag(char** ufrag)
+  {
+    int r,_status;
+    char buf[ICE_UFRAG_LEN+1];
+
+    if(r=nr_ice_random_string(buf,ICE_UFRAG_LEN))
+      ABORT(r);
+    if(!(*ufrag=r_strdup(buf)))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    if(_status) {
+      RFREE(*ufrag);
+      *ufrag = 0;
+    }
+    return(_status);
+  }
+
+int nr_ice_get_new_ice_pwd(char** pwd)
+  {
+    int r,_status;
+    char buf[ICE_PWD_LEN+1];
+
+    if(r=nr_ice_random_string(buf,ICE_PWD_LEN))
+      ABORT(r);
+    if(!(*pwd=r_strdup(buf)))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    if(_status) {
+      RFREE(*pwd);
+      *pwd = 0;
+    }
+    return(_status);
+  }
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_ctx.h b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_ctx.h
new file mode 100644
index 0000000000..8b3081f567
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_ctx.h
@@ -0,0 +1,188 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _ice_ctx_h
+#define _ice_ctx_h
+#ifdef __cplusplus
+using namespace std;
+extern "C" {
+#endif /* __cplusplus */
+
+/* Not good practice but making includes simpler */
+#include "transport_addr.h"
+#include "nr_socket.h"
+#include "nr_resolver.h"
+#include "nr_interface_prioritizer.h"
+#include "nr_socket_wrapper.h"
+#include "stun_client_ctx.h"
+#include "stun_server_ctx.h"
+#include "turn_client_ctx.h"
+
+typedef struct nr_ice_foundation_ {
+  int index;
+
+  nr_transport_addr addr;
+  int type;
+  /* ICE spec says that we only compare IP address, not port */
+  nr_transport_addr stun_server_addr;
+
+  STAILQ_ENTRY(nr_ice_foundation_) entry;
+} nr_ice_foundation;
+
+typedef STAILQ_HEAD(nr_ice_foundation_head_,nr_ice_foundation_) nr_ice_foundation_head;
+
+typedef TAILQ_HEAD(nr_ice_candidate_head_,nr_ice_candidate_) nr_ice_candidate_head;
+typedef TAILQ_HEAD(nr_ice_cand_pair_head_,nr_ice_cand_pair_) nr_ice_cand_pair_head;
+typedef struct nr_ice_component_ nr_ice_component;
+typedef struct nr_ice_media_stream_ nr_ice_media_stream;
+typedef struct nr_ice_ctx_ nr_ice_ctx;
+typedef struct nr_ice_peer_ctx_ nr_ice_peer_ctx;
+typedef struct nr_ice_candidate_ nr_ice_candidate;
+typedef struct nr_ice_cand_pair_ nr_ice_cand_pair;
+typedef void (*nr_ice_trickle_candidate_cb) (void *cb_arg,
+  nr_ice_ctx *ctx, nr_ice_media_stream *stream, int component_id,
+  nr_ice_candidate *candidate);
+
+#include "ice_socket.h"
+#include "ice_component.h"
+#include "ice_media_stream.h"
+#include "ice_candidate.h"
+#include "ice_candidate_pair.h"
+#include "ice_handler.h"
+#include "ice_peer_ctx.h"
+
+typedef struct nr_ice_stun_id_ {
+  UCHAR id[12];
+
+  STAILQ_ENTRY(nr_ice_stun_id_) entry;
+} nr_ice_stun_id;
+
+typedef STAILQ_HEAD(nr_ice_stun_id_head_,nr_ice_stun_id_) nr_ice_stun_id_head;
+
+typedef struct nr_ice_stats_ {
+  UINT2 stun_retransmits;
+  UINT2 turn_401s;
+  UINT2 turn_403s;
+  UINT2 turn_438s;
+} nr_ice_stats;
+
+struct nr_ice_ctx_ {
+  UINT4 flags;
+  char *label;
+
+  UINT4 Ta;
+
+  nr_ice_stun_server *stun_servers_cfg;           /* The list of stun servers */
+  int stun_server_ct_cfg;
+  nr_ice_turn_server *turn_servers_cfg;           /* The list of turn servers */
+  int turn_server_ct_cfg;
+  nr_local_addr *local_addrs;                 /* The list of available local addresses and corresponding interface information */
+  int local_addr_ct;
+
+  nr_resolver *resolver;                      /* The resolver to use */
+  nr_interface_prioritizer *interface_prioritizer;  /* Priority decision logic */
+  nr_socket_factory *socket_factory;
+
+  nr_ice_foundation_head foundations;
+
+  nr_ice_media_stream_head streams;           /* Media streams */
+  int stream_ct;
+  nr_ice_socket_head sockets;                 /* The sockets we're using */
+  int uninitialized_candidates;
+
+  UINT4 gather_rto;
+  UINT4 stun_delay;
+
+  UINT4 test_timer_divider;
+
+  nr_ice_peer_ctx_head peers;
+  nr_ice_stun_id_head ids;
+
+  NR_async_cb done_cb;
+  void *cb_arg;
+
+  nr_ice_trickle_candidate_cb trickle_cb;
+  void *trickle_cb_arg;
+
+  char force_net_interface[MAXIFNAME];
+  nr_ice_stats stats;
+
+  nr_transport_addr *target_for_default_local_address_lookup;
+};
+
+int nr_ice_ctx_create(char *label, UINT4 flags, nr_ice_ctx **ctxp);
+int nr_ice_ctx_create_with_credentials(char *label, UINT4 flags, char* ufrag, char* pwd, nr_ice_ctx **ctxp);
+#define NR_ICE_CTX_FLAGS_AGGRESSIVE_NOMINATION             (1)
+#define NR_ICE_CTX_FLAGS_LITE                              (1<<1)
+#define NR_ICE_CTX_FLAGS_RELAY_ONLY                        (1<<2)
+#define NR_ICE_CTX_FLAGS_DISABLE_HOST_CANDIDATES           (1<<3)
+#define NR_ICE_CTX_FLAGS_ONLY_DEFAULT_ADDRS                (1<<4)
+#define NR_ICE_CTX_FLAGS_ONLY_PROXY                        (1<<5)
+#define NR_ICE_CTX_FLAGS_OBFUSCATE_HOST_ADDRESSES          (1<<6)
+
+void nr_ice_ctx_add_flags(nr_ice_ctx *ctx, UINT4 flags);
+void nr_ice_ctx_remove_flags(nr_ice_ctx *ctx, UINT4 flags);
+void nr_ice_ctx_destroy(nr_ice_ctx** ctxp);
+int nr_ice_set_local_addresses(nr_ice_ctx *ctx, nr_local_addr* stun_addrs, int stun_addr_ct);
+int nr_ice_set_target_for_default_local_address_lookup(nr_ice_ctx *ctx, const char *target_ip, UINT2 target_port);
+int nr_ice_gather(nr_ice_ctx *ctx, NR_async_cb done_cb, void *cb_arg);
+int nr_ice_add_candidate(nr_ice_ctx *ctx, nr_ice_candidate *cand);
+void nr_ice_gather_finished_cb(NR_SOCKET s, int h, void *cb_arg);
+int nr_ice_add_media_stream(nr_ice_ctx *ctx,const char *label,const char *ufrag,const char *pwd,int components, nr_ice_media_stream **streamp);
+int nr_ice_remove_media_stream(nr_ice_ctx *ctx,nr_ice_media_stream **streamp);
+int nr_ice_get_global_attributes(nr_ice_ctx *ctx,char ***attrsp, int *attrctp);
+int nr_ice_ctx_deliver_packet(nr_ice_ctx *ctx, nr_ice_component *comp, nr_transport_addr *source_addr, UCHAR *data, int len);
+int nr_ice_ctx_is_known_id(nr_ice_ctx *ctx, UCHAR id[12]);
+int nr_ice_ctx_remember_id(nr_ice_ctx *ctx, nr_stun_message *msg);
+int nr_ice_ctx_finalize(nr_ice_ctx *ctx, nr_ice_peer_ctx *pctx);
+int nr_ice_ctx_set_stun_servers(nr_ice_ctx *ctx,nr_ice_stun_server *servers, int ct);
+int nr_ice_ctx_set_turn_servers(nr_ice_ctx *ctx,nr_ice_turn_server *servers, int ct);
+int nr_ice_ctx_copy_turn_servers(nr_ice_ctx *ctx, nr_ice_turn_server *servers, int ct);
+int nr_ice_ctx_set_resolver(nr_ice_ctx *ctx, nr_resolver *resolver);
+int nr_ice_ctx_set_interface_prioritizer(nr_ice_ctx *ctx, nr_interface_prioritizer *prioritizer);
+void nr_ice_ctx_set_socket_factory(nr_ice_ctx *ctx, nr_socket_factory *factory);
+int nr_ice_ctx_set_trickle_cb(nr_ice_ctx *ctx, nr_ice_trickle_candidate_cb cb, void *cb_arg);
+int nr_ice_ctx_hide_candidate(nr_ice_ctx *ctx, nr_ice_candidate *cand);
+int nr_ice_get_new_ice_ufrag(char** ufrag);
+int nr_ice_get_new_ice_pwd(char** pwd);
+
+#define NR_ICE_MAX_ATTRIBUTE_SIZE 256
+
+extern int LOG_ICE;
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_handler.h b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_handler.h
new file mode 100644
index 0000000000..5a0690adad
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_handler.h
@@ -0,0 +1,84 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _ice_h
+#define _ice_h
+#ifdef __cplusplus
+using namespace std;
+extern "C" {
+#endif /* __cplusplus */
+
+typedef struct nr_ice_handler_vtbl_ {
+  /* The checks on this media stream are done. The handler needs to
+     select a single pair to proceed with (regular nomination).
+     Once this returns the check starts and the pair can be
+     written on. Use nr_ice_candidate_pair_select() to perform the
+     selection.
+     TODO: !ekr! is this right?
+  */
+  int (*select_pair)(void *obj,nr_ice_media_stream *stream,
+int component_id, nr_ice_cand_pair **potentials,int potential_ct);
+
+  /* This media stream is ready to read/write (aggressive nomination).
+     May be called again if the nominated pair changes due to
+     ICE instability. TODO: !ekr! think about this
+  */
+  int (*stream_ready)(void *obj, nr_ice_media_stream *stream);
+
+  /* This media stream has failed */
+  int (*stream_failed)(void *obj, nr_ice_media_stream *stream);
+
+  /* ICE is connected for this peer ctx */
+  int (*ice_connected)(void *obj, nr_ice_peer_ctx *pctx);
+
+  /* A message was delivered to us */
+  int (*msg_recvd)(void *obj, nr_ice_peer_ctx *pctx, nr_ice_media_stream *stream, int component_id, UCHAR *msg, int len);
+
+  /* ICE has started checking. */
+  int (*ice_checking)(void *obj, nr_ice_peer_ctx *pctx);
+
+  /* ICE detected a (temporary?) disconnect. */
+  int (*ice_disconnected)(void *obj, nr_ice_peer_ctx *pctx);
+} nr_ice_handler_vtbl;
+
+typedef struct nr_ice_handler_ {
+  void *obj;
+  nr_ice_handler_vtbl *vtbl;
+} nr_ice_handler;
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_media_stream.c b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_media_stream.c
new file mode 100644
index 0000000000..62bfbad629
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_media_stream.c
@@ -0,0 +1,1087 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <string.h>
+#include <assert.h>
+#include <nr_api.h>
+#include <r_assoc.h>
+#include <async_timer.h>
+#include "ice_util.h"
+#include "ice_ctx.h"
+
+static char *nr_ice_media_stream_states[]={"INVALID",
+  "UNPAIRED","FROZEN","ACTIVE","CONNECTED","FAILED"
+};
+
+int nr_ice_media_stream_set_state(nr_ice_media_stream *str, int state);
+
+int nr_ice_media_stream_create(nr_ice_ctx *ctx,const char *label,const char *ufrag,const char *pwd,int components, nr_ice_media_stream **streamp)
+  {
+    int r,_status;
+    nr_ice_media_stream *stream=0;
+    nr_ice_component *comp=0;
+    int i;
+
+    if(!(stream=RCALLOC(sizeof(nr_ice_media_stream))))
+      ABORT(R_NO_MEMORY);
+
+    if(!(stream->label=r_strdup(label)))
+      ABORT(R_NO_MEMORY);
+
+    if(!(stream->ufrag=r_strdup(ufrag)))
+      ABORT(R_NO_MEMORY);
+
+    if(!(stream->pwd=r_strdup(pwd)))
+      ABORT(R_NO_MEMORY);
+
+    stream->ctx=ctx;
+
+    STAILQ_INIT(&stream->components);
+    for(i=0;i<components;i++){
+      /* component-id must be > 0, so increment by 1 */
+      if(r=nr_ice_component_create(stream, i+1, &comp))
+        ABORT(r);
+
+    }
+
+    TAILQ_INIT(&stream->check_list);
+    TAILQ_INIT(&stream->trigger_check_queue);
+
+    stream->disconnected = 0;
+    stream->component_ct=components;
+    stream->ice_state = NR_ICE_MEDIA_STREAM_UNPAIRED;
+    stream->obsolete = 0;
+    stream->r2l_user = 0;
+    stream->l2r_user = 0;
+    stream->flags = ctx->flags;
+    if(ctx->stun_server_ct_cfg) {
+      if(!(stream->stun_servers=RCALLOC(sizeof(nr_ice_stun_server)*(ctx->stun_server_ct_cfg))))
+        ABORT(R_NO_MEMORY);
+
+      memcpy(stream->stun_servers,ctx->stun_servers_cfg,sizeof(nr_ice_stun_server)*(ctx->stun_server_ct_cfg));
+      stream->stun_server_ct = ctx->stun_server_ct_cfg;
+    }
+
+    if(ctx->turn_server_ct_cfg) {
+      if(!(stream->turn_servers=RCALLOC(sizeof(nr_ice_turn_server)*(ctx->turn_server_ct_cfg))))
+        ABORT(R_NO_MEMORY);
+
+      for(int i = 0; i < ctx->turn_server_ct_cfg; ++i) {
+        nr_ice_turn_server *dst = &stream->turn_servers[i];
+        nr_ice_turn_server *src = &ctx->turn_servers_cfg[i];
+        memcpy(&dst->turn_server, &src->turn_server, sizeof(nr_ice_stun_server));
+        dst->username = r_strdup(src->username);
+        r_data_create(&dst->password, src->password->data, src->password->len);
+      }
+      stream->turn_server_ct = ctx->turn_server_ct_cfg;
+    }
+
+    r_log(LOG_ICE,LOG_DEBUG,"ICE-STREAM(%s): flags %d",stream->label,stream->flags);
+    *streamp=stream;
+
+    _status=0;
+  abort:
+    if(_status){
+      nr_ice_media_stream_destroy(&stream);
+    }
+    return(_status);
+  }
+
+int nr_ice_media_stream_destroy(nr_ice_media_stream **streamp)
+  {
+    nr_ice_media_stream *stream;
+    nr_ice_component *c1,*c2;
+    nr_ice_cand_pair *p1,*p2;
+    if(!streamp || !*streamp)
+      return(0);
+
+    stream=*streamp;
+    *streamp=0;
+
+    STAILQ_FOREACH_SAFE(c1, &stream->components, entry, c2){
+      STAILQ_REMOVE(&stream->components,c1,nr_ice_component_,entry);
+      nr_ice_component_destroy(&c1);
+    }
+
+    /* Note: all the entries from the trigger check queue are held in here as
+     *       well, so we only clean up the super set. */
+    TAILQ_FOREACH_SAFE(p1, &stream->check_list, check_queue_entry, p2){
+      TAILQ_REMOVE(&stream->check_list,p1,check_queue_entry);
+      nr_ice_candidate_pair_destroy(&p1);
+    }
+
+    RFREE(stream->label);
+
+    RFREE(stream->ufrag);
+    RFREE(stream->pwd);
+    RFREE(stream->r2l_user);
+    RFREE(stream->l2r_user);
+    r_data_zfree(&stream->r2l_pass);
+    r_data_zfree(&stream->l2r_pass);
+
+    RFREE(stream->stun_servers);
+    for (int i = 0; i < stream->turn_server_ct; i++) {
+        RFREE(stream->turn_servers[i].username);
+        r_data_destroy(&stream->turn_servers[i].password);
+    }
+    RFREE(stream->turn_servers);
+
+    if(stream->timer)
+      NR_async_timer_cancel(stream->timer);
+
+    RFREE(stream);
+
+    return(0);
+  }
+
+int nr_ice_media_stream_initialize(nr_ice_ctx *ctx, nr_ice_media_stream *stream)
+  {
+    int r,_status;
+    nr_ice_component *comp;
+
+    assert(!stream->obsolete);
+
+    comp=STAILQ_FIRST(&stream->components);
+    while(comp){
+      if(r=nr_ice_component_initialize(ctx,comp))
+        ABORT(r);
+      comp=STAILQ_NEXT(comp,entry);
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+
+int nr_ice_media_stream_get_attributes(nr_ice_media_stream *stream, char ***attrsp, int *attrctp)
+  {
+    int attrct=2;
+    nr_ice_component *comp;
+    char **attrs=0;
+    int index=0;
+    nr_ice_candidate *cand;
+    int r,_status;
+    char *tmp=0;
+
+    *attrctp=0;
+
+    /* First find out how many attributes we need */
+    comp=STAILQ_FIRST(&stream->components);
+    while(comp){
+      if (comp->state != NR_ICE_COMPONENT_DISABLED) {
+        cand = TAILQ_FIRST(&comp->candidates);
+        while(cand){
+          if (!nr_ice_ctx_hide_candidate(stream->ctx, cand)) {
+            ++attrct;
+          }
+
+          cand = TAILQ_NEXT(cand, entry_comp);
+        }
+      }
+      comp=STAILQ_NEXT(comp,entry);
+    }
+
+    /* Make the array we'll need */
+    if(!(attrs=RCALLOC(sizeof(char *)*attrct)))
+      ABORT(R_NO_MEMORY);
+    for(index=0;index<attrct;index++){
+      if(!(attrs[index]=RMALLOC(NR_ICE_MAX_ATTRIBUTE_SIZE)))
+        ABORT(R_NO_MEMORY);
+    }
+
+    index=0;
+    /* Now format the attributes */
+    comp=STAILQ_FIRST(&stream->components);
+    while(comp){
+      if (comp->state != NR_ICE_COMPONENT_DISABLED) {
+        nr_ice_candidate *cand;
+
+        cand=TAILQ_FIRST(&comp->candidates);
+        while(cand){
+          if (!nr_ice_ctx_hide_candidate(stream->ctx, cand)) {
+            assert(index < attrct);
+
+            if (index >= attrct)
+              ABORT(R_INTERNAL);
+
+            if(r=nr_ice_format_candidate_attribute(cand, attrs[index],NR_ICE_MAX_ATTRIBUTE_SIZE, 0))
+              ABORT(r);
+
+            index++;
+          }
+
+          cand=TAILQ_NEXT(cand,entry_comp);
+        }
+      }
+      comp=STAILQ_NEXT(comp,entry);
+    }
+
+    /* Now, ufrag and pwd */
+    if(!(tmp=RMALLOC(100)))
+      ABORT(R_NO_MEMORY);
+    snprintf(tmp,100,"ice-ufrag:%s",stream->ufrag);
+    attrs[index++]=tmp;
+
+    if(!(tmp=RMALLOC(100)))
+      ABORT(R_NO_MEMORY);
+    snprintf(tmp,100,"ice-pwd:%s",stream->pwd);
+    attrs[index++]=tmp;
+
+    *attrsp=attrs;
+    *attrctp=attrct;
+
+    _status=0;
+  abort:
+    if(_status){
+      if(attrs){
+        for(index=0;index<attrct;index++){
+          RFREE(attrs[index]);
+        }
+        RFREE(attrs);
+      }
+    }
+    return(_status);
+  }
+
+/* Get a default candidate per 4.1.4 */
+int nr_ice_media_stream_get_default_candidate(nr_ice_media_stream *stream, int component, nr_ice_candidate **candp)
+  {
+    int r,_status;
+    nr_ice_component *comp;
+
+    comp=STAILQ_FIRST(&stream->components);
+    while(comp){
+      if (comp->component_id == component)
+        break;
+
+      comp=STAILQ_NEXT(comp,entry);
+    }
+
+    if (!comp)
+      ABORT(R_NOT_FOUND);
+
+    /* If there aren't any IPV4 candidates, try IPV6 */
+    if((r=nr_ice_component_get_default_candidate(comp, candp, NR_IPV4)) &&
+       (r=nr_ice_component_get_default_candidate(comp, candp, NR_IPV6))) {
+      ABORT(r);
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+
+int nr_ice_media_stream_pair_candidates(nr_ice_peer_ctx *pctx,nr_ice_media_stream *lstream,nr_ice_media_stream *pstream)
+  {
+    int r,_status;
+    nr_ice_component *pcomp,*lcomp;
+
+    pcomp=STAILQ_FIRST(&pstream->components);
+    lcomp=STAILQ_FIRST(&lstream->components);
+    while(pcomp){
+      if ((lcomp->state != NR_ICE_COMPONENT_DISABLED) &&
+          (pcomp->state != NR_ICE_COMPONENT_DISABLED)) {
+        if(r=nr_ice_component_pair_candidates(pctx,lcomp,pcomp))
+          ABORT(r);
+      }
+
+      lcomp=STAILQ_NEXT(lcomp,entry);
+      pcomp=STAILQ_NEXT(pcomp,entry);
+    };
+
+    if (pstream->ice_state == NR_ICE_MEDIA_STREAM_UNPAIRED) {
+      nr_ice_media_stream_set_state(pstream, NR_ICE_MEDIA_STREAM_CHECKS_FROZEN);
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_ice_media_stream_service_pre_answer_requests(nr_ice_peer_ctx *pctx, nr_ice_media_stream *lstream, nr_ice_media_stream *pstream, int *serviced)
+  {
+    nr_ice_component *pcomp;
+    int r,_status;
+    char *user = 0;
+
+    if (serviced)
+      *serviced = 0;
+
+    pcomp=STAILQ_FIRST(&pstream->components);
+    while(pcomp){
+      int serviced_inner=0;
+
+      /* Flush all the pre-answer requests */
+      if(r=nr_ice_component_service_pre_answer_requests(pctx, pcomp, pstream->r2l_user, &serviced_inner))
+        ABORT(r);
+      if (serviced)
+        *serviced += serviced_inner;
+
+      pcomp=STAILQ_NEXT(pcomp,entry);
+    }
+
+    _status=0;
+   abort:
+    RFREE(user);
+    return(_status);
+  }
+
+/* S 5.8 -- run the first pair from the triggered check queue (even after
+ * checks have completed S 8.1.2) or run the highest priority WAITING pair or
+ * if not available FROZEN pair from the check queue */
+static void nr_ice_media_stream_check_timer_cb(NR_SOCKET s, int h, void *cb_arg)
+  {
+    int r,_status;
+    nr_ice_media_stream *stream=cb_arg;
+    nr_ice_cand_pair *pair = 0;
+    int timer_multiplier=stream->pctx->active_streams ? stream->pctx->active_streams : 1;
+    int timer_val=stream->pctx->ctx->Ta*timer_multiplier;
+
+    assert(timer_val>0);
+
+    r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s): check timer expired for media stream %s",stream->pctx->label,stream->label);
+    stream->timer=0;
+
+    /* The trigger check queue has the highest priority */
+    pair=TAILQ_FIRST(&stream->trigger_check_queue);
+    while(pair){
+      if(pair->state==NR_ICE_PAIR_STATE_WAITING){
+        /* Remove the pair from he trigger check queue */
+        r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s): Removing pair from trigger check queue %s",stream->pctx->label,pair->as_string);
+        TAILQ_REMOVE(&stream->trigger_check_queue,pair,triggered_check_queue_entry);
+        break;
+      }
+      pair=TAILQ_NEXT(pair,triggered_check_queue_entry);
+    }
+
+    if (stream->ice_state != NR_ICE_MEDIA_STREAM_CHECKS_CONNECTED) {
+      if(!pair){
+        /* Find the highest priority WAITING check and move it to RUNNING */
+        pair=TAILQ_FIRST(&stream->check_list);
+        while(pair){
+          if(pair->state==NR_ICE_PAIR_STATE_WAITING)
+            break;
+          pair=TAILQ_NEXT(pair,check_queue_entry);
+        }
+      }
+
+      /* Hmmm... No WAITING. Let's look for FROZEN */
+      if(!pair){
+        pair=TAILQ_FIRST(&stream->check_list);
+
+        while(pair){
+          if(pair->state==NR_ICE_PAIR_STATE_FROZEN){
+            if(r=nr_ice_candidate_pair_unfreeze(stream->pctx,pair))
+              ABORT(r);
+            break;
+          }
+          pair=TAILQ_NEXT(pair,check_queue_entry);
+        }
+      }
+    }
+
+    if(pair){
+      nr_ice_candidate_pair_start(pair->pctx,pair); /* Ignore failures */
+      NR_ASYNC_TIMER_SET(timer_val,nr_ice_media_stream_check_timer_cb,cb_arg,&stream->timer);
+    }
+    else {
+      r_log(LOG_ICE,LOG_INFO,"ICE-PEER(%s): no FROZEN/WAITING pairs for %s",stream->pctx->label,stream->label);
+    }
+
+    _status=0;
+  abort:
+    if (_status) {
+      // cb doesn't return anything, but we should probably log that we aborted
+      // This also quiets the unused variable warnings.
+      r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s): check timer cb for media stream %s abort with status: %d",
+        stream->pctx->label,stream->label, _status);
+    }
+    return;
+  }
+
+/* Start checks for this media stream (aka check list) */
+int nr_ice_media_stream_start_checks(nr_ice_peer_ctx *pctx, nr_ice_media_stream *stream)
+  {
+    int r,_status;
+
+    /* Don't start the check timer if the stream is failed */
+    if (stream->ice_state == NR_ICE_MEDIA_STREAM_CHECKS_FAILED) {
+      assert(0);
+      ABORT(R_INTERNAL);
+    }
+
+    if (stream->local_stream->obsolete) {
+      assert(0);
+      ABORT(R_INTERNAL);
+    }
+
+    /* Even if the stream is completed already remote can still create a new
+     * triggered check request which needs to fire, but not change our stream
+     * state. */
+    if (stream->ice_state != NR_ICE_MEDIA_STREAM_CHECKS_CONNECTED) {
+      if(r=nr_ice_media_stream_set_state(stream,NR_ICE_MEDIA_STREAM_CHECKS_ACTIVE)) {
+        ABORT(r);
+      }
+    }
+
+    if (!stream->timer) {
+      r_log(LOG_ICE,LOG_INFO,"ICE-PEER(%s)/ICE-STREAM(%s): Starting check timer for stream.",pctx->label,stream->label);
+      nr_ice_media_stream_check_timer_cb(0,0,stream);
+    }
+
+    nr_ice_peer_ctx_stream_started_checks(pctx, stream);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+/* Start checks for this media stream (aka check list) S 5.7 */
+int nr_ice_media_stream_unfreeze_pairs(nr_ice_peer_ctx *pctx, nr_ice_media_stream *stream)
+  {
+    int r,_status;
+    r_assoc *assoc=0;
+    nr_ice_cand_pair *pair=0;
+
+    /* Already seen assoc */
+    if(r=r_assoc_create(&assoc,r_assoc_crc32_hash_compute,5))
+      ABORT(r);
+
+    /* S 5.7.4. Set the highest priority pairs in each foundation to WAITING */
+    pair=TAILQ_FIRST(&stream->check_list);
+    while(pair){
+      void *v;
+
+      if(r=r_assoc_fetch(assoc,pair->foundation,strlen(pair->foundation),&v)){
+        if(r!=R_NOT_FOUND)
+          ABORT(r);
+        if(r=nr_ice_candidate_pair_unfreeze(pctx,pair))
+          ABORT(r);
+
+        if(r=r_assoc_insert(assoc,pair->foundation,strlen(pair->foundation),
+          0,0,0,R_ASSOC_NEW))
+          ABORT(r);
+      }
+
+      /* Already exists... fall through */
+      pair=TAILQ_NEXT(pair,check_queue_entry);
+    }
+
+    _status=0;
+  abort:
+    r_assoc_destroy(&assoc);
+    return(_status);
+  }
+
+static int nr_ice_media_stream_unfreeze_pairs_match(nr_ice_media_stream *stream, char *foundation)
+  {
+    nr_ice_cand_pair *pair;
+    int r,_status;
+    int unfroze=0;
+
+    pair=TAILQ_FIRST(&stream->check_list);
+    while(pair){
+      if(pair->state==NR_ICE_PAIR_STATE_FROZEN &&
+        !strcmp(foundation,pair->foundation)){
+        if(r=nr_ice_candidate_pair_unfreeze(stream->pctx,pair))
+          ABORT(r);
+        unfroze++;
+      }
+      pair=TAILQ_NEXT(pair,check_queue_entry);
+    }
+
+    if(!unfroze)
+      return(R_NOT_FOUND);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+/* S 7.1.2.2 */
+int nr_ice_media_stream_unfreeze_pairs_foundation(nr_ice_media_stream *stream, char *foundation)
+  {
+    int r,_status;
+    nr_ice_media_stream *str;
+
+    /* 1. Unfreeze all frozen pairs with the same foundation
+       in this stream */
+    if(r=nr_ice_media_stream_unfreeze_pairs_match(stream,foundation)){
+      if(r!=R_NOT_FOUND)
+        ABORT(r);
+    }
+
+    /* Now go through the check lists for the other streams */
+    str=STAILQ_FIRST(&stream->pctx->peer_streams);
+    while(str){
+      if(str!=stream && !str->local_stream->obsolete){
+        switch(str->ice_state){
+          case NR_ICE_MEDIA_STREAM_CHECKS_ACTIVE:
+            /* Unfreeze matching pairs */
+            if(r=nr_ice_media_stream_unfreeze_pairs_match(str,foundation)){
+              if(r!=R_NOT_FOUND)
+                ABORT(r);
+            }
+            break;
+          case NR_ICE_MEDIA_STREAM_CHECKS_FROZEN:
+            /* Unfreeze matching pairs if any */
+            r=nr_ice_media_stream_unfreeze_pairs_match(str,foundation);
+            if(r){
+              if(r!=R_NOT_FOUND)
+                ABORT(r);
+
+              /* OK, no matching pairs: execute the algorithm from 5.7
+                 for this stream */
+              if(r=nr_ice_media_stream_unfreeze_pairs(str->pctx,str))
+                ABORT(r);
+            }
+            if(r=nr_ice_media_stream_start_checks(str->pctx,str))
+              ABORT(r);
+
+            break;
+          default:
+            break;
+        }
+      }
+
+      str=STAILQ_NEXT(str,entry);
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+
+void nr_ice_media_stream_dump_state(nr_ice_peer_ctx *pctx, nr_ice_media_stream *stream, int log_level)
+  {
+    nr_ice_cand_pair *pair;
+    nr_ice_component *comp;
+
+    if(stream->local_stream){
+      /* stream has a corresponding local_stream */
+      nr_ice_media_stream_dump_state(stream->local_stream->pctx,stream->local_stream, log_level);
+      r_log(LOG_ICE,log_level,"ICE-PEER(%s)/STREAM(%s): state dump", stream->pctx->label, stream->label);
+    } else {
+      r_log(LOG_ICE,log_level,"ICE(%s)/STREAM(%s): state dump", stream->ctx->label, stream->label);
+    }
+
+    pair=TAILQ_FIRST(&stream->check_list);
+    while(pair){
+      nr_ice_candidate_pair_dump_state(pair, log_level);
+      pair=TAILQ_NEXT(pair,check_queue_entry);
+    }
+
+    comp=STAILQ_FIRST(&stream->components);
+    while(comp){
+      nr_ice_component_dump_state(comp, log_level);
+      comp=STAILQ_NEXT(comp,entry);
+    }
+  }
+
+int nr_ice_media_stream_set_state(nr_ice_media_stream *str, int state)
+  {
+    /* Make no-change a no-op */
+    if (state == str->ice_state)
+      return 0;
+
+    assert((size_t)state < sizeof(nr_ice_media_stream_states)/sizeof(char *));
+    assert((size_t)str->ice_state < sizeof(nr_ice_media_stream_states)/sizeof(char *));
+
+    r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s): stream %s state %s->%s",
+      str->pctx->label,str->label,
+      nr_ice_media_stream_states[str->ice_state],
+      nr_ice_media_stream_states[state]);
+
+    if(state == NR_ICE_MEDIA_STREAM_CHECKS_ACTIVE)
+      str->pctx->active_streams++;
+    if(str->ice_state == NR_ICE_MEDIA_STREAM_CHECKS_ACTIVE)
+      str->pctx->active_streams--;
+
+    r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s): %d active streams",
+      str->pctx->label, str->pctx->active_streams);
+
+    str->ice_state=state;
+    if (state == NR_ICE_MEDIA_STREAM_CHECKS_FAILED) {
+      nr_ice_media_stream_dump_state(str->pctx,str,LOG_ERR);
+    }
+
+    return(0);
+  }
+
+void nr_ice_media_stream_stop_checking(nr_ice_media_stream *str)
+  {
+    nr_ice_cand_pair *p;
+    nr_ice_component *comp;
+
+    /* Cancel candidate pairs */
+    p=TAILQ_FIRST(&str->check_list);
+    while(p){
+      nr_ice_candidate_pair_cancel(p->pctx,p,0);
+      p=TAILQ_NEXT(p,check_queue_entry);
+    }
+
+    if(str->timer) {
+      NR_async_timer_cancel(str->timer);
+      str->timer = 0;
+    }
+
+    /* Cancel consent timers in case it is running already */
+    comp=STAILQ_FIRST(&str->components);
+    while(comp){
+      nr_ice_component_consent_destroy(comp);
+      comp=STAILQ_NEXT(comp,entry);
+    }
+  }
+
+void nr_ice_media_stream_set_obsolete(nr_ice_media_stream *str)
+  {
+    nr_ice_component *c1,*c2;
+    str->obsolete = 1;
+
+    STAILQ_FOREACH_SAFE(c1, &str->components, entry, c2){
+      nr_ice_component_stop_gathering(c1);
+    }
+
+    nr_ice_media_stream_stop_checking(str);
+  }
+
+int nr_ice_media_stream_is_done_gathering(nr_ice_media_stream *str)
+  {
+    nr_ice_component *comp;
+    comp=STAILQ_FIRST(&str->components);
+    while(comp){
+      if(!nr_ice_component_is_done_gathering(comp)) {
+        return 0;
+      }
+      comp=STAILQ_NEXT(comp,entry);
+    }
+    return 1;
+  }
+
+void nr_ice_media_stream_refresh_consent_all(nr_ice_media_stream *stream)
+  {
+    nr_ice_component *comp;
+
+    comp=STAILQ_FIRST(&stream->components);
+    while(comp){
+      if(comp->disconnected) {
+        nr_ice_component_refresh_consent_now(comp);
+      }
+
+      comp=STAILQ_NEXT(comp,entry);
+    }
+  }
+
+void nr_ice_media_stream_disconnect_all_components(nr_ice_media_stream *stream)
+  {
+    nr_ice_component *comp;
+
+    comp=STAILQ_FIRST(&stream->components);
+    while(comp){
+      comp->disconnected = 1;
+
+      comp=STAILQ_NEXT(comp,entry);
+    }
+  }
+
+void nr_ice_media_stream_set_disconnected(nr_ice_media_stream *stream, int disconnected)
+  {
+    if (stream->disconnected == disconnected) {
+      return;
+    }
+
+    if (stream->ice_state != NR_ICE_MEDIA_STREAM_CHECKS_CONNECTED) {
+      return;
+    }
+    stream->disconnected = disconnected;
+
+    if (disconnected == NR_ICE_MEDIA_STREAM_DISCONNECTED) {
+      if (!stream->local_stream->obsolete) {
+        nr_ice_peer_ctx_disconnected(stream->pctx);
+      }
+    } else {
+      nr_ice_peer_ctx_check_if_connected(stream->pctx);
+    }
+  }
+
+int nr_ice_media_stream_check_if_connected(nr_ice_media_stream *stream)
+  {
+    nr_ice_component *comp;
+
+    comp=STAILQ_FIRST(&stream->components);
+    while(comp){
+      if((comp->state != NR_ICE_COMPONENT_DISABLED) &&
+         (comp->local_component->state != NR_ICE_COMPONENT_DISABLED) &&
+         comp->disconnected)
+        break;
+
+      comp=STAILQ_NEXT(comp,entry);
+    }
+
+    /* At least one disconnected component */
+    if(comp)
+      goto done;
+
+    nr_ice_media_stream_set_disconnected(stream, NR_ICE_MEDIA_STREAM_CONNECTED);
+
+  done:
+    return(0);
+  }
+
+/* S OK, this component has a nominated. If every component has a nominated,
+   the stream is ready */
+void nr_ice_media_stream_component_nominated(nr_ice_media_stream *stream,nr_ice_component *component)
+  {
+    nr_ice_component *comp;
+
+    comp=STAILQ_FIRST(&stream->components);
+    while(comp){
+      if((comp->state != NR_ICE_COMPONENT_DISABLED) &&
+         (comp->local_component->state != NR_ICE_COMPONENT_DISABLED) &&
+         !comp->nominated)
+        break;
+
+      comp=STAILQ_NEXT(comp,entry);
+    }
+
+    /* At least one un-nominated component */
+    if(comp)
+      return;
+
+    /* All done... */
+    r_log(LOG_ICE,LOG_INFO,"ICE-PEER(%s)/ICE-STREAM(%s): all active components have nominated candidate pairs",stream->pctx->label,stream->label);
+    nr_ice_media_stream_set_state(stream,NR_ICE_MEDIA_STREAM_CHECKS_CONNECTED);
+
+    /* Cancel our timer */
+    if(stream->timer){
+      NR_async_timer_cancel(stream->timer);
+      stream->timer=0;
+    }
+
+    if (stream->pctx->handler && !stream->local_stream->obsolete) {
+      stream->pctx->handler->vtbl->stream_ready(stream->pctx->handler->obj,stream->local_stream);
+    }
+
+    /* Now tell the peer_ctx that we're connected */
+    nr_ice_peer_ctx_check_if_connected(stream->pctx);
+  }
+
+void nr_ice_media_stream_component_failed(nr_ice_media_stream *stream,nr_ice_component *component)
+  {
+    component->state=NR_ICE_COMPONENT_FAILED;
+
+    /* at least one component failed in this media stream, so the entire
+     * media stream is marked failed */
+
+    nr_ice_media_stream_set_state(stream,NR_ICE_MEDIA_STREAM_CHECKS_FAILED);
+
+    nr_ice_media_stream_stop_checking(stream);
+
+    if (stream->pctx->handler && !stream->local_stream->obsolete) {
+      stream->pctx->handler->vtbl->stream_failed(stream->pctx->handler->obj,stream->local_stream);
+    }
+
+    /* Now tell the peer_ctx that we've failed */
+    nr_ice_peer_ctx_check_if_connected(stream->pctx);
+  }
+
+int nr_ice_media_stream_get_best_candidate(nr_ice_media_stream *str, int component, nr_ice_candidate **candp)
+  {
+    nr_ice_candidate *cand;
+    nr_ice_candidate *best_cand=0;
+    nr_ice_component *comp;
+    int r,_status;
+
+    if(r=nr_ice_media_stream_find_component(str,component,&comp))
+      ABORT(r);
+
+    cand=TAILQ_FIRST(&comp->candidates);
+    while(cand){
+      if(cand->state==NR_ICE_CAND_STATE_INITIALIZED){
+        if(!best_cand || (cand->priority>best_cand->priority))
+          best_cand=cand;
+
+      }
+      cand=TAILQ_NEXT(cand,entry_comp);
+    }
+
+    if(!best_cand)
+      ABORT(R_NOT_FOUND);
+
+    *candp=best_cand;
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+
+/* OK, we have the stream the user created, but that reflects the base
+   ICE ctx, not the peer_ctx. So, find the related stream in the pctx,
+   and then find the component */
+int nr_ice_media_stream_find_component(nr_ice_media_stream *str, int comp_id, nr_ice_component **compp)
+  {
+    int _status;
+    nr_ice_component *comp;
+
+    comp=STAILQ_FIRST(&str->components);
+    while(comp){
+      if(comp->component_id==comp_id)
+        break;
+
+      comp=STAILQ_NEXT(comp,entry);
+    }
+    if(!comp)
+      ABORT(R_NOT_FOUND);
+
+    *compp=comp;
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_ice_media_stream_send(nr_ice_peer_ctx *pctx, nr_ice_media_stream *str, int component, UCHAR *data, int len)
+  {
+    int r,_status;
+    nr_ice_component *comp;
+
+    /* First find the peer component */
+    if(r=nr_ice_peer_ctx_find_component(pctx, str, component, &comp))
+      ABORT(r);
+
+    /* Do we have an active pair yet? We should... */
+    if(!comp->active)
+      ABORT(R_NOT_FOUND);
+
+    /* Does fresh ICE consent exist? */
+    if(!comp->can_send)
+      ABORT(R_FAILED);
+
+    /* OK, write to that pair, which means:
+       1. Use the socket on our local side.
+       2. Use the address on the remote side
+    */
+    if(r=nr_socket_sendto(comp->active->local->osock,data,len,0,
+                          &comp->active->remote->addr)) {
+      if ((r==R_IO_ERROR) || (r==R_EOD)) {
+        nr_ice_component_disconnected(comp);
+      }
+      ABORT(r);
+    }
+
+    // accumulate the sent bytes for the active candidate pair
+    comp->active->bytes_sent += len;
+    gettimeofday(&comp->active->last_sent, 0);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+/* Returns R_REJECTED if the component is unpaired or has been disabled. */
+int nr_ice_media_stream_get_active(nr_ice_peer_ctx *pctx, nr_ice_media_stream *str, int component, nr_ice_candidate **local, nr_ice_candidate **remote)
+  {
+    int r,_status;
+    nr_ice_component *comp;
+
+    /* First find the peer component */
+    if(r=nr_ice_peer_ctx_find_component(pctx, str, component, &comp))
+      ABORT(r);
+
+    if (comp->state == NR_ICE_COMPONENT_UNPAIRED ||
+        comp->state == NR_ICE_COMPONENT_DISABLED)
+      ABORT(R_REJECTED);
+
+    if(!comp->active)
+      ABORT(R_NOT_FOUND);
+
+    if (local) *local = comp->active->local;
+    if (remote) *remote = comp->active->remote;
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_ice_media_stream_addrs(nr_ice_peer_ctx *pctx, nr_ice_media_stream *str, int component, nr_transport_addr *local, nr_transport_addr *remote)
+  {
+    int r,_status;
+    nr_ice_component *comp;
+
+    /* First find the peer component */
+    if(r=nr_ice_peer_ctx_find_component(pctx, str, component, &comp))
+      ABORT(r);
+
+    /* Do we have an active pair yet? We should... */
+    if(!comp->active)
+      ABORT(R_BAD_ARGS);
+
+    /* Use the socket on our local side */
+    if(r=nr_socket_getaddr(comp->active->local->osock,local))
+      ABORT(r);
+
+    /* Use the address on the remote side */
+    if(r=nr_transport_addr_copy(remote,&comp->active->remote->addr))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+
+
+int nr_ice_media_stream_finalize(nr_ice_media_stream *lstr,nr_ice_media_stream *rstr)
+  {
+    nr_ice_component *lcomp,*rcomp;
+
+    r_log(LOG_ICE,LOG_DEBUG,"Finalizing media stream %s, peer=%s",lstr->label,
+      rstr?rstr->label:"NONE");
+
+    lcomp=STAILQ_FIRST(&lstr->components);
+    if(rstr)
+      rcomp=STAILQ_FIRST(&rstr->components);
+    else
+      rcomp=0;
+
+    while(lcomp){
+      nr_ice_component_finalize(lcomp,rcomp);
+
+      lcomp=STAILQ_NEXT(lcomp,entry);
+      if(rcomp){
+        rcomp=STAILQ_NEXT(rcomp,entry);
+      }
+    }
+
+    return(0);
+  }
+
+int nr_ice_media_stream_pair_new_trickle_candidate(nr_ice_peer_ctx *pctx, nr_ice_media_stream *pstream, nr_ice_candidate *cand)
+  {
+    int r,_status;
+    nr_ice_component *comp;
+
+    if ((r=nr_ice_media_stream_find_component(pstream, cand->component_id, &comp)))
+      ABORT(R_NOT_FOUND);
+
+    if (r=nr_ice_component_pair_candidate(pctx, comp, cand, 1))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_ice_media_stream_get_consent_status(nr_ice_media_stream *stream, int
+component_id, int *can_send, struct timeval *ts)
+  {
+    int r,_status;
+    nr_ice_component *comp;
+
+    if ((r=nr_ice_media_stream_find_component(stream, component_id, &comp)))
+      ABORT(r);
+
+    *can_send = comp->can_send;
+    ts->tv_sec = comp->consent_last_seen.tv_sec;
+    ts->tv_usec = comp->consent_last_seen.tv_usec;
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_ice_media_stream_disable_component(nr_ice_media_stream *stream, int component_id)
+  {
+    int r,_status;
+    nr_ice_component *comp;
+
+    if (stream->ice_state != NR_ICE_MEDIA_STREAM_UNPAIRED)
+      ABORT(R_FAILED);
+
+    if ((r=nr_ice_media_stream_find_component(stream, component_id, &comp)))
+      ABORT(r);
+
+    /* Can only disable before pairing */
+    if (comp->state != NR_ICE_COMPONENT_UNPAIRED &&
+        comp->state != NR_ICE_COMPONENT_DISABLED)
+      ABORT(R_FAILED);
+
+    comp->state = NR_ICE_COMPONENT_DISABLED;
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+void nr_ice_media_stream_role_change(nr_ice_media_stream *stream)
+  {
+    nr_ice_cand_pair *pair,*temp_pair;
+    /* Changing role causes candidate pair priority to change, which requires
+     * re-sorting the check list. */
+    nr_ice_cand_pair_head old_checklist;
+
+    /* Move check_list to old_checklist (not POD, have to do the hard way) */
+    TAILQ_INIT(&old_checklist);
+    TAILQ_FOREACH_SAFE(pair,&stream->check_list,check_queue_entry,temp_pair) {
+      TAILQ_REMOVE(&stream->check_list,pair,check_queue_entry);
+      TAILQ_INSERT_TAIL(&old_checklist,pair,check_queue_entry);
+    }
+
+    /* Re-insert into the check list */
+    TAILQ_FOREACH_SAFE(pair,&old_checklist,check_queue_entry,temp_pair) {
+      TAILQ_REMOVE(&old_checklist,pair,check_queue_entry);
+      nr_ice_candidate_pair_role_change(pair);
+      nr_ice_candidate_pair_insert(&stream->check_list,pair);
+    }
+  }
+
+int nr_ice_media_stream_find_pair(nr_ice_media_stream *str, nr_ice_candidate *lcand, nr_ice_candidate *rcand, nr_ice_cand_pair **pair)
+  {
+    nr_ice_cand_pair_head *head = &str->check_list;
+    nr_ice_cand_pair *c1;
+
+    c1=TAILQ_FIRST(head);
+    while(c1){
+      if(c1->local == lcand &&
+         c1->remote == rcand) {
+        *pair=c1;
+        return(0);
+      }
+
+      c1=TAILQ_NEXT(c1,check_queue_entry);
+    }
+
+    return(R_NOT_FOUND);
+  }
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_media_stream.h b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_media_stream.h
new file mode 100644
index 0000000000..99f906c100
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_media_stream.h
@@ -0,0 +1,146 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _ice_media_stream_h
+#define _ice_media_stream_h
+#ifdef __cplusplus
+using namespace std;
+extern "C" {
+#endif /* __cplusplus */
+
+#include "transport_addr.h"
+
+typedef struct nr_ice_stun_server_ {
+  nr_transport_addr addr;
+  int id;
+} nr_ice_stun_server;
+
+typedef struct nr_ice_turn_server_ {
+    nr_ice_stun_server    turn_server;
+    char                 *username;
+    Data                 *password;
+} nr_ice_turn_server;
+
+struct nr_ice_media_stream_ {
+  char *label;
+  struct nr_ice_ctx_ *ctx;
+  struct nr_ice_peer_ctx_ *pctx;
+
+  struct nr_ice_media_stream_ *local_stream; /* used when this is a peer */
+  int component_ct;
+  nr_ice_component_head components;
+
+  char *ufrag;    /* ICE username */
+  char *pwd;    /* ICE password */
+  char *r2l_user;  /* The username for incoming requests */
+  char *l2r_user;  /* The username for outgoing requests */
+  Data r2l_pass;   /* The password for incoming requests */
+  Data l2r_pass;   /* The password for outcoming requests */
+  int ice_state;
+  /* The stream is being replaced by another, so it will not continue any ICE
+   * processing. If this stream is connected already, traffic can continue to
+   * flow for a limited time while the new stream gets ready. */
+  int obsolete;
+
+#define NR_ICE_MEDIA_STREAM_UNPAIRED           1
+#define NR_ICE_MEDIA_STREAM_CHECKS_FROZEN      2
+#define NR_ICE_MEDIA_STREAM_CHECKS_ACTIVE      3
+#define NR_ICE_MEDIA_STREAM_CHECKS_CONNECTED   4
+#define NR_ICE_MEDIA_STREAM_CHECKS_FAILED      5
+
+  int disconnected;
+
+#define NR_ICE_MEDIA_STREAM_CONNECTED    0
+#define NR_ICE_MEDIA_STREAM_DISCONNECTED 1
+
+  /* Copy of flags field from nr_ice_ctx */
+  int flags;
+
+  /* Copy of STUN/TURN servers from nr_ice_ctx */
+  nr_ice_stun_server *stun_servers;           /* The list of stun servers */
+  int stun_server_ct;
+  nr_ice_turn_server *turn_servers;           /* The list of turn servers */
+  int turn_server_ct;
+
+  nr_ice_cand_pair_head check_list;
+  nr_ice_cand_pair_head trigger_check_queue;
+  void *timer;  /* Check list periodic timer */
+
+/*  nr_ice_cand_pair_head valid_list; */
+
+  STAILQ_ENTRY(nr_ice_media_stream_) entry;
+};
+
+typedef STAILQ_HEAD(nr_ice_media_stream_head_,nr_ice_media_stream_) nr_ice_media_stream_head;
+
+int nr_ice_media_stream_create(struct nr_ice_ctx_ *ctx,const char *label,const char *ufrag,const char *pwd,int components, nr_ice_media_stream **streamp);
+int nr_ice_media_stream_destroy(nr_ice_media_stream **streamp);
+int nr_ice_media_stream_finalize(nr_ice_media_stream *lstr,nr_ice_media_stream *rstr);
+int nr_ice_media_stream_initialize(struct nr_ice_ctx_ *ctx, nr_ice_media_stream *stream);
+int nr_ice_media_stream_get_attributes(nr_ice_media_stream *stream, char ***attrsp,int *attrctp);
+int nr_ice_media_stream_get_default_candidate(nr_ice_media_stream *stream, int component, nr_ice_candidate **candp);
+int nr_ice_media_stream_pair_candidates(nr_ice_peer_ctx *pctx,nr_ice_media_stream *lstream,nr_ice_media_stream *pstream);
+int nr_ice_media_stream_start_checks(nr_ice_peer_ctx *pctx, nr_ice_media_stream *stream);
+int nr_ice_media_stream_service_pre_answer_requests(nr_ice_peer_ctx *pctx,nr_ice_media_stream *lstream,nr_ice_media_stream *pstream, int *serviced);
+int nr_ice_media_stream_unfreeze_pairs(nr_ice_peer_ctx *pctx, nr_ice_media_stream *stream);
+int nr_ice_media_stream_unfreeze_pairs_foundation(nr_ice_media_stream *stream, char *foundation);
+void nr_ice_media_stream_dump_state(nr_ice_peer_ctx *pctx, nr_ice_media_stream *stream, int log_level);
+void nr_ice_media_stream_component_nominated(nr_ice_media_stream *stream,nr_ice_component *component);
+void nr_ice_media_stream_component_failed(nr_ice_media_stream *stream,nr_ice_component *component);
+void nr_ice_media_stream_refresh_consent_all(nr_ice_media_stream *stream);
+void nr_ice_media_stream_disconnect_all_components(nr_ice_media_stream *stream);
+void nr_ice_media_stream_set_disconnected(nr_ice_media_stream *stream, int disconnected);
+int nr_ice_media_stream_check_if_connected(nr_ice_media_stream *stream);
+int nr_ice_media_stream_set_state(nr_ice_media_stream *str, int state);
+void nr_ice_media_stream_stop_checking(nr_ice_media_stream *str);
+void nr_ice_media_stream_set_obsolete(nr_ice_media_stream *str);
+int nr_ice_media_stream_is_done_gathering(nr_ice_media_stream *str);
+int nr_ice_media_stream_get_best_candidate(nr_ice_media_stream *str, int component, nr_ice_candidate **candp);
+int nr_ice_media_stream_send(nr_ice_peer_ctx *pctx, nr_ice_media_stream *str, int component, UCHAR *data, int len);
+int nr_ice_media_stream_get_active(nr_ice_peer_ctx *pctx, nr_ice_media_stream *str, int component, nr_ice_candidate **local, nr_ice_candidate **remote);
+int nr_ice_media_stream_find_component(nr_ice_media_stream *str, int comp_id, nr_ice_component **compp);
+int nr_ice_media_stream_find_pair(nr_ice_media_stream *str, nr_ice_candidate *local, nr_ice_candidate *remote, nr_ice_cand_pair **pair);
+int nr_ice_media_stream_addrs(nr_ice_peer_ctx *pctx, nr_ice_media_stream *str, int component, nr_transport_addr *local, nr_transport_addr *remote);
+int
+nr_ice_peer_ctx_parse_media_stream_attribute(nr_ice_peer_ctx *pctx, nr_ice_media_stream *stream, char *attr);
+int nr_ice_media_stream_get_consent_status(nr_ice_media_stream *stream, int component_id, int *can_send, struct timeval *ts);
+int nr_ice_media_stream_disable_component(nr_ice_media_stream *stream, int component_id);
+int nr_ice_media_stream_pair_new_trickle_candidate(nr_ice_peer_ctx *pctx, nr_ice_media_stream *pstream, nr_ice_candidate *cand);
+void nr_ice_media_stream_role_change(nr_ice_media_stream *stream);
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_parser.c b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_parser.c
new file mode 100644
index 0000000000..25cda3364a
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_parser.c
@@ -0,0 +1,564 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <csi_platform.h>
+#include <sys/types.h>
+#ifdef WIN32
+#include <winsock2.h>
+#else
+#include <sys/socket.h>
+#include <netinet/in.h>
+#include <arpa/inet.h>
+#include <strings.h>
+#endif
+#include <string.h>
+#include <assert.h>
+#include <ctype.h>
+#include "nr_api.h"
+#include "ice_ctx.h"
+#include "ice_candidate.h"
+#include "ice_reg.h"
+
+static void
+skip_whitespace(char **str)
+{
+    char *c = *str;
+    while (*c == ' ')
+        ++c;
+
+    *str = c;
+}
+
+static void
+fast_forward(char **str, int skip)
+{
+    char *c = *str;
+    while (*c != '\0' && skip-- > 0)
+        ++c;
+
+    *str = c;
+}
+
+static void
+skip_to_past_space(char **str)
+{
+    char *c = *str;
+    while (*c != ' ' && *c != '\0')
+        ++c;
+
+    *str = c;
+
+    skip_whitespace(str);
+}
+
+static int
+grab_token(char **str, char **out)
+{
+    int _status;
+    char *c = *str;
+    int len;
+    char *tmp;
+
+    while (*c != ' ' && *c != '\0')
+        ++c;
+
+    len = c - *str;
+
+    tmp = RMALLOC(len + 1);
+    if (!tmp)
+        ABORT(R_NO_MEMORY);
+
+    memcpy(tmp, *str, len);
+    tmp[len] = '\0';
+
+    *str = c;
+    *out = tmp;
+
+    _status = 0;
+abort:
+    return _status;
+}
+
+int
+nr_ice_peer_candidate_from_attribute(nr_ice_ctx *ctx,char *orig,nr_ice_media_stream *stream,nr_ice_candidate **candp)
+{
+    int r,_status;
+    char* str = orig;
+    nr_ice_candidate *cand;
+    char *connection_address=0;
+    unsigned int port;
+    int i;
+    unsigned int component_id;
+    char *rel_addr=0;
+    unsigned char transport;
+
+    if(!(cand=RCALLOC(sizeof(nr_ice_candidate))))
+        ABORT(R_NO_MEMORY);
+
+    if(!(cand->label=r_strdup(orig)))
+        ABORT(R_NO_MEMORY);
+
+    cand->ctx=ctx;
+    cand->isock=0;
+    cand->state=NR_ICE_CAND_PEER_CANDIDATE_UNPAIRED;
+    cand->stream=stream;
+    skip_whitespace(&str);
+
+    /* Skip a= if present */
+    if (!strncmp(str, "a=", 2))
+        str += 2;
+
+    /* Candidate attr */
+    if (strncasecmp(str, "candidate:", 10))
+        ABORT(R_BAD_DATA);
+
+    fast_forward(&str, 10);
+    if (*str == '\0')
+        ABORT(R_BAD_DATA);
+
+    skip_whitespace(&str);
+    if (*str == '\0')
+        ABORT(R_BAD_DATA);
+
+    /* Foundation */
+    if ((r=grab_token(&str, &cand->foundation)))
+        ABORT(r);
+
+    if (*str == '\0')
+        ABORT(R_BAD_DATA);
+
+    skip_whitespace(&str);
+    if (*str == '\0')
+        ABORT(R_BAD_DATA);
+
+    /* component */
+    if (sscanf(str, "%u", &component_id) != 1)
+        ABORT(R_BAD_DATA);
+
+    if (component_id < 1 || component_id > 256)
+        ABORT(R_BAD_DATA);
+
+    cand->component_id = (UCHAR)component_id;
+
+    skip_to_past_space(&str);
+    if (*str == '\0')
+        ABORT(R_BAD_DATA);
+
+    /* Protocol */
+    if (!strncasecmp(str, "UDP", 3))
+      transport=IPPROTO_UDP;
+    else if (!strncasecmp(str, "TCP", 3))
+      transport=IPPROTO_TCP;
+    else
+      ABORT(R_BAD_DATA);
+
+    fast_forward(&str, 3);
+    if (*str == '\0')
+        ABORT(R_BAD_DATA);
+
+    skip_whitespace(&str);
+    if (*str == '\0')
+        ABORT(R_BAD_DATA);
+
+    /* priority */
+    if (sscanf(str, "%u", &cand->priority) != 1)
+        ABORT(R_BAD_DATA);
+
+    if (cand->priority < 1)
+        ABORT(R_BAD_DATA);
+
+    skip_to_past_space(&str);
+    if (*str == '\0')
+        ABORT(R_BAD_DATA);
+
+    /* Peer address/port */
+    if ((r=grab_token(&str, &connection_address)))
+        ABORT(r);
+
+    if (*str == '\0')
+        ABORT(R_BAD_DATA);
+
+    skip_whitespace(&str);
+    if (*str == '\0')
+        ABORT(R_BAD_DATA);
+
+    if (sscanf(str, "%u", &port) != 1)
+        ABORT(R_BAD_DATA);
+
+    if (port < 1 || port > 0x0FFFF)
+        ABORT(R_BAD_DATA);
+
+    if ((r=nr_str_port_to_transport_addr(connection_address,port,transport,&cand->addr)))
+      ABORT(r);
+
+    skip_to_past_space(&str);
+    if (*str == '\0')
+        ABORT(R_BAD_DATA);
+
+    /* Type */
+    if (strncasecmp("typ", str, 3))
+        ABORT(R_BAD_DATA);
+
+    fast_forward(&str, 3);
+    if (*str == '\0')
+        ABORT(R_BAD_DATA);
+
+    skip_whitespace(&str);
+    if (*str == '\0')
+        ABORT(R_BAD_DATA);
+
+    assert(nr_ice_candidate_type_names[0] == 0);
+
+    for (i = 1; nr_ice_candidate_type_names[i]; ++i) {
+        if(!strncasecmp(nr_ice_candidate_type_names[i], str, strlen(nr_ice_candidate_type_names[i]))) {
+            cand->type=i;
+            break;
+        }
+    }
+    if (nr_ice_candidate_type_names[i] == 0)
+        ABORT(R_BAD_DATA);
+
+    fast_forward(&str, strlen(nr_ice_candidate_type_names[i]));
+
+    /* Look for the other side's raddr, rport */
+    /* raddr, rport */
+    switch (cand->type) {
+    case HOST:
+        break;
+    case SERVER_REFLEXIVE:
+    case PEER_REFLEXIVE:
+    case RELAYED:
+
+        skip_whitespace(&str);
+        if (*str == '\0')
+            ABORT(R_BAD_DATA);
+
+        if (strncasecmp("raddr", str, 5))
+            ABORT(R_BAD_DATA);
+
+        fast_forward(&str, 5);
+        if (*str == '\0')
+            ABORT(R_BAD_DATA);
+
+        skip_whitespace(&str);
+        if (*str == '\0')
+            ABORT(R_BAD_DATA);
+
+        if ((r=grab_token(&str, &rel_addr)))
+            ABORT(r);
+
+        if (*str == '\0')
+            ABORT(R_BAD_DATA);
+
+        skip_whitespace(&str);
+        if (*str == '\0')
+            ABORT(R_BAD_DATA);
+
+        if (strncasecmp("rport", str, 5))
+              ABORT(R_BAD_DATA);
+
+        fast_forward(&str, 5);
+        if (*str == '\0')
+            ABORT(R_BAD_DATA);
+
+        skip_whitespace(&str);
+        if (*str == '\0')
+            ABORT(R_BAD_DATA);
+
+        if (sscanf(str, "%u", &port) != 1)
+            ABORT(R_BAD_DATA);
+
+        if (port > 0x0FFFF)
+            ABORT(R_BAD_DATA);
+
+        if ((r=nr_str_port_to_transport_addr(rel_addr,port,transport,&cand->base)))
+          ABORT(r);
+
+        skip_to_past_space(&str);
+        /* it's expected to be at EOD at this point */
+
+        break;
+    default:
+        ABORT(R_INTERNAL);
+        break;
+    }
+
+    skip_whitespace(&str);
+
+    if (transport == IPPROTO_TCP && cand->type != RELAYED) {
+      /* Parse tcptype extension per RFC 6544 S 4.5 */
+      if (strncasecmp("tcptype ", str, 8))
+        ABORT(R_BAD_DATA);
+
+      fast_forward(&str, 8);
+      skip_whitespace(&str);
+
+      for (i = 1; nr_ice_candidate_tcp_type_names[i]; ++i) {
+        if(!strncasecmp(nr_ice_candidate_tcp_type_names[i], str, strlen(nr_ice_candidate_tcp_type_names[i]))) {
+          cand->tcp_type=i;
+          fast_forward(&str, strlen(nr_ice_candidate_tcp_type_names[i]));
+          break;
+        }
+      }
+
+      if (cand->tcp_type == 0)
+        ABORT(R_BAD_DATA);
+
+      if (*str && *str != ' ')
+        ABORT(R_BAD_DATA);
+    }
+    /* Ignore extensions per RFC 5245 S 15.1 */
+#if 0
+    /* This used to be an assert, but we don't want to exit on invalid
+       remote data */
+    if (strlen(str) != 0) {
+      ABORT(R_BAD_DATA);
+    }
+#endif
+
+    nr_ice_candidate_compute_codeword(cand);
+
+    *candp=cand;
+
+    _status=0;
+  abort:
+    if (_status){
+        r_log(LOG_ICE,LOG_WARNING,"ICE(%s): Error parsing attribute: %s",ctx->label,orig);
+        nr_ice_candidate_destroy(&cand);
+    }
+
+    RFREE(connection_address);
+    RFREE(rel_addr);
+    return(_status);
+}
+
+
+int
+nr_ice_peer_ctx_parse_media_stream_attribute(nr_ice_peer_ctx *pctx, nr_ice_media_stream *stream, char *attr)
+{
+    int r,_status;
+    char *orig = 0;
+    char *str;
+
+    orig = str = attr;
+
+    if (!strncasecmp(str, "ice-ufrag:", 10)) {
+      fast_forward(&str, 10);
+      if (*str == '\0')
+        ABORT(R_BAD_DATA);
+
+      skip_whitespace(&str);
+      if (*str == '\0')
+        ABORT(R_BAD_DATA);
+
+      RFREE(stream->ufrag);
+      if ((r=grab_token(&str, &stream->ufrag)))
+        ABORT(r);
+    }
+    else if (!strncasecmp(str, "ice-pwd:", 8)) {
+      fast_forward(&str, 8);
+      if (*str == '\0')
+        ABORT(R_BAD_DATA);
+
+      skip_whitespace(&str);
+      if (*str == '\0')
+        ABORT(R_BAD_DATA);
+
+      RFREE(stream->pwd);
+      if ((r=grab_token(&str, &stream->pwd)))
+        ABORT(r);
+    }
+    else {
+      ABORT(R_BAD_DATA);
+    }
+
+    skip_whitespace(&str);
+
+    /* RFC 5245 grammar doesn't have an extension point for ice-pwd or
+       ice-ufrag: if there's anything left on the line, we treat it as bad. */
+    if (str[0] != '\0') {
+      ABORT(R_BAD_DATA);
+    }
+
+    _status=0;
+  abort:
+    if (_status) {
+      if (orig)
+        r_log(LOG_ICE,LOG_WARNING,"ICE-PEER(%s): Error parsing attribute: %s",pctx->label,orig);
+    }
+
+    return(_status);
+}
+
+int
+nr_ice_peer_ctx_parse_global_attributes(nr_ice_peer_ctx *pctx, char **attrs, int attr_ct)
+{
+    int r,_status;
+    int i;
+    char *orig = 0;
+    char *str;
+    char *component_id = 0;
+    char *connection_address = 0;
+    unsigned int port;
+    in_addr_t addr;
+    char *ice_option_tag = 0;
+
+    for(i=0;i<attr_ct;i++){
+        orig = str = attrs[i];
+
+        component_id = 0;
+        connection_address = 0;
+        ice_option_tag = 0;
+
+        if (!strncasecmp(str, "remote-candidates:", 18)) {
+            fast_forward(&str, 18);
+            skip_whitespace(&str);
+
+            while (*str != '\0') {
+                if ((r=grab_token(&str, &component_id)))
+                    ABORT(r);
+
+                if (*str == '\0')
+                    ABORT(R_BAD_DATA);
+
+                skip_whitespace(&str);
+                if (*str == '\0')
+                    ABORT(R_BAD_DATA);
+
+                if ((r=grab_token(&str, &connection_address)))
+                    ABORT(r);
+
+                if (*str == '\0')
+                    ABORT(R_BAD_DATA);
+
+                addr = inet_addr(connection_address);
+                if (addr == INADDR_NONE)
+                    ABORT(R_BAD_DATA);
+
+                skip_whitespace(&str);
+                if (*str == '\0')
+                    ABORT(R_BAD_DATA);
+
+                if (sscanf(str, "%u", &port) != 1)
+                    ABORT(R_BAD_DATA);
+
+                if (port < 1 || port > 0x0FFFF)
+                    ABORT(R_BAD_DATA);
+
+                skip_to_past_space(&str);
+
+#if 0
+                /* TODO: !nn! just drop on the floor for now, later put somewhere */
+                /* Assume v4 for now */
+                if(r=nr_ip4_port_to_transport_addr(ntohl(addr),port,IPPROTO_UDP,&candidate->base))
+                  ABORT(r);
+
+                TAILQ_INSERT_TAIL(head, elm, field);
+#endif
+
+                component_id = 0;  /* prevent free */
+                RFREE(connection_address);
+                connection_address = 0;  /* prevent free */
+            }
+        }
+        else if (!strncasecmp(str, "ice-lite", 8)) {
+            pctx->peer_lite = 1;
+            pctx->controlling = 1;
+
+            fast_forward(&str, 8);
+        }
+        else if (!strncasecmp(str, "ice-mismatch", 12)) {
+            pctx->peer_ice_mismatch = 1;
+
+            fast_forward(&str, 12);
+        }
+        else if (!strncasecmp(str, "ice-ufrag:", 10)) {
+            fast_forward(&str, 10);
+            if (*str == '\0')
+                ABORT(R_BAD_DATA);
+
+            skip_whitespace(&str);
+            if (*str == '\0')
+                ABORT(R_BAD_DATA);
+        }
+        else if (!strncasecmp(str, "ice-pwd:", 8)) {
+            fast_forward(&str, 8);
+            if (*str == '\0')
+                ABORT(R_BAD_DATA);
+
+            skip_whitespace(&str);
+            if (*str == '\0')
+                ABORT(R_BAD_DATA);
+        }
+        else if (!strncasecmp(str, "ice-options:", 12)) {
+            fast_forward(&str, 12);
+            skip_whitespace(&str);
+
+            while (*str != '\0') {
+                if ((r=grab_token(&str, &ice_option_tag)))
+                    ABORT(r);
+
+                skip_whitespace(&str);
+
+                //TODO: for now, just throw away; later put somewhere
+                RFREE(ice_option_tag);
+
+                ice_option_tag = 0;  /* prevent free */
+            }
+        }
+        else {
+            ABORT(R_BAD_DATA);
+        }
+
+        skip_whitespace(&str);
+
+      /* RFC 5245 grammar doesn't have an extension point for any of the
+         preceding attributes: if there's anything left on the line, we
+         treat it as bad data. */
+      if (str[0] != '\0') {
+        ABORT(R_BAD_DATA);
+      }
+    }
+
+    _status=0;
+  abort:
+    if (_status) {
+      if (orig)
+        r_log(LOG_ICE,LOG_WARNING,"ICE-PEER(%s): Error parsing attribute: %s",pctx->label,orig);
+    }
+
+    RFREE(connection_address);
+    RFREE(component_id);
+    RFREE(ice_option_tag);
+    return(_status);
+}
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_peer_ctx.c b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_peer_ctx.c
new file mode 100644
index 0000000000..0bf97eb984
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_peer_ctx.c
@@ -0,0 +1,875 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <string.h>
+#include <assert.h>
+#include <registry.h>
+#include <nr_api.h>
+#include "ice_ctx.h"
+#include "ice_peer_ctx.h"
+#include "ice_media_stream.h"
+#include "ice_util.h"
+#include "nr_crypto.h"
+#include "async_timer.h"
+#include "ice_reg.h"
+
+static void nr_ice_peer_ctx_parse_stream_attributes_int(nr_ice_peer_ctx *pctx, nr_ice_media_stream *stream, nr_ice_media_stream *pstream, char **attrs, int attr_ct);
+static int nr_ice_ctx_parse_candidate(nr_ice_peer_ctx *pctx, nr_ice_media_stream *pstream, char *candidate, int trickled, const char *mdns_addr);
+static void nr_ice_peer_ctx_start_trickle_timer(nr_ice_peer_ctx *pctx);
+
+int nr_ice_peer_ctx_create(nr_ice_ctx *ctx, nr_ice_handler *handler,char *label, nr_ice_peer_ctx **pctxp)
+  {
+    int r,_status;
+    nr_ice_peer_ctx *pctx=0;
+
+    if(!(pctx=RCALLOC(sizeof(nr_ice_peer_ctx))))
+      ABORT(R_NO_MEMORY);
+
+    pctx->state = NR_ICE_PEER_STATE_UNPAIRED;
+
+    if(!(pctx->label=r_strdup(label)))
+      ABORT(R_NO_MEMORY);
+
+    pctx->ctx=ctx;
+    pctx->handler=handler;
+
+    /* Decide controlling vs. controlled */
+    if(ctx->flags & NR_ICE_CTX_FLAGS_LITE){
+      pctx->controlling=0;
+    } else {
+      pctx->controlling=1;
+    }
+    if(r=nr_crypto_random_bytes((UCHAR *)&pctx->tiebreaker,8))
+      ABORT(r);
+
+    STAILQ_INIT(&pctx->peer_streams);
+
+    STAILQ_INSERT_TAIL(&ctx->peers,pctx,entry);
+
+    *pctxp=pctx;
+
+    _status = 0;
+  abort:
+    if(_status){
+      nr_ice_peer_ctx_destroy(&pctx);
+    }
+    return(_status);
+  }
+
+
+
+int nr_ice_peer_ctx_parse_stream_attributes(nr_ice_peer_ctx *pctx, nr_ice_media_stream *stream, char **attrs, int attr_ct)
+  {
+    nr_ice_media_stream *pstream=0;
+    nr_ice_component *comp,*comp2;
+    char *lufrag,*rufrag;
+    char *lpwd,*rpwd;
+    int r,_status;
+
+    /*
+      Note: use component_ct from our own stream since components other
+      than this offered by the other side are unusable */
+    if(r=nr_ice_media_stream_create(pctx->ctx,stream->label,"","",stream->component_ct,&pstream))
+      ABORT(r);
+
+    /* Match up the local and remote components */
+    comp=STAILQ_FIRST(&stream->components);
+    comp2=STAILQ_FIRST(&pstream->components);
+    while(comp){
+      comp2->local_component=comp;
+
+      comp=STAILQ_NEXT(comp,entry);
+      comp2=STAILQ_NEXT(comp2,entry);
+    }
+
+    pstream->local_stream=stream;
+    pstream->pctx=pctx;
+
+    nr_ice_peer_ctx_parse_stream_attributes_int(pctx,stream,pstream,attrs,attr_ct);
+
+    /* Now that we have the ufrag and password, compute all the username/password
+       pairs */
+    lufrag=stream->ufrag;
+    lpwd=stream->pwd;
+    assert(lufrag);
+    assert(lpwd);
+    rufrag=pstream->ufrag;
+    rpwd=pstream->pwd;
+    if (!rufrag || !rpwd)
+      ABORT(R_BAD_DATA);
+
+    if(r=nr_concat_strings(&pstream->r2l_user,lufrag,":",rufrag,NULL))
+      ABORT(r);
+    if(r=nr_concat_strings(&pstream->l2r_user,rufrag,":",lufrag,NULL))
+      ABORT(r);
+    if(r=r_data_make(&pstream->r2l_pass, (UCHAR *)lpwd, strlen(lpwd)))
+      ABORT(r);
+    if(r=r_data_make(&pstream->l2r_pass, (UCHAR *)rpwd, strlen(rpwd)))
+      ABORT(r);
+
+    STAILQ_INSERT_TAIL(&pctx->peer_streams,pstream,entry);
+    pstream=0;
+
+    _status=0;
+  abort:
+    if (_status) {
+      nr_ice_media_stream_destroy(&pstream);
+    }
+    return(_status);
+  }
+
+static void nr_ice_peer_ctx_parse_stream_attributes_int(nr_ice_peer_ctx *pctx, nr_ice_media_stream *stream, nr_ice_media_stream *pstream, char **attrs, int attr_ct)
+  {
+    int r;
+    int i;
+
+    for(i=0;i<attr_ct;i++){
+      if(!strncmp(attrs[i],"ice-",4)){
+        if(r=nr_ice_peer_ctx_parse_media_stream_attribute(pctx,pstream,attrs[i])) {
+          r_log(LOG_ICE,LOG_WARNING,"ICE(%s): peer (%s) specified bogus ICE attribute",pctx->ctx->label,pctx->label);
+          continue;
+        }
+      }
+      else if (!strncmp(attrs[i],"candidate",9)){
+        if(r=nr_ice_ctx_parse_candidate(pctx,pstream,attrs[i],0,0)) {
+          r_log(LOG_ICE,LOG_WARNING,"ICE(%s): peer (%s) specified bogus candidate",pctx->ctx->label,pctx->label);
+          continue;
+        }
+      }
+      else {
+        r_log(LOG_ICE,LOG_WARNING,"ICE(%s): peer (%s) specified bogus attribute: %s",pctx->ctx->label,pctx->label,attrs[i]);
+      }
+    }
+
+    /* Doesn't fail because we just skip errors */
+  }
+
+static int nr_ice_ctx_parse_candidate(nr_ice_peer_ctx *pctx, nr_ice_media_stream *pstream, char *candidate, int trickled, const char *mdns_addr)
+  {
+    nr_ice_candidate *cand=0;
+    nr_ice_component *comp;
+    int j;
+    int r, _status;
+
+    if(r=nr_ice_peer_candidate_from_attribute(pctx->ctx,candidate,pstream,&cand))
+      ABORT(r);
+
+    /* set the trickled flag on the candidate */
+    cand->trickled = trickled;
+
+    if (mdns_addr) {
+      cand->mdns_addr = r_strdup(mdns_addr);
+      if (!cand->mdns_addr) {
+        ABORT(R_NO_MEMORY);
+      }
+    }
+
+    /* Not the fastest way to find a component, but it's what we got */
+    j=1;
+    for(comp=STAILQ_FIRST(&pstream->components);comp;comp=STAILQ_NEXT(comp,entry)){
+      if(j==cand->component_id)
+        break;
+
+      j++;
+    }
+
+    if(!comp){
+      /* Very common for the answerer when it uses rtcp-mux */
+      r_log(LOG_ICE,LOG_INFO,"ICE(%s): peer (%s) no such component for candidate %s",pctx->ctx->label,pctx->label, candidate);
+      ABORT(R_REJECTED);
+    }
+
+    if (comp->state == NR_ICE_COMPONENT_DISABLED) {
+      r_log(LOG_ICE,LOG_WARNING,"Peer offered candidate for disabled remote component: %s", candidate);
+      ABORT(R_BAD_DATA);
+    }
+    if (comp->local_component->state == NR_ICE_COMPONENT_DISABLED) {
+      r_log(LOG_ICE,LOG_WARNING,"Peer offered candidate for disabled local component: %s", candidate);
+      ABORT(R_BAD_DATA);
+    }
+
+    cand->component=comp;
+
+    TAILQ_INSERT_TAIL(&comp->candidates,cand,entry_comp);
+
+    r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s)/CAND(%s): creating peer candidate",
+      pctx->label,cand->label);
+
+    _status=0;
+ abort:
+    if (_status) {
+      nr_ice_candidate_destroy(&cand);
+    }
+    return(_status);
+  }
+
+int nr_ice_peer_ctx_find_pstream(nr_ice_peer_ctx *pctx, nr_ice_media_stream *stream, nr_ice_media_stream **pstreamp)
+  {
+    int _status;
+    nr_ice_media_stream *pstream;
+
+    /* Because we don't have forward pointers, iterate through all the
+       peer streams to find one that matches us */
+     pstream=STAILQ_FIRST(&pctx->peer_streams);
+
+     if(!pstream) {
+       /* No peer streams at all, presumably because they do not exist yet.
+        * Don't log a warning here. */
+       ABORT(R_NOT_FOUND);
+     }
+
+     while(pstream) {
+       if (pstream->local_stream == stream)
+         break;
+
+       pstream = STAILQ_NEXT(pstream, entry);
+     }
+
+     if (!pstream) {
+       r_log(LOG_ICE,LOG_WARNING,"ICE(%s): peer (%s) has no stream matching stream %s",pctx->ctx->label,pctx->label,stream->label);
+       ABORT(R_NOT_FOUND);
+     }
+
+    *pstreamp = pstream;
+
+    _status=0;
+ abort:
+    return(_status);
+  }
+
+int nr_ice_peer_ctx_remove_pstream(nr_ice_peer_ctx *pctx, nr_ice_media_stream **pstreamp)
+  {
+    int r,_status;
+
+    STAILQ_REMOVE(&pctx->peer_streams,*pstreamp,nr_ice_media_stream_,entry);
+
+    if(r=nr_ice_media_stream_destroy(pstreamp)) {
+      ABORT(r);
+    }
+
+    _status=0;
+ abort:
+    return(_status);
+  }
+
+int nr_ice_peer_ctx_parse_trickle_candidate(nr_ice_peer_ctx *pctx, nr_ice_media_stream *stream, char *candidate, const char *mdns_addr)
+  {
+    nr_ice_media_stream *pstream;
+    int r,_status;
+    int needs_pairing = 0;
+
+    if (stream->obsolete) {
+      return 0;
+    }
+
+    r_log(LOG_ICE,LOG_DEBUG,"ICE(%s): peer (%s) parsing trickle ICE candidate %s",pctx->ctx->label,pctx->label,candidate);
+    r = nr_ice_peer_ctx_find_pstream(pctx, stream, &pstream);
+    if (r)
+      ABORT(r);
+
+    switch(pstream->ice_state) {
+      case NR_ICE_MEDIA_STREAM_UNPAIRED:
+        break;
+      case NR_ICE_MEDIA_STREAM_CHECKS_FROZEN:
+      case NR_ICE_MEDIA_STREAM_CHECKS_ACTIVE:
+        needs_pairing = 1;
+        break;
+      default:
+        r_log(LOG_ICE,LOG_ERR,"ICE(%s): peer (%s), stream(%s) tried to trickle ICE in inappropriate state %d",pctx->ctx->label,pctx->label,stream->label,pstream->ice_state);
+        ABORT(R_ALREADY);
+        break;
+    }
+
+    if(r=nr_ice_ctx_parse_candidate(pctx,pstream,candidate,1,mdns_addr)){
+      ABORT(r);
+    }
+
+    /* If ICE is running (i.e., we are in FROZEN or ACTIVE states)
+       then we need to pair this new candidate. For now we
+       just re-pair the stream which is inefficient but still
+       fine because we suppress duplicate pairing */
+    if (needs_pairing) {
+      /* Start the remote trickle grace timeout if it hasn't been started by
+         another trickled candidate or from the SDP. */
+      if (!pctx->trickle_grace_period_timer) {
+        nr_ice_peer_ctx_start_trickle_timer(pctx);
+      }
+
+      if(r=nr_ice_media_stream_pair_candidates(pctx, stream, pstream)) {
+        r_log(LOG_ICE,LOG_ERR,"ICE(%s): peer (%s), stream(%s) failed to pair trickle ICE candidates",pctx->ctx->label,pctx->label,stream->label);
+        ABORT(r);
+      }
+
+      /* Start checks if this stream is not checking yet or if it has checked
+         all the available candidates but not had a completed check for all
+         components.
+
+         Note that this is not compliant with RFC 5245, but consistent with
+         the libjingle trickle ICE behavior. Note that we will not restart
+         checks if either (a) the stream has failed or (b) all components
+         have a successful pair because the switch statement above jumps
+         will in both states.
+
+         TODO(ekr@rtfm.com): restart checks.
+         TODO(ekr@rtfm.com): update when the trickle ICE RFC is published
+      */
+      if (!pstream->timer) {
+        if(r=nr_ice_media_stream_start_checks(pctx, pstream)) {
+          r_log(LOG_ICE,LOG_ERR,"ICE(%s): peer (%s), stream(%s) failed to start checks",pctx->ctx->label,pctx->label,stream->label);
+          ABORT(r);
+        }
+      }
+    }
+
+    _status=0;
+ abort:
+    return(_status);
+
+  }
+
+
+static void nr_ice_peer_ctx_trickle_wait_cb(NR_SOCKET s, int how, void *cb_arg)
+  {
+    nr_ice_peer_ctx *pctx=cb_arg;
+    nr_ice_media_stream *stream;
+    nr_ice_component *comp;
+
+    pctx->trickle_grace_period_timer=0;
+
+    r_log(LOG_ICE,LOG_INFO,"ICE(%s): peer (%s) Trickle grace period is over; marking every component with only failed pairs as failed.",pctx->ctx->label,pctx->label);
+
+    stream=STAILQ_FIRST(&pctx->peer_streams);
+    while(stream){
+      comp=STAILQ_FIRST(&stream->components);
+      while(comp){
+        nr_ice_component_check_if_failed(comp);
+        comp=STAILQ_NEXT(comp,entry);
+      }
+      stream=STAILQ_NEXT(stream,entry);
+    }
+  }
+
+static void nr_ice_peer_ctx_start_trickle_timer(nr_ice_peer_ctx *pctx)
+  {
+    UINT4 grace_period_timeout=0;
+
+    if(pctx->trickle_grace_period_timer) {
+      NR_async_timer_cancel(pctx->trickle_grace_period_timer);
+      pctx->trickle_grace_period_timer=0;
+    }
+
+    NR_reg_get_uint4(NR_ICE_REG_TRICKLE_GRACE_PERIOD,&grace_period_timeout);
+
+    if (grace_period_timeout) {
+      r_log(LOG_ICE,LOG_INFO,"ICE(%s): peer (%s) starting grace period timer for %u ms",pctx->ctx->label,pctx->label, grace_period_timeout);
+      /* If we're doing trickle, we need to allow a grace period for new
+       * trickle candidates to arrive in case the pairs we have fail quickly. */
+       NR_ASYNC_TIMER_SET(grace_period_timeout,nr_ice_peer_ctx_trickle_wait_cb,pctx,&pctx->trickle_grace_period_timer);
+    }
+  }
+
+int nr_ice_peer_ctx_pair_candidates(nr_ice_peer_ctx *pctx)
+  {
+    nr_ice_media_stream *stream;
+    int r,_status;
+
+    if(pctx->peer_lite && !pctx->controlling) {
+      if(pctx->ctx->flags & NR_ICE_CTX_FLAGS_LITE){
+        r_log(LOG_ICE,LOG_ERR,"Both sides are ICE-Lite");
+        ABORT(R_BAD_DATA);
+      }
+      nr_ice_peer_ctx_switch_controlling_role(pctx);
+    }
+
+    r_log(LOG_ICE,LOG_DEBUG,"ICE(%s): peer (%s) pairing candidates",pctx->ctx->label,pctx->label);
+
+    if(STAILQ_EMPTY(&pctx->peer_streams)) {
+        r_log(LOG_ICE,LOG_ERR,"ICE(%s): peer (%s) received no media stream attributes",pctx->ctx->label,pctx->label);
+        ABORT(R_FAILED);
+    }
+
+    /* Set this first; if we fail partway through, we do not want to end
+     * up in UNPAIRED after creating some pairs. */
+    pctx->state = NR_ICE_PEER_STATE_PAIRED;
+
+    stream=STAILQ_FIRST(&pctx->peer_streams);
+    while(stream){
+      if(!stream->local_stream->obsolete) {
+        if(r=nr_ice_media_stream_pair_candidates(pctx, stream->local_stream,
+          stream))
+          ABORT(r);
+      }
+
+      stream=STAILQ_NEXT(stream,entry);
+    }
+
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+
+int nr_ice_peer_ctx_pair_new_trickle_candidate(nr_ice_ctx *ctx, nr_ice_peer_ctx *pctx, nr_ice_candidate *cand)
+  {
+    int r, _status;
+    nr_ice_media_stream *pstream;
+
+    r_log(LOG_ICE,LOG_ERR,"ICE(%s): peer (%s) pairing local trickle ICE candidate %s",pctx->ctx->label,pctx->label,cand->label);
+    if ((r = nr_ice_peer_ctx_find_pstream(pctx, cand->stream, &pstream)))
+      ABORT(r);
+
+    /* Start the remote trickle grace timeout if it hasn't been started
+       already. */
+    if (!pctx->trickle_grace_period_timer) {
+      nr_ice_peer_ctx_start_trickle_timer(pctx);
+    }
+
+    if ((r = nr_ice_media_stream_pair_new_trickle_candidate(pctx, pstream, cand)))
+      ABORT(r);
+
+    _status=0;
+ abort:
+    return _status;
+  }
+
+int nr_ice_peer_ctx_disable_component(nr_ice_peer_ctx *pctx, nr_ice_media_stream *lstream, int component_id)
+  {
+    int r, _status;
+    nr_ice_media_stream *pstream;
+    nr_ice_component *component;
+
+    if ((r=nr_ice_peer_ctx_find_pstream(pctx, lstream, &pstream)))
+      ABORT(r);
+
+    /* We shouldn't be calling this after we have started pairing */
+    if (pstream->ice_state != NR_ICE_MEDIA_STREAM_UNPAIRED)
+      ABORT(R_FAILED);
+
+    if ((r=nr_ice_media_stream_find_component(pstream, component_id,
+                                              &component)))
+      ABORT(r);
+
+    component->state = NR_ICE_COMPONENT_DISABLED;
+
+    _status=0;
+ abort:
+    return(_status);
+  }
+
+  void nr_ice_peer_ctx_destroy(nr_ice_peer_ctx** pctxp) {
+    if (!pctxp || !*pctxp) return;
+
+    nr_ice_peer_ctx* pctx = *pctxp;
+    nr_ice_media_stream *str1,*str2;
+
+    /* Stop calling the handler */
+    pctx->handler = 0;
+
+    NR_async_timer_cancel(pctx->connected_cb_timer);
+    RFREE(pctx->label);
+
+    STAILQ_FOREACH_SAFE(str1, &pctx->peer_streams, entry, str2){
+      STAILQ_REMOVE(&pctx->peer_streams,str1,nr_ice_media_stream_,entry);
+      nr_ice_media_stream_destroy(&str1);
+    }
+    assert(pctx->ctx);
+    if (pctx->ctx)
+      STAILQ_REMOVE(&pctx->ctx->peers, pctx, nr_ice_peer_ctx_, entry);
+
+    if(pctx->trickle_grace_period_timer) {
+      NR_async_timer_cancel(pctx->trickle_grace_period_timer);
+      pctx->trickle_grace_period_timer=0;
+    }
+
+    RFREE(pctx);
+
+    *pctxp=0;
+  }
+
+/* Start the checks for the first media stream (S 5.7)
+   The rest remain FROZEN */
+int nr_ice_peer_ctx_start_checks(nr_ice_peer_ctx *pctx)
+  {
+    return nr_ice_peer_ctx_start_checks2(pctx, 0);
+  }
+
+/* Start checks for some media stream.
+
+   If allow_non_first == 0, then we only look at the first stream,
+   which is 5245-complaint.
+
+   If allow_non_first == 1 then we find the first non-empty stream
+   This is not compliant with RFC 5245 but is necessary to make trickle ICE
+   work plausibly
+*/
+int nr_ice_peer_ctx_start_checks2(nr_ice_peer_ctx *pctx, int allow_non_first)
+  {
+    int r,_status;
+    nr_ice_media_stream *stream;
+    int started = 0;
+
+    /* Ensure that grace period timer is running. We might cancel this if we
+     * didn't actually start any pairs. */
+    nr_ice_peer_ctx_start_trickle_timer(pctx);
+
+    /* Might have added some streams */
+    pctx->reported_connected = 0;
+    NR_async_timer_cancel(pctx->connected_cb_timer);
+    pctx->connected_cb_timer = 0;
+    pctx->checks_started = 0;
+
+    nr_ice_peer_ctx_check_if_connected(pctx);
+
+    if (pctx->reported_connected) {
+      r_log(LOG_ICE,LOG_ERR,"ICE(%s): peer (%s) in %s all streams were done",pctx->ctx->label,pctx->label,__FUNCTION__);
+      return (0);
+    }
+
+    stream=STAILQ_FIRST(&pctx->peer_streams);
+    if(!stream)
+      ABORT(R_FAILED);
+
+    while (stream) {
+      if(!stream->local_stream->obsolete) {
+        assert(stream->ice_state != NR_ICE_MEDIA_STREAM_UNPAIRED);
+
+        if (stream->ice_state == NR_ICE_MEDIA_STREAM_CHECKS_FROZEN) {
+          if(!TAILQ_EMPTY(&stream->check_list))
+            break;
+
+          if(!allow_non_first){
+            /* This test applies if:
+
+               1. allow_non_first is 0 (i.e., non-trickle ICE)
+               2. the first stream has an empty check list.
+
+               But in the non-trickle ICE case, the other side should have provided
+               some candidates or ICE is pretty much not going to work and we're
+               just going to fail. Hence R_FAILED as opposed to R_NOT_FOUND and
+               immediate termination here.
+            */
+            r_log(LOG_ICE,LOG_ERR,"ICE(%s): peer (%s) first stream has empty check list",pctx->ctx->label,pctx->label);
+            ABORT(R_FAILED);
+          }
+        }
+      }
+
+      stream=STAILQ_NEXT(stream, entry);
+    }
+
+    if (!stream) {
+      /*
+         We fail above if we aren't doing trickle, and this is not all that
+         unusual in the trickle case.
+       */
+      r_log(LOG_ICE,LOG_NOTICE,"ICE(%s): peer (%s) no streams with non-empty check lists",pctx->ctx->label,pctx->label);
+    }
+    else if (stream->ice_state == NR_ICE_MEDIA_STREAM_CHECKS_FROZEN) {
+      if(r=nr_ice_media_stream_unfreeze_pairs(pctx,stream))
+        ABORT(r);
+      if(r=nr_ice_media_stream_start_checks(pctx,stream))
+        ABORT(r);
+      ++started;
+    }
+
+    stream=STAILQ_FIRST(&pctx->peer_streams);
+    while (stream) {
+      int serviced = 0;
+      if (r=nr_ice_media_stream_service_pre_answer_requests(pctx, stream->local_stream, stream, &serviced))
+        ABORT(r);
+
+      if (serviced) {
+        ++started;
+      }
+      else {
+        r_log(LOG_ICE,LOG_NOTICE,"ICE(%s): peer (%s) no streams with pre-answer requests",pctx->ctx->label,pctx->label);
+      }
+
+
+      stream=STAILQ_NEXT(stream, entry);
+    }
+
+    if (!started && pctx->ctx->uninitialized_candidates) {
+      r_log(LOG_ICE,LOG_INFO,"ICE(%s): peer (%s) no checks to start, but gathering is not done yet, cancelling grace period timer",pctx->ctx->label,pctx->label);
+      /* Never mind on the grace period timer */
+      NR_async_timer_cancel(pctx->trickle_grace_period_timer);
+      pctx->trickle_grace_period_timer=0;
+      ABORT(R_NOT_FOUND);
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+void nr_ice_peer_ctx_stream_started_checks(nr_ice_peer_ctx *pctx, nr_ice_media_stream *stream)
+  {
+    if (!pctx->checks_started) {
+      r_log(LOG_ICE,LOG_NOTICE,"ICE(%s): peer (%s) is now checking",pctx->ctx->label,pctx->label);
+      pctx->checks_started = 1;
+      if (pctx->handler && pctx->handler->vtbl->ice_checking) {
+        pctx->handler->vtbl->ice_checking(pctx->handler->obj, pctx);
+      }
+    }
+  }
+
+void nr_ice_peer_ctx_dump_state(nr_ice_peer_ctx *pctx, int log_level)
+  {
+    nr_ice_media_stream *stream;
+
+    r_log(LOG_ICE,log_level,"PEER %s STATE DUMP",pctx->label);
+    r_log(LOG_ICE,log_level,"==========================================");
+    stream=STAILQ_FIRST(&pctx->peer_streams);
+    while(stream){
+      nr_ice_media_stream_dump_state(pctx,stream,log_level);
+    }
+    r_log(LOG_ICE,log_level,"==========================================");
+  }
+
+void nr_ice_peer_ctx_refresh_consent_all_streams(nr_ice_peer_ctx *pctx)
+  {
+    nr_ice_media_stream *str;
+
+    r_log(LOG_ICE,LOG_INFO,"ICE-PEER(%s): refreshing consent on all streams",pctx->label);
+
+    str=STAILQ_FIRST(&pctx->peer_streams);
+    while(str) {
+      nr_ice_media_stream_refresh_consent_all(str);
+      str=STAILQ_NEXT(str,entry);
+    }
+  }
+
+void nr_ice_peer_ctx_disconnected(nr_ice_peer_ctx *pctx)
+  {
+    if (pctx->reported_connected &&
+        pctx->handler &&
+        pctx->handler->vtbl->ice_disconnected) {
+      pctx->handler->vtbl->ice_disconnected(pctx->handler->obj, pctx);
+
+      pctx->reported_connected = 0;
+    }
+  }
+
+void nr_ice_peer_ctx_disconnect_all_streams(nr_ice_peer_ctx *pctx)
+  {
+    nr_ice_media_stream *str;
+
+    r_log(LOG_ICE,LOG_INFO,"ICE-PEER(%s): disconnecting all streams",pctx->label);
+
+    str=STAILQ_FIRST(&pctx->peer_streams);
+    while(str) {
+      nr_ice_media_stream_disconnect_all_components(str);
+
+      /* The first stream to be disconnected will cause the peer ctx to signal
+         the disconnect up. */
+      nr_ice_media_stream_set_disconnected(str, NR_ICE_MEDIA_STREAM_DISCONNECTED);
+
+      str=STAILQ_NEXT(str,entry);
+    }
+  }
+
+void nr_ice_peer_ctx_connected(nr_ice_peer_ctx *pctx)
+  {
+    /* Fire the handler callback to say we're done */
+    if (pctx->reported_connected &&
+        pctx->handler &&
+        pctx->handler->vtbl->ice_connected) {
+      pctx->handler->vtbl->ice_connected(pctx->handler->obj, pctx);
+    }
+  }
+
+static void nr_ice_peer_ctx_fire_connected(NR_SOCKET s, int how, void *cb_arg)
+  {
+    nr_ice_peer_ctx *pctx=cb_arg;
+
+    pctx->connected_cb_timer=0;
+
+    nr_ice_peer_ctx_connected(pctx);
+  }
+
+/* Examine all the streams to see if we're
+   maybe miraculously connected */
+void nr_ice_peer_ctx_check_if_connected(nr_ice_peer_ctx *pctx)
+  {
+    nr_ice_media_stream *str;
+    int failed=0;
+    int succeeded=0;
+
+    str=STAILQ_FIRST(&pctx->peer_streams);
+    while(str){
+      if (!str->local_stream->obsolete){
+        if(str->ice_state==NR_ICE_MEDIA_STREAM_CHECKS_CONNECTED){
+          succeeded++;
+        }
+        else if(str->ice_state==NR_ICE_MEDIA_STREAM_CHECKS_FAILED){
+          failed++;
+        }
+        else{
+          break;
+        }
+      }
+      str=STAILQ_NEXT(str,entry);
+    }
+
+    if(str)
+      return;  /* Something isn't done */
+
+    /* OK, we're finished, one way or another */
+    r_log(LOG_ICE,LOG_INFO,"ICE-PEER(%s): all checks completed success=%d fail=%d",pctx->label,succeeded,failed);
+
+    /* Make sure grace period timer is cancelled */
+    if(pctx->trickle_grace_period_timer) {
+      r_log(LOG_ICE,LOG_INFO,"ICE(%s): peer (%s) cancelling grace period timer",pctx->ctx->label,pctx->label);
+      NR_async_timer_cancel(pctx->trickle_grace_period_timer);
+      pctx->trickle_grace_period_timer=0;
+    }
+
+    /* Schedule a connected notification for the first connected event.
+       IMPORTANT: This is done in a callback because we expect destructors
+       of various kinds to be fired from here */
+    if (!pctx->reported_connected) {
+      pctx->reported_connected = 1;
+      assert(!pctx->connected_cb_timer);
+      NR_ASYNC_TIMER_SET(0,nr_ice_peer_ctx_fire_connected,pctx,&pctx->connected_cb_timer);
+    }
+  }
+
+
+/* Given a component in the main ICE ctx, find the relevant component in
+   the peer_ctx */
+int nr_ice_peer_ctx_find_component(nr_ice_peer_ctx *pctx, nr_ice_media_stream *str, int component_id, nr_ice_component **compp)
+  {
+    nr_ice_media_stream *pstr;
+    int r,_status;
+
+    pstr=STAILQ_FIRST(&pctx->peer_streams);
+    while(pstr){
+      if(pstr->local_stream==str)
+        break;
+
+      pstr=STAILQ_NEXT(pstr,entry);
+    }
+    if(!pstr)
+      ABORT(R_BAD_ARGS);
+
+    if(r=nr_ice_media_stream_find_component(pstr,component_id,compp))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+/*
+   This packet may be for us.
+
+   1. Find the matching peer component
+   2. Examine the packet source address to see if it matches
+   one of the peer candidates.
+   3. Fire the relevant callback handler if there is a match
+
+   Return 0 if match, R_REJECTED if no match, other errors
+   if we can't even find the component or something like that.
+*/
+
+int nr_ice_peer_ctx_deliver_packet_maybe(nr_ice_peer_ctx *pctx, nr_ice_component *comp, nr_transport_addr *source_addr, UCHAR *data, int len)
+  {
+    nr_ice_component *peer_comp;
+    nr_ice_candidate *cand;
+    int r,_status;
+
+    if(r=nr_ice_peer_ctx_find_component(pctx, comp->stream, comp->component_id,
+      &peer_comp))
+      ABORT(r);
+
+    /* OK, we've found the component, now look for matches */
+    cand=TAILQ_FIRST(&peer_comp->candidates);
+    while(cand){
+      if(!nr_transport_addr_cmp(source_addr,&cand->addr,
+        NR_TRANSPORT_ADDR_CMP_MODE_ALL))
+        break;
+
+      cand=TAILQ_NEXT(cand,entry_comp);
+    }
+
+    if(!cand)
+      ABORT(R_REJECTED);
+
+    // accumulate the received bytes for the active candidate pair
+    if (peer_comp->active) {
+      peer_comp->active->bytes_recvd += len;
+      gettimeofday(&peer_comp->active->last_recvd, 0);
+    }
+
+    /* OK, there's a match. Call the handler */
+
+    if (pctx->handler) {
+      r_log(LOG_ICE,LOG_DEBUG,"ICE-PEER(%s): Delivering data", pctx->label);
+
+      pctx->handler->vtbl->msg_recvd(pctx->handler->obj,
+        pctx,comp->stream,comp->component_id,data,len);
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+void nr_ice_peer_ctx_switch_controlling_role(nr_ice_peer_ctx *pctx)
+  {
+    int controlling = !(pctx->controlling);
+    if(pctx->controlling_conflict_resolved) {
+      r_log(LOG_ICE,LOG_WARNING,"ICE(%s): peer (%s) %s called more than once; "
+            "this probably means the peer is confused. Not switching roles.",
+            pctx->ctx->label,pctx->label,__FUNCTION__);
+      return;
+    }
+
+    r_log(LOG_ICE,LOG_INFO,"ICE-PEER(%s): detected "
+          "role conflict. Switching to %s",
+          pctx->label,
+          controlling ? "controlling" : "controlled");
+
+    pctx->controlling = controlling;
+    pctx->controlling_conflict_resolved = 1;
+
+    if(pctx->state == NR_ICE_PEER_STATE_PAIRED) {
+      /*  We have formed candidate pairs. We need to inform them. */
+      nr_ice_media_stream *pstream=STAILQ_FIRST(&pctx->peer_streams);
+      while(pstream) {
+        nr_ice_media_stream_role_change(pstream);
+        pstream = STAILQ_NEXT(pstream, entry);
+      }
+    }
+  }
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_peer_ctx.h b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_peer_ctx.h
new file mode 100644
index 0000000000..ec73d96a03
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_peer_ctx.h
@@ -0,0 +1,101 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _ice_peer_ctx_h
+#define _ice_peer_ctx_h
+#ifdef __cplusplus
+using namespace std;
+extern "C" {
+#endif /* __cplusplus */
+
+struct nr_ice_peer_ctx_ {
+  int state;
+#define NR_ICE_PEER_STATE_UNPAIRED 1
+#define NR_ICE_PEER_STATE_PAIRED   2
+
+  char *label;
+  nr_ice_ctx *ctx;
+  nr_ice_handler *handler;
+
+  UCHAR controlling; /* 1 for controlling, 0 for controlled */
+  UCHAR controlling_conflict_resolved;
+  UINT8 tiebreaker;
+
+  int peer_lite;
+  int peer_ice_mismatch;
+
+  nr_ice_media_stream_head peer_streams;
+  int active_streams;
+  int waiting_pairs;
+  UCHAR checks_started;
+
+  void *connected_cb_timer;
+  UCHAR reported_connected;
+  void *trickle_grace_period_timer;
+
+  STAILQ_ENTRY(nr_ice_peer_ctx_) entry;
+};
+
+typedef STAILQ_HEAD(nr_ice_peer_ctx_head_, nr_ice_peer_ctx_) nr_ice_peer_ctx_head;
+
+int nr_ice_peer_ctx_create(nr_ice_ctx *ctx, nr_ice_handler *handler,char *label, nr_ice_peer_ctx **pctxp);
+void nr_ice_peer_ctx_destroy(nr_ice_peer_ctx** pctxp);
+int nr_ice_peer_ctx_parse_stream_attributes(nr_ice_peer_ctx *pctx, nr_ice_media_stream *stream, char **attrs, int attr_ct);
+int nr_ice_peer_ctx_find_pstream(nr_ice_peer_ctx *pctx, nr_ice_media_stream *stream, nr_ice_media_stream **pstreamp);
+int nr_ice_peer_ctx_remove_pstream(nr_ice_peer_ctx *pctx, nr_ice_media_stream **pstreamp);
+int nr_ice_peer_ctx_parse_trickle_candidate(nr_ice_peer_ctx *pctx, nr_ice_media_stream *stream, char *cand, const char *mdns_addr);
+
+int nr_ice_peer_ctx_pair_candidates(nr_ice_peer_ctx *pctx);
+int nr_ice_peer_ctx_parse_global_attributes(nr_ice_peer_ctx *pctx, char **attrs, int attr_ct);
+int nr_ice_peer_ctx_start_checks(nr_ice_peer_ctx *pctx);
+int nr_ice_peer_ctx_start_checks2(nr_ice_peer_ctx *pctx, int allow_non_first);
+void nr_ice_peer_ctx_stream_started_checks(nr_ice_peer_ctx *pctx, nr_ice_media_stream *stream);
+void nr_ice_peer_ctx_refresh_consent_all_streams(nr_ice_peer_ctx *pctx);
+void nr_ice_peer_ctx_disconnect_all_streams(nr_ice_peer_ctx *pctx);
+void nr_ice_peer_ctx_disconnected(nr_ice_peer_ctx *pctx);
+void nr_ice_peer_ctx_connected(nr_ice_peer_ctx *pctx);
+void nr_ice_peer_ctx_dump_state(nr_ice_peer_ctx *pctx, int log_level);
+int nr_ice_peer_ctx_log_state(nr_ice_peer_ctx *pctx);
+void nr_ice_peer_ctx_check_if_connected(nr_ice_peer_ctx *pctx);
+int nr_ice_peer_ctx_find_component(nr_ice_peer_ctx *pctx, nr_ice_media_stream *str, int component_id, nr_ice_component **compp);
+int nr_ice_peer_ctx_deliver_packet_maybe(nr_ice_peer_ctx *pctx, nr_ice_component *comp, nr_transport_addr *source_addr, UCHAR *data, int len);
+int nr_ice_peer_ctx_disable_component(nr_ice_peer_ctx *pctx, nr_ice_media_stream *lstream, int component_id);
+int nr_ice_peer_ctx_pair_new_trickle_candidate(nr_ice_ctx *ctx, nr_ice_peer_ctx *pctx, nr_ice_candidate *cand);
+void nr_ice_peer_ctx_switch_controlling_role(nr_ice_peer_ctx *pctx);
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_reg.h b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_reg.h
new file mode 100644
index 0000000000..3acc02360a
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_reg.h
@@ -0,0 +1,81 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _ice_reg_h
+#define _ice_reg_h
+#ifdef __cplusplus
+using namespace std;
+extern "C" {
+#endif /* __cplusplus */
+
+#define NR_ICE_REG_PREF_TYPE_HOST           "ice.pref.type.host"
+#define NR_ICE_REG_PREF_TYPE_RELAYED        "ice.pref.type.relayed"
+#define NR_ICE_REG_PREF_TYPE_SRV_RFLX       "ice.pref.type.srv_rflx"
+#define NR_ICE_REG_PREF_TYPE_PEER_RFLX      "ice.pref.type.peer_rflx"
+#define NR_ICE_REG_PREF_TYPE_HOST_TCP       "ice.pref.type.host_tcp"
+#define NR_ICE_REG_PREF_TYPE_RELAYED_TCP    "ice.pref.type.relayed_tcp"
+#define NR_ICE_REG_PREF_TYPE_SRV_RFLX_TCP   "ice.pref.type.srv_rflx_tcp"
+#define NR_ICE_REG_PREF_TYPE_PEER_RFLX_TCP  "ice.pref.type.peer_rflx_tcp"
+
+#define NR_ICE_REG_PREF_INTERFACE_PRFX      "ice.pref.interface"
+#define NR_ICE_REG_SUPPRESS_INTERFACE_PRFX  "ice.suppress.interface"
+
+#define NR_ICE_REG_STUN_SRV_PRFX            "ice.stun.server"
+#define NR_ICE_REG_STUN_SRV_ADDR            "addr"
+#define NR_ICE_REG_STUN_SRV_PORT            "port"
+
+#define NR_ICE_REG_TURN_SRV_PRFX            "ice.turn.server"
+#define NR_ICE_REG_TURN_SRV_ADDR            "addr"
+#define NR_ICE_REG_TURN_SRV_PORT            "port"
+#define NR_ICE_REG_TURN_SRV_BANDWIDTH       "bandwidth"
+#define NR_ICE_REG_TURN_SRV_LIFETIME        "lifetime"
+#define NR_ICE_REG_TURN_SRV_USERNAME        "username"
+#define NR_ICE_REG_TURN_SRV_PASSWORD        "password"
+
+#define NR_ICE_REG_ICE_TCP_DISABLE          "ice.tcp.disable"
+#define NR_ICE_REG_ICE_TCP_SO_SOCK_COUNT    "ice.tcp.so_sock_count"
+#define NR_ICE_REG_ICE_TCP_LISTEN_BACKLOG   "ice.tcp.listen_backlog"
+
+#define NR_ICE_REG_KEEPALIVE_TIMER          "ice.keepalive_timer"
+
+#define NR_ICE_REG_TRICKLE_GRACE_PERIOD     "ice.trickle_grace_period"
+#define NR_ICE_REG_PREF_FORCE_INTERFACE_NAME "ice.forced_interface_name"
+#define NR_ICE_REG_USE_NR_RESOLVER_FOR_TCP  "ice.tcp.use_nr_resolver"
+#define NR_ICE_REG_USE_NR_RESOLVER_FOR_UDP  "ice.udp.use_nr_resolver"
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_socket.c b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_socket.c
new file mode 100644
index 0000000000..a6c8513300
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_socket.c
@@ -0,0 +1,404 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <assert.h>
+#include <string.h>
+#include "nr_api.h"
+#include "ice_ctx.h"
+#include "stun.h"
+#include "nr_socket_buffered_stun.h"
+#include "nr_socket_multi_tcp.h"
+
+static void nr_ice_socket_readable_cb(NR_SOCKET s, int how, void *cb_arg)
+  {
+    int r;
+    nr_ice_stun_ctx *sc1,*sc2;
+    nr_ice_socket *sock=cb_arg;
+    UCHAR buf[9216];
+    char string[256];
+    nr_transport_addr addr;
+    int len;
+    size_t len_s;
+    int is_stun;
+    int is_req;
+    int is_ind;
+    int processed_indication=0;
+
+    nr_socket *stun_srv_sock=sock->sock;
+
+    r_log(LOG_ICE,LOG_DEBUG,"ICE(%s): Socket ready to read",sock->ctx->label);
+
+    if(r=nr_socket_recvfrom(sock->sock,buf,sizeof(buf),&len_s,0,&addr)){
+      if (r != R_WOULDBLOCK && (sock->type != NR_ICE_SOCKET_TYPE_DGRAM)) {
+        /* Report this error upward. Bug 946423 */
+        r_log(LOG_ICE,LOG_ERR,"ICE(%s): Error %d on reliable socket(%p). Abandoning.",sock->ctx->label, r, s);
+        nr_ice_socket_failed(sock);
+        return;
+      }
+    }
+
+    if (sock->type != NR_ICE_SOCKET_TYPE_STREAM_TCP) {
+      r_log(LOG_ICE,LOG_DEBUG,"ICE(%s): rearming",sock->ctx->label);
+      NR_ASYNC_WAIT(s,how,nr_ice_socket_readable_cb,cb_arg);
+    }
+
+    if (r) {
+      return;
+    }
+
+    /* Deal with the fact that sizeof(int) and sizeof(size_t) may not
+       be the same */
+    if (len_s > (size_t)INT_MAX)
+      return;
+
+    len = (int)len_s;
+
+#ifdef USE_TURN
+  re_process:
+#endif /* USE_TURN */
+    r_log(LOG_ICE,LOG_DEBUG,"ICE(%s): Read %d bytes %sfrom %s",sock->ctx->label,len,(processed_indication ? "relayed " : ""),addr.as_string);
+
+    /* First question: is this STUN or not? */
+    is_stun=nr_is_stun_message(buf,len);
+
+    if(is_stun){
+      is_req=nr_is_stun_request_message(buf,len);
+      is_ind=is_req?0:nr_is_stun_indication_message(buf,len);
+
+      snprintf(string, sizeof(string)-1, "ICE(%s): Message is STUN (%s)",sock->ctx->label,
+               is_req ? "request" : (is_ind ? "indication" : "other"));
+      r_dump(NR_LOG_STUN, LOG_DEBUG, string, (char*)buf, len);
+
+
+      /* We need to offer it to all of our stun contexts
+         to see who bites */
+      sc1=TAILQ_FIRST(&sock->stun_ctxs);
+      while(sc1){
+        sc2=TAILQ_NEXT(sc1,entry);
+
+        r=-1;
+        switch(sc1->type){
+          /* This has been deleted, prune... */
+          case NR_ICE_STUN_NONE:
+            TAILQ_REMOVE(&sock->stun_ctxs,sc1,entry);
+            RFREE(sc1);
+            break;
+
+          case NR_ICE_STUN_CLIENT:
+            if(!(is_req||is_ind)){
+                r=nr_stun_client_process_response(sc1->u.client,buf,len,&addr);
+            }
+            break;
+
+          case NR_ICE_STUN_SERVER:
+            if(is_req){
+              r=nr_stun_server_process_request(sc1->u.server,stun_srv_sock,(char *)buf,len,&addr,NR_STUN_AUTH_RULE_SHORT_TERM);
+            }
+            break;
+#ifdef USE_TURN
+          case NR_ICE_TURN_CLIENT:
+            /* data indications are ok, so don't ignore those */
+            /* Check that this is from the right TURN server address. Else
+               skip */
+            if (nr_transport_addr_cmp(
+                    &sc1->u.turn_client.turn_client->turn_server_addr,
+                    &addr, NR_TRANSPORT_ADDR_CMP_MODE_ALL))
+              break;
+
+            if(!is_req){
+              if(!is_ind)
+                r=nr_turn_client_process_response(sc1->u.turn_client.turn_client,buf,len,&addr);
+              else{
+                nr_transport_addr n_addr;
+                size_t n_len;
+
+                if (processed_indication) {
+                  /* Don't allow recursively wrapped indications */
+                  r_log(LOG_ICE, LOG_WARNING,
+                        "ICE(%s): discarding recursively wrapped indication",
+                        sock->ctx->label);
+                  break;
+                }
+                /* This is a bit of a hack. If it's a data indication, strip
+                   off the TURN framing and re-enter. This works because
+                   all STUN processing is on the same physical socket.
+                   We don't care about other kinds of indication */
+                r=nr_turn_client_parse_data_indication(
+                    sc1->u.turn_client.turn_client, &addr,
+                    buf, len, buf, &n_len, len, &n_addr);
+                if(!r){
+                  r_log(LOG_ICE,LOG_DEBUG,"Unwrapped a data indication.");
+                  len=n_len;
+                  nr_transport_addr_copy(&addr,&n_addr);
+                  stun_srv_sock=sc1->u.turn_client.turn_sock;
+                  processed_indication=1;
+                  goto re_process;
+                }
+              }
+            }
+            break;
+#endif /* USE_TURN */
+
+          default:
+            assert(0); /* Can't happen */
+            return;
+        }
+        if(!r) {
+          break;
+        }
+
+        sc1=sc2;
+      }
+      if(!sc1){
+        if (nr_ice_ctx_is_known_id(sock->ctx,((nr_stun_message_header*)buf)->id.octet))
+            r_log(LOG_ICE,LOG_DEBUG,"ICE(%s): Message is a retransmit",sock->ctx->label);
+        else
+            r_log(LOG_ICE,LOG_NOTICE,"ICE(%s): Message does not correspond to any registered stun ctx",sock->ctx->label);
+      }
+    }
+    else{
+      r_log(LOG_ICE,LOG_DEBUG,"ICE(%s): Message is not STUN",sock->ctx->label);
+
+      nr_ice_ctx_deliver_packet(sock->ctx, sock->component, &addr, buf, len);
+    }
+
+    return;
+  }
+
+int nr_ice_socket_create(nr_ice_ctx *ctx,nr_ice_component *comp, nr_socket *nsock, int type, nr_ice_socket **sockp)
+  {
+    nr_ice_socket *sock=0;
+    NR_SOCKET fd;
+    nr_transport_addr addr;
+    int r,_status;
+
+    if(!(sock=RCALLOC(sizeof(nr_ice_socket))))
+      ABORT(R_NO_MEMORY);
+
+    sock->sock=nsock;
+    sock->ctx=ctx;
+    sock->component=comp;
+
+    if(r=nr_socket_getaddr(nsock, &addr))
+      ABORT(r);
+
+    if (type == NR_ICE_SOCKET_TYPE_DGRAM) {
+      assert(addr.protocol == IPPROTO_UDP);
+    }
+    else {
+      assert(addr.protocol == IPPROTO_TCP);
+    }
+    sock->type=type;
+
+    TAILQ_INIT(&sock->candidates);
+    TAILQ_INIT(&sock->stun_ctxs);
+
+    if (sock->type == NR_ICE_SOCKET_TYPE_DGRAM){
+      if((r=nr_socket_getfd(nsock,&fd)))
+        ABORT(r);
+      NR_ASYNC_WAIT(fd,NR_ASYNC_WAIT_READ,nr_ice_socket_readable_cb,sock);
+    }
+    else if (sock->type == NR_ICE_SOCKET_TYPE_STREAM_TURN) {
+      /* some OS's (e.g. Linux) don't like to see un-connected TCP sockets in
+       * the poll socket set. */
+      nr_socket_buffered_stun_set_readable_cb(nsock,nr_ice_socket_readable_cb,sock);
+    }
+    else if (sock->type == NR_ICE_SOCKET_TYPE_STREAM_TCP) {
+      /* in this case we can't hook up using NR_ASYNC_WAIT, because nr_socket_multi_tcp
+         consists of multiple nr_sockets and file descriptors. */
+      if((r=nr_socket_multi_tcp_set_readable_cb(nsock,nr_ice_socket_readable_cb,sock)))
+        ABORT(r);
+    }
+
+    *sockp=sock;
+
+    _status=0;
+  abort:
+    if(_status) RFREE(sock);
+    return(_status);
+  }
+
+
+int nr_ice_socket_destroy(nr_ice_socket **isockp)
+  {
+    nr_ice_stun_ctx *s1,*s2;
+    nr_ice_socket *isock;
+
+    if(!isockp || !*isockp)
+      return(0);
+
+    isock=*isockp;
+    *isockp=0;
+
+    /* Close the socket */
+    nr_ice_socket_close(isock);
+
+    /* The STUN server */
+    nr_stun_server_ctx_destroy(&isock->stun_server);
+
+    /* Now clean up the STUN ctxs */
+    TAILQ_FOREACH_SAFE(s1, &isock->stun_ctxs, entry, s2){
+      TAILQ_REMOVE(&isock->stun_ctxs, s1, entry);
+      RFREE(s1);
+    }
+
+    RFREE(isock);
+
+    return(0);
+  }
+
+int nr_ice_socket_close(nr_ice_socket *isock)
+  {
+#ifdef NR_SOCKET_IS_VOID_PTR
+    NR_SOCKET fd=NULL;
+    NR_SOCKET no_socket = NULL;
+#else
+    NR_SOCKET fd=-1;
+    NR_SOCKET no_socket = -1;
+#endif
+
+    if (!isock||!isock->sock)
+      return(0);
+
+    if (isock->type != NR_ICE_SOCKET_TYPE_STREAM_TCP){
+      nr_socket_getfd(isock->sock,&fd);
+      assert(isock->sock!=0);
+      if(fd != no_socket){
+        NR_ASYNC_CANCEL(fd,NR_ASYNC_WAIT_READ);
+        NR_ASYNC_CANCEL(fd,NR_ASYNC_WAIT_WRITE);
+      }
+    }
+    nr_socket_destroy(&isock->sock);
+
+    return(0);
+  }
+
+int nr_ice_socket_register_stun_client(nr_ice_socket *sock, nr_stun_client_ctx *srv,void **handle)
+  {
+    nr_ice_stun_ctx *sc=0;
+    int _status;
+
+    if(!(sc=RCALLOC(sizeof(nr_ice_stun_ctx))))
+      ABORT(R_NO_MEMORY);
+
+    sc->type=NR_ICE_STUN_CLIENT;
+    sc->u.client=srv;
+
+    TAILQ_INSERT_TAIL(&sock->stun_ctxs,sc,entry);
+
+    *handle=sc;
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_ice_socket_register_stun_server(nr_ice_socket *sock, nr_stun_server_ctx *srv,void **handle)
+  {
+    nr_ice_stun_ctx *sc=0;
+    int _status;
+
+    if(!(sc=RCALLOC(sizeof(nr_ice_stun_ctx))))
+      ABORT(R_NO_MEMORY);
+
+    sc->type=NR_ICE_STUN_SERVER;
+    sc->u.server=srv;
+
+    TAILQ_INSERT_TAIL(&sock->stun_ctxs,sc,entry);
+
+    *handle=sc;
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_ice_socket_register_turn_client(nr_ice_socket *sock, nr_turn_client_ctx *srv,
+                                       nr_socket *turn_socket, void **handle)
+  {
+    nr_ice_stun_ctx *sc=0;
+    int _status;
+
+    if(!(sc=RCALLOC(sizeof(nr_ice_stun_ctx))))
+      ABORT(R_NO_MEMORY);
+
+    sc->type=NR_ICE_TURN_CLIENT;
+    sc->u.turn_client.turn_client=srv;
+    sc->u.turn_client.turn_sock=turn_socket;
+
+    TAILQ_INSERT_TAIL(&sock->stun_ctxs,sc,entry);
+
+    *handle=sc;
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+/* Just mark it deregistered. Don't delete it now because it's not safe
+   in the CB, which is where this is likely to be called */
+int nr_ice_socket_deregister(nr_ice_socket *sock, void *handle)
+  {
+    nr_ice_stun_ctx *sc=handle;
+
+    if(!sc)
+      return(0);
+
+    sc->type=NR_ICE_STUN_NONE;
+
+    return(0);
+  }
+
+void nr_ice_socket_failed(nr_ice_socket *sock)
+  {
+    nr_ice_stun_ctx *s1,*s2;
+    TAILQ_FOREACH_SAFE(s1, &sock->stun_ctxs, entry, s2){
+      switch (s1->type) {
+        case NR_ICE_STUN_NONE:
+          break;
+        case NR_ICE_STUN_CLIENT:
+          nr_stun_client_failed(s1->u.client);
+          break;
+        case NR_ICE_STUN_SERVER:
+          /* Nothing to do here? */
+          break;
+#ifdef USE_TURN
+        case NR_ICE_TURN_CLIENT:
+          nr_turn_client_failed(s1->u.turn_client.turn_client);
+          break;
+#endif
+        default:
+          assert(0);
+      }
+    }
+  }
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_socket.h b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_socket.h
new file mode 100644
index 0000000000..9b73a2690e
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/ice/ice_socket.h
@@ -0,0 +1,98 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _ice_socket_h
+#define _ice_socket_h
+#ifdef __cplusplus
+using namespace std;
+extern "C" {
+#endif /* __cplusplus */
+
+typedef struct nr_ice_stun_ctx_ {
+  int type;
+#define NR_ICE_STUN_NONE    0 /* Deregistered */
+#define NR_ICE_STUN_CLIENT  1
+#define NR_ICE_STUN_SERVER  2
+#define NR_ICE_TURN_CLIENT  3
+
+  union {
+    nr_stun_client_ctx *client;
+    nr_stun_server_ctx *server;
+    struct {
+      nr_turn_client_ctx *turn_client;
+      nr_socket *turn_sock;  /* The nr_socket_turn wrapped around
+                                turn_client */
+    } turn_client;
+  } u;
+
+  TAILQ_ENTRY(nr_ice_stun_ctx_) entry;
+} nr_ice_stun_ctx;
+
+
+typedef struct nr_ice_socket_ {
+  int type;
+#define NR_ICE_SOCKET_TYPE_DGRAM       1
+#define NR_ICE_SOCKET_TYPE_STREAM_TURN 2
+#define NR_ICE_SOCKET_TYPE_STREAM_TCP  3
+
+  nr_socket *sock;
+  nr_ice_ctx *ctx;
+
+  nr_ice_candidate_head candidates;
+  nr_ice_component *component;
+
+  TAILQ_HEAD(nr_ice_stun_ctx_head_,nr_ice_stun_ctx_) stun_ctxs;
+
+  nr_stun_server_ctx *stun_server;
+  void *stun_server_handle;
+
+  STAILQ_ENTRY(nr_ice_socket_) entry;
+} nr_ice_socket;
+
+typedef STAILQ_HEAD(nr_ice_socket_head_,nr_ice_socket_) nr_ice_socket_head;
+
+int nr_ice_socket_create(struct nr_ice_ctx_ *ctx, struct nr_ice_component_ *comp, nr_socket *nsock, int type, nr_ice_socket **sockp);
+int nr_ice_socket_destroy(nr_ice_socket **isock);
+int nr_ice_socket_close(nr_ice_socket *isock);
+int nr_ice_socket_register_stun_client(nr_ice_socket *sock, nr_stun_client_ctx *srv,void **handle);
+int nr_ice_socket_register_stun_server(nr_ice_socket *sock, nr_stun_server_ctx *srv,void **handle);
+int nr_ice_socket_register_turn_client(nr_ice_socket *sock, nr_turn_client_ctx *srv,nr_socket *turn_socket, void **handle);
+int nr_ice_socket_deregister(nr_ice_socket *sock, void *handle);
+void nr_ice_socket_failed(nr_ice_socket *sock);
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/net/local_addr.c b/dom/media/webrtc/transport/third_party/nICEr/src/net/local_addr.c
new file mode 100644
index 0000000000..a0896d5e9d
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/net/local_addr.c
@@ -0,0 +1,70 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+Copyright (c) 2013, Mozilla
+
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance, Mozilla nor
+  the names of its contributors may be used to endorse or promote
+  products derived from this software without specific prior written
+  permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <nr_api.h>
+#include <string.h>
+#include "local_addr.h"
+
+int nr_local_addr_copy(nr_local_addr *to, nr_local_addr *from)
+  {
+    int r,_status;
+
+    if (r=nr_transport_addr_copy(&(to->addr), &(from->addr))) {
+      ABORT(r);
+    }
+    to->interface = from->interface;
+    to->flags = from->flags;
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_local_addr_fmt_info_string(nr_local_addr *addr, char *buf, int len)
+  {
+    int addr_type = addr->interface.type;
+    const char *vpn = (addr_type & NR_INTERFACE_TYPE_VPN) ? "VPN on " : "";
+
+    const char *type = (addr_type & NR_INTERFACE_TYPE_WIRED) ? "wired" :
+                       (addr_type & NR_INTERFACE_TYPE_WIFI) ? "wifi" :
+                       (addr_type & NR_INTERFACE_TYPE_MOBILE) ? "mobile" :
+                       "unknown";
+
+    snprintf(buf, len, "%s%s, estimated speed: %d kbps %s",
+             vpn, type, addr->interface.estimated_speed,
+             (addr->flags & NR_ADDR_FLAG_TEMPORARY ? "temporary" : ""));
+    buf[len - 1] = '\0';
+    return (0);
+  }
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/net/local_addr.h b/dom/media/webrtc/transport/third_party/nICEr/src/net/local_addr.h
new file mode 100644
index 0000000000..fb963e1115
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/net/local_addr.h
@@ -0,0 +1,62 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+Copyright (c) 2013, Mozilla
+
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance, Mozilla nor
+  the names of its contributors may be used to endorse or promote
+  products derived from this software without specific prior written
+  permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef _local_addr_h
+#define _local_addr_h
+
+#include "transport_addr.h"
+
+typedef struct nr_interface_ {
+  int type;
+#define NR_INTERFACE_TYPE_UNKNOWN 0
+#define NR_INTERFACE_TYPE_WIRED   1
+#define NR_INTERFACE_TYPE_WIFI    1 << 1
+#define NR_INTERFACE_TYPE_MOBILE  1 << 2
+#define NR_INTERFACE_TYPE_VPN     1 << 3
+#define NR_INTERFACE_TYPE_TEREDO  1 << 4
+  int estimated_speed; /* Speed in kbps */
+} nr_interface;
+
+typedef struct nr_local_addr_ {
+  nr_transport_addr addr;
+  nr_interface interface;
+#define NR_ADDR_FLAG_TEMPORARY 0x1
+  int flags;
+} nr_local_addr;
+
+int nr_local_addr_copy(nr_local_addr *to, nr_local_addr *from);
+int nr_local_addr_fmt_info_string(nr_local_addr *addr, char *buf, int len);
+
+#endif
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_interface_prioritizer.c b/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_interface_prioritizer.c
new file mode 100644
index 0000000000..75e5f95467
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_interface_prioritizer.c
@@ -0,0 +1,88 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+Copyright (c) 2013, Mozilla
+
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance, Mozilla nor
+  the names of its contributors may be used to endorse or promote
+  products derived from this software without specific prior written
+  permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include "nr_api.h"
+#include "nr_interface_prioritizer.h"
+#include "transport_addr.h"
+
+int nr_interface_prioritizer_create_int(void *obj,
+  nr_interface_prioritizer_vtbl *vtbl,nr_interface_prioritizer **ifpp)
+  {
+    int _status;
+    nr_interface_prioritizer *ifp=0;
+
+    if(!(ifp=RCALLOC(sizeof(nr_interface_prioritizer))))
+      ABORT(R_NO_MEMORY);
+
+    ifp->obj = obj;
+    ifp->vtbl = vtbl;
+
+    *ifpp = ifp;
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_interface_prioritizer_destroy(nr_interface_prioritizer **ifpp)
+  {
+    nr_interface_prioritizer *ifp;
+
+    if (!ifpp || !*ifpp)
+      return(0);
+
+    ifp = *ifpp;
+    *ifpp = 0;
+    ifp->vtbl->destroy(&ifp->obj);
+    RFREE(ifp);
+    return(0);
+  }
+
+int nr_interface_prioritizer_add_interface(nr_interface_prioritizer *ifp,
+  nr_local_addr *addr)
+  {
+    return ifp->vtbl->add_interface(ifp->obj, addr);
+  }
+
+int nr_interface_prioritizer_get_priority(nr_interface_prioritizer *ifp,
+  const char *key, UCHAR *interface_preference)
+  {
+    return ifp->vtbl->get_priority(ifp->obj,key,interface_preference);
+  }
+
+int nr_interface_prioritizer_sort_preference(nr_interface_prioritizer *ifp)
+  {
+    return ifp->vtbl->sort_preference(ifp->obj);
+  }
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_interface_prioritizer.h b/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_interface_prioritizer.h
new file mode 100644
index 0000000000..c8a36526cc
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_interface_prioritizer.h
@@ -0,0 +1,66 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+Copyright (c) 2013, Mozilla
+
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance, Mozilla nor
+  the names of its contributors may be used to endorse or promote
+  products derived from this software without specific prior written
+  permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef _nr_interface_prioritizer
+#define _nr_interface_prioritizer
+
+#include "transport_addr.h"
+#include "local_addr.h"
+
+typedef struct nr_interface_prioritizer_vtbl_ {
+  int (*add_interface)(void *obj, nr_local_addr *iface);
+  int (*get_priority)(void *obj, const char *key, UCHAR *pref);
+  int (*sort_preference)(void *obj);
+  int (*destroy)(void **obj);
+} nr_interface_prioritizer_vtbl;
+
+typedef struct nr_interface_prioritizer_ {
+  void *obj;
+  nr_interface_prioritizer_vtbl *vtbl;
+} nr_interface_prioritizer;
+
+int nr_interface_prioritizer_create_int(void *obj, nr_interface_prioritizer_vtbl *vtbl,
+                                        nr_interface_prioritizer **prioritizer);
+
+int nr_interface_prioritizer_destroy(nr_interface_prioritizer **prioritizer);
+
+int nr_interface_prioritizer_add_interface(nr_interface_prioritizer *prioritizer,
+                                           nr_local_addr *addr);
+
+int nr_interface_prioritizer_get_priority(nr_interface_prioritizer *prioritizer,
+                                          const char *key, UCHAR *interface_preference);
+
+int nr_interface_prioritizer_sort_preference(nr_interface_prioritizer *prioritizer);
+#endif
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_resolver.c b/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_resolver.c
new file mode 100644
index 0000000000..4dbf1bbe91
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_resolver.c
@@ -0,0 +1,85 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+Copyright (c) 2013, Mozilla
+
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance, Mozilla nor
+  the names of its contributors may be used to endorse or promote
+  products derived from this software without specific prior written
+  permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <nr_api.h>
+#include "nr_resolver.h"
+
+int nr_resolver_create_int(void *obj, nr_resolver_vtbl *vtbl, nr_resolver **resolverp)
+{
+  int _status;
+  nr_resolver *resolver=0;
+
+  if (!(resolver=RCALLOC(sizeof(nr_resolver))))
+    ABORT(R_NO_MEMORY);
+
+  resolver->obj=obj;
+  resolver->vtbl=vtbl;
+
+  *resolverp=resolver;
+  _status=0;
+abort:
+  return(_status);
+}
+
+int nr_resolver_destroy(nr_resolver **resolverp)
+{
+  nr_resolver *resolver;
+
+  if(!resolverp || !*resolverp)
+    return(0);
+
+  resolver=*resolverp;
+  *resolverp=0;
+
+  resolver->vtbl->destroy(&resolver->obj);
+
+  RFREE(resolver);
+
+  return(0);
+}
+
+int nr_resolver_resolve(nr_resolver *resolver,
+                        nr_resolver_resource *resource,
+                        int (*cb)(void *cb_arg, nr_transport_addr *addr),
+                        void *cb_arg,
+                        void **handle)
+{
+  return resolver->vtbl->resolve(resolver->obj, resource, cb, cb_arg, handle);
+}
+
+int nr_resolver_cancel(nr_resolver *resolver, void *handle)
+{
+  return resolver->vtbl->cancel(resolver->obj, handle);
+}
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_resolver.h b/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_resolver.h
new file mode 100644
index 0000000000..376ba9998b
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_resolver.h
@@ -0,0 +1,96 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+Copyright (c) 2013, Mozilla
+
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance, Mozilla nor
+  the names of its contributors may be used to endorse or promote
+  products derived from this software without specific prior written
+  permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef _nr_resolver_h
+#define _nr_resolver_h
+
+#include "transport_addr.h"
+
+#define NR_RESOLVE_PROTOCOL_STUN 1
+#define NR_RESOLVE_PROTOCOL_TURN 2
+
+typedef struct nr_resolver_resource_ {
+  const char *domain_name;
+  UINT2 port;
+  int stun_turn;
+  UCHAR transport_protocol;
+  UCHAR address_family;
+} nr_resolver_resource;
+
+typedef struct nr_resolver_vtbl_ {
+  int (*destroy)(void **obj);
+  int (*resolve)(void *obj,
+                 nr_resolver_resource *resource,
+                 int (*cb)(void *cb_arg, nr_transport_addr *addr),
+                 void *cb_arg,
+                 void **handle);
+  int (*cancel)(void *obj, void *handle);
+} nr_resolver_vtbl;
+
+typedef struct nr_resolver_ {
+  void *obj;
+  nr_resolver_vtbl *vtbl;
+} nr_resolver;
+
+
+/*
+  The convention here is that the provider of this interface
+  must generate a void *obj, and a vtbl and then call
+  nr_resolver_create_int() to allocate the generic wrapper
+  object.
+
+  The vtbl must contain implementations for all the functions
+  listed.
+
+  The nr_resolver_destroy() function (and hence vtbl->destroy)
+  will be called when the consumer of the resolver is done
+  with it. That is the signal that it is safe to clean up
+  the resources associated with obj. No other function will
+  be called afterwards.
+*/
+int nr_resolver_create_int(void *obj, nr_resolver_vtbl *vtbl,
+                           nr_resolver **resolverp);
+int nr_resolver_destroy(nr_resolver **resolverp);
+
+/* Request resolution of a domain */
+int nr_resolver_resolve(nr_resolver *resolver,
+                        nr_resolver_resource *resource,
+                        int (*cb)(void *cb_arg, nr_transport_addr *addr),
+                        void *cb_arg,
+                        void **handle);
+
+/* Cancel a requested resolution. No callback will fire. */
+int nr_resolver_cancel(nr_resolver *resolver, void *handle);
+#endif
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_socket.c b/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_socket.c
new file mode 100644
index 0000000000..c9867610a6
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_socket.c
@@ -0,0 +1,187 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <assert.h>
+#include <nr_api.h>
+#include "nr_socket.h"
+#include "local_addr.h"
+
+#define CHECK_DEFINED(f) assert(sock->vtbl->f); if (!sock->vtbl->f) ERETURN(R_INTERNAL);
+int nr_socket_create_int(void *obj, nr_socket_vtbl *vtbl, nr_socket **sockp)
+  {
+    int _status;
+    nr_socket *sock=0;
+
+    if(!(sock=RCALLOC(sizeof(nr_socket))))
+      ABORT(R_NO_MEMORY);
+
+    assert(vtbl->version >= 1 && vtbl->version <= 2);
+    if (vtbl->version < 1 || vtbl->version > 2)
+       ABORT(R_INTERNAL);
+
+    sock->obj=obj;
+    sock->vtbl=vtbl;
+
+    *sockp=sock;
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_socket_destroy(nr_socket **sockp)
+  {
+    nr_socket *sock;
+
+    if(!sockp || !*sockp)
+      return(0);
+
+
+    sock=*sockp;
+    *sockp=0;
+
+    CHECK_DEFINED(destroy);
+
+    assert(sock->vtbl);
+    if (sock->vtbl)
+      sock->vtbl->destroy(&sock->obj);
+
+    RFREE(sock);
+
+    return(0);
+  }
+
+int nr_socket_sendto(nr_socket *sock,const void *msg, size_t len, int flags,
+  const nr_transport_addr *addr)
+  {
+    CHECK_DEFINED(ssendto);
+    return sock->vtbl->ssendto(sock->obj,msg,len,flags,addr);
+  }
+
+int nr_socket_recvfrom(nr_socket *sock,void * restrict buf, size_t maxlen,
+  size_t *len, int flags, nr_transport_addr *addr)
+  {
+    CHECK_DEFINED(srecvfrom);
+    return sock->vtbl->srecvfrom(sock->obj, buf, maxlen, len, flags, addr);
+  }
+
+int nr_socket_getfd(nr_socket *sock, NR_SOCKET *fd)
+  {
+    CHECK_DEFINED(getfd);
+    return sock->vtbl->getfd(sock->obj, fd);
+  }
+
+int nr_socket_getaddr(nr_socket *sock, nr_transport_addr *addrp)
+  {
+    CHECK_DEFINED(getaddr);
+    return sock->vtbl->getaddr(sock->obj, addrp);
+  }
+
+int nr_socket_close(nr_socket *sock)
+  {
+    CHECK_DEFINED(close);
+    return sock->vtbl->close(sock->obj);
+  }
+
+int nr_socket_connect(nr_socket *sock, const nr_transport_addr *addr)
+  {
+    CHECK_DEFINED(connect);
+    return sock->vtbl->connect(sock->obj, addr);
+  }
+
+int nr_socket_write(nr_socket *sock,const void *msg, size_t len, size_t *written, int flags)
+  {
+    CHECK_DEFINED(swrite);
+    return sock->vtbl->swrite(sock->obj, msg, len, written);
+  }
+
+
+int nr_socket_read(nr_socket *sock,void * restrict buf, size_t maxlen,
+  size_t *len, int flags)
+  {
+    CHECK_DEFINED(sread);
+    return sock->vtbl->sread(sock->obj, buf, maxlen, len);
+  }
+
+int nr_socket_listen(nr_socket *sock, int backlog)
+  {
+    assert(sock->vtbl->version >=2 );
+    CHECK_DEFINED(listen);
+    return sock->vtbl->listen(sock->obj, backlog);
+  }
+
+int nr_socket_accept(nr_socket *sock, nr_transport_addr *addrp, nr_socket **sockp)
+{
+  assert(sock->vtbl->version >= 2);
+  CHECK_DEFINED(accept);
+  return sock->vtbl->accept(sock->obj, addrp, sockp);
+}
+
+
+int nr_socket_factory_create_int(void *obj,
+  nr_socket_factory_vtbl *vtbl, nr_socket_factory **factorypp)
+  {
+    int _status;
+    nr_socket_factory *factoryp=0;
+
+    if(!(factoryp=RCALLOC(sizeof(nr_socket_factory))))
+      ABORT(R_NO_MEMORY);
+
+    factoryp->obj = obj;
+    factoryp->vtbl = vtbl;
+
+    *factorypp = factoryp;
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_socket_factory_destroy(nr_socket_factory **factorypp)
+  {
+    nr_socket_factory *factoryp;
+
+    if (!factorypp || !*factorypp)
+      return (0);
+
+    factoryp = *factorypp;
+    *factorypp = NULL;
+    factoryp->vtbl->destroy(&factoryp->obj);
+    RFREE(factoryp);
+    return (0);
+  }
+
+int nr_socket_factory_create_socket(nr_socket_factory *factory, nr_transport_addr *addr, nr_socket **sockp)
+  {
+    return factory->vtbl->create_socket(factory->obj, addr, sockp);
+  }
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_socket.h b/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_socket.h
new file mode 100644
index 0000000000..777837f6cc
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_socket.h
@@ -0,0 +1,123 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _nr_socket_h
+#define _nr_socket_h
+
+#include <sys/types.h>
+#ifdef WIN32
+#include <winsock2.h>
+#include <ws2tcpip.h>
+#else
+#include <sys/socket.h>
+#endif
+
+#include "transport_addr.h"
+#include "csi_platform.h"
+
+#ifdef __cplusplus
+#define restrict
+#elif defined(WIN32)
+/* Undef before defining to avoid a compiler warning */
+#undef restrict
+#define restrict __restrict
+#endif
+
+typedef enum {
+  TCP_TYPE_NONE=0,
+  TCP_TYPE_ACTIVE,
+  TCP_TYPE_PASSIVE,
+  TCP_TYPE_SO,
+  TCP_TYPE_MAX
+} nr_socket_tcp_type;
+
+typedef struct nr_socket_ nr_socket;
+
+typedef struct nr_socket_vtbl_ {
+  UINT4 version;   /* Currently 2 */
+  int (*destroy)(void **obj);
+  int (*ssendto)(void *obj,const void *msg, size_t len, int flags,
+    const nr_transport_addr *addr);
+  int (*srecvfrom)(void *obj,void * restrict buf, size_t maxlen, size_t *len, int flags,
+    nr_transport_addr *addr);
+  int (*getfd)(void *obj, NR_SOCKET *fd);
+  int (*getaddr)(void *obj, nr_transport_addr *addrp);
+  int (*connect)(void *obj, const nr_transport_addr *addr);
+  int (*swrite)(void *obj,const void *msg, size_t len, size_t *written);
+  int (*sread)(void *obj,void * restrict buf, size_t maxlen, size_t *len);
+  int (*close)(void *obj);
+
+  /* available since version 2 */
+  int (*listen)(void *obj, int backlog);
+  int (*accept)(void *obj, nr_transport_addr *addrp, nr_socket **sockp);
+} nr_socket_vtbl;
+
+
+struct nr_socket_ {
+  void *obj;
+  nr_socket_vtbl *vtbl;
+};
+
+typedef struct nr_socket_factory_vtbl_ {
+  int (*create_socket)(void *obj, nr_transport_addr *addr, nr_socket **sockp);
+  int (*destroy)(void **obj);
+} nr_socket_factory_vtbl;
+
+typedef struct nr_socket_factory_ {
+  void *obj;
+  nr_socket_factory_vtbl *vtbl;
+} nr_socket_factory;
+
+/* To be called by constructors */
+int nr_socket_create_int(void *obj, nr_socket_vtbl *vtbl, nr_socket **sockp);
+int nr_socket_destroy(nr_socket **sockp);
+int nr_socket_sendto(nr_socket *sock,const void *msg, size_t len,
+  int flags, const nr_transport_addr *addr);
+int nr_socket_recvfrom(nr_socket *sock,void * restrict buf, size_t maxlen,
+  size_t *len, int flags, nr_transport_addr *addr);
+int nr_socket_getfd(nr_socket *sock, NR_SOCKET *fd);
+int nr_socket_getaddr(nr_socket *sock, nr_transport_addr *addrp);
+int nr_socket_close(nr_socket *sock);
+int nr_socket_connect(nr_socket *sock, const nr_transport_addr *addr);
+int nr_socket_write(nr_socket *sock,const void *msg, size_t len, size_t *written, int flags);
+int nr_socket_read(nr_socket *sock, void * restrict buf, size_t maxlen, size_t *len, int flags);
+int nr_socket_listen(nr_socket *sock, int backlog);
+int nr_socket_accept(nr_socket *sock, nr_transport_addr *addrp, nr_socket **sockp);
+
+int nr_socket_factory_create_int(void *obj, nr_socket_factory_vtbl *vtbl, nr_socket_factory **factorypp);
+int nr_socket_factory_destroy(nr_socket_factory **factoryp);
+int nr_socket_factory_create_socket(nr_socket_factory *factory, nr_transport_addr *addr, nr_socket **sockp);
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_socket_local.h b/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_socket_local.h
new file mode 100644
index 0000000000..a2f813ff66
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_socket_local.h
@@ -0,0 +1,41 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _nr_socket_local_h
+#define _nr_socket_local_h
+
+int nr_socket_local_create(void *obj, nr_transport_addr *addr, nr_socket **sockp);
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_socket_multi_tcp.c b/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_socket_multi_tcp.c
new file mode 100644
index 0000000000..9b2489b214
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_socket_multi_tcp.c
@@ -0,0 +1,642 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+Copyright (c) 2014, Mozilla
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <assert.h>
+#include <sys/types.h>
+
+#include "nr_api.h"
+#include "ice_ctx.h"
+#include "nr_socket.h"
+#include "nr_socket_local.h"
+#include "nr_socket_multi_tcp.h"
+#include "nr_socket_buffered_stun.h"
+#include "async_timer.h"
+
+typedef struct nr_tcp_socket_ctx_ {
+     nr_socket * inner;
+     nr_transport_addr remote_addr;
+     int is_framed;
+
+     TAILQ_ENTRY(nr_tcp_socket_ctx_) entry;
+} nr_tcp_socket_ctx;
+
+typedef TAILQ_HEAD(nr_tcp_socket_head_,nr_tcp_socket_ctx_) nr_tcp_socket_head;
+
+static void nr_tcp_socket_readable_cb(NR_SOCKET s, int how, void *arg);
+
+static int nr_tcp_socket_ctx_destroy(nr_tcp_socket_ctx **objp)
+  {
+    nr_tcp_socket_ctx *sock;
+
+    if (!objp || !*objp)
+      return(0);
+
+    sock=*objp;
+    *objp=0;
+
+    nr_socket_destroy(&sock->inner);
+
+    RFREE(sock);
+
+    return(0);
+  }
+
+/* This takes ownership of nrsock whether it fails or not. */
+static int nr_tcp_socket_ctx_create(nr_socket *nrsock, int is_framed,
+  int max_pending, nr_tcp_socket_ctx **sockp)
+  {
+    int r, _status;
+    nr_tcp_socket_ctx *sock = 0;
+    nr_socket *tcpsock;
+
+    if (!(sock = RCALLOC(sizeof(nr_tcp_socket_ctx)))) {
+      nr_socket_destroy(&nrsock);
+      ABORT(R_NO_MEMORY);
+    }
+
+    if ((r=nr_socket_buffered_stun_create(nrsock, max_pending, is_framed ? ICE_TCP_FRAMING : TURN_TCP_FRAMING, &tcpsock))){
+      nr_socket_destroy(&nrsock);
+      ABORT(r);
+    }
+
+    sock->inner=tcpsock;
+    sock->is_framed=is_framed;
+
+    if ((r=nr_ip4_port_to_transport_addr(ntohl(INADDR_ANY), 0, IPPROTO_TCP, &sock->remote_addr)))
+      ABORT(r);
+
+    *sockp=sock;
+
+    _status=0;
+abort:
+    if (_status) {
+      r_log(LOG_ICE,LOG_DEBUG,"%s:%d function %s failed with error %d",__FILE__,__LINE__,__FUNCTION__,_status);
+      nr_tcp_socket_ctx_destroy(&sock);
+    }
+    return(_status);
+  }
+
+static int nr_tcp_socket_ctx_initialize(nr_tcp_socket_ctx *tcpsock,
+  const nr_transport_addr *addr, void* cb_arg)
+  {
+    int r, _status;
+    NR_SOCKET fd;
+
+    if ((r=nr_transport_addr_copy(&tcpsock->remote_addr, addr)))
+      ABORT(r);
+    if ((r=nr_socket_getfd(tcpsock->inner, &fd)))
+      ABORT(r);
+    NR_ASYNC_WAIT(fd, NR_ASYNC_WAIT_READ, nr_tcp_socket_readable_cb, cb_arg);
+
+    _status=0;
+  abort:
+    if (_status)
+      r_log(LOG_ICE,LOG_DEBUG,"%s:%d function %s(addr:%s) failed with error %d",__FILE__,__LINE__,__FUNCTION__,addr->as_string,_status);
+    return(_status);
+  }
+
+typedef struct nr_socket_multi_tcp_ {
+  nr_ice_ctx *ctx;
+  nr_socket *listen_socket;
+  nr_tcp_socket_head sockets;
+  nr_socket_tcp_type tcp_type;
+  nr_transport_addr addr;
+  NR_async_cb readable_cb;
+  void *readable_cb_arg;
+  int max_pending;
+} nr_socket_multi_tcp;
+
+static int nr_socket_multi_tcp_destroy(void **objp);
+static int nr_socket_multi_tcp_sendto(void *obj,const void *msg, size_t len,
+  int flags, const nr_transport_addr *to);
+static int nr_socket_multi_tcp_recvfrom(void *obj,void * restrict buf,
+  size_t maxlen, size_t *len, int flags, nr_transport_addr *from);
+static int nr_socket_multi_tcp_getaddr(void *obj, nr_transport_addr *addrp);
+static int nr_socket_multi_tcp_close(void *obj);
+static int nr_socket_multi_tcp_connect(void *sock, const nr_transport_addr *addr);
+static int nr_socket_multi_tcp_listen(void *obj, int backlog);
+
+static nr_socket_vtbl nr_socket_multi_tcp_vtbl={
+  2,
+  nr_socket_multi_tcp_destroy,
+  nr_socket_multi_tcp_sendto,
+  nr_socket_multi_tcp_recvfrom,
+  0,
+  nr_socket_multi_tcp_getaddr,
+  nr_socket_multi_tcp_connect,
+  0,
+  0,
+  nr_socket_multi_tcp_close,
+  nr_socket_multi_tcp_listen,
+  0
+};
+
+static int nr_socket_multi_tcp_create_stun_server_socket(
+  nr_socket_multi_tcp *sock, nr_ice_stun_server * stun_server,
+  nr_transport_addr *addr, int max_pending)
+  {
+    int r, _status;
+    nr_tcp_socket_ctx *tcp_socket_ctx=0;
+    nr_socket * nrsock;
+
+    if (stun_server->addr.protocol != IPPROTO_TCP) {
+      r_log(LOG_ICE, LOG_INFO,
+            "%s:%d function %s skipping UDP STUN server(addr:%s)", __FILE__,
+            __LINE__, __FUNCTION__, stun_server->addr.as_string);
+      ABORT(R_BAD_ARGS);
+    }
+
+    if (nr_transport_addr_cmp(&stun_server->addr, addr,
+                              NR_TRANSPORT_ADDR_CMP_MODE_VERSION)) {
+      r_log(LOG_ICE, LOG_INFO,
+            "%s:%d function %s skipping STUN with different IP version (%u) "
+            "than local socket (%u),",
+            __FILE__, __LINE__, __FUNCTION__, stun_server->addr.ip_version,
+            addr->ip_version);
+      ABORT(R_BAD_ARGS);
+    }
+
+    if ((r=nr_socket_factory_create_socket(sock->ctx->socket_factory,addr, &nrsock)))
+      ABORT(r);
+
+    /* This takes ownership of nrsock whether it fails or not. */
+    if ((r=nr_tcp_socket_ctx_create(nrsock, 0, max_pending, &tcp_socket_ctx)))
+      ABORT(r);
+
+    nr_transport_addr stun_server_addr;
+
+    nr_transport_addr_copy(&stun_server_addr, &stun_server->addr);
+    r = nr_socket_connect(tcp_socket_ctx->inner, &stun_server_addr);
+    if (r && r != R_WOULDBLOCK) {
+      r_log(LOG_ICE, LOG_WARNING,
+            "%s:%d function %s connect to STUN server(addr:%s) failed with "
+            "error %d",
+            __FILE__, __LINE__, __FUNCTION__, stun_server_addr.as_string, r);
+      ABORT(r);
+    }
+
+    if ((r = nr_tcp_socket_ctx_initialize(tcp_socket_ctx, &stun_server_addr,
+                                          sock)))
+      ABORT(r);
+
+    TAILQ_INSERT_TAIL(&sock->sockets, tcp_socket_ctx, entry);
+
+    _status=0;
+  abort:
+    if (_status) {
+      nr_tcp_socket_ctx_destroy(&tcp_socket_ctx);
+      r_log(LOG_ICE,LOG_DEBUG,"%s:%d function %s(addr:%s) failed with error %d",__FILE__,__LINE__,__FUNCTION__,addr->as_string,_status);
+    }
+    return(_status);
+  }
+
+int nr_socket_multi_tcp_create(struct nr_ice_ctx_ *ctx,
+    struct nr_ice_component_ *component,
+    nr_transport_addr *addr, nr_socket_tcp_type tcp_type,
+    int precreated_so_count, int max_pending, nr_socket **sockp)
+  {
+    int i=0;
+    int r, _status;
+    nr_socket_multi_tcp *sock=0;
+    nr_tcp_socket_ctx *tcp_socket_ctx;
+    nr_socket * nrsock;
+
+    if (!(sock = RCALLOC(sizeof(nr_socket_multi_tcp))))
+      ABORT(R_NO_MEMORY);
+
+    TAILQ_INIT(&sock->sockets);
+
+    sock->ctx=ctx;
+    sock->max_pending=max_pending;
+    sock->tcp_type=tcp_type;
+    nr_transport_addr_copy(&sock->addr, addr);
+
+    if((tcp_type==TCP_TYPE_PASSIVE) &&
+      ((r=nr_socket_factory_create_socket(sock->ctx->socket_factory, addr, &sock->listen_socket))))
+      ABORT(r);
+
+    if (tcp_type!=TCP_TYPE_ACTIVE) {
+      nr_ice_stun_server *stun_servers;
+      nr_ice_turn_server *turn_servers;
+      int stun_server_ct, turn_server_ct;
+      if (component) {
+        stun_servers = component->stream->stun_servers;
+        turn_servers = component->stream->turn_servers;
+        stun_server_ct = component->stream->stun_server_ct;
+        turn_server_ct = component->stream->turn_server_ct;
+      } else {
+        /* Mainly for unit-testing */
+        stun_servers = ctx->stun_servers_cfg;
+        turn_servers = ctx->turn_servers_cfg;
+        stun_server_ct = ctx->stun_server_ct_cfg;
+        turn_server_ct = ctx->turn_server_ct_cfg;
+      }
+      if (stun_servers) {
+        for (i=0; i<stun_server_ct; ++i) {
+          if ((r=nr_socket_multi_tcp_create_stun_server_socket(sock,
+              stun_servers+i, addr, max_pending))) {
+            if (r!=R_BAD_ARGS) {
+              r_log(LOG_ICE,LOG_WARNING,"%s:%d function %s failed to connect STUN server from addr:%s with error %d",__FILE__,__LINE__,__FUNCTION__,addr->as_string,r);
+            }
+          }
+        }
+      }
+      if (turn_servers) {
+        for (i=0; i<turn_server_ct; ++i) {
+          if ((r=nr_socket_multi_tcp_create_stun_server_socket(sock,
+              &(turn_servers[i]).turn_server, addr, max_pending))) {
+            if (r!=R_BAD_ARGS) {
+              r_log(LOG_ICE,LOG_WARNING,"%s:%d function %s failed to connect TURN server from addr:%s with error %d",__FILE__,__LINE__,__FUNCTION__,addr->as_string,r);
+            }
+          }
+        }
+      }
+    }
+
+    if ((tcp_type==TCP_TYPE_SO)) {
+      for (i=0; i<precreated_so_count; ++i) {
+
+        if ((r=nr_socket_factory_create_socket(sock->ctx->socket_factory, addr, &nrsock)))
+          ABORT(r);
+
+        /* This takes ownership of nrsock whether it fails or not. */
+        if ((r=nr_tcp_socket_ctx_create(nrsock, 1, max_pending, &tcp_socket_ctx))){
+          ABORT(r);
+        }
+        TAILQ_INSERT_TAIL(&sock->sockets, tcp_socket_ctx, entry);
+      }
+    }
+
+    if((r=nr_socket_create_int(sock, &nr_socket_multi_tcp_vtbl, sockp)))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    if (_status) {
+      r_log(LOG_ICE,LOG_DEBUG,"%s:%d function %s(addr:%s) failed with error %d",__FILE__,__LINE__,__FUNCTION__,addr->as_string,_status);
+      nr_socket_multi_tcp_destroy((void**)&sock);
+    }
+    return(_status);
+  }
+
+int nr_socket_multi_tcp_set_readable_cb(nr_socket *sock,
+  NR_async_cb readable_cb, void *readable_cb_arg)
+  {
+    nr_socket_multi_tcp *mtcp_sock = (nr_socket_multi_tcp *)sock->obj;
+
+    mtcp_sock->readable_cb=readable_cb;
+    mtcp_sock->readable_cb_arg=readable_cb_arg;
+
+    return 0;
+  }
+
+#define PREALLOC_CONNECT_FRAMED          0
+#define PREALLOC_CONNECT_NON_FRAMED      1
+#define PREALLOC_DONT_CONNECT_UNLESS_SO  2
+
+static int nr_socket_multi_tcp_get_sock_connected_to(nr_socket_multi_tcp *sock,
+  const nr_transport_addr *to, int preallocated_connect_mode, nr_socket **ret_sock)
+  {
+    int r, _status;
+    nr_tcp_socket_ctx *tcp_sock_ctx;
+    nr_socket * nrsock;
+
+    TAILQ_FOREACH(tcp_sock_ctx, &sock->sockets, entry) {
+      if (!nr_transport_addr_is_wildcard(&tcp_sock_ctx->remote_addr)) {
+        if (!nr_transport_addr_cmp(to, &tcp_sock_ctx->remote_addr, NR_TRANSPORT_ADDR_CMP_MODE_ALL)) {
+          *ret_sock=tcp_sock_ctx->inner;
+          return(0);
+        }
+      }
+    }
+
+    tcp_sock_ctx=NULL;
+    /* not connected yet */
+    if (sock->tcp_type != TCP_TYPE_ACTIVE) {
+      if (preallocated_connect_mode == PREALLOC_DONT_CONNECT_UNLESS_SO && sock->tcp_type != TCP_TYPE_SO)
+        ABORT(R_FAILED);
+
+      /* find free preallocated socket and connect */
+      TAILQ_FOREACH(tcp_sock_ctx, &sock->sockets, entry) {
+        if (nr_transport_addr_is_wildcard(&tcp_sock_ctx->remote_addr)) {
+          if (preallocated_connect_mode == PREALLOC_CONNECT_NON_FRAMED && tcp_sock_ctx->is_framed)
+            continue;
+          if (preallocated_connect_mode != PREALLOC_CONNECT_NON_FRAMED && !tcp_sock_ctx->is_framed)
+            continue;
+
+          if ((r=nr_socket_connect(tcp_sock_ctx->inner, to))){
+            if (r!=R_WOULDBLOCK)
+               ABORT(r);
+          }
+
+          if ((r=nr_tcp_socket_ctx_initialize(tcp_sock_ctx, to, sock)))
+            ABORT(r);
+
+          *ret_sock=tcp_sock_ctx->inner;
+
+          return(0);
+        }
+      }
+      tcp_sock_ctx=NULL;
+      ABORT(R_FAILED);
+    }
+
+    /* if active type - create new socket for each new remote addr */
+    assert(sock->tcp_type == TCP_TYPE_ACTIVE);
+
+    if ((r=nr_socket_factory_create_socket(sock->ctx->socket_factory, &sock->addr, &nrsock)))
+      ABORT(r);
+
+    /* This takes ownership of nrsock whether it fails or not. */
+    if ((r=nr_tcp_socket_ctx_create(nrsock, 1, sock->max_pending, &tcp_sock_ctx))){
+      ABORT(r);
+    }
+
+    TAILQ_INSERT_TAIL(&sock->sockets, tcp_sock_ctx, entry);
+
+    if ((r=nr_socket_connect(tcp_sock_ctx->inner, to))){
+      if (r!=R_WOULDBLOCK)
+        ABORT(r);
+    }
+
+    if ((r=nr_tcp_socket_ctx_initialize(tcp_sock_ctx, to, sock)))
+      ABORT(r);
+
+    *ret_sock=tcp_sock_ctx->inner;
+    tcp_sock_ctx=NULL;
+
+    _status=0;
+  abort:
+    if (_status) {
+      if (tcp_sock_ctx) {
+        r_log(LOG_ICE,LOG_DEBUG,"%s:%d function %s failed with error %d, tcp_sock_ctx remote_addr: %s",__FILE__,__LINE__,__FUNCTION__,_status, tcp_sock_ctx->remote_addr.as_string);
+        TAILQ_REMOVE(&sock->sockets, tcp_sock_ctx, entry);
+        nr_tcp_socket_ctx_destroy(&tcp_sock_ctx);
+      } else {
+        r_log(LOG_ICE,LOG_DEBUG,"%s:%d function %s failed with error %d, tcp_sock_ctx=NULL",__FILE__,__LINE__,__FUNCTION__,_status);
+      }
+    }
+
+    return(_status);
+  }
+
+int nr_socket_multi_tcp_stun_server_connect(nr_socket *sock,
+  const nr_transport_addr *addr)
+  {
+    int r, _status;
+    nr_socket_multi_tcp *mtcp_sock = (nr_socket_multi_tcp *)sock->obj;
+    nr_socket *nrsock;
+
+    assert(mtcp_sock->tcp_type != TCP_TYPE_ACTIVE);
+    if (mtcp_sock->tcp_type == TCP_TYPE_ACTIVE)
+      ABORT(R_INTERNAL);
+
+    if ((r=nr_socket_multi_tcp_get_sock_connected_to(mtcp_sock,addr,PREALLOC_CONNECT_NON_FRAMED,&nrsock)))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    if (_status)
+      r_log(LOG_ICE,LOG_DEBUG,"%s:%d function %s(addr:%s) failed with error %d",__FILE__,__LINE__,__FUNCTION__,addr->as_string,_status);
+    return(_status);
+  }
+
+static int nr_socket_multi_tcp_destroy(void **objp)
+  {
+    nr_socket_multi_tcp *sock;
+    nr_tcp_socket_ctx *tcpsock;
+    NR_SOCKET fd;
+
+    if (!objp || !*objp)
+      return 0;
+
+    sock=(nr_socket_multi_tcp *)*objp;
+    *objp=0;
+
+    /* Cancel waiting on the socket */
+    if (sock->listen_socket && !nr_socket_getfd(sock->listen_socket, &fd)) {
+      NR_ASYNC_CANCEL(fd, NR_ASYNC_WAIT_READ);
+    }
+
+    nr_socket_destroy(&sock->listen_socket);
+
+    while (!TAILQ_EMPTY(&sock->sockets)) {
+
+      tcpsock = TAILQ_FIRST(&sock->sockets);
+      TAILQ_REMOVE(&sock->sockets, tcpsock, entry);
+
+      if (!nr_socket_getfd(tcpsock->inner, &fd)) {
+        NR_ASYNC_CANCEL(fd, NR_ASYNC_WAIT_READ);
+      }
+
+      nr_tcp_socket_ctx_destroy(&tcpsock);
+    }
+
+    RFREE(sock);
+
+    return 0;
+  }
+
+static int nr_socket_multi_tcp_sendto(void *obj, const void *msg, size_t len,
+  int flags, const nr_transport_addr *to)
+  {
+    int r, _status;
+    nr_socket_multi_tcp *sock=(nr_socket_multi_tcp *)obj;
+    nr_socket *nrsock;
+
+    if ((r=nr_socket_multi_tcp_get_sock_connected_to(sock, to,
+      PREALLOC_DONT_CONNECT_UNLESS_SO, &nrsock)))
+      ABORT(r);
+
+    if((r=nr_socket_sendto(nrsock, msg, len, flags, to)))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    if (_status)
+      r_log(LOG_ICE,LOG_DEBUG,"%s:%d function %s(to:%s) failed with error %d",__FILE__,__LINE__,__FUNCTION__,to->as_string,_status);
+
+    return(_status);
+}
+
+static int nr_socket_multi_tcp_recvfrom(void *obj,void * restrict buf,
+  size_t maxlen, size_t *len, int flags, nr_transport_addr *from)
+  {
+    int r, _status = 0;
+    nr_socket_multi_tcp *sock=(nr_socket_multi_tcp *)obj;
+    nr_tcp_socket_ctx *tcpsock;
+
+    if (TAILQ_EMPTY(&sock->sockets))
+      ABORT(R_FAILED);
+
+    TAILQ_FOREACH(tcpsock, &sock->sockets, entry) {
+      if (nr_transport_addr_is_wildcard(&tcpsock->remote_addr))
+        continue;
+      r=nr_socket_recvfrom(tcpsock->inner, buf, maxlen, len, flags, from);
+      if (!r)
+        return 0;
+
+      if (r!=R_WOULDBLOCK) {
+        NR_SOCKET fd;
+        r_log(LOG_ICE,LOG_DEBUG,
+              "%s:%d function %s(to:%s) failed with error %d",__FILE__,
+              __LINE__,__FUNCTION__,tcpsock->remote_addr.as_string,r);
+        if (!nr_socket_getfd(tcpsock->inner, &fd)) {
+          NR_ASYNC_CANCEL(fd, NR_ASYNC_WAIT_READ);
+          NR_ASYNC_CANCEL(fd, NR_ASYNC_WAIT_WRITE);
+        }
+
+        TAILQ_REMOVE(&sock->sockets, tcpsock, entry);
+        nr_tcp_socket_ctx_destroy(&tcpsock);
+        ABORT(r);
+      }
+    }
+
+    /* this also gets returned if all tcpsocks have wildcard remote_addr */
+    _status=R_WOULDBLOCK;
+  abort:
+
+    return(_status);
+  }
+
+static int nr_socket_multi_tcp_getaddr(void *obj, nr_transport_addr *addrp)
+  {
+    nr_socket_multi_tcp *sock=(nr_socket_multi_tcp *)obj;
+
+    return nr_transport_addr_copy(addrp,&sock->addr);
+  }
+
+static int nr_socket_multi_tcp_close(void *obj)
+  {
+    nr_socket_multi_tcp *sock=(nr_socket_multi_tcp *)obj;
+    nr_tcp_socket_ctx *tcpsock;
+
+    if(sock->listen_socket)
+      nr_socket_close(sock->listen_socket);
+
+    TAILQ_FOREACH(tcpsock, &sock->sockets, entry) {
+      nr_socket_close(tcpsock->inner); //ignore errors
+    }
+
+    return 0;
+  }
+
+static void nr_tcp_socket_readable_cb(NR_SOCKET s, int how, void *arg)
+  {
+    nr_socket_multi_tcp *sock=(nr_socket_multi_tcp *)arg;
+
+    // rearm
+    NR_ASYNC_WAIT(s, NR_ASYNC_WAIT_READ, nr_tcp_socket_readable_cb, arg);
+
+    if (sock->readable_cb)
+      sock->readable_cb(s, how, sock->readable_cb_arg);
+  }
+
+static int nr_socket_multi_tcp_connect(void *obj, const nr_transport_addr *addr)
+  {
+    int r, _status;
+    nr_socket_multi_tcp *sock=(nr_socket_multi_tcp *)obj;
+    nr_socket *nrsock;
+
+    if ((r=nr_socket_multi_tcp_get_sock_connected_to(sock,addr,PREALLOC_CONNECT_FRAMED,&nrsock)))
+      ABORT(r);
+
+    _status=0;
+abort:
+    if (_status)
+      r_log(LOG_ICE,LOG_DEBUG,"%s:%d function %s(addr:%s) failed with error %d",__FILE__,__LINE__,__FUNCTION__,addr->as_string,_status);
+
+    return(_status);
+  }
+
+static void nr_tcp_multi_lsocket_readable_cb(NR_SOCKET s, int how, void *arg)
+  {
+    nr_socket_multi_tcp *sock=(nr_socket_multi_tcp *)arg;
+    nr_socket *newsock;
+    nr_transport_addr remote_addr;
+    nr_tcp_socket_ctx *tcp_sock_ctx;
+    int r, _status;
+
+    // rearm
+    NR_ASYNC_WAIT(s, NR_ASYNC_WAIT_READ, nr_tcp_multi_lsocket_readable_cb, arg);
+
+    /* accept */
+    if ((r=nr_socket_accept(sock->listen_socket, &remote_addr, &newsock)))
+      ABORT(r);
+
+    /* This takes ownership of newsock whether it fails or not. */
+    if ((r=nr_tcp_socket_ctx_create(newsock, 1, sock->max_pending, &tcp_sock_ctx)))
+      ABORT(r);
+
+    nr_socket_buffered_set_connected_to(tcp_sock_ctx->inner, &remote_addr);
+
+    if ((r=nr_tcp_socket_ctx_initialize(tcp_sock_ctx, &remote_addr, sock))) {
+      nr_tcp_socket_ctx_destroy(&tcp_sock_ctx);
+      ABORT(r);
+    }
+
+    TAILQ_INSERT_HEAD(&sock->sockets, tcp_sock_ctx, entry);
+
+    _status=0;
+abort:
+    if (_status) {
+      r_log(LOG_ICE,LOG_WARNING,"%s:%d %s failed to accept new TCP connection: %d",__FILE__,__LINE__,__FUNCTION__,_status);
+    } else {
+      r_log(LOG_ICE,LOG_INFO,"%s:%d %s accepted new TCP connection from %s",__FILE__,__LINE__,__FUNCTION__,remote_addr.as_string);
+    }
+  }
+
+static int nr_socket_multi_tcp_listen(void *obj, int backlog)
+  {
+    int r, _status;
+    nr_socket_multi_tcp *sock=(nr_socket_multi_tcp *)obj;
+    NR_SOCKET fd;
+
+    if(!sock->listen_socket)
+      ABORT(R_FAILED);
+
+    if ((r=nr_socket_listen(sock->listen_socket, backlog)))
+      ABORT(r);
+
+    if ((r=nr_socket_getfd(sock->listen_socket, &fd)))
+      ABORT(r);
+
+    NR_ASYNC_WAIT(fd, NR_ASYNC_WAIT_READ, nr_tcp_multi_lsocket_readable_cb, sock);
+
+    _status=0;
+  abort:
+    if (_status)
+      r_log(LOG_ICE,LOG_WARNING,"%s:%d function %s failed with error %d",__FILE__,__LINE__,__FUNCTION__,_status);
+
+    return(_status);
+  }
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_socket_multi_tcp.h b/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_socket_multi_tcp.h
new file mode 100644
index 0000000000..8413e67293
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_socket_multi_tcp.h
@@ -0,0 +1,53 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+Copyright (c) 2014, Mozilla
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef _nr_socket_multi_tcp_h
+#define _nr_socket_multi_tcp_h
+
+#include "nr_socket.h"
+
+/* Argument use_framing is 0 only in call from test code (STUN TCP server
+   listening socket). For other purposes it should be always set to true */
+
+int nr_socket_multi_tcp_create(struct nr_ice_ctx_ *ctx,
+  struct nr_ice_component_ *component,
+  nr_transport_addr *addr,  nr_socket_tcp_type tcp_type,
+  int precreated_so_count, int max_pending, nr_socket **sockp);
+
+int nr_socket_multi_tcp_set_readable_cb(nr_socket *sock,
+  NR_async_cb readable_cb,void *readable_cb_arg);
+
+int nr_socket_multi_tcp_stun_server_connect(nr_socket *sock,
+  const nr_transport_addr *addr);
+
+#endif
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_socket_wrapper.c b/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_socket_wrapper.c
new file mode 100644
index 0000000000..4ad59527c1
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_socket_wrapper.c
@@ -0,0 +1,84 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+Copyright (c) 2013, Mozilla
+
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance, Mozilla nor
+  the names of its contributors may be used to endorse or promote
+  products derived from this software without specific prior written
+  permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <nr_api.h>
+#include "nr_socket_wrapper.h"
+
+#include <assert.h>
+
+int nr_socket_wrapper_factory_create_int(void *obj, nr_socket_wrapper_factory_vtbl *vtbl,
+                                         nr_socket_wrapper_factory **wrapperp)
+{
+  int _status;
+  nr_socket_wrapper_factory *wrapper=0;
+
+  if (!(wrapper=RCALLOC(sizeof(nr_socket_wrapper_factory))))
+    ABORT(R_NO_MEMORY);
+
+  wrapper->obj=obj;
+  wrapper->vtbl=vtbl;
+
+  *wrapperp=wrapper;
+  _status=0;
+abort:
+  return(_status);
+}
+
+int nr_socket_wrapper_factory_wrap(nr_socket_wrapper_factory *wrapper,
+                                   nr_socket *inner,
+                                   nr_socket **socketp)
+{
+  return wrapper->vtbl->wrap(wrapper->obj, inner, socketp);
+}
+
+int nr_socket_wrapper_factory_destroy(nr_socket_wrapper_factory **wrapperp)
+{
+  nr_socket_wrapper_factory *wrapper;
+
+  if (!wrapperp || !*wrapperp)
+    return 0;
+
+  wrapper = *wrapperp;
+  *wrapperp = 0;
+
+  assert(wrapper->vtbl);
+  if (wrapper->vtbl)
+    wrapper->vtbl->destroy(&wrapper->obj);
+
+  RFREE(wrapper);
+
+  return 0;
+}
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_socket_wrapper.h b/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_socket_wrapper.h
new file mode 100644
index 0000000000..717518e23e
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/net/nr_socket_wrapper.h
@@ -0,0 +1,63 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+Copyright (c) 2013, Mozilla
+
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance, Mozilla nor
+  the names of its contributors may be used to endorse or promote
+  products derived from this software without specific prior written
+  permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef _nr_socket_wrapper_h
+#define _nr_socket_wrapper_h
+
+#include "nr_socket.h"
+
+typedef struct nr_socket_wrapper_factory_vtbl_ {
+  int (*wrap)(void *obj,
+                nr_socket *socket,
+                nr_socket **socketp);
+  int (*destroy)(void **obj);
+} nr_socket_wrapper_factory_vtbl;
+
+typedef struct nr_socket_wrapper_factory_ {
+  void *obj;
+  nr_socket_wrapper_factory_vtbl *vtbl;
+} nr_socket_wrapper_factory;
+
+
+int nr_socket_wrapper_factory_create_int(void *obj, nr_socket_wrapper_factory_vtbl *vtbl,
+                                         nr_socket_wrapper_factory **wrapperp);
+
+
+int nr_socket_wrapper_factory_wrap(nr_socket_wrapper_factory *wrapper, nr_socket *inner,
+                                   nr_socket **socketp);
+
+int nr_socket_wrapper_factory_destroy(nr_socket_wrapper_factory **wrapperp);
+
+#endif
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/net/transport_addr.c b/dom/media/webrtc/transport/third_party/nICEr/src/net/transport_addr.c
new file mode 100644
index 0000000000..efedb3782a
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/net/transport_addr.c
@@ -0,0 +1,559 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <csi_platform.h>
+#include <stdio.h>
+#include <memory.h>
+#include <sys/types.h>
+#include <errno.h>
+#ifdef WIN32
+#include <winsock2.h>
+#else
+#include <unistd.h>
+#include <sys/socket.h>
+#include <netinet/in.h>
+#include <arpa/inet.h>
+#endif
+#include <assert.h>
+#include "nr_api.h"
+#include "util.h"
+#include "transport_addr.h"
+
+int nr_transport_addr_fmt_addr_string(nr_transport_addr *addr)
+  {
+    int _status;
+    /* Max length for normalized IPv6 address string representation is 39 */
+    char buffer[40];
+    const char *protocol;
+
+    switch(addr->protocol){
+      case IPPROTO_TCP:
+        if (addr->tls) {
+          protocol = "TLS";
+        } else {
+          protocol = "TCP";
+        }
+        break;
+      case IPPROTO_UDP:
+        protocol = "UDP";
+        break;
+      default:
+        ABORT(R_INTERNAL);
+    }
+
+    switch(addr->ip_version){
+      case NR_IPV4:
+        if (!inet_ntop(AF_INET, &addr->u.addr4.sin_addr,buffer,sizeof(buffer)))
+          strcpy(buffer, "[error]");
+        snprintf(addr->as_string,sizeof(addr->as_string),"IP4:%s:%d/%s",buffer,(int)ntohs(addr->u.addr4.sin_port),protocol);
+        break;
+      case NR_IPV6:
+        if (!inet_ntop(AF_INET6, &addr->u.addr6.sin6_addr,buffer,sizeof(buffer)))
+          strcpy(buffer, "[error]");
+        snprintf(addr->as_string,sizeof(addr->as_string),"IP6:[%s]:%d/%s",buffer,(int)ntohs(addr->u.addr6.sin6_port),protocol);
+        break;
+      default:
+        ABORT(R_INTERNAL);
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_transport_addr_fmt_ifname_addr_string(const nr_transport_addr *addr, char *buf, int len)
+  {
+    int _status;
+    /* leave room for a fully-expanded IPV4-mapped IPV6 address */
+    char buffer[46];
+
+    switch(addr->ip_version){
+      case NR_IPV4:
+        if (!inet_ntop(AF_INET, &addr->u.addr4.sin_addr,buffer,sizeof(buffer))) {
+           strncpy(buffer, "[error]", sizeof(buffer));
+        }
+        break;
+      case NR_IPV6:
+        if (!inet_ntop(AF_INET6, &addr->u.addr6.sin6_addr,buffer,sizeof(buffer))) {
+           strncpy(buffer, "[error]", sizeof(buffer));
+        }
+        break;
+      default:
+        ABORT(R_INTERNAL);
+    }
+    buffer[sizeof(buffer) - 1] = '\0';
+
+    snprintf(buf,len,"%s:%s",addr->ifname,buffer);
+    buf[len - 1] = '\0';
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_sockaddr_to_transport_addr(struct sockaddr *saddr, int protocol, int keep, nr_transport_addr *addr)
+  {
+    int r,_status;
+
+    if(!keep) memset(addr,0,sizeof(nr_transport_addr));
+
+    switch(protocol){
+      case IPPROTO_TCP:
+      case IPPROTO_UDP:
+        break;
+      default:
+        ABORT(R_BAD_ARGS);
+    }
+
+    addr->protocol=protocol;
+
+    if(saddr->sa_family==AF_INET){
+      addr->ip_version=NR_IPV4;
+
+      memcpy(&addr->u.addr4,saddr,sizeof(struct sockaddr_in));
+    }
+    else if(saddr->sa_family==AF_INET6){
+      addr->ip_version=NR_IPV6;
+
+      memcpy(&addr->u.addr6, saddr, sizeof(struct sockaddr_in6));
+    }
+    else
+      ABORT(R_BAD_ARGS);
+
+    if(r=nr_transport_addr_fmt_addr_string(addr))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+
+int nr_transport_addr_copy(nr_transport_addr *to, const nr_transport_addr *from)
+  {
+    memcpy(to,from,sizeof(nr_transport_addr));
+    return 0;
+  }
+
+int nr_transport_addr_copy_keep_ifname(nr_transport_addr *to, const nr_transport_addr *from)
+  {
+    int r,_status;
+    char save_ifname[MAXIFNAME];
+
+    strncpy(save_ifname, to->ifname, MAXIFNAME);
+    save_ifname[MAXIFNAME-1]=0;  /* Ensure null termination */
+
+    if (r=nr_transport_addr_copy(to, from))
+      ABORT(r);
+
+    strncpy(to->ifname, save_ifname, MAXIFNAME);
+
+    if (r=nr_transport_addr_fmt_addr_string(to))
+      ABORT(r);
+
+    _status=0;
+ abort:
+    return _status;
+  }
+
+int nr_transport_addr_copy_addrport(nr_transport_addr *to, const nr_transport_addr *from)
+  {
+    int r,_status;
+
+    switch (from->ip_version) {
+      case NR_IPV4:
+        memcpy(&to->u.addr4, &from->u.addr4, sizeof(to->u.addr4));
+        break;
+      case NR_IPV6:
+        memcpy(&to->u.addr6, &from->u.addr6, sizeof(to->u.addr6));
+        break;
+      default:
+        ABORT(R_BAD_ARGS);
+    }
+
+    to->ip_version = from->ip_version;
+
+    if (r=nr_transport_addr_fmt_addr_string(to)) {
+      ABORT(r);
+    }
+
+    _status=0;
+ abort:
+    return _status;
+  }
+
+/* Convenience fxn. Is this the right API?*/
+int nr_ip4_port_to_transport_addr(UINT4 ip4, UINT2 port, int protocol, nr_transport_addr *addr)
+  {
+    int r,_status;
+
+    memset(addr, 0, sizeof(nr_transport_addr));
+
+    addr->ip_version=NR_IPV4;
+    addr->protocol=protocol;
+#ifdef HAVE_SIN_LEN
+    addr->u.addr4.sin_len=sizeof(struct sockaddr_in);
+#endif
+    addr->u.addr4.sin_family=PF_INET;
+    addr->u.addr4.sin_port=htons(port);
+    addr->u.addr4.sin_addr.s_addr=htonl(ip4);
+
+    if(r=nr_transport_addr_fmt_addr_string(addr))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_str_port_to_transport_addr(const char *ip, UINT2 port, int protocol, nr_transport_addr *addr_out)
+  {
+    int r,_status;
+    struct in_addr addr;
+    struct in6_addr addr6;
+
+    if (inet_pton(AF_INET, ip, &addr) == 1) {
+      if(r=nr_ip4_port_to_transport_addr(ntohl(addr.s_addr),port,protocol,addr_out))
+        ABORT(r);
+    } else if (inet_pton(AF_INET6, ip, &addr6) == 1) {
+      if(r=nr_ip6_port_to_transport_addr(&addr6,port,protocol,addr_out))
+        ABORT(r);
+    } else {
+      ABORT(R_BAD_DATA);
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_ip6_port_to_transport_addr(struct in6_addr* addr6, UINT2 port, int protocol, nr_transport_addr *addr)
+  {
+    int r,_status;
+
+    memset(addr, 0, sizeof(nr_transport_addr));
+
+    addr->ip_version=NR_IPV6;
+    addr->protocol=protocol;
+    addr->u.addr6.sin6_family=PF_INET6;
+    addr->u.addr6.sin6_port=htons(port);
+    memcpy(addr->u.addr6.sin6_addr.s6_addr, addr6->s6_addr, sizeof(addr6->s6_addr));
+
+    if(r=nr_transport_addr_fmt_addr_string(addr))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_transport_addr_get_addrstring(const nr_transport_addr *addr, char *str, int maxlen)
+  {
+    int _status;
+
+    if (addr->fqdn[0]) {
+      strncpy(str, addr->fqdn, maxlen);
+    } else {
+      const char* res;
+      switch (addr->ip_version) {
+        case NR_IPV4:
+          res = inet_ntop(AF_INET, &addr->u.addr4.sin_addr, str, maxlen);
+          break;
+        case NR_IPV6:
+          res = inet_ntop(AF_INET6, &addr->u.addr6.sin6_addr, str, maxlen);
+          break;
+        default:
+          ABORT(R_INTERNAL);
+      }
+
+      if (!res) {
+        if (errno == ENOSPC) {
+          ABORT(R_BAD_ARGS);
+        }
+        ABORT(R_INTERNAL);
+      }
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_transport_addr_get_port(const nr_transport_addr *addr, int *port)
+  {
+    int _status;
+
+    switch(addr->ip_version){
+      case NR_IPV4:
+        *port=ntohs(addr->u.addr4.sin_port);
+        break;
+      case NR_IPV6:
+        *port=ntohs(addr->u.addr6.sin6_port);
+        break;
+      default:
+        ABORT(R_INTERNAL);
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_transport_addr_set_port(nr_transport_addr *addr, int port)
+  {
+    int _status;
+
+    switch(addr->ip_version){
+      case NR_IPV4:
+        addr->u.addr4.sin_port=htons(port);
+        break;
+      case NR_IPV6:
+        addr->u.addr6.sin6_port=htons(port);
+        break;
+      default:
+        ABORT(R_INTERNAL);
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+/* memcmp() may not work if, for instance, the string or interface
+   haven't been made. Hmmm.. */
+int nr_transport_addr_cmp(const nr_transport_addr *addr1,const nr_transport_addr *addr2,int mode)
+  {
+    assert(mode);
+
+    if(addr1->ip_version != addr2->ip_version)
+      return(1);
+
+    if(mode < NR_TRANSPORT_ADDR_CMP_MODE_PROTOCOL)
+      return(0);
+
+    if(addr1->protocol != addr2->protocol)
+      return(1);
+
+    if(mode < NR_TRANSPORT_ADDR_CMP_MODE_ADDR)
+      return(0);
+
+    switch(addr1->ip_version){
+      case NR_IPV4:
+        if(addr1->u.addr4.sin_addr.s_addr != addr2->u.addr4.sin_addr.s_addr)
+          return(1);
+        if(mode < NR_TRANSPORT_ADDR_CMP_MODE_ALL)
+          return(0);
+        if(addr1->u.addr4.sin_port != addr2->u.addr4.sin_port)
+          return(1);
+        break;
+      case NR_IPV6:
+        if(memcmp(addr1->u.addr6.sin6_addr.s6_addr,addr2->u.addr6.sin6_addr.s6_addr,sizeof(struct in6_addr)))
+          return(1);
+        if(mode < NR_TRANSPORT_ADDR_CMP_MODE_ALL)
+          return(0);
+        if(addr1->u.addr6.sin6_port != addr2->u.addr6.sin6_port)
+          return(1);
+        break;
+      default:
+        abort();
+    }
+
+    return(0);
+  }
+
+int nr_transport_addr_is_loopback(const nr_transport_addr *addr)
+  {
+    switch(addr->ip_version){
+      case NR_IPV4:
+        switch(addr->u.addr4.sin_family){
+          case AF_INET:
+            if (((ntohl(addr->u.addr4.sin_addr.s_addr)>>24)&0xff)==0x7f)
+              return 1;
+            break;
+          default:
+            NR_UNIMPLEMENTED;
+            break;
+        }
+        break;
+
+      case NR_IPV6:
+        if(!memcmp(addr->u.addr6.sin6_addr.s6_addr,in6addr_loopback.s6_addr,sizeof(struct in6_addr)))
+          return(1);
+        break;
+      default:
+        NR_UNIMPLEMENTED;
+    }
+
+    return(0);
+  }
+
+int nr_transport_addr_is_link_local(const nr_transport_addr *addr)
+  {
+    switch(addr->ip_version){
+      case NR_IPV4:
+        /* RFC3927: 169.254/16 */
+        if ((ntohl(addr->u.addr4.sin_addr.s_addr) & 0xFFFF0000) == 0xA9FE0000)
+          return(1);
+        break;
+      case NR_IPV6:
+        {
+          UINT4* addrTop = (UINT4*)(addr->u.addr6.sin6_addr.s6_addr);
+          if ((*addrTop & htonl(0xFFC00000)) == htonl(0xFE800000))
+            return(2);
+        }
+        break;
+      default:
+        NR_UNIMPLEMENTED;
+    }
+
+    return(0);
+  }
+
+int nr_transport_addr_is_mac_based(const nr_transport_addr *addr)
+  {
+    switch(addr->ip_version){
+      case NR_IPV4:
+        // IPv4 has no MAC based self assigned IP addresses
+        return(0);
+      case NR_IPV6:
+        {
+          // RFC 2373, Appendix A: lower 64bit 0x020000FFFE000000
+          // indicates a MAC based IPv6 address
+          UINT4* macCom = (UINT4*)(addr->u.addr6.sin6_addr.s6_addr + 8);
+          UINT4* macExt = (UINT4*)(addr->u.addr6.sin6_addr.s6_addr + 12);
+          if ((*macCom & htonl(0x020000FF)) == htonl(0x020000FF) &&
+              (*macExt & htonl(0xFF000000)) == htonl(0xFE000000)) {
+            return(1);
+          }
+        }
+        break;
+      default:
+        NR_UNIMPLEMENTED;
+    }
+    return(0);
+  }
+
+int nr_transport_addr_is_teredo(const nr_transport_addr *addr)
+  {
+    switch(addr->ip_version){
+      case NR_IPV4:
+        return(0);
+      case NR_IPV6:
+        {
+          UINT4* addrTop = (UINT4*)(addr->u.addr6.sin6_addr.s6_addr);
+          if ((*addrTop & htonl(0xFFFFFFFF)) == htonl(0x20010000))
+            return(1);
+        }
+        break;
+      default:
+        NR_UNIMPLEMENTED;
+    }
+
+    return(0);
+  }
+
+int nr_transport_addr_check_compatibility(const nr_transport_addr *addr1, const nr_transport_addr *addr2)
+  {
+    // first make sure we're comparing the same ip versions and protocols
+    if ((addr1->ip_version != addr2->ip_version) ||
+        (addr1->protocol != addr2->protocol)) {
+      return(1);
+    }
+
+    if (!addr1->fqdn[0] && !addr2->fqdn[0]) {
+      // now make sure the link local status matches
+      if (nr_transport_addr_is_link_local(addr1) !=
+          nr_transport_addr_is_link_local(addr2)) {
+        return(1);
+      }
+    }
+    return(0);
+  }
+
+int nr_transport_addr_is_wildcard(const nr_transport_addr *addr)
+  {
+    switch(addr->ip_version){
+      case NR_IPV4:
+        if(addr->u.addr4.sin_addr.s_addr==INADDR_ANY)
+          return(1);
+        if(addr->u.addr4.sin_port==0)
+          return(1);
+        break;
+      case NR_IPV6:
+        if(!memcmp(addr->u.addr6.sin6_addr.s6_addr,in6addr_any.s6_addr,sizeof(struct in6_addr)))
+          return(1);
+        if(addr->u.addr6.sin6_port==0)
+          return(1);
+        break;
+      default:
+        NR_UNIMPLEMENTED;
+    }
+
+    return(0);
+  }
+
+nr_transport_addr_mask nr_private_ipv4_addrs[] = {
+  /* RFC1918: 10/8 */
+  {0x0A000000, 0xFF000000},
+  /* RFC1918: 172.16/12 */
+  {0xAC100000, 0xFFF00000},
+  /* RFC1918: 192.168/16 */
+  {0xC0A80000, 0xFFFF0000},
+  /* RFC6598: 100.64/10 */
+  {0x64400000, 0xFFC00000}
+};
+
+int nr_transport_addr_get_private_addr_range(const nr_transport_addr *addr)
+  {
+    switch(addr->ip_version){
+      case NR_IPV4:
+        {
+          UINT4 ip = ntohl(addr->u.addr4.sin_addr.s_addr);
+          for (size_t i=0; i<(sizeof(nr_private_ipv4_addrs)/sizeof(nr_transport_addr_mask)); i++) {
+            if ((ip & nr_private_ipv4_addrs[i].mask) == nr_private_ipv4_addrs[i].addr)
+              return i + 1;
+          }
+        }
+        break;
+      case NR_IPV6:
+        return(0);
+      default:
+        NR_UNIMPLEMENTED;
+    }
+
+    return(0);
+  }
+
+int nr_transport_addr_is_reliable_transport(const nr_transport_addr *addr)
+  {
+    return addr->protocol == IPPROTO_TCP;
+  }
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/net/transport_addr.h b/dom/media/webrtc/transport/third_party/nICEr/src/net/transport_addr.h
new file mode 100644
index 0000000000..e8679a7e5a
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/net/transport_addr.h
@@ -0,0 +1,128 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _transport_addr_h
+#define _transport_addr_h
+
+#include <stdbool.h>
+#include <sys/types.h>
+#ifdef WIN32
+
+// FIXME: This is dangerous, but exactly the pattern used in
+// nrappkit/src/port/win32/include/csi_platform.h
+// Not good because INT8 are typedefed to different values in
+// <winsock2.h> and <r_types.h>.
+// {
+typedef unsigned char UBLAH_IGNORE_ME_PLEASE;
+typedef signed char BLAH_IGNORE_ME_PLEASE;
+#define UINT8 UBLAH_IGNORE_ME_PLEASE
+#define INT8 BLAH_IGNORE_ME_PLEASE
+#include <winsock2.h>
+#undef UINT8
+#undef INT8
+#include <r_types.h>
+
+// }
+
+#include <ws2tcpip.h>
+#else
+#include <sys/socket.h>
+#include <netinet/in.h>
+#endif
+
+#include "r_types.h"
+
+/* Length of a string  hex representation of a MD5 hash */
+#define MAXIFNAME 33
+
+/* Generic transport address
+
+   This spans both sockaddr_in and sockaddr_in6
+ */
+typedef struct nr_transport_addr_ {
+  UCHAR ip_version;  /* 4 or 6 */
+#define NR_IPV4 4
+#define NR_IPV6 6
+  UCHAR protocol;    /* IPPROTO_TCP, IPPROTO_UDP */
+  union {
+    struct sockaddr_in addr4;
+    struct sockaddr_in6 addr6;
+  } u;
+  char ifname[MAXIFNAME];
+  /* A string version.
+     56 = 5 ("IP6:[") + 39 (ipv6 address) + 2 ("]:") + 5 (port) + 4 (/UDP) + 1 (null) */
+  char as_string[56];
+  char fqdn[256];
+  bool is_proxied;
+  bool tls;
+} nr_transport_addr;
+
+typedef struct nr_transport_addr_mask_ {
+  UINT4 addr;
+  UINT4 mask;
+} nr_transport_addr_mask;
+
+int nr_sockaddr_to_transport_addr(struct sockaddr *saddr, int protocol, int keep, nr_transport_addr *addr);
+
+// addresses, ports in local byte order
+int nr_ip4_port_to_transport_addr(UINT4 ip4, UINT2 port, int protocol, nr_transport_addr *addr);
+int nr_str_port_to_transport_addr(const char *str, UINT2 port, int protocol, nr_transport_addr *addr);
+int nr_ip6_port_to_transport_addr(struct in6_addr* addr6, UINT2 port, int protocol, nr_transport_addr *addr);
+
+int nr_transport_addr_get_addrstring(const nr_transport_addr *addr, char *str, int maxlen);
+int nr_transport_addr_get_port(const nr_transport_addr *addr, int *port);
+int nr_transport_addr_cmp(const nr_transport_addr *addr1,const nr_transport_addr *addr2,int mode);
+#define NR_TRANSPORT_ADDR_CMP_MODE_VERSION   1
+#define NR_TRANSPORT_ADDR_CMP_MODE_PROTOCOL  2
+#define NR_TRANSPORT_ADDR_CMP_MODE_ADDR      3
+#define NR_TRANSPORT_ADDR_CMP_MODE_ALL       4
+
+int nr_transport_addr_is_wildcard(const nr_transport_addr *addr);
+int nr_transport_addr_is_loopback(const nr_transport_addr *addr);
+int nr_transport_addr_get_private_addr_range(const nr_transport_addr *addr);
+int nr_transport_addr_is_link_local(const nr_transport_addr *addr);
+int nr_transport_addr_is_mac_based(const nr_transport_addr *addr);
+int nr_transport_addr_is_teredo(const nr_transport_addr *addr);
+int nr_transport_addr_check_compatibility(const nr_transport_addr *addr1, const nr_transport_addr *addr2);
+int nr_transport_addr_copy(nr_transport_addr *to, const nr_transport_addr *from);
+int nr_transport_addr_copy_keep_ifname(nr_transport_addr *to, const nr_transport_addr *from);
+/* Copies _just_ the address and port (also handles IP version) */
+int nr_transport_addr_copy_addrport(nr_transport_addr *to, const nr_transport_addr *from);
+int nr_transport_addr_fmt_addr_string(nr_transport_addr *addr);
+int nr_transport_addr_fmt_ifname_addr_string(const nr_transport_addr *addr, char *buf, int len);
+int nr_transport_addr_set_port(nr_transport_addr *addr, int port);
+int nr_transport_addr_is_reliable_transport(const nr_transport_addr *addr);
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/net/transport_addr_reg.c b/dom/media/webrtc/transport/third_party/nICEr/src/net/transport_addr_reg.c
new file mode 100644
index 0000000000..10f93f1947
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/net/transport_addr_reg.c
@@ -0,0 +1,230 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <csi_platform.h>
+#include <stdio.h>
+#include <string.h>
+#include <memory.h>
+#include <sys/types.h>
+#ifdef WIN32
+#include <winsock2.h>
+#else
+#include <strings.h>
+#include <unistd.h>
+#include <sys/socket.h>
+#include <netinet/in.h>
+#include <arpa/inet.h>
+#endif
+#include <assert.h>
+#include "nr_api.h"
+#include "util.h"
+#include "transport_addr.h"
+#include "transport_addr_reg.h"
+
+#ifndef INET6_ADDRSTRLEN
+#define INET6_ADDRSTRLEN 46 /* Value used by linux/BSD */
+#endif
+
+int
+nr_reg_get_transport_addr(NR_registry prefix, int keep, nr_transport_addr *addr)
+{
+    int r,_status;
+    unsigned int count;
+    char *address = 0;
+    UINT2 port = 0;
+    char *ifname = 0;
+    char *protocol = 0;
+    int p;
+
+    if ((r=NR_reg_get_child_count(prefix, &count)))
+        ABORT(r);
+
+    if (count == 0)
+        ABORT(R_NOT_FOUND);
+
+    if ((r=NR_reg_alloc2_string(prefix, "address", &address))) {
+        if (r != R_NOT_FOUND)
+            ABORT(r);
+        address = 0;
+    }
+
+    if ((r=NR_reg_alloc2_string(prefix, "ifname", &ifname))) {
+        if (r != R_NOT_FOUND)
+            ABORT(r);
+        ifname = 0;
+    }
+
+    if ((r=NR_reg_get2_uint2(prefix, "port", &port))) {
+        if (r != R_NOT_FOUND)
+            ABORT(r);
+        port = 0;
+    }
+
+    if ((r=NR_reg_alloc2_string(prefix, "protocol", &protocol))) {
+        if (r != R_NOT_FOUND)
+            ABORT(r);
+        p = IPPROTO_UDP;
+
+        protocol = 0;
+    }
+    else {
+        if (!strcasecmp("tcp", protocol))
+            p = IPPROTO_TCP;
+        else if (!strcasecmp("udp", protocol))
+            p = IPPROTO_UDP;
+        else
+            ABORT(R_BAD_DATA);
+    }
+
+    if (!keep) memset(addr, 0, sizeof(*addr));
+
+    if ((r=nr_str_port_to_transport_addr(address?address:"0.0.0.0", port, p, addr)))
+        ABORT(r);
+
+    if (ifname) {
+        (void)strlcpy(addr->ifname, ifname, sizeof(addr->ifname));
+    }
+
+    _status=0;
+  abort:
+    RFREE(protocol);
+    RFREE(ifname);
+    RFREE(address);
+    return(_status);
+}
+
+int
+nr_reg_set_transport_addr(NR_registry prefix, int keep, nr_transport_addr *addr)
+{
+    int r,_status;
+
+    if (! keep) {
+        if ((r=NR_reg_del(prefix)))
+            ABORT(r);
+    }
+
+    switch (addr->ip_version) {
+    case NR_IPV4:
+        if (!nr_transport_addr_is_wildcard(addr)) {
+            if ((r=NR_reg_set2_string(prefix, "address", inet_ntoa(addr->u.addr4.sin_addr))))
+                ABORT(r);
+        }
+
+        if (addr->u.addr4.sin_port != 0) {
+            if ((r=NR_reg_set2_uint2(prefix, "port", ntohs(addr->u.addr4.sin_port))))
+                ABORT(r);
+        }
+        break;
+
+    case NR_IPV6:
+        if (!nr_transport_addr_is_wildcard(addr)) {
+          char address[INET6_ADDRSTRLEN];
+          if(!inet_ntop(AF_INET6, &addr->u.addr6.sin6_addr,address,sizeof(address))) {
+            ABORT(R_BAD_DATA);
+          }
+
+          if ((r=NR_reg_set2_string(prefix, "address", address))) {
+            ABORT(r);
+          }
+        }
+
+        if (addr->u.addr6.sin6_port != 0) {
+            if ((r=NR_reg_set2_uint2(prefix, "port", ntohs(addr->u.addr6.sin6_port))))
+                ABORT(r);
+        }
+        break;
+    default:
+        ABORT(R_INTERNAL);
+        break;
+    }
+
+    /* We abort if neither NR_IPV4 or NR_IPV6 above */
+    switch (addr->protocol) {
+      case IPPROTO_TCP:
+        if ((r=NR_reg_set2_string(prefix, "protocol", "tcp")))
+          ABORT(r);
+        break;
+      case IPPROTO_UDP:
+        if ((r=NR_reg_set2_string(prefix, "protocol", "udp")))
+          ABORT(r);
+        break;
+      default:
+        NR_UNIMPLEMENTED;
+        break;
+    }
+
+    if (strlen(addr->ifname) > 0) {
+      if ((r=NR_reg_set2_string(prefix, "ifname", addr->ifname)))
+        ABORT(r);
+    }
+
+    _status=0;
+  abort:
+    if (_status)
+        NR_reg_del(prefix);
+    return _status;
+}
+
+int
+nr_reg_get_transport_addr2(NR_registry prefix, char *name, int keep, nr_transport_addr *addr)
+{
+    int r, _status;
+    NR_registry registry;
+
+    if ((r=NR_reg_make_registry(prefix, name, registry)))
+        ABORT(r);
+
+    if ((r=nr_reg_get_transport_addr(registry, keep, addr)))
+        ABORT(r);
+
+    _status = 0;
+abort:
+    return _status;
+}
+
+int
+nr_reg_set_transport_addr2(NR_registry prefix, char *name, int keep, nr_transport_addr *addr)
+{
+    int r, _status;
+    NR_registry registry;
+
+    if ((r=NR_reg_make_registry(prefix, name, registry)))
+        ABORT(r);
+
+    if ((r=nr_reg_set_transport_addr(registry, keep, addr)))
+        ABORT(r);
+
+    _status = 0;
+abort:
+    return _status;
+}
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/net/transport_addr_reg.h b/dom/media/webrtc/transport/third_party/nICEr/src/net/transport_addr_reg.h
new file mode 100644
index 0000000000..761953a9ce
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/net/transport_addr_reg.h
@@ -0,0 +1,46 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _transport_addr_reg_h
+#define _transport_addr_reg_h
+
+#include "registry.h"
+
+int nr_reg_get_transport_addr(NR_registry prefix, int keep, nr_transport_addr *addr);
+int nr_reg_set_transport_addr(NR_registry prefix, int keep, nr_transport_addr *addr);
+int nr_reg_get_transport_addr2(NR_registry prefix, char *name, int keep, nr_transport_addr *addr);
+int nr_reg_set_transport_addr2(NR_registry prefix, char *name, int keep, nr_transport_addr *addr);
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/addrs-bsd.c b/dom/media/webrtc/transport/third_party/nICEr/src/stun/addrs-bsd.c
new file mode 100644
index 0000000000..ec2d084445
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/addrs-bsd.c
@@ -0,0 +1,110 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#if defined(BSD) || defined(DARWIN)
+#include "addrs-bsd.h"
+#include <csi_platform.h>
+#include <assert.h>
+#include <string.h>
+#include "util.h"
+#include "stun_util.h"
+#include "util.h"
+#include <r_macros.h>
+
+#include <sys/types.h> /* getifaddrs */
+#include <ifaddrs.h> /* getifaddrs */
+#include <sys/socket.h>
+#include <sys/ioctl.h>
+#include <errno.h>
+#include <netinet/in.h>
+#include <netinet6/in6_var.h>
+
+static int
+stun_ifaddr_get_v6_flags(struct ifaddrs *ifaddr)
+{
+  if (ifaddr->ifa_addr->sa_family != AF_INET6) {
+    return 0;
+  }
+
+  int flags = 0;
+  int s = socket(AF_INET6, SOCK_DGRAM, 0);
+  if (!s) {
+    r_log(NR_LOG_STUN, LOG_ERR, "socket(AF_INET6, SOCK_DGRAM, 0) failed, errno=%d", errno);
+    assert(0);
+    return 0;
+  }
+  struct in6_ifreq ifr6;
+  memset(&ifr6, 0, sizeof(ifr6));
+  strncpy(ifr6.ifr_name, ifaddr->ifa_name, sizeof(ifr6.ifr_name));
+  /* ifr_addr is a sockaddr_in6, ifa_addr is a sockaddr* */
+  struct sockaddr_in6 *sin6 = (struct sockaddr_in6*)ifaddr->ifa_addr;
+  ifr6.ifr_addr = *sin6;
+  if (ioctl(s, SIOCGIFAFLAG_IN6, &ifr6) != -1) {
+    flags = ifr6.ifr_ifru.ifru_flags6;
+  } else {
+    r_log(NR_LOG_STUN, LOG_ERR, "ioctl(SIOCGIFAFLAG_IN6) failed, errno=%d", errno);
+    assert(0);
+  }
+  close(s);
+  return flags;
+}
+
+static int
+stun_ifaddr_is_disallowed_v6(int flags) {
+  return flags & (IN6_IFF_ANYCAST | IN6_IFF_TENTATIVE | IN6_IFF_DUPLICATED | IN6_IFF_DETACHED | IN6_IFF_DEPRECATED);
+}
+
+int stun_getaddrs_filtered(nr_local_addr addrs[], int maxaddrs, int *count)
+{
+  int r,_status,flags;
+  struct ifaddrs* if_addrs_head=NULL;
+  struct ifaddrs* if_addr;
+
+  *count = 0;
+
+  if (maxaddrs <= 0)
+    ABORT(R_BAD_ARGS);
+
+  if (getifaddrs(&if_addrs_head) == -1) {
+    r_log(NR_LOG_STUN, LOG_ERR, "getifaddrs error e = %d", errno);
+    ABORT(R_INTERNAL);
+  }
+
+  if_addr = if_addrs_head;
+
+  while (if_addr && *count < maxaddrs) {
+    /* This can be null */
+    if (if_addr->ifa_addr) {
+      switch (if_addr->ifa_addr->sa_family) {
+        case AF_INET:
+        case AF_INET6:
+          flags = stun_ifaddr_get_v6_flags(if_addr);
+          if (!stun_ifaddr_is_disallowed_v6(flags)) {
+            if (r=nr_sockaddr_to_transport_addr(if_addr->ifa_addr, IPPROTO_UDP, 0, &(addrs[*count].addr))) {
+              r_log(NR_LOG_STUN, LOG_ERR, "nr_sockaddr_to_transport_addr error r = %d", r);
+            } else {
+              if (flags & IN6_IFF_TEMPORARY) {
+                addrs[*count].flags |= NR_ADDR_FLAG_TEMPORARY;
+              }
+              (void)strlcpy(addrs[*count].addr.ifname, if_addr->ifa_name, sizeof(addrs[*count].addr.ifname));
+              ++(*count);
+            }
+          }
+          break;
+        default:
+          ;
+      }
+    }
+
+    if_addr = if_addr->ifa_next;
+  }
+
+  _status=0;
+abort:
+  if (if_addrs_head) {
+    freeifaddrs(if_addrs_head);
+  }
+  return(_status);
+}
+#endif
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/addrs-bsd.h b/dom/media/webrtc/transport/third_party/nICEr/src/stun/addrs-bsd.h
new file mode 100644
index 0000000000..b575586f48
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/addrs-bsd.h
@@ -0,0 +1,13 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef STUN_IFADDRS_BSD_H_
+#define STUN_IFADDRS_BSD_H_
+
+#include "local_addr.h"
+
+int stun_getaddrs_filtered(nr_local_addr addrs[], int maxaddrs, int *count);
+
+#endif  /* STUN_IFADDRS_BSD_H_ */
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/addrs-netlink.c b/dom/media/webrtc/transport/third_party/nICEr/src/stun/addrs-netlink.c
new file mode 100644
index 0000000000..8d22e5979e
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/addrs-netlink.c
@@ -0,0 +1,285 @@
+/*
+Copyright (c) 2011, The WebRTC project authors. All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+  * Redistributions of source code must retain the above copyright
+    notice, this list of conditions and the following disclaimer.
+
+  * Redistributions in binary form must reproduce the above copyright
+    notice, this list of conditions and the following disclaimer in
+    the documentation and/or other materials provided with the
+    distribution.
+
+  * Neither the name of Google nor the names of its contributors may
+    be used to endorse or promote products derived from this software
+    without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+HOLDER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#if defined(LINUX)
+#include <net/if.h>
+#include "addrs-netlink.h"
+#include <csi_platform.h>
+#include <assert.h>
+#include <string.h>
+#include "util.h"
+#include "stun_util.h"
+#include "util.h"
+#include <r_macros.h>
+
+#include <stdlib.h>
+#include <sys/types.h>
+#include <sys/socket.h>
+#include <sys/utsname.h>
+#include <sys/ioctl.h>
+#include <netinet/in.h>
+#include <unistd.h>
+#include <errno.h>
+#include <linux/netlink.h>
+#include <linux/rtnetlink.h>
+
+#ifdef ANDROID
+/* Work around an Android NDK < r8c bug */
+#undef __unused
+#else
+#include <linux/if.h> /* struct ifreq, IFF_POINTTOPOINT */
+#include <linux/wireless.h> /* struct iwreq */
+#include <linux/ethtool.h> /* struct ethtool_cmd */
+#include <linux/sockios.h> /* SIOCETHTOOL */
+#endif /* ANDROID */
+
+
+struct netlinkrequest {
+  struct nlmsghdr header;
+  struct ifaddrmsg msg;
+};
+
+static const int kMaxReadSize = 4096;
+
+static void set_ifname(nr_local_addr *addr, struct ifaddrmsg* msg) {
+  assert(sizeof(addr->addr.ifname) > IF_NAMESIZE);
+  if_indextoname(msg->ifa_index, addr->addr.ifname);
+}
+
+static int get_siocgifflags(nr_local_addr *addr) {
+  int fd = socket(AF_INET, SOCK_DGRAM, 0);
+  if (fd == -1) {
+    assert(0);
+    return 0;
+  }
+  struct ifreq ifr;
+  memset(&ifr, 0, sizeof(ifr));
+  strncpy(ifr.ifr_name, addr->addr.ifname, IFNAMSIZ - 1);
+  int rc = ioctl(fd, SIOCGIFFLAGS, &ifr);
+  close(fd);
+  if (rc == -1) {
+    assert(0);
+    return 0;
+  }
+  return ifr.ifr_flags;
+}
+
+static int set_sockaddr(nr_local_addr *addr, struct ifaddrmsg* msg, struct rtattr* rta) {
+  assert(rta->rta_type == IFA_ADDRESS || rta->rta_type == IFA_LOCAL);
+  void *data = RTA_DATA(rta);
+  size_t len = RTA_PAYLOAD(rta);
+  if (msg->ifa_family == AF_INET) {
+    struct sockaddr_in sa;
+    memset(&sa, 0, sizeof(struct sockaddr_in));
+    sa.sin_family = AF_INET;
+    memcpy(&sa.sin_addr, data, len);
+    return nr_sockaddr_to_transport_addr((struct sockaddr*)&sa, IPPROTO_UDP, 0, &(addr->addr));
+  } else if (msg->ifa_family == AF_INET6) {
+    struct sockaddr_in6 sa;
+    memset(&sa, 0, sizeof(struct sockaddr_in6));
+    sa.sin6_family = AF_INET6;
+    /* We do not set sin6_scope_id to ifa_index, because that is only valid for
+     * link local addresses, and we don't use those anyway */
+    memcpy(&sa.sin6_addr, data, len);
+    return nr_sockaddr_to_transport_addr((struct sockaddr*)&sa, IPPROTO_UDP, 0, &(addr->addr));
+  }
+
+  return R_BAD_ARGS;
+}
+
+static int
+stun_ifaddr_is_disallowed_v6(int flags) {
+  return flags & (IFA_F_TENTATIVE | IFA_F_OPTIMISTIC | IFA_F_DADFAILED | IFA_F_DEPRECATED);
+}
+
+static int
+stun_convert_netlink(nr_local_addr *addr, struct ifaddrmsg *address_msg, struct rtattr* rta)
+{
+  int r = set_sockaddr(addr, address_msg, rta);
+  if (r) {
+    r_log(NR_LOG_STUN, LOG_ERR, "set_sockaddr error r = %d", r);
+    return r;
+  }
+
+  set_ifname(addr, address_msg);
+
+  if (address_msg->ifa_flags & IFA_F_TEMPORARY) {
+    addr->flags |= NR_ADDR_FLAG_TEMPORARY;
+  }
+
+  int flags = get_siocgifflags(addr);
+  if (flags & IFF_POINTOPOINT)
+  {
+    addr->interface.type = NR_INTERFACE_TYPE_UNKNOWN | NR_INTERFACE_TYPE_VPN;
+    /* TODO (Bug 896913): find backend network type of this VPN */
+  }
+
+#if defined(LINUX) && !defined(ANDROID)
+  struct ethtool_cmd ecmd;
+  struct ifreq ifr;
+  struct iwreq wrq;
+  int e;
+  int s = socket(AF_INET, SOCK_DGRAM, 0);
+
+  strncpy(ifr.ifr_name, addr->addr.ifname, sizeof(ifr.ifr_name));
+  /* TODO (Bug 896851): interface property for Android */
+  /* Getting ethtool for ethernet information. */
+  ecmd.cmd = ETHTOOL_GSET;
+  /* In/out param */
+  ifr.ifr_data = (void*)&ecmd;
+
+  e = ioctl(s, SIOCETHTOOL, &ifr);
+  if (e == 0)
+  {
+    /* For wireless network, we won't get ethtool, it's a wired
+     * connection */
+    addr->interface.type = NR_INTERFACE_TYPE_WIRED;
+#ifdef DONT_HAVE_ETHTOOL_SPEED_HI
+    addr->interface.estimated_speed = ecmd.speed;
+#else
+    addr->interface.estimated_speed = ((ecmd.speed_hi << 16) | ecmd.speed) * 1000;
+#endif
+  }
+
+  strncpy(wrq.ifr_name, addr->addr.ifname, sizeof(wrq.ifr_name));
+  e = ioctl(s, SIOCGIWRATE, &wrq);
+  if (e == 0)
+  {
+    addr->interface.type = NR_INTERFACE_TYPE_WIFI;
+    addr->interface.estimated_speed = wrq.u.bitrate.value / 1000;
+  }
+
+  close(s);
+
+#else
+  addr->interface.type = NR_INTERFACE_TYPE_UNKNOWN;
+  addr->interface.estimated_speed = 0;
+#endif
+  return 0;
+}
+
+int
+stun_getaddrs_filtered(nr_local_addr addrs[], int maxaddrs, int *count)
+{
+  int _status;
+  int fd = 0;
+
+  /* Scope everything else since we're using ABORT. */
+  {
+    *count = 0;
+
+    if (maxaddrs <= 0)
+      ABORT(R_BAD_ARGS);
+
+    fd = socket(PF_NETLINK, SOCK_RAW, NETLINK_ROUTE);
+    if (fd < 0) {
+      ABORT(R_INTERNAL);
+    }
+
+    struct netlinkrequest ifaddr_request;
+    memset(&ifaddr_request, 0, sizeof(ifaddr_request));
+    ifaddr_request.header.nlmsg_flags = NLM_F_ROOT | NLM_F_REQUEST;
+    ifaddr_request.header.nlmsg_type = RTM_GETADDR;
+    ifaddr_request.header.nlmsg_len = NLMSG_LENGTH(sizeof(struct ifaddrmsg));
+
+    ssize_t bytes = send(fd, &ifaddr_request, ifaddr_request.header.nlmsg_len, 0);
+    if ((size_t)bytes != ifaddr_request.header.nlmsg_len) {
+      ABORT(R_INTERNAL);
+    }
+
+    char buf[kMaxReadSize];
+    ssize_t amount_read = recv(fd, &buf, kMaxReadSize, 0);
+    while ((amount_read > 0) && (*count != maxaddrs)) {
+      struct nlmsghdr* header = (struct nlmsghdr*)&buf[0];
+      size_t header_size = (size_t)amount_read;
+      for ( ; NLMSG_OK(header, header_size) && (*count != maxaddrs);
+            header = NLMSG_NEXT(header, header_size)) {
+        switch (header->nlmsg_type) {
+          case NLMSG_DONE:
+            /* Success. Return. */
+            close(fd);
+            return 0;
+          case NLMSG_ERROR:
+            ABORT(R_INTERNAL);
+          case RTM_NEWADDR: {
+            struct ifaddrmsg* address_msg =
+                (struct ifaddrmsg*)NLMSG_DATA(header);
+            struct rtattr* rta = IFA_RTA(address_msg);
+            ssize_t payload_len = IFA_PAYLOAD(header);
+            bool found = false;
+            while (RTA_OK(rta, payload_len)) {
+              // This is a bit convoluted. IFA_ADDRESS and IFA_LOCAL are the
+              // same thing except when using a POINTTOPOINT interface, in
+              // which case IFA_LOCAL is the local address, and IFA_ADDRESS is
+              // the remote address. In a reasonable world, that would mean we
+              // could just use IFA_LOCAL all the time. Sadly, IFA_LOCAL is not
+              // always set (IPv6 in particular). So, we have to be on the
+              // lookout for both, and prefer IFA_LOCAL when present.
+              if (rta->rta_type == IFA_ADDRESS || rta->rta_type == IFA_LOCAL) {
+                int family = address_msg->ifa_family;
+                if ((family == AF_INET || family == AF_INET6) &&
+                    !stun_ifaddr_is_disallowed_v6(address_msg->ifa_flags) &&
+                    !stun_convert_netlink(&addrs[*count], address_msg, rta)) {
+                  found = true;
+                  if (rta->rta_type == IFA_LOCAL) {
+                    // IFA_LOCAL is what we really want; if we find it we're
+                    // done. If this is IFA_ADDRESS instead, we do not proceed
+                    // yet, and allow a subsequent IFA_LOCAL to overwrite what
+                    // we just put in |addrs|.
+                    break;
+                  }
+                }
+              }
+              /* TODO: Use IFA_LABEL instead of if_indextoname? We would need
+               * to remember how many nr_local_addr we've converted for this
+               * ifaddrmsg, and set the label on all of them. */
+              rta = RTA_NEXT(rta, payload_len);
+            }
+
+            if (found) {
+              ++(*count);
+            }
+            break;
+          }
+        }
+      }
+      amount_read = recv(fd, &buf, kMaxReadSize, 0);
+    }
+  }
+
+  _status=0;
+abort:
+  close(fd);
+  return(_status);
+}
+
+#endif  /* defined(LINUX) */
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/addrs-netlink.h b/dom/media/webrtc/transport/third_party/nICEr/src/stun/addrs-netlink.h
new file mode 100644
index 0000000000..b2b07bddc9
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/addrs-netlink.h
@@ -0,0 +1,45 @@
+/*
+Copyright (c) 2011, The WebRTC project authors. All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+  * Redistributions of source code must retain the above copyright
+    notice, this list of conditions and the following disclaimer.
+
+  * Redistributions in binary form must reproduce the above copyright
+    notice, this list of conditions and the following disclaimer in
+    the documentation and/or other materials provided with the
+    distribution.
+
+  * Neither the name of Google nor the names of its contributors may
+    be used to endorse or promote products derived from this software
+    without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+HOLDER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifndef WEBRTC_BASE_IFADDRS_NETLINK_H_
+#define WEBRTC_BASE_IFADDRS_NETLINK_H_
+
+#include <stdio.h>
+#include <sys/socket.h>
+#include "local_addr.h"
+
+/* for platforms with netlink (android and linux) */
+/* Filters out things like deprecated addresses, and stuff that doesn't pass
+ * IPv6 duplicate address detection */
+int stun_getaddrs_filtered(nr_local_addr addrs[], int maxaddrs, int *count);
+#endif  /* WEBRTC_BASE_IFADDRS_NETLINK_H_ */
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/addrs-win32.c b/dom/media/webrtc/transport/third_party/nICEr/src/stun/addrs-win32.c
new file mode 100644
index 0000000000..4fdf5e5d44
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/addrs-win32.c
@@ -0,0 +1,210 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifdef WIN32
+
+#include "addrs-win32.h"
+#include <csi_platform.h>
+#include <assert.h>
+#include <string.h>
+#include "util.h"
+#include "stun_util.h"
+#include "util.h"
+#include <r_macros.h>
+#include "nr_crypto.h"
+
+#include <winsock2.h>
+#include <iphlpapi.h>
+#include <tchar.h>
+
+#define WIN32_MAX_NUM_INTERFACES  20
+
+#define NR_MD5_HASH_LENGTH 16
+
+#define _NR_MAX_KEY_LENGTH 256
+#define _NR_MAX_NAME_LENGTH 512
+
+#define _ADAPTERS_BASE_REG "SYSTEM\\CurrentControlSet\\Control\\Network\\{4D36E972-E325-11CE-BFC1-08002BE10318}"
+
+static int nr_win32_get_adapter_friendly_name(char *adapter_GUID, char **friendly_name)
+{
+    int r,_status;
+    HKEY adapter_reg;
+    TCHAR adapter_key[_NR_MAX_KEY_LENGTH];
+    TCHAR keyval_buf[_NR_MAX_KEY_LENGTH];
+    TCHAR adapter_GUID_tchar[_NR_MAX_NAME_LENGTH];
+    DWORD keyval_len, key_type;
+    size_t converted_chars, newlen;
+    char *my_fn = 0;
+
+#ifdef _UNICODE
+    mbstowcs_s(&converted_chars, adapter_GUID_tchar, strlen(adapter_GUID)+1,
+               adapter_GUID, _TRUNCATE);
+#else
+    strlcpy(adapter_GUID_tchar, adapter_GUID, _NR_MAX_NAME_LENGTH);
+#endif
+
+    _tcscpy_s(adapter_key, _NR_MAX_KEY_LENGTH, TEXT(_ADAPTERS_BASE_REG));
+    _tcscat_s(adapter_key, _NR_MAX_KEY_LENGTH, TEXT("\\"));
+    _tcscat_s(adapter_key, _NR_MAX_KEY_LENGTH, adapter_GUID_tchar);
+    _tcscat_s(adapter_key, _NR_MAX_KEY_LENGTH, TEXT("\\Connection"));
+
+    r = RegOpenKeyEx(HKEY_LOCAL_MACHINE, adapter_key, 0, KEY_READ, &adapter_reg);
+
+    if (r != ERROR_SUCCESS) {
+      r_log(NR_LOG_STUN, LOG_ERR, "Got error %d opening adapter reg key\n", r);
+      ABORT(R_INTERNAL);
+    }
+
+    keyval_len = sizeof(keyval_buf);
+    r = RegQueryValueEx(adapter_reg, TEXT("Name"), NULL, &key_type,
+                        (BYTE *)keyval_buf, &keyval_len);
+
+    RegCloseKey(adapter_reg);
+
+#ifdef UNICODE
+    newlen = wcslen(keyval_buf)+1;
+    my_fn = (char *) RCALLOC(newlen);
+    if (!my_fn) {
+      ABORT(R_NO_MEMORY);
+    }
+    wcstombs_s(&converted_chars, my_fn, newlen, keyval_buf, _TRUNCATE);
+#else
+    my_fn = r_strdup(keyval_buf);
+#endif
+
+    *friendly_name = my_fn;
+    _status=0;
+
+abort:
+    if (_status) {
+      if (my_fn) free(my_fn);
+    }
+    return(_status);
+}
+
+static int stun_win32_address_disallowed(IP_ADAPTER_UNICAST_ADDRESS *addr)
+{
+  return (addr->DadState != NldsPreferred) &&
+          (addr->DadState != IpDadStatePreferred);
+}
+
+static int stun_win32_address_temp_v6(IP_ADAPTER_UNICAST_ADDRESS *addr)
+{
+  return (addr->Address.lpSockaddr->sa_family == AF_INET6) &&
+      (addr->SuffixOrigin == IpSuffixOriginRandom);
+}
+
+int
+stun_getaddrs_filtered(nr_local_addr addrs[], int maxaddrs, int *count)
+{
+    int r, _status;
+    PIP_ADAPTER_ADDRESSES AdapterAddresses = NULL, tmpAddress = NULL;
+    // recomended per https://msdn.microsoft.com/en-us/library/windows/desktop/aa365915(v=vs.85).aspx
+    static const ULONG initialBufLen = 15000;
+    ULONG buflen = initialBufLen;
+    char bin_hashed_ifname[NR_MD5_HASH_LENGTH];
+    char hex_hashed_ifname[MAXIFNAME];
+    int n = 0;
+
+    *count = 0;
+
+    if (maxaddrs <= 0)
+      ABORT(R_BAD_ARGS);
+
+    /* According to MSDN (see above) we have try GetAdapterAddresses() multiple times */
+    for (n = 0; n < 5; n++) {
+      AdapterAddresses = (PIP_ADAPTER_ADDRESSES) RMALLOC(buflen);
+      if (AdapterAddresses == NULL) {
+        r_log(NR_LOG_STUN, LOG_ERR, "Error allocating buf for GetAdaptersAddresses()");
+        ABORT(R_NO_MEMORY);
+      }
+
+      r = GetAdaptersAddresses(AF_UNSPEC, GAA_FLAG_SKIP_MULTICAST | GAA_FLAG_SKIP_DNS_SERVER, NULL, AdapterAddresses, &buflen);
+      if (r == NO_ERROR) {
+        break;
+      }
+      r_log(NR_LOG_STUN, LOG_ERR, "GetAdaptersAddresses() returned error (%d)", r);
+      RFREE(AdapterAddresses);
+      AdapterAddresses = NULL;
+    }
+
+    if (n >= 5) {
+      r_log(NR_LOG_STUN, LOG_ERR, "5 failures calling GetAdaptersAddresses()");
+      ABORT(R_INTERNAL);
+    }
+
+    n = 0;
+
+    /* Loop through the adapters */
+
+    for (tmpAddress = AdapterAddresses; tmpAddress != NULL; tmpAddress = tmpAddress->Next) {
+
+      if (tmpAddress->OperStatus != IfOperStatusUp)
+        continue;
+
+      if ((tmpAddress->IfIndex != 0) || (tmpAddress->Ipv6IfIndex != 0)) {
+        IP_ADAPTER_UNICAST_ADDRESS *u = 0;
+
+        if(r=nr_crypto_md5((UCHAR *)tmpAddress->FriendlyName,
+                           wcslen(tmpAddress->FriendlyName) * sizeof(wchar_t),
+                           bin_hashed_ifname))
+          ABORT(r);
+        if(r=nr_bin2hex(bin_hashed_ifname, sizeof(bin_hashed_ifname),
+          hex_hashed_ifname))
+          ABORT(r);
+
+        for (u = tmpAddress->FirstUnicastAddress; u != 0; u = u->Next) {
+          SOCKET_ADDRESS *sa_addr = &u->Address;
+
+          if ((sa_addr->lpSockaddr->sa_family != AF_INET) &&
+              (sa_addr->lpSockaddr->sa_family != AF_INET6)) {
+            r_log(NR_LOG_STUN, LOG_DEBUG, "Unrecognized sa_family for address on adapter %lu", tmpAddress->IfIndex);
+            continue;
+          }
+
+          if (stun_win32_address_disallowed(u)) {
+            continue;
+          }
+
+          if ((r=nr_sockaddr_to_transport_addr((struct sockaddr*)sa_addr->lpSockaddr, IPPROTO_UDP, 0, &(addrs[n].addr)))) {
+            ABORT(r);
+          }
+
+          strlcpy(addrs[n].addr.ifname, hex_hashed_ifname, sizeof(addrs[n].addr.ifname));
+          if (tmpAddress->IfType == IF_TYPE_ETHERNET_CSMACD) {
+            addrs[n].interface.type = NR_INTERFACE_TYPE_WIRED;
+          } else if (tmpAddress->IfType == IF_TYPE_IEEE80211) {
+            /* Note: this only works for >= Win Vista */
+            addrs[n].interface.type = NR_INTERFACE_TYPE_WIFI;
+          } else {
+            addrs[n].interface.type = NR_INTERFACE_TYPE_UNKNOWN;
+          }
+#if (_WIN32_WINNT >= 0x0600)
+          /* Note: only >= Vista provide link speed information */
+          addrs[n].interface.estimated_speed = tmpAddress->TransmitLinkSpeed / 1000;
+#else
+          addrs[n].interface.estimated_speed = 0;
+#endif
+          if (stun_win32_address_temp_v6(u)) {
+            addrs[n].flags |= NR_ADDR_FLAG_TEMPORARY;
+          }
+
+          if (++n >= maxaddrs)
+            goto done;
+        }
+      }
+    }
+
+   done:
+    *count = n;
+    _status = 0;
+
+  abort:
+    RFREE(AdapterAddresses);
+    return _status;
+}
+
+#endif //WIN32
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/addrs-win32.h b/dom/media/webrtc/transport/third_party/nICEr/src/stun/addrs-win32.h
new file mode 100644
index 0000000000..a00802192a
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/addrs-win32.h
@@ -0,0 +1,13 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef STUN_IFADDRS_WIN32_H_
+#define STUN_IFADDRS_WIN32_H_
+
+#include "local_addr.h"
+
+int stun_getaddrs_filtered(nr_local_addr addrs[], int maxaddrs, int *count);
+
+#endif  /* STUN_IFADDRS_WIN32_H_ */
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/addrs.c b/dom/media/webrtc/transport/third_party/nICEr/src/stun/addrs.c
new file mode 100644
index 0000000000..362b7d828e
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/addrs.c
@@ -0,0 +1,176 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <csi_platform.h>
+#include <assert.h>
+#include <string.h>
+
+#ifdef WIN32
+#include "addrs-win32.h"
+#elif defined(BSD) || defined(DARWIN)
+#include "addrs-bsd.h"
+#else
+#include "addrs-netlink.h"
+#endif
+
+#include "stun.h"
+#include "addrs.h"
+#include "util.h"
+
+static int
+nr_stun_is_duplicate_addr(nr_local_addr addrs[], int count, nr_local_addr *addr)
+{
+    int i;
+    int different;
+
+    for (i = 0; i < count; ++i) {
+        different = nr_transport_addr_cmp(&addrs[i].addr, &(addr->addr),
+          NR_TRANSPORT_ADDR_CMP_MODE_ALL);
+        if (!different)
+            return 1;  /* duplicate */
+    }
+
+    return 0;
+}
+
+int
+nr_stun_filter_addrs(nr_local_addr addrs[], int remove_loopback, int remove_link_local, int *count)
+{
+    int r, _status;
+    nr_local_addr *tmp = 0;
+    int i;
+    int n;
+    /* We prefer temp ipv6 for their privacy properties. If we cannot get
+     * that, we prefer ipv6 that are not based on mac address. */
+    int filter_mac_ipv6 = 0;
+    int filter_teredo_ipv6 = 0;
+    int filter_non_temp_ipv6 = 0;
+
+    tmp = RMALLOC(*count * sizeof(*tmp));
+    if (!tmp)
+        ABORT(R_NO_MEMORY);
+
+    for (i = 0; i < *count; ++i) {
+      if (addrs[i].addr.ip_version == NR_IPV6) {
+        if (nr_transport_addr_is_teredo(&addrs[i].addr)) {
+            addrs[i].interface.type |= NR_INTERFACE_TYPE_TEREDO;
+            /* Prefer teredo over mac-based address. Probably will never see
+             * both. */
+            filter_mac_ipv6 = 1;
+        } else {
+          filter_teredo_ipv6 = 1;
+        }
+
+        if (!nr_transport_addr_is_mac_based(&addrs[i].addr)) {
+          filter_mac_ipv6 = 1;
+        }
+
+        if (addrs[i].flags & NR_ADDR_FLAG_TEMPORARY) {
+          filter_non_temp_ipv6 = 1;
+        }
+      }
+    }
+
+    n = 0;
+    for (i = 0; i < *count; ++i) {
+        if (nr_stun_is_duplicate_addr(tmp, n, &addrs[i])) {
+            /* skip addrs[i], it's a duplicate */
+        }
+        else if (remove_loopback && nr_transport_addr_is_loopback(&addrs[i].addr)) {
+            /* skip addrs[i], it's a loopback */
+        }
+        else if (remove_link_local &&
+                 nr_transport_addr_is_link_local(&addrs[i].addr)) {
+            /* skip addrs[i], it's a link-local address */
+        }
+        else if (filter_mac_ipv6 &&
+                 nr_transport_addr_is_mac_based(&addrs[i].addr)) {
+            /* skip addrs[i], it's MAC based */
+        }
+        else if (filter_teredo_ipv6 &&
+                 nr_transport_addr_is_teredo(&addrs[i].addr)) {
+            /* skip addrs[i], it's a Teredo address */
+        }
+        else if (filter_non_temp_ipv6 &&
+                 (addrs[i].addr.ip_version == NR_IPV6) &&
+                 !(addrs[i].flags & NR_ADDR_FLAG_TEMPORARY)) {
+            /* skip addrs[i], it's a non-temporary ipv6, and we have a temporary */
+        }
+        else {
+            /* otherwise, copy it to the temporary array */
+            if ((r=nr_local_addr_copy(&tmp[n], &addrs[i])))
+                ABORT(r);
+            ++n;
+        }
+    }
+
+    *count = n;
+
+    memset(addrs, 0, *count * sizeof(*addrs));
+    /* copy temporary array into passed in/out array */
+    for (i = 0; i < *count; ++i) {
+        if ((r=nr_local_addr_copy(&addrs[i], &tmp[i])))
+            ABORT(r);
+    }
+
+    _status = 0;
+  abort:
+    RFREE(tmp);
+    return _status;
+}
+
+#ifndef USE_PLATFORM_NR_STUN_GET_ADDRS
+
+int
+nr_stun_get_addrs(nr_local_addr addrs[], int maxaddrs, int *count)
+{
+    int _status=0;
+    int i;
+    char typestr[100];
+
+    // Ensure output records are always fully defined.  See bug 1589990.
+    if (maxaddrs > 0) {
+       memset(addrs, 0, maxaddrs * sizeof(nr_local_addr));
+    }
+
+    _status = stun_getaddrs_filtered(addrs, maxaddrs, count);
+
+    for (i = 0; i < *count; ++i) {
+      nr_local_addr_fmt_info_string(addrs+i,typestr,sizeof(typestr));
+      r_log(NR_LOG_STUN, LOG_DEBUG, "Address %d: %s on %s, type: %s\n",
+            i,addrs[i].addr.as_string,addrs[i].addr.ifname,typestr);
+    }
+
+    return _status;
+}
+
+#endif
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/addrs.h b/dom/media/webrtc/transport/third_party/nICEr/src/stun/addrs.h
new file mode 100644
index 0000000000..522bd92fd5
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/addrs.h
@@ -0,0 +1,43 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+#ifndef _addrs_h_
+#define _addrs_h_
+
+#include "transport_addr.h"
+#include "local_addr.h"
+
+int nr_stun_get_addrs(nr_local_addr addrs[], int maxaddrs, int *count);
+int nr_stun_filter_addrs(nr_local_addr addrs[], int remove_loopback, int remove_link_local, int *count);
+
+#endif
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/nr_socket_buffered_stun.c b/dom/media/webrtc/transport/third_party/nICEr/src/stun/nr_socket_buffered_stun.c
new file mode 100644
index 0000000000..4b5b92fb75
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/nr_socket_buffered_stun.c
@@ -0,0 +1,656 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+Copyright (c) 2013, Mozilla
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <nr_api.h>
+
+#include <stdio.h>
+#include <string.h>
+#include <sys/types.h>
+#include <sys/queue.h>
+#include <assert.h>
+#include <inttypes.h>
+
+#include "p_buf.h"
+#include "nr_socket.h"
+#include "stun.h"
+#include "nr_socket_buffered_stun.h"
+
+#define NR_MAX_FRAME_SIZE 0xFFFF
+
+typedef struct nr_frame_header_ {
+  UINT2 frame_length;
+  char  data[0];
+} nr_frame_header;
+
+typedef struct nr_socket_buffered_stun_ {
+  nr_socket *inner;
+  nr_transport_addr remote_addr;
+  int connected;
+
+  /* Read state */
+  int read_state;
+#define NR_ICE_SOCKET_READ_NONE   0
+#define NR_ICE_SOCKET_READ_HDR    1
+#define NR_ICE_SOCKET_READ_FAILED 2
+  UCHAR *buffer;
+  size_t buffer_size;
+  size_t bytes_needed;
+  size_t bytes_read;
+  NR_async_cb readable_cb;
+  void *readable_cb_arg;
+
+  /* Write state */
+  nr_p_buf_ctx *p_bufs;
+  nr_p_buf_head pending_writes;
+  size_t pending;
+  size_t max_pending;
+  nr_framing_type framing_type;
+} nr_socket_buffered_stun;
+
+static int nr_socket_buffered_stun_destroy(void **objp);
+static int nr_socket_buffered_stun_sendto(void *obj,const void *msg, size_t len,
+  int flags, const nr_transport_addr *to);
+static int nr_socket_buffered_stun_recvfrom(void *obj,void * restrict buf,
+  size_t maxlen, size_t *len, int flags, nr_transport_addr *from);
+static int nr_socket_buffered_stun_getfd(void *obj, NR_SOCKET *fd);
+static int nr_socket_buffered_stun_getaddr(void *obj, nr_transport_addr *addrp);
+static int nr_socket_buffered_stun_close(void *obj);
+static int nr_socket_buffered_stun_connect(void *sock, const nr_transport_addr *addr);
+static int nr_socket_buffered_stun_write(void *obj,const void *msg, size_t len, size_t *written);
+static void nr_socket_buffered_stun_writable_cb(NR_SOCKET s, int how, void *arg);
+static int nr_socket_buffered_stun_listen(void *obj, int backlog);
+static int nr_socket_buffered_stun_accept(void *obj, nr_transport_addr *addrp, nr_socket **sockp);
+
+static nr_socket_vtbl nr_socket_buffered_stun_vtbl={
+  2,
+  nr_socket_buffered_stun_destroy,
+  nr_socket_buffered_stun_sendto,
+  nr_socket_buffered_stun_recvfrom,
+  nr_socket_buffered_stun_getfd,
+  nr_socket_buffered_stun_getaddr,
+  nr_socket_buffered_stun_connect,
+  0,
+  0,
+  nr_socket_buffered_stun_close,
+  nr_socket_buffered_stun_listen,
+  nr_socket_buffered_stun_accept
+};
+
+void nr_socket_buffered_stun_set_readable_cb(nr_socket *sock,
+  NR_async_cb readable_cb, void *readable_cb_arg)
+{
+  nr_socket_buffered_stun *buf_sock = (nr_socket_buffered_stun *)sock->obj;
+
+  buf_sock->readable_cb = readable_cb;
+  buf_sock->readable_cb_arg = readable_cb_arg;
+}
+
+int nr_socket_buffered_set_connected_to(nr_socket *sock, nr_transport_addr *remote_addr)
+{
+  nr_socket_buffered_stun *buf_sock = (nr_socket_buffered_stun *)sock->obj;
+  int r, _status;
+
+  if ((r=nr_transport_addr_copy(&buf_sock->remote_addr, remote_addr)))
+    ABORT(r);
+
+  buf_sock->connected = 1;
+
+  _status=0;
+abort:
+  return(_status);
+}
+
+int nr_socket_buffered_stun_create(nr_socket *inner, int max_pending,
+  nr_framing_type framing_type, nr_socket **sockp)
+{
+  int r, _status;
+  nr_socket_buffered_stun *sock = 0;
+  size_t frame_size;
+
+  if (!(sock = RCALLOC(sizeof(nr_socket_buffered_stun))))
+    ABORT(R_NO_MEMORY);
+
+  sock->inner = inner;
+  sock->framing_type = framing_type;
+
+  if ((r=nr_ip4_port_to_transport_addr(INADDR_ANY, 0, IPPROTO_TCP, &sock->remote_addr)))
+    ABORT(r);
+
+  switch (framing_type) {
+    case ICE_TCP_FRAMING:
+      frame_size = sizeof(nr_frame_header);
+      sock->buffer_size = sizeof(nr_frame_header) + NR_MAX_FRAME_SIZE;
+      sock->bytes_needed = sizeof(nr_frame_header);
+      break;
+    case TURN_TCP_FRAMING:
+      frame_size = 0;
+      sock->buffer_size = NR_STUN_MAX_MESSAGE_SIZE;
+      sock->bytes_needed = sizeof(nr_stun_message_header);
+      break;
+    default:
+      assert(0);
+      ABORT(R_BAD_ARGS);
+  }
+
+  /* TODO(ekr@rtfm.com): Check this */
+  if (!(sock->buffer = RMALLOC(sock->buffer_size)))
+    ABORT(R_NO_MEMORY);
+
+  sock->read_state = NR_ICE_SOCKET_READ_NONE;
+  sock->connected = 0;
+
+  STAILQ_INIT(&sock->pending_writes);
+  if ((r=nr_p_buf_ctx_create(sock->buffer_size, &sock->p_bufs)))
+    ABORT(r);
+  sock->max_pending = max_pending + frame_size;
+
+  if ((r=nr_socket_create_int(sock, &nr_socket_buffered_stun_vtbl, sockp)))
+    ABORT(r);
+
+  _status=0;
+abort:
+  if (_status && sock) {
+    void *sock_v = sock;
+    sock->inner = 0;  /* Give up ownership so we don't destroy */
+    nr_socket_buffered_stun_destroy(&sock_v);
+  }
+  return(_status);
+}
+
+/* Note: This destroys the inner socket */
+int nr_socket_buffered_stun_destroy(void **objp)
+{
+  nr_socket_buffered_stun *sock;
+  NR_SOCKET fd;
+
+  if (!objp || !*objp)
+    return 0;
+
+  sock = (nr_socket_buffered_stun *)*objp;
+  *objp = 0;
+
+  /* Free the buffer if needed */
+  RFREE(sock->buffer);
+
+  /* Cancel waiting on the socket */
+  if (sock->inner && !nr_socket_getfd(sock->inner, &fd)) {
+    NR_ASYNC_CANCEL(fd, NR_ASYNC_WAIT_WRITE);
+    NR_ASYNC_CANCEL(fd, NR_ASYNC_WAIT_READ);
+  }
+
+  nr_p_buf_free_chain(sock->p_bufs, &sock->pending_writes);
+  nr_p_buf_ctx_destroy(&sock->p_bufs);
+  nr_socket_destroy(&sock->inner);
+  RFREE(sock);
+
+  return 0;
+}
+
+static int nr_socket_buffered_stun_sendto(void *obj,const void *msg, size_t len,
+  int flags, const nr_transport_addr *to)
+{
+  nr_socket_buffered_stun *sock = (nr_socket_buffered_stun *)obj;
+  int r, _status;
+  size_t written;
+  nr_frame_header *frame = NULL;
+
+  /* Check that we are writing to the connected address if
+     connected */
+  if (!nr_transport_addr_is_wildcard(&sock->remote_addr)) {
+    if (nr_transport_addr_cmp(&sock->remote_addr, to, NR_TRANSPORT_ADDR_CMP_MODE_ALL)) {
+      r_log(LOG_GENERIC, LOG_ERR, "Sendto on connected socket doesn't match");
+      ABORT(R_BAD_DATA);
+    }
+  }
+
+  if (sock->framing_type == ICE_TCP_FRAMING) {
+
+    assert(len <= NR_MAX_FRAME_SIZE);
+    if (len > NR_MAX_FRAME_SIZE)
+      ABORT(R_FAILED);
+
+    if (!(frame = RMALLOC(len + sizeof(nr_frame_header))))
+      ABORT(R_NO_MEMORY);
+
+    frame->frame_length = htons(len);
+    memcpy(frame->data, msg, len);
+    len += sizeof(nr_frame_header);
+    msg = frame;
+  }
+
+  if ((r=nr_socket_buffered_stun_write(obj, msg, len, &written)))
+    ABORT(r);
+
+  if (len != written)
+    ABORT(R_IO_ERROR);
+
+  _status=0;
+abort:
+  RFREE(frame);
+  return _status;
+}
+
+static void nr_socket_buffered_stun_failed(nr_socket_buffered_stun *sock)
+  {
+    NR_SOCKET fd;
+
+    sock->read_state = NR_ICE_SOCKET_READ_FAILED;
+
+    /* Cancel waiting on the socket */
+    if (sock->inner && !nr_socket_getfd(sock->inner, &fd)) {
+      NR_ASYNC_CANCEL(fd, NR_ASYNC_WAIT_WRITE);
+      NR_ASYNC_CANCEL(fd, NR_ASYNC_WAIT_READ);
+    }
+  }
+
+static int nr_socket_buffered_stun_recvfrom(void *obj,void * restrict buf,
+  size_t maxlen, size_t *len, int flags, nr_transport_addr *from)
+{
+  int r, _status;
+  size_t bytes_read;
+  nr_socket_buffered_stun *sock = (nr_socket_buffered_stun *)obj;
+  nr_frame_header *frame = (nr_frame_header *)sock->buffer;
+  size_t skip_hdr_size = (sock->framing_type == ICE_TCP_FRAMING) ? sizeof(nr_frame_header) : 0;
+
+  if (sock->read_state == NR_ICE_SOCKET_READ_FAILED) {
+    ABORT(R_FAILED);
+  }
+
+  while (sock->bytes_needed) {
+    /* Read all the expected bytes */
+    assert(sock->bytes_needed <= sock->buffer_size - sock->bytes_read);
+
+    if(r=nr_socket_read(sock->inner,
+                        sock->buffer + sock->bytes_read,
+                        sock->bytes_needed, &bytes_read, 0))
+      ABORT(r);
+
+    assert(bytes_read <= sock->bytes_needed);
+    sock->bytes_needed -= bytes_read;
+    sock->bytes_read += bytes_read;
+
+    /* Unfinished */
+    if (sock->bytes_needed)
+      ABORT(R_WOULDBLOCK);
+
+    /* No more bytes expected */
+    if (sock->read_state == NR_ICE_SOCKET_READ_NONE) {
+      size_t remaining_length;
+      if (sock->framing_type == ICE_TCP_FRAMING) {
+        if (sock->bytes_read < sizeof(nr_frame_header))
+          ABORT(R_BAD_DATA);
+        remaining_length = ntohs(frame->frame_length);
+      } else {
+        int tmp_length;
+
+        /* Parse the header */
+        if (r = nr_stun_message_length(sock->buffer, sock->bytes_read, &tmp_length))
+          ABORT(r);
+        assert(tmp_length >= 0);
+        if (tmp_length < 0)
+          ABORT(R_BAD_DATA);
+        remaining_length = tmp_length;
+
+      }
+      /* Check to see if we have enough room */
+      if ((sock->buffer_size - sock->bytes_read) < remaining_length)
+        ABORT(R_BAD_DATA);
+
+      sock->read_state = NR_ICE_SOCKET_READ_HDR;
+      /* Set ourselves up to read the rest of the data */
+      sock->bytes_needed = remaining_length;
+    }
+  }
+
+  assert(skip_hdr_size <= sock->bytes_read);
+  if (skip_hdr_size > sock->bytes_read)
+    ABORT(R_BAD_DATA);
+  sock->bytes_read -= skip_hdr_size;
+
+  if (maxlen < sock->bytes_read)
+    ABORT(R_BAD_ARGS);
+
+  *len = sock->bytes_read;
+  memcpy(buf, sock->buffer + skip_hdr_size, sock->bytes_read);
+
+  sock->bytes_read = 0;
+  sock->read_state = NR_ICE_SOCKET_READ_NONE;
+  sock->bytes_needed = (sock->framing_type == ICE_TCP_FRAMING) ? sizeof(nr_frame_header) : sizeof(nr_stun_message_header);
+
+  if ((r = nr_transport_addr_copy(from, &sock->remote_addr))) ABORT(r);
+
+  _status=0;
+abort:
+  if (_status && (_status != R_WOULDBLOCK)) {
+    nr_socket_buffered_stun_failed(sock);
+  }
+
+  return(_status);
+}
+
+static int nr_socket_buffered_stun_getfd(void *obj, NR_SOCKET *fd)
+{
+  nr_socket_buffered_stun *sock = (nr_socket_buffered_stun *)obj;
+
+  return nr_socket_getfd(sock->inner, fd);
+}
+
+static int nr_socket_buffered_stun_getaddr(void *obj, nr_transport_addr *addrp)
+{
+  nr_socket_buffered_stun *sock = (nr_socket_buffered_stun *)obj;
+
+  return nr_socket_getaddr(sock->inner, addrp);
+}
+
+static int nr_socket_buffered_stun_close(void *obj)
+{
+  nr_socket_buffered_stun *sock = (nr_socket_buffered_stun *)obj;
+  NR_SOCKET fd;
+
+  /* Cancel waiting on the socket */
+  if (sock->inner && !nr_socket_getfd(sock->inner, &fd)) {
+    NR_ASYNC_CANCEL(fd, NR_ASYNC_WAIT_WRITE);
+  }
+
+  return nr_socket_close(sock->inner);
+}
+
+static int nr_socket_buffered_stun_listen(void *obj, int backlog)
+{
+  int r, _status;
+  nr_socket_buffered_stun *sock = (nr_socket_buffered_stun *)obj;
+
+  if (!sock->inner)
+    ABORT(R_FAILED);
+
+  if ((r=nr_socket_listen(sock->inner, backlog)))
+    ABORT(r);
+
+  _status=0;
+abort:
+  return(_status);
+}
+
+
+static int nr_socket_buffered_stun_accept(void *obj, nr_transport_addr *addrp, nr_socket **sockp)
+{
+  nr_socket_buffered_stun *bsock = (nr_socket_buffered_stun *)obj;
+
+  return nr_socket_accept(bsock->inner, addrp, sockp);
+}
+
+static void nr_socket_buffered_stun_connected_cb(NR_SOCKET s, int how, void *arg)
+{
+  nr_socket_buffered_stun *sock = (nr_socket_buffered_stun *)arg;
+  int r, _status;
+  NR_SOCKET fd;
+
+  assert(!sock->connected);
+
+  sock->connected = 1;
+
+  if ((r=nr_socket_getfd(sock->inner, &fd)))
+    ABORT(r);
+  NR_ASYNC_CANCEL(fd, NR_ASYNC_WAIT_WRITE);
+
+  // once connected arm for read
+  if (sock->readable_cb) {
+    NR_ASYNC_WAIT(fd, NR_ASYNC_WAIT_READ, sock->readable_cb, sock->readable_cb_arg);
+  }
+
+  if (sock->pending) {
+    r_log(LOG_GENERIC, LOG_INFO, "Invoking writable_cb on connected (%u)", (uint32_t) sock->pending);
+    nr_socket_buffered_stun_writable_cb(s, how, arg);
+  }
+
+  _status=0;
+abort:
+  if (_status) {
+    r_log(LOG_GENERIC, LOG_ERR, "Failure in nr_socket_buffered_stun_connected_cb: %d", _status);
+
+  }
+}
+
+static int nr_socket_buffered_stun_connect(void *obj, const nr_transport_addr *addr)
+{
+  nr_socket_buffered_stun *sock = (nr_socket_buffered_stun *)obj;
+  int r, _status;
+
+  if ((r=nr_transport_addr_copy(&sock->remote_addr, addr)))
+    ABORT(r);
+
+  if ((r=nr_socket_connect(sock->inner, addr))) {
+    if (r == R_WOULDBLOCK) {
+      NR_SOCKET fd;
+
+      if ((r=nr_socket_getfd(sock->inner, &fd)))
+        ABORT(r);
+
+      NR_ASYNC_WAIT(fd, NR_ASYNC_WAIT_WRITE, nr_socket_buffered_stun_connected_cb, sock);
+      ABORT(R_WOULDBLOCK);
+    }
+    ABORT(r);
+  } else {
+    r_log(LOG_GENERIC, LOG_INFO, "Connected without blocking");
+    sock->connected = 1;
+  }
+
+  _status=0;
+abort:
+  return(_status);
+}
+
+static int nr_socket_buffered_stun_arm_writable_cb(nr_socket_buffered_stun *sock)
+{
+  int r, _status;
+  NR_SOCKET fd;
+
+  if ((r=nr_socket_getfd(sock->inner, &fd)))
+    ABORT(r);
+
+  NR_ASYNC_WAIT(fd, NR_ASYNC_WAIT_WRITE, nr_socket_buffered_stun_writable_cb, sock);
+
+  _status=0;
+abort:
+  return(_status);
+}
+
+static int nr_socket_buffered_stun_write(void *obj,const void *msg, size_t len, size_t *written)
+{
+  nr_socket_buffered_stun *sock = (nr_socket_buffered_stun *)obj;
+  int already_armed = 0;
+  int r,_status;
+  size_t written2 = 0;
+  size_t original_len = len;
+
+  /* Buffers are close to full, report error. Do this now so we never
+     get partial writes */
+  if ((sock->pending + len) > sock->max_pending) {
+    r_log(LOG_GENERIC, LOG_INFO, "Write buffer for %s full (%u + %u > %u) - re-arming @%p",
+          sock->remote_addr.as_string, (uint32_t)sock->pending, (uint32_t)len, (uint32_t)sock->max_pending,
+          &(sock->pending));
+    ABORT(R_WOULDBLOCK);
+  }
+
+
+  if (sock->connected && !sock->pending) {
+    r = nr_socket_write(sock->inner, msg, len, &written2, 0);
+    if (r) {
+      if (r != R_WOULDBLOCK) {
+        r_log(LOG_GENERIC, LOG_ERR, "Write error for %s - %d",
+              sock->remote_addr.as_string, r);
+        ABORT(r);
+      }
+      r_log(LOG_GENERIC, LOG_INFO, "Write of %" PRIu64 " blocked for %s",
+            (uint64_t) len, sock->remote_addr.as_string);
+
+      written2=0;
+    }
+  } else {
+    already_armed = 1;
+  }
+
+  /* Buffer what's left */
+  len -= written2;
+
+  if (len) {
+    if ((r=nr_p_buf_write_to_chain(sock->p_bufs, &sock->pending_writes,
+                                   ((UCHAR *)msg) + written2, len))) {
+      r_log(LOG_GENERIC, LOG_ERR, "Write_to_chain error for %s - %d",
+            sock->remote_addr.as_string, r);
+
+      ABORT(r);
+    }
+
+    sock->pending += len;
+  }
+
+  if (sock->pending) {
+    if (!already_armed) {
+      if ((r=nr_socket_buffered_stun_arm_writable_cb(sock)))
+        ABORT(r);
+    }
+    r_log(LOG_GENERIC, LOG_INFO, "Write buffer not empty for %s  %u - %s armed (@%p),%s connected",
+          sock->remote_addr.as_string, (uint32_t)sock->pending,
+          already_armed ? "already" : "", &sock->pending,
+          sock->connected ? "" : " not");
+  }
+
+  *written = original_len;
+
+  _status=0;
+abort:
+  return _status;
+}
+
+static void nr_socket_buffered_stun_writable_cb(NR_SOCKET s, int how, void *arg)
+{
+  nr_socket_buffered_stun *sock = (nr_socket_buffered_stun *)arg;
+  int r,_status;
+  nr_p_buf *n1, *n2;
+
+  if (sock->read_state == NR_ICE_SOCKET_READ_FAILED) {
+    ABORT(R_FAILED);
+  }
+
+  /* Try to flush */
+  STAILQ_FOREACH_SAFE(n1, &sock->pending_writes, entry, n2) {
+    size_t written = 0;
+
+    if ((r=nr_socket_write(sock->inner, n1->data + n1->r_offset,
+                           n1->length - n1->r_offset,
+                           &written, 0))) {
+
+      r_log(LOG_GENERIC, LOG_ERR, "Write error for %s - %d",
+            sock->remote_addr.as_string, r);
+      ABORT(r);
+    }
+
+    n1->r_offset += written;
+    assert(sock->pending >= written);
+    sock->pending -= written;
+
+    if (n1->r_offset < n1->length) {
+      /* We wrote something, but not everything */
+      r_log(LOG_GENERIC, LOG_INFO, "Write in callback didn't write all (remaining %u of %u) for %s",
+            n1->length - n1->r_offset, n1->length,
+            sock->remote_addr.as_string);
+      ABORT(R_WOULDBLOCK);
+    }
+
+    /* We are done with this p_buf */
+    STAILQ_REMOVE_HEAD(&sock->pending_writes, entry);
+    nr_p_buf_free(sock->p_bufs, n1);
+  }
+
+  assert(!sock->pending);
+  _status=0;
+abort:
+  r_log(LOG_GENERIC, LOG_INFO, "Writable_cb %s (%u (%p) pending)",
+        sock->remote_addr.as_string, (uint32_t)sock->pending, &(sock->pending));
+  if (_status && _status != R_WOULDBLOCK) {
+    r_log(LOG_GENERIC, LOG_ERR, "Failure in writable_cb: %d", _status);
+    nr_socket_buffered_stun_failed(sock);
+    /* Report this failure up; the only way to do this is a readable callback.
+     * Once the user tries to read (using nr_socket_buffered_stun_recvfrom), it
+     * will notice that there has been a failure. */
+    if (sock->readable_cb) {
+      sock->readable_cb(s, NR_ASYNC_WAIT_READ, sock->readable_cb_arg);
+    }
+  } else if (sock->pending) {
+    nr_socket_buffered_stun_arm_writable_cb(sock);
+  }
+}
+
+int nr_socket_buffered_stun_reset(nr_socket* sock_arg, nr_socket* new_inner) {
+  int r, _status;
+  NR_SOCKET fd;
+
+  nr_socket_buffered_stun* sock = (nr_socket_buffered_stun*)sock_arg->obj;
+
+  if (sock->inner && !nr_socket_getfd(sock->inner, &fd)) {
+    r_log(LOG_GENERIC, LOG_DEBUG, "In %s, canceling wait on old socket", __FUNCTION__);
+    NR_ASYNC_CANCEL(fd, NR_ASYNC_WAIT_WRITE);
+    NR_ASYNC_CANCEL(fd, NR_ASYNC_WAIT_READ);
+  }
+
+  nr_socket_destroy(&sock->inner);
+  sock->inner = new_inner;
+
+  sock->read_state = NR_ICE_SOCKET_READ_NONE;
+  sock->connected = 0;
+
+  sock->bytes_read = 0;
+  sock->bytes_needed = (sock->framing_type == ICE_TCP_FRAMING)
+                           ? sizeof(nr_frame_header)
+                           : sizeof(nr_stun_message_header);
+  sock->pending = 0;
+
+  nr_p_buf_free_chain(sock->p_bufs, &sock->pending_writes);
+  nr_p_buf_ctx_destroy(&sock->p_bufs);
+
+  STAILQ_INIT(&sock->pending_writes);
+
+  if ((r = nr_p_buf_ctx_create(sock->buffer_size, &sock->p_bufs))) {
+    ABORT(r);
+  }
+
+  if ((r = nr_ip4_port_to_transport_addr(INADDR_ANY, 0, IPPROTO_TCP,
+                                         &sock->remote_addr))) {
+    ABORT(r);
+  }
+
+  _status = 0;
+abort:
+  return (_status);
+}
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/nr_socket_buffered_stun.h b/dom/media/webrtc/transport/third_party/nICEr/src/stun/nr_socket_buffered_stun.h
new file mode 100644
index 0000000000..c635ee393d
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/nr_socket_buffered_stun.h
@@ -0,0 +1,66 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+Copyright (c) 2013, Mozilla
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _nr_socket_buffered_stun_h
+#define _nr_socket_buffered_stun_h
+
+#include "nr_socket.h"
+
+/* Wrapper socket which provides buffered STUN-oriented I/O
+
+   1. Writes don't block and are automatically flushed when needed.
+   2. All reads are in units of STUN messages
+
+   This socket takes ownership of the inner socket |sock|.
+ */
+
+typedef enum {
+  TURN_TCP_FRAMING=0,
+  ICE_TCP_FRAMING
+} nr_framing_type;
+
+void nr_socket_buffered_stun_set_readable_cb(nr_socket *sock,
+  NR_async_cb readable_cb, void *readable_cb_arg);
+
+int nr_socket_buffered_stun_create(nr_socket *inner, int max_pending,
+  nr_framing_type framing_type, nr_socket **sockp);
+
+int nr_socket_buffered_set_connected_to(nr_socket *sock,
+    nr_transport_addr *remote_addr);
+
+int nr_socket_buffered_stun_reset(nr_socket *sock, nr_socket *new_inner);
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/nr_socket_turn.c b/dom/media/webrtc/transport/third_party/nICEr/src/stun/nr_socket_turn.c
new file mode 100644
index 0000000000..1a0162b13e
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/nr_socket_turn.c
@@ -0,0 +1,195 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifdef USE_TURN
+
+#include <csi_platform.h>
+#include <stdio.h>
+#include <string.h>
+#include <sys/types.h>
+#include <assert.h>
+
+#include "stun.h"
+#include "turn_client_ctx.h"
+#include "nr_socket_turn.h"
+
+
+static char *nr_socket_turn_magic_cookie = "nr_socket_turn";
+
+typedef struct nr_socket_turn_ {
+  char *magic_cookie;
+  nr_turn_client_ctx *turn;
+} nr_socket_turn;
+
+
+static int nr_socket_turn_destroy(void **objp);
+static int nr_socket_turn_sendto(void *obj,const void *msg, size_t len,
+  int flags, const nr_transport_addr *to);
+static int nr_socket_turn_recvfrom(void *obj,void * restrict buf,
+  size_t maxlen, size_t *len, int flags, nr_transport_addr *from);
+static int nr_socket_turn_getfd(void *obj, NR_SOCKET *fd);
+static int nr_socket_turn_getaddr(void *obj, nr_transport_addr *addrp);
+static int nr_socket_turn_close(void *obj);
+
+static nr_socket_vtbl nr_socket_turn_vtbl={
+  2,
+  nr_socket_turn_destroy,
+  nr_socket_turn_sendto,
+  nr_socket_turn_recvfrom,
+  nr_socket_turn_getfd,
+  nr_socket_turn_getaddr,
+  0,
+  0,
+  0,
+  nr_socket_turn_close,
+  0,
+  0
+};
+
+int nr_socket_turn_create(nr_socket **sockp)
+  {
+    int r,_status;
+    nr_socket_turn *sturn=0;
+
+    if(!(sturn=RCALLOC(sizeof(nr_socket_turn))))
+      ABORT(R_NO_MEMORY);
+
+    sturn->magic_cookie = nr_socket_turn_magic_cookie;
+
+    if(r=nr_socket_create_int(sturn, &nr_socket_turn_vtbl, sockp))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    if(_status){
+      nr_socket_turn_destroy((void **)&sturn);
+    }
+    return(_status);
+  }
+
+static int nr_socket_turn_destroy(void **objp)
+  {
+    int _status;
+    nr_socket_turn *sturn;
+
+    if(!objp || !*objp)
+      return(0);
+
+    sturn=*objp;
+    *objp=0;
+
+    assert(sturn->magic_cookie == nr_socket_turn_magic_cookie);
+
+    /* we don't own the socket, so don't destroy it */
+
+    RFREE(sturn);
+
+    _status=0;
+    return(_status);
+  }
+
+static int nr_socket_turn_sendto(void *obj,const void *msg, size_t len,
+  int flags, const nr_transport_addr *addr)
+  {
+    int r,_status;
+    nr_socket_turn *sturn=obj;
+
+    assert(sturn->magic_cookie == nr_socket_turn_magic_cookie);
+    assert(sturn->turn);
+
+    if ((r = nr_turn_client_send_indication(sturn->turn, msg, len, flags,
+                                            addr)))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+static int nr_socket_turn_recvfrom(void *obj,void * restrict buf,
+  size_t maxlen, size_t *len, int flags, nr_transport_addr *addr)
+  {
+    /* Reading from TURN sockets is done by the indication
+       processing code in turn_client_ctx. */
+    assert(0);
+
+    return(R_INTERNAL);
+  }
+
+static int nr_socket_turn_getfd(void *obj, NR_SOCKET *fd)
+  {
+    /* You should never directly be touching this fd. */
+    assert(0);
+
+    return(R_INTERNAL);
+  }
+
+static int nr_socket_turn_getaddr(void *obj, nr_transport_addr *addrp)
+  {
+    nr_socket_turn *sturn=obj;
+    int r, _status;
+
+    assert(sturn->magic_cookie == nr_socket_turn_magic_cookie);
+    assert(sturn->turn);
+
+    /* This returns the relayed address */
+    if ((r=nr_turn_client_get_relayed_address(sturn->turn, addrp)))
+      ABORT(r);
+
+    _status=0;
+ abort:
+    return(_status);
+  }
+
+static int nr_socket_turn_close(void *obj)
+  {
+    /* No-op */
+#ifndef NDEBUG
+    nr_socket_turn *sturn=obj;
+    assert(sturn->magic_cookie == nr_socket_turn_magic_cookie);
+#endif
+
+    return 0;
+  }
+
+int nr_socket_turn_set_ctx(nr_socket *sock, nr_turn_client_ctx *ctx)
+{
+  nr_socket_turn *sturn=(nr_socket_turn*)sock->obj;
+  assert(sturn->magic_cookie == nr_socket_turn_magic_cookie);
+  assert(!sturn->turn);
+
+  sturn->turn = ctx;
+
+  return 0;
+}
+
+#endif /* USE_TURN */
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/nr_socket_turn.h b/dom/media/webrtc/transport/third_party/nICEr/src/stun/nr_socket_turn.h
new file mode 100644
index 0000000000..13506045a8
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/nr_socket_turn.h
@@ -0,0 +1,48 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _nr_socket_turn_h
+#define _nr_socket_turn_h
+
+#include "nr_socket.h"
+
+/* This is a partial implementation of an nr_socket wrapped
+   around TURN. It implements only the nr_socket features
+   actually used by the ICE stack. You can't, for instance,
+   read off the socket */
+int nr_socket_turn_create(nr_socket **sockp);
+int nr_socket_turn_set_ctx(nr_socket *sock, nr_turn_client_ctx *ctx);
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun.h b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun.h
new file mode 100644
index 0000000000..a32751d795
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun.h
@@ -0,0 +1,218 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+#ifndef _STUN_H
+#define _STUN_H
+
+#ifdef WIN32
+#include <winsock2.h>
+#else
+#include <sys/param.h>
+#include <sys/socket.h>
+#ifndef LINUX
+#include <net/if.h>
+#ifdef DARWIN
+#include <net/if_var.h>
+#endif
+#include <net/if_dl.h>
+#include <net/if_types.h>
+#else
+#include <linux/if.h>
+#endif
+#ifndef BSD
+#include <net/route.h>
+#endif
+#include <netinet/in.h>
+#ifndef LINUX
+#include <netinet/in_var.h>
+#endif
+#include <arpa/inet.h>
+#include <netdb.h>
+#endif
+#include <time.h>
+
+#include "nr_api.h"
+#include "stun_msg.h"
+#include "stun_build.h"
+#include "stun_codec.h"
+#include "stun_hint.h"
+#include "stun_util.h"
+#include "nr_socket.h"
+#include "stun_client_ctx.h"
+#include "stun_server_ctx.h"
+#include "stun_proc.h"
+
+#define NR_STUN_VERSION                 "rfc3489bis-11"
+#define NR_STUN_PORT                    3478
+
+/* STUN attributes */
+#define NR_STUN_ATTR_MAPPED_ADDRESS          0x0001
+#define NR_STUN_ATTR_USERNAME                0x0006
+#define NR_STUN_ATTR_MESSAGE_INTEGRITY       0x0008
+#define NR_STUN_ATTR_ERROR_CODE              0x0009
+#define NR_STUN_ATTR_UNKNOWN_ATTRIBUTES      0x000A
+#define NR_STUN_ATTR_REALM                   0x0014
+#define NR_STUN_ATTR_NONCE                   0x0015
+#define NR_STUN_ATTR_XOR_MAPPED_ADDRESS      0x0020
+#define NR_STUN_ATTR_SERVER                  0x8022
+#define NR_STUN_ATTR_ALTERNATE_SERVER        0x8023
+#define NR_STUN_ATTR_FINGERPRINT             0x8028
+
+/* for backwards compatibility with obsolete versions of the STUN spec */
+#define NR_STUN_ATTR_OLD_XOR_MAPPED_ADDRESS  0x8020
+
+#ifdef USE_STUND_0_96
+#define NR_STUN_ATTR_OLD_CHANGE_REQUEST      0x0003
+#endif /* USE_STUND_0_96 */
+
+#ifdef USE_RFC_3489_BACKWARDS_COMPATIBLE
+/* for backwards compatibility with obsolete versions of the STUN spec */
+#define NR_STUN_ATTR_OLD_PASSWORD            0x0007
+#define NR_STUN_ATTR_OLD_RESPONSE_ADDRESS    0x0002
+#define NR_STUN_ATTR_OLD_SOURCE_ADDRESS      0x0004
+#define NR_STUN_ATTR_OLD_CHANGED_ADDRESS     0x0005
+#endif /* USE_RFC_3489_BACKWARDS_COMPATIBLE */
+
+#ifdef USE_ICE
+/* ICE attributes */
+#define NR_STUN_ATTR_PRIORITY                0x0024
+#define NR_STUN_ATTR_USE_CANDIDATE           0x0025
+#define NR_STUN_ATTR_ICE_CONTROLLED          0x8029
+#define NR_STUN_ATTR_ICE_CONTROLLING         0x802A
+#endif /* USE_ICE */
+
+#ifdef USE_TURN
+/* TURN attributes */
+#define NR_STUN_ATTR_LIFETIME                0x000d
+/* from an expired draft defined as optional, but in the required range */
+#define NR_STUN_ATTR_BANDWIDTH               0x0010
+#define NR_STUN_ATTR_XOR_PEER_ADDRESS        0x0012
+#define NR_STUN_ATTR_DATA                    0x0013
+#define NR_STUN_ATTR_XOR_RELAY_ADDRESS       0x0016
+#define NR_STUN_ATTR_REQUESTED_TRANSPORT     0x0019
+
+#define NR_STUN_ATTR_REQUESTED_TRANSPORT_UDP     17
+#endif /* USE_TURN */
+
+/*
+ *                +-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ *                |M|M|M|M|M|C|M|M|M|C|M|M|M|M|
+ *                |1|1|9|8|7|1|6|5|4|0|3|2|1|0|
+ *                |1|0| | | | | | | | | | | | |
+ *                +-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+ *
+ *      Figure 3: Format of STUN Message Type Field
+ */
+#define NR_STUN_METHOD_TYPE_BITS(m) \
+        ((((m) & 0xf80) << 2) | (((m) & 0x070) << 1) | ((m) & 0x00f))
+
+#define NR_STUN_CLASS_TYPE_BITS(c) \
+        ((((c) & 0x002) << 7) | (((c) & 0x001) << 4))
+
+#define NR_STUN_GET_TYPE_METHOD(t) \
+        ((((t) >> 2) & 0xf80) | (((t) >> 1) & 0x070) | ((t) & 0x00f))
+
+#define NR_STUN_GET_TYPE_CLASS(t) \
+        ((((t) >> 7) & 0x002) | (((t) >> 4) & 0x001))
+
+#define NR_STUN_TYPE(m,c)  (NR_STUN_METHOD_TYPE_BITS((m)) | NR_STUN_CLASS_TYPE_BITS((c)))
+
+/* building blocks for message types */
+#define NR_METHOD_BINDING          0x001
+#define NR_CLASS_REQUEST           0x0
+#define NR_CLASS_INDICATION        0x1
+#define NR_CLASS_RESPONSE          0x2
+#define NR_CLASS_ERROR_RESPONSE    0x3
+
+/* define types for STUN messages */
+#define NR_STUN_MSG_BINDING_REQUEST                 NR_STUN_TYPE(NR_METHOD_BINDING, \
+                                                                 NR_CLASS_REQUEST)
+#define NR_STUN_MSG_BINDING_INDICATION              NR_STUN_TYPE(NR_METHOD_BINDING, \
+                                                                 NR_CLASS_INDICATION)
+#define NR_STUN_MSG_BINDING_RESPONSE                NR_STUN_TYPE(NR_METHOD_BINDING, \
+                                                                 NR_CLASS_RESPONSE)
+#define NR_STUN_MSG_BINDING_ERROR_RESPONSE          NR_STUN_TYPE(NR_METHOD_BINDING, \
+                                                                 NR_CLASS_ERROR_RESPONSE)
+
+#ifdef USE_TURN
+/* building blocks for TURN message types */
+#define NR_METHOD_ALLOCATE                 0x003
+#define NR_METHOD_REFRESH                  0x004
+
+#define NR_METHOD_SEND                     0x006
+#define NR_METHOD_DATA                     0x007
+#define NR_METHOD_CREATE_PERMISSION        0x008
+#define NR_METHOD_CHANNEL_BIND             0x009
+
+/* define types for a TURN message */
+#define NR_STUN_MSG_ALLOCATE_REQUEST                NR_STUN_TYPE(NR_METHOD_ALLOCATE, \
+                                                                 NR_CLASS_REQUEST)
+#define NR_STUN_MSG_ALLOCATE_RESPONSE               NR_STUN_TYPE(NR_METHOD_ALLOCATE, \
+                                                                 NR_CLASS_RESPONSE)
+#define NR_STUN_MSG_ALLOCATE_ERROR_RESPONSE         NR_STUN_TYPE(NR_METHOD_ALLOCATE, \
+                                                                 NR_CLASS_ERROR_RESPONSE)
+#define NR_STUN_MSG_REFRESH_REQUEST                 NR_STUN_TYPE(NR_METHOD_REFRESH, \
+                                                                 NR_CLASS_REQUEST)
+#define NR_STUN_MSG_REFRESH_RESPONSE                NR_STUN_TYPE(NR_METHOD_REFRESH, \
+                                                                 NR_CLASS_RESPONSE)
+#define NR_STUN_MSG_REFRESH_ERROR_RESPONSE          NR_STUN_TYPE(NR_METHOD_REFRESH, \
+                                                                 NR_CLASS_ERROR_RESPONSE)
+
+#define NR_STUN_MSG_SEND_INDICATION                 NR_STUN_TYPE(NR_METHOD_SEND, \
+                                                                 NR_CLASS_INDICATION)
+#define NR_STUN_MSG_DATA_INDICATION                 NR_STUN_TYPE(NR_METHOD_DATA, \
+                                                                 NR_CLASS_INDICATION)
+
+#define NR_STUN_MSG_PERMISSION_REQUEST                 NR_STUN_TYPE(NR_METHOD_CREATE_PERMISSION, \
+                                                                    NR_CLASS_REQUEST)
+#define NR_STUN_MSG_PERMISSION_RESPONSE                NR_STUN_TYPE(NR_METHOD_CREATE_PERMISSION, \
+                                                                    NR_CLASS_RESPONSE)
+#define NR_STUN_MSG_PERMISSION_ERROR_RESPONSE          NR_STUN_TYPE(NR_METHOD_CREATE_PERMISSION, \
+                                                                    NR_CLASS_ERROR_RESPONSE)
+
+#define NR_STUN_MSG_CHANNEL_BIND_REQUEST                 NR_STUN_TYPE(NR_METHOD_CHANNEL_BIND, \
+                                                                      NR_CLASS_REQUEST)
+#define NR_STUN_MSG_CHANNEL_BIND_RESPONSE                NR_STUN_TYPE(NR_METHOD_CHANNEL_BIND, \
+                                                                      NR_CLASS_RESPONSE)
+#define NR_STUN_MSG_CHANNEL_BIND_ERROR_RESPONSE          NR_STUN_TYPE(NR_METHOD_CHANNEL_BIND, \
+                                                                      NR_CLASS_ERROR_RESPONSE)
+
+
+#endif /* USE_TURN */
+
+
+#define NR_STUN_AUTH_RULE_OPTIONAL      (1<<0)
+#define NR_STUN_AUTH_RULE_SHORT_TERM    (1<<8)
+#define NR_STUN_AUTH_RULE_LONG_TERM     (1<<9)
+
+#endif
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_build.c b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_build.c
new file mode 100644
index 0000000000..001b38e7c4
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_build.c
@@ -0,0 +1,611 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <csi_platform.h>
+#include <assert.h>
+#include <errno.h>
+#include <stdio.h>
+#include <string.h>
+#include <ctype.h>
+
+#include "nr_api.h"
+#include "stun.h"
+#include "registry.h"
+#include "stun_reg.h"
+#include "nr_crypto.h"
+
+
+/* draft-ietf-behave-rfc3489bis-10.txt S 7.1 */
+/* draft-ietf-behave-rfc3489bis-10.txt S 10.1.1 */
+/* note that S 10.1.1 states the message MUST include MESSAGE-INTEGRITY
+ * and USERNAME, but that's not correct -- for instance ICE keepalive
+ * messages don't include these (See draft-ietf-mmusic-ice-18.txt S 10:
+ * "If STUN is being used for keepalives, a STUN Binding Indication is
+ * used.  The Indication MUST NOT utilize any authentication mechanism")
+ */
+int
+nr_stun_form_request_or_indication(int mode, int msg_type, nr_stun_message **msg)
+{
+   int r,_status;
+   nr_stun_message *req = 0;
+
+   assert(NR_STUN_GET_TYPE_CLASS(msg_type) == NR_CLASS_REQUEST
+       || NR_STUN_GET_TYPE_CLASS(msg_type) == NR_CLASS_INDICATION);
+
+   *msg = 0;
+
+   if ((r=nr_stun_message_create(&req)))
+       ABORT(r);
+
+   req->header.type = msg_type;
+
+   nr_crypto_random_bytes((UCHAR*)&req->header.id,sizeof(req->header.id));
+
+   switch (mode) {
+   default:
+       if ((r=nr_stun_message_add_fingerprint_attribute(req)))
+           ABORT(r);
+       /* fall through */
+   case NR_STUN_MODE_STUN_NO_AUTH:
+       req->header.magic_cookie = NR_STUN_MAGIC_COOKIE;
+       break;
+
+#ifdef USE_STUND_0_96
+   case NR_STUN_MODE_STUND_0_96:
+       req->header.magic_cookie = NR_STUN_MAGIC_COOKIE2;
+
+       /* actually, stund 0.96 just ignores the fingerprint
+        * attribute, but don't bother to send it */
+
+       break;
+#endif /* USE_STUND_0_96 */
+
+   }
+
+   *msg = req;
+
+   _status=0;
+ abort:
+   if (_status) RFREE(req);
+   return _status;
+}
+
+int
+nr_stun_build_req_lt_auth(nr_stun_client_stun_binding_request_params *params, nr_stun_message **msg)
+{
+   int r,_status;
+   nr_stun_message *req = 0;
+
+   if ((r=nr_stun_form_request_or_indication(NR_STUN_MODE_STUN, NR_STUN_MSG_BINDING_REQUEST, &req)))
+       ABORT(r);
+
+   if ((r=nr_stun_message_add_username_attribute(req, params->username)))
+       ABORT(r);
+
+   if (params->realm && params->nonce) {
+       if ((r=nr_stun_message_add_realm_attribute(req, params->realm)))
+           ABORT(r);
+
+       if ((r=nr_stun_message_add_nonce_attribute(req, params->nonce)))
+           ABORT(r);
+
+       if ((r=nr_stun_message_add_message_integrity_attribute(req, params->password)))
+           ABORT(r);
+   }
+
+   *msg = req;
+
+   _status=0;
+ abort:
+   if (_status) nr_stun_message_destroy(&req);
+   return _status;
+}
+
+int
+nr_stun_build_req_st_auth(nr_stun_client_stun_binding_request_params *params, nr_stun_message **msg)
+{
+   int r,_status;
+   nr_stun_message *req = 0;
+
+   if ((r=nr_stun_form_request_or_indication(NR_STUN_MODE_STUN, NR_STUN_MSG_BINDING_REQUEST, &req)))
+       ABORT(r);
+
+   if ((r=nr_stun_message_add_username_attribute(req, params->username)))
+       ABORT(r);
+
+   if (params->password) {
+       if ((r=nr_stun_message_add_message_integrity_attribute(req, params->password)))
+           ABORT(r);
+   }
+
+   *msg = req;
+
+   _status=0;
+ abort:
+   if (_status) nr_stun_message_destroy(&req);
+   return _status;
+}
+
+int
+nr_stun_build_req_no_auth(nr_stun_client_stun_binding_request_params *params, nr_stun_message **msg)
+{
+   int r,_status;
+   nr_stun_message *req = 0;
+
+   if ((r=nr_stun_form_request_or_indication(NR_STUN_MODE_STUN_NO_AUTH, NR_STUN_MSG_BINDING_REQUEST, &req)))
+       ABORT(r);
+
+   *msg = req;
+
+   _status=0;
+ abort:
+   if (_status) nr_stun_message_destroy(&req);
+   return _status;
+}
+
+int
+nr_stun_build_keepalive(nr_stun_client_stun_keepalive_params *params, nr_stun_message **msg)
+{
+   int r,_status;
+   nr_stun_message *ind = 0;
+
+   if ((r=nr_stun_form_request_or_indication(NR_STUN_MODE_STUN, NR_STUN_MSG_BINDING_INDICATION, &ind)))
+       ABORT(r);
+
+   *msg = ind;
+
+   _status=0;
+ abort:
+   if (_status) nr_stun_message_destroy(&ind);
+   return _status;
+}
+
+#ifdef USE_STUND_0_96
+int
+nr_stun_build_req_stund_0_96(nr_stun_client_stun_binding_request_stund_0_96_params *params, nr_stun_message **msg)
+{
+    int r,_status;
+    nr_stun_message *req = 0;
+
+    if ((r=nr_stun_form_request_or_indication(NR_STUN_MODE_STUND_0_96, NR_STUN_MSG_BINDING_REQUEST, &req)))
+        ABORT(r);
+
+    if ((r=nr_stun_message_add_change_request_attribute(req, 0)))
+        ABORT(r);
+
+    assert(! nr_stun_message_has_attribute(req, NR_STUN_ATTR_USERNAME, 0));
+    assert(! nr_stun_message_has_attribute(req, NR_STUN_ATTR_MESSAGE_INTEGRITY, 0));
+
+    *msg = req;
+
+   _status=0;
+ abort:
+   if (_status) nr_stun_message_destroy(&req);
+   return _status;
+}
+#endif /* USE_STUND_0_96 */
+
+#ifdef USE_ICE
+int
+nr_stun_build_use_candidate(nr_stun_client_ice_binding_request_params *params, nr_stun_message **msg)
+{
+   int r,_status;
+   nr_stun_message *req = 0;
+
+   if ((r=nr_stun_form_request_or_indication(NR_STUN_MODE_STUN, NR_STUN_MSG_BINDING_REQUEST, &req)))
+       ABORT(r);
+
+   if ((r=nr_stun_message_add_username_attribute(req, params->username)))
+       ABORT(r);
+
+   if ((r=nr_stun_message_add_message_integrity_attribute(req, &params->password)))
+       ABORT(r);
+
+   if ((r=nr_stun_message_add_use_candidate_attribute(req)))
+       ABORT(r);
+
+   if ((r=nr_stun_message_add_priority_attribute(req, params->priority)))
+       ABORT(r);
+
+   if ((r=nr_stun_message_add_ice_controlling_attribute(req, params->tiebreaker)))
+     ABORT(r);
+
+   *msg = req;
+
+   _status=0;
+ abort:
+   if (_status) nr_stun_message_destroy(&req);
+   return _status;
+}
+
+int
+nr_stun_build_req_ice(nr_stun_client_ice_binding_request_params *params, nr_stun_message **msg)
+{
+   int r,_status;
+   nr_stun_message *req = 0;
+
+   if ((r=nr_stun_form_request_or_indication(NR_STUN_MODE_STUN, NR_STUN_MSG_BINDING_REQUEST, &req)))
+       ABORT(r);
+
+   if ((r=nr_stun_message_add_username_attribute(req, params->username)))
+       ABORT(r);
+
+   if ((r=nr_stun_message_add_message_integrity_attribute(req, &params->password)))
+       ABORT(r);
+
+   if ((r=nr_stun_message_add_priority_attribute(req, params->priority)))
+       ABORT(r);
+
+   switch (params->control) {
+   case NR_ICE_CONTROLLING:
+       if ((r=nr_stun_message_add_ice_controlling_attribute(req, params->tiebreaker)))
+           ABORT(r);
+       break;
+   case NR_ICE_CONTROLLED:
+       if ((r=nr_stun_message_add_ice_controlled_attribute(req, params->tiebreaker)))
+           ABORT(r);
+       break;
+   default:
+       assert(0);
+       ABORT(R_INTERNAL);
+   }
+
+   *msg = req;
+
+   _status=0;
+ abort:
+   if (_status) nr_stun_message_destroy(&req);
+   return _status;
+}
+#endif /* USE_ICE */
+
+#ifdef USE_TURN
+
+#ifndef __isascii
+#define __isascii(c) (((c) & ~0x7F) == 0)
+#endif
+
+/* Long-term passwords are computed over the key:
+
+            key = MD5(username ":" realm ":" SASLprep(password))
+
+   Per RFC 5389 S 15.4
+*/
+int
+nr_stun_compute_lt_message_integrity_password(const char *username, const char *realm,
+                                              Data *password, Data *hmac_key)
+{
+  char digest_input[1000];
+  size_t i;
+  int r, _status;
+  size_t len;
+
+  /* First check that the password is ASCII. We are supposed to
+     SASLprep but we don't support this yet
+     TODO(ekr@rtfm.com): Add SASLprep for password.
+ */
+  for (i=0; i<password->len; i++) {
+    if (!__isascii(password->data[i]))
+      ABORT(R_BAD_DATA);
+  }
+
+  if (hmac_key->len < 16)
+    ABORT(R_BAD_ARGS);
+
+  snprintf(digest_input, sizeof(digest_input), "%s:%s:", username, realm);
+  if ((sizeof(digest_input) - strlen(digest_input)) < password->len)
+    ABORT(R_BAD_DATA);
+
+  len = strlen(digest_input);
+  memcpy(digest_input + len, password->data, password->len);
+
+
+  if (r=nr_crypto_md5((UCHAR *)digest_input, len + password->len, hmac_key->data))
+    ABORT(r);
+  hmac_key->len=16;
+
+  _status=0;
+abort:
+  return(_status);
+}
+
+static int
+nr_stun_build_auth_params(nr_stun_client_auth_params *auth, nr_stun_message *req)
+{
+  int r, _status;
+  UCHAR hmac_key_d[16];
+  Data hmac_key;
+
+  ATTACH_DATA(hmac_key, hmac_key_d);
+
+  if (!auth->authenticate)
+    goto done;
+
+  assert(auth->username);
+  assert(auth->password.len);
+  assert(auth->realm);
+  assert(auth->nonce);
+
+  if (r=nr_stun_compute_lt_message_integrity_password(auth->username,
+                                                      auth->realm,
+                                                      &auth->password,
+                                                      &hmac_key))
+    ABORT(r);
+
+  if (!auth->username) {
+    r_log(NR_LOG_STUN, LOG_WARNING, "STUN authentication requested but no username provided");
+    ABORT(R_INTERNAL);
+  }
+
+  if (!auth->password.len) {
+    r_log(NR_LOG_STUN, LOG_WARNING, "STUN authentication requested but no password provided");
+    ABORT(R_INTERNAL);
+  }
+
+  if (!auth->realm) {
+    r_log(NR_LOG_STUN, LOG_WARNING, "STUN authentication requested but no realm provided");
+    ABORT(R_INTERNAL);
+  }
+
+  if (!auth->nonce) {
+    r_log(NR_LOG_STUN, LOG_WARNING, "STUN authentication requested but no nonce provided");
+    ABORT(R_INTERNAL);
+  }
+
+  if ((r=nr_stun_message_add_username_attribute(req, auth->username)))
+    ABORT(r);
+
+  if ((r=nr_stun_message_add_realm_attribute(req, auth->realm)))
+    ABORT(r);
+
+  if ((r=nr_stun_message_add_nonce_attribute(req, auth->nonce)))
+    ABORT(r);
+
+  if ((r=nr_stun_message_add_message_integrity_attribute(req, &hmac_key)))
+    ABORT(r);
+
+done:
+  _status=0;
+abort:
+  return(_status);
+}
+
+int
+nr_stun_build_allocate_request(nr_stun_client_auth_params *auth, nr_stun_client_allocate_request_params *params, nr_stun_message **msg)
+{
+   int r,_status;
+   nr_stun_message *req = 0;
+
+   if ((r=nr_stun_form_request_or_indication(NR_STUN_MODE_STUN, NR_STUN_MSG_ALLOCATE_REQUEST, &req)))
+       ABORT(r);
+
+   if ((r=nr_stun_message_add_requested_transport_attribute(req, NR_STUN_ATTR_REQUESTED_TRANSPORT_UDP)))
+       ABORT(r);
+
+  if ((r=nr_stun_message_add_lifetime_attribute(req, params->lifetime_secs)))
+       ABORT(r);
+
+  /* TODO(ekr@rtfm.com): Add the SOFTWARE attribute (Firefox bug 857666) */
+
+  if ((r=nr_stun_build_auth_params(auth, req)))
+       ABORT(r);
+
+   *msg = req;
+
+   _status=0;
+ abort:
+   if (_status) nr_stun_message_destroy(&req);
+   return _status;
+}
+
+
+int nr_stun_build_refresh_request(nr_stun_client_auth_params *auth, nr_stun_client_refresh_request_params *params, nr_stun_message **msg)
+{
+   int r,_status;
+   nr_stun_message *req = 0;
+
+   if ((r=nr_stun_form_request_or_indication(NR_STUN_MODE_STUN, NR_STUN_MSG_REFRESH_REQUEST, &req)))
+       ABORT(r);
+
+  if ((r=nr_stun_message_add_lifetime_attribute(req, params->lifetime_secs)))
+       ABORT(r);
+
+
+  /* TODO(ekr@rtfm.com): Add the SOFTWARE attribute (Firefox bug 857666) */
+
+  if ((r=nr_stun_build_auth_params(auth, req)))
+       ABORT(r);
+
+  *msg = req;
+
+   _status=0;
+ abort:
+   if (_status) nr_stun_message_destroy(&req);
+   return _status;
+}
+
+
+int nr_stun_build_permission_request(nr_stun_client_auth_params *auth, nr_stun_client_permission_request_params *params, nr_stun_message **msg)
+{
+   int r,_status;
+   nr_stun_message *req = 0;
+
+   if ((r=nr_stun_form_request_or_indication(NR_STUN_MODE_STUN, NR_STUN_MSG_PERMISSION_REQUEST, &req)))
+       ABORT(r);
+
+   if ((r=nr_stun_message_add_xor_peer_address_attribute(req, &params->remote_addr)))
+       ABORT(r);
+
+   if ((r=nr_stun_build_auth_params(auth, req)))
+       ABORT(r);
+
+   *msg = req;
+
+   _status=0;
+ abort:
+   if (_status) nr_stun_message_destroy(&req);
+   return _status;
+}
+
+int
+nr_stun_build_send_indication(nr_stun_client_send_indication_params *params, nr_stun_message **msg)
+{
+   int r,_status;
+   nr_stun_message *ind = 0;
+
+   if ((r=nr_stun_form_request_or_indication(NR_STUN_MODE_STUN, NR_STUN_MSG_SEND_INDICATION, &ind)))
+       ABORT(r);
+
+   if ((r=nr_stun_message_add_xor_peer_address_attribute(ind, &params->remote_addr)))
+       ABORT(r);
+
+   if ((r=nr_stun_message_add_data_attribute(ind, params->data.data, params->data.len)))
+       ABORT(r);
+
+   *msg = ind;
+
+   _status=0;
+ abort:
+   if (_status) nr_stun_message_destroy(&ind);
+   return _status;
+}
+
+int
+nr_stun_build_data_indication(nr_stun_client_data_indication_params *params, nr_stun_message **msg)
+{
+   int r,_status;
+   nr_stun_message *ind = 0;
+
+   if ((r=nr_stun_form_request_or_indication(NR_STUN_MODE_STUN, NR_STUN_MSG_DATA_INDICATION, &ind)))
+       ABORT(r);
+
+   if ((r=nr_stun_message_add_xor_peer_address_attribute(ind, &params->remote_addr)))
+       ABORT(r);
+
+   if ((r=nr_stun_message_add_data_attribute(ind, params->data.data, params->data.len)))
+       ABORT(r);
+
+   *msg = ind;
+
+   _status=0;
+ abort:
+   if (_status) nr_stun_message_destroy(&ind);
+   return _status;
+}
+
+#endif /* USE_TURN */
+
+/* draft-ietf-behave-rfc3489bis-10.txt S 7.3.1.1 */
+int
+nr_stun_form_success_response(nr_stun_message *req, nr_transport_addr *from, Data *password, nr_stun_message *res)
+{
+    int r,_status;
+    int request_method;
+    char server_name[NR_STUN_MAX_SERVER_BYTES+1]; /* +1 for \0 */
+
+    /* set up information for default response */
+
+    request_method = NR_STUN_GET_TYPE_METHOD(req->header.type);
+    res->header.type = NR_STUN_TYPE(request_method, NR_CLASS_RESPONSE);
+    res->header.magic_cookie = req->header.magic_cookie;
+    memcpy(&res->header.id, &req->header.id, sizeof(res->header.id));
+
+    r_log(NR_LOG_STUN, LOG_DEBUG, "Mapped Address = %s", from->as_string);
+
+    if ((r=nr_stun_message_add_xor_mapped_address_attribute(res, from)))
+        ABORT(r);
+
+    if (!NR_reg_get_string(NR_STUN_REG_PREF_SERVER_NAME, server_name, sizeof(server_name))) {
+        if ((r=nr_stun_message_add_server_attribute(res, server_name)))
+            ABORT(r);
+    }
+
+    if (res->header.magic_cookie == NR_STUN_MAGIC_COOKIE) {
+        if (password != 0) {
+            if ((r=nr_stun_message_add_message_integrity_attribute(res, password)))
+                ABORT(r);
+        }
+
+        if ((r=nr_stun_message_add_fingerprint_attribute(res)))
+            ABORT(r);
+    }
+
+    _status=0;
+ abort:
+    return _status;
+}
+
+/* draft-ietf-behave-rfc3489bis-10.txt S 7.3.1.1 */
+void
+nr_stun_form_error_response(nr_stun_message *req, nr_stun_message* res, int number, char* msg)
+{
+    char *str;
+    int request_method;
+    char server_name[NR_STUN_MAX_SERVER_BYTES+1]; /* +1 for \0 */
+
+    if (number < 300 || number > 699)
+        number = 500;
+
+    r_log(NR_LOG_STUN, LOG_INFO, "Responding with error %d: %s", number, msg);
+
+    request_method = NR_STUN_GET_TYPE_METHOD(req->header.type);
+    res->header.type = NR_STUN_TYPE(request_method, NR_CLASS_ERROR_RESPONSE);
+    res->header.magic_cookie = req->header.magic_cookie;
+    memcpy(&res->header.id, &req->header.id, sizeof(res->header.id));
+
+    /* during development we should never see 500s (hopefully not in deployment either) */
+
+    str = 0;
+    switch (number) {
+    case 300:  str = "Try Alternate";       break;
+    case 400:  str = "Bad Request";         break;
+    case 401:  str = "Unauthorized";        break;
+    case 420:  str = "Unknown Attribute";   break;
+    case 438:  str = "Stale Nonce";         break;
+#ifdef USE_ICE
+    case 487:  str = "Role Conflict";       break;
+#endif
+    case 500:  str = "Server Error";        break;
+    }
+    if (str == 0) {
+        str = "Unknown";
+    }
+
+    if (nr_stun_message_add_error_code_attribute(res, number, str)) {
+         assert(0);  /* should never happen */
+    }
+
+    if (!NR_reg_get_string(NR_STUN_REG_PREF_SERVER_NAME, server_name, sizeof(server_name))) {
+        nr_stun_message_add_server_attribute(res, server_name);
+    }
+}
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_build.h b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_build.h
new file mode 100644
index 0000000000..c3f91a87b0
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_build.h
@@ -0,0 +1,147 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+#ifndef _stun_build_h
+#define _stun_build_h
+
+#include "stun.h"
+
+#define NR_STUN_MODE_STUN               1
+#ifdef USE_STUND_0_96
+#define NR_STUN_MODE_STUND_0_96         2    /* backwards compatibility mode */
+#endif /* USE_STUND_0_96 */
+#define NR_STUN_MODE_STUN_NO_AUTH       3
+int nr_stun_form_request_or_indication(int mode, int msg_type, nr_stun_message **msg);
+
+typedef struct nr_stun_client_stun_binding_request_params_ {
+    char *username;
+    Data *password;
+    char *nonce;
+    char *realm;
+} nr_stun_client_stun_binding_request_params;
+
+int nr_stun_build_req_lt_auth(nr_stun_client_stun_binding_request_params *params, nr_stun_message **msg);
+int nr_stun_build_req_st_auth(nr_stun_client_stun_binding_request_params *params, nr_stun_message **msg);
+int nr_stun_build_req_no_auth(nr_stun_client_stun_binding_request_params *params, nr_stun_message **msg);
+
+
+typedef struct nr_stun_client_stun_keepalive_params_ {
+#if defined(WIN32) || defined(__clang__)
+    // VC++ and clang give error and warning respectively if no members
+    int dummy;
+#endif
+} nr_stun_client_stun_keepalive_params;
+
+int nr_stun_build_keepalive(nr_stun_client_stun_keepalive_params *params, nr_stun_message **msg);
+
+
+#ifdef USE_STUND_0_96
+typedef struct nr_stun_client_stun_binding_request_stund_0_96_params_ {
+#ifdef WIN32  // silly VC++ gives error if no members
+    int dummy;
+#endif
+} nr_stun_client_stun_binding_request_stund_0_96_params;
+
+int nr_stun_build_req_stund_0_96(nr_stun_client_stun_binding_request_stund_0_96_params *params, nr_stun_message **msg);
+#endif /* USE_STUND_0_96 */
+
+
+#ifdef USE_ICE
+typedef struct nr_stun_client_ice_binding_request_params_ {
+    char *username;
+    Data password;
+    UINT4 priority;
+    int control;
+#define NR_ICE_CONTROLLING  1
+#define NR_ICE_CONTROLLED   2
+    UINT8 tiebreaker;
+} nr_stun_client_ice_binding_request_params;
+
+int nr_stun_build_use_candidate(nr_stun_client_ice_binding_request_params *params, nr_stun_message **msg);
+
+int nr_stun_build_req_ice(nr_stun_client_ice_binding_request_params *params, nr_stun_message **msg);
+#endif /* USE_ICE */
+
+
+typedef struct nr_stun_client_auth_params_ {
+    char authenticate;
+    char *username;
+    char *realm;
+    char *nonce;
+    Data password;
+} nr_stun_client_auth_params;
+
+#ifdef USE_TURN
+typedef struct nr_stun_client_allocate_request_params_ {
+    UINT4 lifetime_secs;
+} nr_stun_client_allocate_request_params;
+
+int nr_stun_build_allocate_request(nr_stun_client_auth_params *auth, nr_stun_client_allocate_request_params *params, nr_stun_message **msg);
+
+
+typedef struct nr_stun_client_refresh_request_params_ {
+    UINT4 lifetime_secs;
+} nr_stun_client_refresh_request_params;
+
+int nr_stun_build_refresh_request(nr_stun_client_auth_params *auth, nr_stun_client_refresh_request_params *params, nr_stun_message **msg);
+
+
+
+typedef struct nr_stun_client_permission_request_params_ {
+    nr_transport_addr remote_addr;
+} nr_stun_client_permission_request_params;
+
+int nr_stun_build_permission_request(nr_stun_client_auth_params *auth, nr_stun_client_permission_request_params *params, nr_stun_message **msg);
+
+
+typedef struct nr_stun_client_send_indication_params_ {
+    nr_transport_addr remote_addr;
+    Data data;
+} nr_stun_client_send_indication_params;
+
+int nr_stun_build_send_indication(nr_stun_client_send_indication_params *params, nr_stun_message **msg);
+
+typedef struct nr_stun_client_data_indication_params_ {
+    nr_transport_addr remote_addr;
+    Data data;
+} nr_stun_client_data_indication_params;
+
+int nr_stun_build_data_indication(nr_stun_client_data_indication_params *params, nr_stun_message **msg);
+#endif /* USE_TURN */
+
+int nr_stun_form_success_response(nr_stun_message *req, nr_transport_addr *from, Data *password, nr_stun_message *res);
+void nr_stun_form_error_response(nr_stun_message *request, nr_stun_message* response, int number, char* msg);
+int nr_stun_compute_lt_message_integrity_password(const char *username, const char *realm,
+                                                  Data *password, Data *hmac_key);
+
+#endif
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_client_ctx.c b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_client_ctx.c
new file mode 100644
index 0000000000..50b9f74a5b
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_client_ctx.c
@@ -0,0 +1,888 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <assert.h>
+#include <string.h>
+#include <math.h>
+
+#include <nr_api.h>
+#include "stun.h"
+#include "async_timer.h"
+#include "registry.h"
+#include "stun_reg.h"
+#include "nr_crypto.h"
+#include "r_time.h"
+
+static int nr_stun_client_send_request(nr_stun_client_ctx *ctx);
+static void nr_stun_client_timer_expired_cb(NR_SOCKET s, int b, void *cb_arg);
+static int nr_stun_client_get_password(void *arg, nr_stun_message *msg, Data **password);
+
+#define NR_STUN_TRANSPORT_ADDR_CHECK_WILDCARD 1
+#define NR_STUN_TRANSPORT_ADDR_CHECK_LOOPBACK 2
+
+int nr_stun_client_ctx_create(char *label, nr_socket *sock, nr_transport_addr *peer, UINT4 RTO, nr_stun_client_ctx **ctxp)
+  {
+    nr_stun_client_ctx *ctx=0;
+    char allow_loopback;
+    int r,_status;
+
+    if ((r=nr_stun_startup()))
+      ABORT(r);
+
+    if(!(ctx=RCALLOC(sizeof(nr_stun_client_ctx))))
+      ABORT(R_NO_MEMORY);
+
+    ctx->state=NR_STUN_CLIENT_STATE_INITTED;
+
+    if(!(ctx->label=r_strdup(label)))
+      ABORT(R_NO_MEMORY);
+
+    ctx->sock=sock;
+
+    nr_socket_getaddr(sock,&ctx->my_addr);
+    nr_transport_addr_copy(&ctx->peer_addr,peer);
+    assert(ctx->my_addr.protocol==ctx->peer_addr.protocol);
+
+    if (RTO != 0) {
+      ctx->rto_ms = RTO;
+    } else if (NR_reg_get_uint4(NR_STUN_REG_PREF_CLNT_RETRANSMIT_TIMEOUT, &ctx->rto_ms)) {
+      ctx->rto_ms = 100;
+    }
+
+    if (NR_reg_get_double(NR_STUN_REG_PREF_CLNT_RETRANSMIT_BACKOFF, &ctx->retransmission_backoff_factor))
+      ctx->retransmission_backoff_factor = 2.0;
+
+    if (NR_reg_get_uint4(NR_STUN_REG_PREF_CLNT_MAXIMUM_TRANSMITS, &ctx->maximum_transmits))
+      ctx->maximum_transmits = 7;
+
+    if (NR_reg_get_uint4(NR_STUN_REG_PREF_CLNT_FINAL_RETRANSMIT_BACKOFF, &ctx->maximum_transmits_timeout_ms))
+      ctx->maximum_transmits_timeout_ms = 16 * ctx->rto_ms;
+
+    ctx->mapped_addr_check_mask = NR_STUN_TRANSPORT_ADDR_CHECK_WILDCARD;
+    if (NR_reg_get_char(NR_STUN_REG_PREF_ALLOW_LOOPBACK_ADDRS, &allow_loopback) ||
+        !allow_loopback) {
+      ctx->mapped_addr_check_mask |= NR_STUN_TRANSPORT_ADDR_CHECK_LOOPBACK;
+    }
+
+    if (ctx->my_addr.protocol == IPPROTO_TCP) {
+      /* Because TCP is reliable there is only one final timeout value.
+       * We store the timeout value for TCP in here, because timeout_ms gets
+       * reset to 0 in client_reset() which gets called from client_start() */
+      ctx->maximum_transmits_timeout_ms = ctx->rto_ms *
+                        pow(ctx->retransmission_backoff_factor,
+                            ctx->maximum_transmits);
+      ctx->maximum_transmits = 1;
+    }
+
+    *ctxp=ctx;
+
+    _status=0;
+  abort:
+    if(_status){
+      nr_stun_client_ctx_destroy(&ctx);
+    }
+    return(_status);
+  }
+
+static void nr_stun_client_fire_finished_cb(nr_stun_client_ctx *ctx)
+  {
+    if (ctx->finished_cb) {
+      NR_async_cb finished_cb = ctx->finished_cb;
+      ctx->finished_cb = 0;  /* prevent 2nd call */
+      /* finished_cb call must be absolutely last thing in function
+       * because as a side effect this ctx may be operated on in the
+       * callback */
+      finished_cb(0,0,ctx->cb_arg);
+    }
+  }
+
+int nr_stun_client_start(nr_stun_client_ctx *ctx, int mode, NR_async_cb finished_cb, void *cb_arg)
+  {
+    int r,_status;
+
+    if (ctx->state != NR_STUN_CLIENT_STATE_INITTED)
+        ABORT(R_NOT_PERMITTED);
+
+    ctx->mode=mode;
+
+    ctx->state=NR_STUN_CLIENT_STATE_RUNNING;
+    ctx->finished_cb=finished_cb;
+    ctx->cb_arg=cb_arg;
+
+    if(mode!=NR_STUN_CLIENT_MODE_KEEPALIVE){
+      if(r=nr_stun_client_send_request(ctx))
+        ABORT(r);
+    }
+
+    _status=0;
+  abort:
+   if (ctx->state != NR_STUN_CLIENT_STATE_RUNNING) {
+     nr_stun_client_fire_finished_cb(ctx);
+    }
+
+    return(_status);
+  }
+
+  int nr_stun_client_restart(nr_stun_client_ctx* ctx,
+                             const nr_transport_addr* peer_addr) {
+    int r,_status;
+    int mode;
+    NR_async_cb finished_cb;
+    void *cb_arg;
+    if (ctx->state != NR_STUN_CLIENT_STATE_RUNNING)
+        ABORT(R_NOT_PERMITTED);
+
+    mode = ctx->mode;
+    finished_cb = ctx->finished_cb;
+    cb_arg = ctx->cb_arg;
+
+    nr_stun_client_reset(ctx);
+    nr_transport_addr_copy(&ctx->peer_addr, peer_addr);
+
+    if (r=nr_stun_client_start(ctx, mode, finished_cb, cb_arg))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int
+nr_stun_client_reset(nr_stun_client_ctx *ctx)
+{
+    /* Cancel the timer firing */
+    if (ctx->timer_handle){
+        NR_async_timer_cancel(ctx->timer_handle);
+        ctx->timer_handle=0;
+    }
+
+    nr_stun_message_destroy(&ctx->request);
+    ctx->request   = 0;
+
+    nr_stun_message_destroy(&ctx->response);
+    ctx->response  = 0;
+
+    memset(&ctx->results, 0, sizeof(ctx->results));
+
+    ctx->mode          = 0;
+    ctx->finished_cb   = 0;
+    ctx->cb_arg        = 0;
+    ctx->request_ct    = 0;
+    ctx->timeout_ms    = 0;
+
+    ctx->state = NR_STUN_CLIENT_STATE_INITTED;
+
+    return 0;
+}
+
+static void nr_stun_client_timer_expired_cb(NR_SOCKET s, int b, void *cb_arg)
+  {
+    int _status;
+    nr_stun_client_ctx *ctx=cb_arg;
+    struct timeval now;
+    INT8 ms_waited;
+
+    /* Prevent this timer from being cancelled later */
+    ctx->timer_handle=0;
+
+    /* Shouldn't happen */
+    if(ctx->state==NR_STUN_CLIENT_STATE_CANCELLED)
+      ABORT(R_REJECTED);
+
+    gettimeofday(&now, 0);
+    if (r_timeval_diff_ms(&now, &ctx->timer_set, &ms_waited)) {
+        r_log(NR_LOG_STUN,LOG_DEBUG,"STUN-CLIENT(%s): Timer expired",ctx->label);
+    }
+    else {
+        r_log(NR_LOG_STUN,LOG_DEBUG,"STUN-CLIENT(%s): Timer expired (after %llu ms)",ctx->label, ms_waited);
+    }
+
+    if (ctx->request_ct >= ctx->maximum_transmits) {
+        r_log(NR_LOG_STUN,LOG_INFO,"STUN-CLIENT(%s): Timed out",ctx->label);
+        ctx->state=NR_STUN_CLIENT_STATE_TIMED_OUT;
+        ABORT(R_FAILED);
+    }
+
+    if (ctx->state != NR_STUN_CLIENT_STATE_RUNNING)
+        ABORT(R_NOT_PERMITTED);
+
+    // track retransmits for ice telemetry
+    nr_accumulate_count(&(ctx->retransmit_ct), 1);
+
+    /* as a side effect will reset the timer */
+    nr_stun_client_send_request(ctx);
+
+    _status = 0;
+  abort:
+    if (ctx->state != NR_STUN_CLIENT_STATE_RUNNING) {
+        /* Cancel the timer firing */
+        if (ctx->timer_handle){
+            NR_async_timer_cancel(ctx->timer_handle);
+            ctx->timer_handle=0;
+        }
+
+        nr_stun_client_fire_finished_cb(ctx);
+    }
+    if (_status) {
+      // cb doesn't return anything, but we should probably log that we aborted
+      // This also quiets the unused variable warnings.
+      r_log(NR_LOG_STUN,LOG_DEBUG,"STUN-CLIENT(%s): Timer expired cb abort with status: %d",
+        ctx->label, _status);
+    }
+    return;
+  }
+
+int nr_stun_client_force_retransmit(nr_stun_client_ctx *ctx)
+  {
+    int r,_status;
+
+    if (ctx->state != NR_STUN_CLIENT_STATE_RUNNING)
+        ABORT(R_NOT_PERMITTED);
+
+    if (ctx->request_ct > ctx->maximum_transmits) {
+        r_log(NR_LOG_STUN,LOG_INFO,"STUN-CLIENT(%s): Too many retransmit attempts",ctx->label);
+        ABORT(R_FAILED);
+    }
+
+    /* if there is a scheduled retransimt, get rid of the scheduled retransmit
+     * and retransmit immediately */
+    if (ctx->timer_handle) {
+        NR_async_timer_cancel(ctx->timer_handle);
+        ctx->timer_handle=0;
+
+        if (r=nr_stun_client_send_request(ctx))
+            ABORT(r);
+    }
+
+    _status=0;
+  abort:
+
+    return(_status);
+  }
+
+static int nr_stun_client_send_request(nr_stun_client_ctx *ctx)
+  {
+    int r,_status;
+    char string[256];
+
+    if (ctx->state != NR_STUN_CLIENT_STATE_RUNNING)
+        ABORT(R_NOT_PERMITTED);
+
+    r_log(NR_LOG_STUN,LOG_DEBUG,"STUN-CLIENT(%s): Sending check request (my_addr=%s,peer_addr=%s)",ctx->label,ctx->my_addr.as_string,ctx->peer_addr.as_string);
+
+    if (ctx->request == 0) {
+        switch (ctx->mode) {
+        case NR_STUN_CLIENT_MODE_BINDING_REQUEST_LONG_TERM_AUTH:
+            ctx->params.stun_binding_request.nonce = ctx->nonce;
+            ctx->params.stun_binding_request.realm = ctx->realm;
+            assert(0);
+            ABORT(R_INTERNAL);
+            /* TODO(ekr@rtfm.com): Need to implement long-term auth for binding
+               requests */
+            if ((r=nr_stun_build_req_lt_auth(&ctx->params.stun_binding_request, &ctx->request)))
+                ABORT(r);
+            break;
+        case NR_STUN_CLIENT_MODE_BINDING_REQUEST_SHORT_TERM_AUTH:
+            if ((r=nr_stun_build_req_st_auth(&ctx->params.stun_binding_request, &ctx->request)))
+                ABORT(r);
+            break;
+        case NR_STUN_CLIENT_MODE_BINDING_REQUEST_NO_AUTH:
+            if ((r=nr_stun_build_req_no_auth(&ctx->params.stun_binding_request, &ctx->request)))
+                ABORT(r);
+            break;
+        case NR_STUN_CLIENT_MODE_KEEPALIVE:
+            if ((r=nr_stun_build_keepalive(&ctx->params.stun_keepalive, &ctx->request)))
+                ABORT(r);
+            break;
+#ifdef USE_STUND_0_96
+        case NR_STUN_CLIENT_MODE_BINDING_REQUEST_STUND_0_96:
+            if ((r=nr_stun_build_req_stund_0_96(&ctx->params.stun_binding_request_stund_0_96, &ctx->request)))
+                ABORT(r);
+            break;
+#endif /* USE_STUND_0_96 */
+
+#ifdef USE_ICE
+        case NR_ICE_CLIENT_MODE_USE_CANDIDATE:
+            if ((r=nr_stun_build_use_candidate(&ctx->params.ice_binding_request, &ctx->request)))
+                ABORT(r);
+            break;
+        case NR_ICE_CLIENT_MODE_BINDING_REQUEST:
+            if ((r=nr_stun_build_req_ice(&ctx->params.ice_binding_request, &ctx->request)))
+                ABORT(r);
+            break;
+#endif /* USE_ICE */
+
+#ifdef USE_TURN
+        case NR_TURN_CLIENT_MODE_ALLOCATE_REQUEST:
+            if ((r=nr_stun_build_allocate_request(&ctx->auth_params, &ctx->params.allocate_request, &ctx->request)))
+                ABORT(r);
+            break;
+        case NR_TURN_CLIENT_MODE_REFRESH_REQUEST:
+            if ((r=nr_stun_build_refresh_request(&ctx->auth_params, &ctx->params.refresh_request, &ctx->request)))
+                ABORT(r);
+            break;
+        case NR_TURN_CLIENT_MODE_PERMISSION_REQUEST:
+            if ((r=nr_stun_build_permission_request(&ctx->auth_params, &ctx->params.permission_request, &ctx->request)))
+                ABORT(r);
+            break;
+        case NR_TURN_CLIENT_MODE_SEND_INDICATION:
+            if ((r=nr_stun_build_send_indication(&ctx->params.send_indication, &ctx->request)))
+                ABORT(r);
+            break;
+#endif /* USE_TURN */
+
+        default:
+            assert(0);
+            ABORT(R_FAILED);
+            break;
+        }
+    }
+
+    if (ctx->request->length == 0) {
+        if ((r=nr_stun_encode_message(ctx->request)))
+            ABORT(r);
+    }
+
+    snprintf(string, sizeof(string)-1, "STUN-CLIENT(%s): Sending to %s ", ctx->label, ctx->peer_addr.as_string);
+    r_dump(NR_LOG_STUN, LOG_DEBUG, string, (char*)ctx->request->buffer, ctx->request->length);
+
+    assert(ctx->my_addr.protocol==ctx->peer_addr.protocol);
+
+    if(r=nr_socket_sendto(ctx->sock, ctx->request->buffer, ctx->request->length, 0, &ctx->peer_addr)) {
+      if (r != R_WOULDBLOCK) {
+        ABORT(r);
+      }
+      r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): nr_socket_sendto blocked, treating as dropped packet",ctx->label);
+    }
+
+    ctx->request_ct++;
+
+    if (NR_STUN_GET_TYPE_CLASS(ctx->request->header.type) == NR_CLASS_INDICATION) {
+        /* no need to set the timer because indications don't receive a
+         * response */
+    }
+    else {
+        if (ctx->request_ct >= ctx->maximum_transmits) {
+          /* Reliable transport only get here once. Unreliable get here for
+           * their final timeout. */
+          ctx->timeout_ms += ctx->maximum_transmits_timeout_ms;
+        }
+        else if (ctx->timeout_ms) {
+          /* exponential backoff */
+          ctx->timeout_ms *= ctx->retransmission_backoff_factor;
+        }
+        else {
+          /* initial timeout unreliable transports */
+          ctx->timeout_ms = ctx->rto_ms;
+        }
+
+        r_log(NR_LOG_STUN,LOG_DEBUG,"STUN-CLIENT(%s): Next timer will fire in %u ms",ctx->label, ctx->timeout_ms);
+
+        gettimeofday(&ctx->timer_set, 0);
+
+        assert(ctx->timeout_ms);
+        NR_ASYNC_TIMER_SET(ctx->timeout_ms, nr_stun_client_timer_expired_cb, ctx, &ctx->timer_handle);
+    }
+
+    _status=0;
+  abort:
+    if (_status) {
+      nr_stun_client_failed(ctx);
+    }
+    return(_status);
+  }
+
+static int nr_stun_client_get_password(void *arg, nr_stun_message *msg, Data **password)
+{
+    *password = (Data*)arg;
+    if (!arg)
+        return(R_NOT_FOUND);
+    return(0);
+}
+
+int nr_stun_transport_addr_check(nr_transport_addr* addr, UINT4 check)
+  {
+    if((check & NR_STUN_TRANSPORT_ADDR_CHECK_WILDCARD) && nr_transport_addr_is_wildcard(addr))
+      return(R_BAD_DATA);
+
+    if ((check & NR_STUN_TRANSPORT_ADDR_CHECK_LOOPBACK) && nr_transport_addr_is_loopback(addr))
+      return(R_BAD_DATA);
+
+    return(0);
+  }
+
+int nr_stun_client_process_response(nr_stun_client_ctx *ctx, UCHAR *msg, int len, nr_transport_addr *peer_addr)
+  {
+    int r,_status;
+    char string[256];
+    char *username = 0;
+    Data *password = 0;
+    int allow_unauthed_redirect = 0;
+    nr_stun_message_attribute *attr;
+    nr_transport_addr *mapped_addr = 0;
+    int fail_on_error = 0;
+    UCHAR hmac_key_d[16];
+    Data hmac_key;
+    int compute_lt_key=0;
+    /* TODO(bcampen@mozilla.com): Bug 1023619, refactor this. */
+    int response_matched=0;
+
+    ATTACH_DATA(hmac_key, hmac_key_d);
+
+    if ((ctx->state != NR_STUN_CLIENT_STATE_RUNNING) &&
+        (ctx->state != NR_STUN_CLIENT_STATE_WAITING))
+      ABORT(R_REJECTED);
+
+    r_log(NR_LOG_STUN,LOG_DEBUG,"STUN-CLIENT(%s): Inspecting STUN response (my_addr=%s, peer_addr=%s)",ctx->label,ctx->my_addr.as_string,peer_addr->as_string);
+
+    snprintf(string, sizeof(string)-1, "STUN-CLIENT(%s): Received ", ctx->label);
+    r_dump(NR_LOG_STUN, LOG_DEBUG, string, (char*)msg, len);
+
+    /* determine password */
+    switch (ctx->mode) {
+    case NR_STUN_CLIENT_MODE_BINDING_REQUEST_LONG_TERM_AUTH:
+      /* If the STUN server responds with an error, give up, since we don't
+       * want to delay the completion of gathering. */
+      fail_on_error = 1;
+      compute_lt_key = 1;
+      /* Fall through */
+    case NR_STUN_CLIENT_MODE_BINDING_REQUEST_SHORT_TERM_AUTH:
+      password = ctx->params.stun_binding_request.password;
+      break;
+
+    case NR_STUN_CLIENT_MODE_BINDING_REQUEST_NO_AUTH:
+      /* If the STUN server responds with an error, give up, since we don't
+       * want to delay the completion of gathering. */
+      fail_on_error = 1;
+      break;
+
+    case NR_STUN_CLIENT_MODE_BINDING_REQUEST_STUND_0_96:
+      /* If the STUN server responds with an error, give up, since we don't
+       * want to delay the completion of gathering. */
+      fail_on_error = 1;
+      break;
+
+#ifdef USE_ICE
+    case NR_ICE_CLIENT_MODE_BINDING_REQUEST:
+      /* We do not set fail_on_error here. The error might be transient, and
+       * retrying isn't going to cause a slowdown. */
+      password = &ctx->params.ice_binding_request.password;
+      break;
+    case NR_ICE_CLIENT_MODE_USE_CANDIDATE:
+      /* We do not set fail_on_error here. The error might be transient, and
+       * retrying isn't going to cause a slowdown. */
+      password = &ctx->params.ice_binding_request.password;
+      break;
+#endif /* USE_ICE */
+
+#ifdef USE_TURN
+    case NR_TURN_CLIENT_MODE_ALLOCATE_REQUEST:
+      fail_on_error = 1;
+      compute_lt_key = 1;
+      /* Do not require mutual auth on redirect responses to Allocate requests. */
+      allow_unauthed_redirect = 1;
+      username = ctx->auth_params.username;
+      password = &ctx->auth_params.password;
+      /* do nothing */
+      break;
+    case NR_TURN_CLIENT_MODE_REFRESH_REQUEST:
+      fail_on_error = 1;
+      compute_lt_key = 1;
+      username = ctx->auth_params.username;
+      password = &ctx->auth_params.password;
+      /* do nothing */
+      break;
+    case NR_TURN_CLIENT_MODE_PERMISSION_REQUEST:
+      fail_on_error = 1;
+      compute_lt_key = 1;
+      username = ctx->auth_params.username;
+      password = &ctx->auth_params.password;
+      /* do nothing */
+      break;
+    case NR_TURN_CLIENT_MODE_SEND_INDICATION:
+      /* do nothing -- we just got our DATA-INDICATION */
+      break;
+#endif /* USE_TURN */
+
+    default:
+      assert(0);
+      ABORT(R_FAILED);
+      break;
+    }
+
+    if (compute_lt_key) {
+      if (!ctx->realm || !username) {
+        r_log(NR_LOG_STUN, LOG_DEBUG, "Long-term auth required but no realm/username specified. Randomizing key");
+        /* Fill the key with random bytes to guarantee non-match */
+        if (r=nr_crypto_random_bytes(hmac_key_d, sizeof(hmac_key_d)))
+          ABORT(r);
+      }
+      else {
+        if (r=nr_stun_compute_lt_message_integrity_password(username, ctx->realm,
+                                                            password, &hmac_key))
+          ABORT(r);
+      }
+      password = &hmac_key;
+    }
+
+    if (ctx->response) {
+      nr_stun_message_destroy(&ctx->response);
+    }
+
+    /* TODO(bcampen@mozilla.com): Bug 1023619, refactor this. */
+    if ((r=nr_stun_message_create2(&ctx->response, msg, len)))
+        ABORT(r);
+
+    if ((r=nr_stun_decode_message(ctx->response, nr_stun_client_get_password, password))) {
+        r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): error decoding response",ctx->label);
+        ABORT(r);
+    }
+
+    /* This will return an error if request and response don't match,
+       which is how we reject responses that match other contexts. */
+    if ((r=nr_stun_receive_message(ctx->request, ctx->response))) {
+        r_log(NR_LOG_STUN,LOG_DEBUG,"STUN-CLIENT(%s): Response is not for us",ctx->label);
+        ABORT(r);
+    }
+
+    r_log(NR_LOG_STUN,LOG_INFO,
+          "STUN-CLIENT(%s): Received response; processing",ctx->label);
+    response_matched=1;
+
+    if (allow_unauthed_redirect &&
+        nr_stun_message_has_attribute(ctx->response, NR_STUN_ATTR_ERROR_CODE,
+                                      &attr) &&
+        (attr->u.error_code.number / 100 == 3)) {
+      password = 0;
+    }
+
+/* TODO: !nn! currently using password!=0 to mean that auth is required,
+ * TODO: !nn! but we should probably pass that in explicitly via the
+ * TODO: !nn! usage (ctx->mode?) */
+    if (password) {
+        if (nr_stun_message_has_attribute(ctx->response, NR_STUN_ATTR_NONCE, 0)) {
+            if ((r=nr_stun_receive_response_long_term_auth(ctx->response, ctx))) {
+                r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): long term auth failed",ctx->label);
+                ABORT(r);
+            }
+        }
+        else {
+            if ((r=nr_stun_receive_response_short_term_auth(ctx->response))) {
+                r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): short term auth failed",ctx->label);
+                ABORT(r);
+            }
+        }
+    }
+
+    if (NR_STUN_GET_TYPE_CLASS(ctx->response->header.type) == NR_CLASS_RESPONSE) {
+        if ((r=nr_stun_process_success_response(ctx->response))) {
+            r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): nr_stun_process_success_response failed",ctx->label);
+            ABORT(r);
+        }
+    }
+    else {
+      if (fail_on_error) {
+        ctx->state = NR_STUN_CLIENT_STATE_FAILED;
+      }
+        /* Note: most times we call process_error_response, we get r != 0.
+
+           However, if the error is to be discarded, we get r == 0, smash
+           the error code, and just keep going.
+        */
+        if ((r=nr_stun_process_error_response(ctx->response, &ctx->error_code))) {
+            r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): nr_stun_process_error_response failed",ctx->label);
+            ABORT(r);
+        }
+        else {
+          ctx->error_code = 0xffff;
+            /* drop the error on the floor */
+            r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): processed error response",ctx->label);
+            ABORT(R_FAILED);
+        }
+    }
+
+    r_log(NR_LOG_STUN,LOG_DEBUG,"STUN-CLIENT(%s): Successfully parsed mode=%d",ctx->label,ctx->mode);
+
+/* TODO: !nn! this should be moved to individual message receive/processing sections */
+    switch (ctx->mode) {
+    case NR_STUN_CLIENT_MODE_BINDING_REQUEST_LONG_TERM_AUTH:
+    case NR_STUN_CLIENT_MODE_BINDING_REQUEST_SHORT_TERM_AUTH:
+        if (! nr_stun_message_has_attribute(ctx->response, NR_STUN_ATTR_XOR_MAPPED_ADDRESS, 0)) {
+            r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): No XOR-MAPPED-ADDRESS",ctx->label);
+            ABORT(R_BAD_DATA);
+        }
+        if (! nr_stun_message_has_attribute(ctx->response, NR_STUN_ATTR_MESSAGE_INTEGRITY, 0)) {
+            r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): No MESSAGE-INTEGRITY",ctx->label);
+            ABORT(R_BAD_DATA);
+        }
+
+        mapped_addr = &ctx->results.stun_binding_response.mapped_addr;
+        break;
+
+    case NR_STUN_CLIENT_MODE_BINDING_REQUEST_NO_AUTH:
+        if (! nr_stun_message_has_attribute(ctx->response, NR_STUN_ATTR_XOR_MAPPED_ADDRESS, 0)) {
+            if (nr_stun_message_has_attribute(ctx->response, NR_STUN_ATTR_MAPPED_ADDRESS, 0)) {
+              /* Compensate for a bug in Google's STUN servers where they always respond with MAPPED-ADDRESS */
+              r_log(NR_LOG_STUN,LOG_INFO,"STUN-CLIENT(%s): No XOR-MAPPED-ADDRESS but MAPPED-ADDRESS. Falling back (though server is wrong).", ctx->label);
+            }
+            else {
+              r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): No XOR-MAPPED-ADDRESS or MAPPED-ADDRESS",ctx->label);
+              ABORT(R_BAD_DATA);
+            }
+        }
+
+        mapped_addr = &ctx->results.stun_binding_response.mapped_addr;
+        break;
+
+    case NR_STUN_CLIENT_MODE_BINDING_REQUEST_STUND_0_96:
+        if (! nr_stun_message_has_attribute(ctx->response, NR_STUN_ATTR_MAPPED_ADDRESS, 0) && ! nr_stun_message_has_attribute(ctx->response, NR_STUN_ATTR_XOR_MAPPED_ADDRESS, 0)) {
+            r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): No MAPPED-ADDRESS",ctx->label);
+            ABORT(R_BAD_DATA);
+        }
+
+        mapped_addr = &ctx->results.stun_binding_response_stund_0_96.mapped_addr;
+        break;
+
+#ifdef USE_ICE
+    case NR_ICE_CLIENT_MODE_BINDING_REQUEST:
+        if (! nr_stun_message_has_attribute(ctx->response, NR_STUN_ATTR_XOR_MAPPED_ADDRESS, 0)) {
+            r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): No XOR-MAPPED-ADDRESS",ctx->label);
+            ABORT(R_BAD_DATA);
+        }
+        if (! nr_stun_message_has_attribute(ctx->response, NR_STUN_ATTR_MESSAGE_INTEGRITY, 0)) {
+            r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): No MESSAGE-INTEGRITY",ctx->label);
+            ABORT(R_BAD_DATA);
+        }
+
+        mapped_addr = &ctx->results.stun_binding_response.mapped_addr;
+        break;
+    case NR_ICE_CLIENT_MODE_USE_CANDIDATE:
+        if (! nr_stun_message_has_attribute(ctx->response, NR_STUN_ATTR_XOR_MAPPED_ADDRESS, 0)) {
+            r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): No XOR-MAPPED-ADDRESS",ctx->label);
+            ABORT(R_BAD_DATA);
+        }
+        if (! nr_stun_message_has_attribute(ctx->response, NR_STUN_ATTR_MESSAGE_INTEGRITY, 0)) {
+            r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): No MESSAGE-INTEGRITY",ctx->label);
+            ABORT(R_BAD_DATA);
+        }
+
+        mapped_addr = &ctx->results.stun_binding_response.mapped_addr;
+        break;
+#endif /* USE_ICE */
+
+#ifdef USE_TURN
+    case NR_TURN_CLIENT_MODE_ALLOCATE_REQUEST:
+        if (! nr_stun_message_has_attribute(ctx->response, NR_STUN_ATTR_XOR_MAPPED_ADDRESS, 0)) {
+            r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): No XOR-MAPPED-ADDRESS",ctx->label);
+            ABORT(R_BAD_DATA);
+        }
+        if (! nr_stun_message_has_attribute(ctx->response, NR_STUN_ATTR_MESSAGE_INTEGRITY, 0)) {
+            r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): No MESSAGE-INTEGRITY",ctx->label);
+            ABORT(R_BAD_DATA);
+        }
+
+        if (!nr_stun_message_has_attribute(ctx->response, NR_STUN_ATTR_XOR_RELAY_ADDRESS, &attr)) {
+            r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): No XOR-RELAYED-ADDRESS",ctx->label);
+            ABORT(R_BAD_DATA);
+        }
+
+        if ((r=nr_stun_transport_addr_check(&attr->u.relay_address.unmasked,
+                                            ctx->mapped_addr_check_mask))) {
+            r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): nr_stun_transport_addr_check failed",ctx->label);
+            ABORT(r);
+        }
+
+        if ((r=nr_transport_addr_copy(
+                &ctx->results.allocate_response.relay_addr,
+                &attr->u.relay_address.unmasked))) {
+            r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): nr_transport_addr_copy failed",ctx->label);
+            ABORT(r);
+        }
+
+        if (!nr_stun_message_has_attribute(ctx->response, NR_STUN_ATTR_LIFETIME, &attr)) {
+            r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): No LIFETIME",ctx->label);
+            ABORT(R_BAD_DATA);
+        }
+        ctx->results.allocate_response.lifetime_secs=attr->u.lifetime_secs;
+
+        r_log(NR_LOG_STUN,LOG_DEBUG,"STUN-CLIENT(%s): Received relay address: %s", ctx->label, ctx->results.allocate_response.relay_addr.as_string);
+
+        mapped_addr = &ctx->results.allocate_response.mapped_addr;
+
+        break;
+    case NR_TURN_CLIENT_MODE_REFRESH_REQUEST:
+        if (! nr_stun_message_has_attribute(ctx->response, NR_STUN_ATTR_MESSAGE_INTEGRITY, 0)) {
+            r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): No MESSAGE-INTEGRITY",ctx->label);
+            ABORT(R_BAD_DATA);
+        }
+        if (!nr_stun_message_has_attribute(ctx->response, NR_STUN_ATTR_LIFETIME, &attr)) {
+            r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): No LIFETIME",ctx->label);
+            ABORT(R_BAD_DATA);
+        }
+        ctx->results.refresh_response.lifetime_secs=attr->u.lifetime_secs;
+        break;
+    case NR_TURN_CLIENT_MODE_PERMISSION_REQUEST:
+        if (! nr_stun_message_has_attribute(ctx->response, NR_STUN_ATTR_MESSAGE_INTEGRITY, 0)) {
+            r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): No MESSAGE-INTEGRITY",ctx->label);
+            ABORT(R_BAD_DATA);
+        }
+        break;
+#endif /* USE_TURN */
+
+    default:
+        assert(0);
+        ABORT(R_FAILED);
+        break;
+    }
+
+    /* make sure we have the most up-to-date address from this peer */
+    if (nr_transport_addr_cmp(&ctx->peer_addr, peer_addr, NR_TRANSPORT_ADDR_CMP_MODE_ALL)) {
+        r_log(NR_LOG_STUN,LOG_INFO,"STUN-CLIENT(%s): Peer moved from %s to %s", ctx->label, ctx->peer_addr.as_string, peer_addr->as_string);
+        nr_transport_addr_copy(&ctx->peer_addr, peer_addr);
+    }
+
+    if (mapped_addr) {
+        if (nr_stun_message_has_attribute(ctx->response, NR_STUN_ATTR_XOR_MAPPED_ADDRESS, &attr)) {
+            if ((r=nr_stun_transport_addr_check(&attr->u.xor_mapped_address.unmasked,
+                                                ctx->mapped_addr_check_mask))) {
+                r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): XOR-MAPPED-ADDRESS is bogus",ctx->label);
+                ABORT(r);
+            }
+
+            if ((r=nr_transport_addr_copy(mapped_addr, &attr->u.xor_mapped_address.unmasked))) {
+                r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): nr_transport_addr_copy failed",ctx->label);
+                ABORT(r);
+            }
+        }
+        else if (nr_stun_message_has_attribute(ctx->response, NR_STUN_ATTR_MAPPED_ADDRESS, &attr)) {
+            if ((r=nr_stun_transport_addr_check(&attr->u.mapped_address,
+                                                ctx->mapped_addr_check_mask))) {
+                r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): MAPPED-ADDRESS is bogus",ctx->label);
+                ABORT(r);
+            }
+
+            if ((r=nr_transport_addr_copy(mapped_addr, &attr->u.mapped_address))) {
+                r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): nr_transport_addr_copy failed",ctx->label);
+                ABORT(r);
+            }
+        }
+        else {
+            r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): No mapped address!",ctx->label);
+            ABORT(R_BAD_DATA);
+        }
+
+        // STUN doesn't distinguish protocol in mapped address, therefore
+        // assign used protocol from peer_addr
+        if (mapped_addr->protocol!=peer_addr->protocol){
+          mapped_addr->protocol=peer_addr->protocol;
+          nr_transport_addr_fmt_addr_string(mapped_addr);
+        }
+
+        r_log(NR_LOG_STUN,LOG_DEBUG,"STUN-CLIENT(%s): Received mapped address: %s", ctx->label, mapped_addr->as_string);
+    }
+
+    ctx->state=NR_STUN_CLIENT_STATE_DONE;
+
+    _status=0;
+  abort:
+    if(_status && response_matched){
+      r_log(NR_LOG_STUN,LOG_WARNING,"STUN-CLIENT(%s): Error processing response: %s, stun error code %d.", ctx->label, nr_strerror(_status), (int)ctx->error_code);
+    }
+
+    if ((ctx->state != NR_STUN_CLIENT_STATE_RUNNING) &&
+        (ctx->state != NR_STUN_CLIENT_STATE_WAITING)) {
+        /* Cancel the timer firing */
+        if (ctx->timer_handle) {
+            NR_async_timer_cancel(ctx->timer_handle);
+            ctx->timer_handle = 0;
+        }
+
+        nr_stun_client_fire_finished_cb(ctx);
+    }
+
+    return(_status);
+  }
+
+int nr_stun_client_ctx_destroy(nr_stun_client_ctx **ctxp)
+  {
+    nr_stun_client_ctx *ctx;
+
+    if(!ctxp || !*ctxp)
+      return(0);
+
+    ctx=*ctxp;
+    *ctxp=0;
+
+    nr_stun_client_reset(ctx);
+
+    RFREE(ctx->nonce);
+    RFREE(ctx->realm);
+
+    RFREE(ctx->label);
+    RFREE(ctx);
+
+    return(0);
+  }
+
+
+int nr_stun_client_cancel(nr_stun_client_ctx *ctx)
+  {
+    /* Cancel the timer firing */
+    if (ctx->timer_handle){
+      NR_async_timer_cancel(ctx->timer_handle);
+      ctx->timer_handle=0;
+    }
+
+    /* Mark cancelled so we ignore any returned messsages */
+    ctx->state=NR_STUN_CLIENT_STATE_CANCELLED;
+    return(0);
+}
+
+int nr_stun_client_wait(nr_stun_client_ctx *ctx)
+  {
+    nr_stun_client_cancel(ctx);
+    ctx->state=NR_STUN_CLIENT_STATE_WAITING;
+
+    ctx->request_ct = ctx->maximum_transmits;
+    ctx->timeout_ms = ctx->maximum_transmits_timeout_ms;
+    NR_ASYNC_TIMER_SET(ctx->timeout_ms, nr_stun_client_timer_expired_cb, ctx, &ctx->timer_handle);
+
+    return(0);
+  }
+
+int nr_stun_client_failed(nr_stun_client_ctx *ctx)
+  {
+    nr_stun_client_cancel(ctx);
+    ctx->state=NR_STUN_CLIENT_STATE_FAILED;
+    nr_stun_client_fire_finished_cb(ctx);
+    return(0);
+  }
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_client_ctx.h b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_client_ctx.h
new file mode 100644
index 0000000000..0cc9045e2a
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_client_ctx.h
@@ -0,0 +1,200 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _stun_client_ctx_h
+#define _stun_client_ctx_h
+
+/* forward declaration */
+typedef struct nr_stun_client_ctx_ nr_stun_client_ctx;
+
+#include "stun.h"
+
+/* Checklist for adding new STUN transaction types
+
+   1. Add new method type in stun.h (NR_METHOD_*)
+   2. Add new MSGs in stun.h (NR_STUN_MSG_*)
+   3. Add new messages to stun_util.c:nr_stun_msg_type
+   4. Add new request type to stun_build.h
+   4. Add new message builder to stun_build.c
+   5. Add new response type to stun_client_ctx.h
+   6. Add new arm to stun_client_ctx.c:nr_stun_client_send_request
+   7. Add new arms to nr_stun_client_process_response
+   8. Add new arms to stun_hint.c:nr_is_stun_message
+*/
+
+
+
+
+typedef union nr_stun_client_params_ {
+
+    nr_stun_client_stun_binding_request_params               stun_binding_request;
+    nr_stun_client_stun_keepalive_params                     stun_keepalive;
+#ifdef USE_STUND_0_96
+    nr_stun_client_stun_binding_request_stund_0_96_params    stun_binding_request_stund_0_96;
+#endif /* USE_STUND_0_96 */
+
+#ifdef USE_ICE
+    nr_stun_client_ice_binding_request_params                ice_binding_request;
+#endif /* USE_ICE */
+
+#ifdef USE_TURN
+    nr_stun_client_allocate_request_params                   allocate_request;
+    nr_stun_client_refresh_request_params                    refresh_request;
+    nr_stun_client_permission_request_params                 permission_request;
+    nr_stun_client_send_indication_params                    send_indication;
+#endif /* USE_TURN */
+
+} nr_stun_client_params;
+
+typedef struct nr_stun_client_stun_binding_response_results_ {
+    nr_transport_addr mapped_addr;
+} nr_stun_client_stun_binding_response_results;
+
+typedef struct nr_stun_client_stun_binding_response_stund_0_96_results_ {
+    nr_transport_addr mapped_addr;
+} nr_stun_client_stun_binding_response_stund_0_96_results;
+
+#ifdef USE_ICE
+typedef struct nr_stun_client_ice_use_candidate_results_ {
+#ifdef WIN32  // silly VC++ gives error if no members
+    int dummy;
+#endif
+} nr_stun_client_ice_use_candidate_results;
+
+typedef struct nr_stun_client_ice_binding_response_results_ {
+    nr_transport_addr mapped_addr;
+} nr_stun_client_ice_binding_response_results;
+#endif /* USE_ICE */
+
+#ifdef USE_TURN
+typedef struct nr_stun_client_allocate_response_results_ {
+    nr_transport_addr relay_addr;
+    nr_transport_addr mapped_addr;
+    UINT4 lifetime_secs;
+} nr_stun_client_allocate_response_results;
+
+typedef struct nr_stun_client_refresh_response_results_ {
+    UINT4 lifetime_secs;
+} nr_stun_client_refresh_response_results;
+
+typedef struct nr_stun_client_permission_response_results_ {
+    UINT4 lifetime_secs;
+} nr_stun_client_permission_response_results;
+
+#endif /* USE_TURN */
+
+typedef union nr_stun_client_results_ {
+    nr_stun_client_stun_binding_response_results              stun_binding_response;
+    nr_stun_client_stun_binding_response_stund_0_96_results   stun_binding_response_stund_0_96;
+
+#ifdef USE_ICE
+    nr_stun_client_ice_use_candidate_results                  ice_use_candidate;
+    nr_stun_client_ice_binding_response_results               ice_binding_response;
+#endif /* USE_ICE */
+
+#ifdef USE_TURN
+    nr_stun_client_allocate_response_results                   allocate_response;
+    nr_stun_client_refresh_response_results                    refresh_response;
+#endif /* USE_TURN */
+} nr_stun_client_results;
+
+struct nr_stun_client_ctx_ {
+  int state;
+#define NR_STUN_CLIENT_STATE_INITTED         0
+#define NR_STUN_CLIENT_STATE_RUNNING         1
+#define NR_STUN_CLIENT_STATE_DONE            2
+#define NR_STUN_CLIENT_STATE_FAILED          3
+#define NR_STUN_CLIENT_STATE_TIMED_OUT       4
+#define NR_STUN_CLIENT_STATE_CANCELLED       5
+#define NR_STUN_CLIENT_STATE_WAITING         6
+
+  int mode;
+#define NR_STUN_CLIENT_MODE_BINDING_REQUEST_SHORT_TERM_AUTH   1
+#define NR_STUN_CLIENT_MODE_BINDING_REQUEST_LONG_TERM_AUTH    2
+#define NR_STUN_CLIENT_MODE_BINDING_REQUEST_NO_AUTH           3
+#define NR_STUN_CLIENT_MODE_KEEPALIVE                         4
+#define NR_STUN_CLIENT_MODE_BINDING_REQUEST_STUND_0_96        5
+#ifdef USE_ICE
+#define NR_ICE_CLIENT_MODE_USE_CANDIDATE                  10
+#define NR_ICE_CLIENT_MODE_BINDING_REQUEST                11
+#endif /* USE_ICE */
+#ifdef USE_TURN
+#define NR_TURN_CLIENT_MODE_ALLOCATE_REQUEST              20
+#define NR_TURN_CLIENT_MODE_REFRESH_REQUEST               21
+#define NR_TURN_CLIENT_MODE_SEND_INDICATION               22
+#define NR_TURN_CLIENT_MODE_DATA_INDICATION               24
+#define NR_TURN_CLIENT_MODE_PERMISSION_REQUEST            25
+#endif /* USE_TURN */
+
+  char *label;
+  nr_transport_addr my_addr;
+  nr_transport_addr peer_addr;
+  nr_socket *sock;
+  nr_stun_client_auth_params auth_params;
+  nr_stun_client_params params;
+  nr_stun_client_results results;
+  char *nonce;
+  char *realm;
+  void *timer_handle;
+  UINT2 request_ct;
+  UINT2 retransmit_ct;
+  UINT4 rto_ms;    /* retransmission time out */
+  double retransmission_backoff_factor;
+  UINT4 maximum_transmits;
+  UINT4 maximum_transmits_timeout_ms;
+  UINT4 mapped_addr_check_mask;  /* What checks to run on mapped addresses */
+  int timeout_ms;
+  struct timeval timer_set;
+  NR_async_cb finished_cb;
+  void *cb_arg;
+  nr_stun_message *request;
+  nr_stun_message *response;
+  UINT2 error_code;
+};
+
+int nr_stun_client_ctx_create(char *label, nr_socket *sock, nr_transport_addr *peer, UINT4 RTO, nr_stun_client_ctx **ctxp);
+int nr_stun_client_start(nr_stun_client_ctx *ctx, int mode, NR_async_cb finished_cb, void *cb_arg);
+int nr_stun_client_restart(nr_stun_client_ctx* ctx,
+                           const nr_transport_addr* peer_addr);
+int nr_stun_client_force_retransmit(nr_stun_client_ctx *ctx);
+int nr_stun_client_reset(nr_stun_client_ctx *ctx);
+int nr_stun_client_ctx_destroy(nr_stun_client_ctx **ctxp);
+int nr_stun_transport_addr_check(nr_transport_addr* addr, UINT4 mask);
+int nr_stun_client_process_response(nr_stun_client_ctx *ctx, UCHAR *msg, int len, nr_transport_addr *peer_addr);
+int nr_stun_client_cancel(nr_stun_client_ctx *ctx);
+int nr_stun_client_wait(nr_stun_client_ctx *ctx);
+int nr_stun_client_failed(nr_stun_client_ctx *ctx);
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_codec.c b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_codec.c
new file mode 100644
index 0000000000..ae748a667b
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_codec.c
@@ -0,0 +1,1550 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <errno.h>
+#include <csi_platform.h>
+
+#ifdef WIN32
+#include <winsock2.h>
+#include <stdlib.h>
+#include <io.h>
+#include <time.h>
+#else   /* UNIX */
+#include <string.h>
+#endif  /* end UNIX */
+#include <assert.h>
+#include <stddef.h>
+
+#include "nr_api.h"
+#include "stun.h"
+#include "byteorder.h"
+#include "r_crc32.h"
+#include "nr_crypto.h"
+
+#define NR_STUN_IPV4_FAMILY  0x01
+#define NR_STUN_IPV6_FAMILY  0x02
+
+#define SKIP_ATTRIBUTE_DECODE -1
+
+static int nr_stun_find_attr_info(UINT2 type, nr_stun_attr_info **info);
+
+static int nr_stun_fix_attribute_ordering(nr_stun_message *msg);
+
+static int nr_stun_encode_htons(UINT2 data, size_t buflen, UCHAR *buf, size_t *offset);
+static int nr_stun_encode_htonl(UINT4 data, size_t buflen, UCHAR *buf, size_t *offset);
+static int nr_stun_encode_htonll(UINT8 data, size_t buflen, UCHAR *buf, size_t *offset);
+static int nr_stun_encode(UCHAR *data, size_t length, size_t buflen, UCHAR *buf, size_t *offset);
+
+static int nr_stun_decode_htons(UCHAR *buf, size_t buflen, size_t *offset, UINT2 *data);
+static int nr_stun_decode_htonl(UCHAR *buf, size_t buflen, size_t *offset, UINT4 *data);
+static int nr_stun_decode_htonll(UCHAR *buf, size_t buflen, size_t *offset, UINT8 *data);
+static int nr_stun_decode(size_t length, UCHAR *buf, size_t buflen, size_t *offset, UCHAR *data);
+
+static int nr_stun_attr_string_illegal(nr_stun_attr_info *attr_info, size_t len, void *data, size_t max_bytes, size_t max_chars);
+
+static int nr_stun_attr_error_code_illegal(nr_stun_attr_info *attr_info, size_t attrlen, void *data);
+static int nr_stun_attr_nonce_illegal(nr_stun_attr_info *attr_info, size_t attrlen, void *data);
+static int nr_stun_attr_realm_illegal(nr_stun_attr_info *attr_info, size_t attrlen, void *data);
+static int nr_stun_attr_server_illegal(nr_stun_attr_info *attr_info, size_t attrlen, void *data);
+static int nr_stun_attr_username_illegal(nr_stun_attr_info *attr_info, size_t attrlen, void *data);
+static int
+nr_stun_attr_codec_fingerprint_decode(nr_stun_attr_info *attr_info, size_t attrlen, UCHAR *buf, size_t offset, size_t buflen, void *data);
+
+
+int
+nr_stun_encode_htons(UINT2 data, size_t buflen, UCHAR *buf, size_t *offset)
+{
+   UINT2 d = htons(data);
+
+   if (*offset + sizeof(d) >= buflen) {
+      r_log(NR_LOG_STUN, LOG_WARNING, "Attempted buffer overrun: %d + %zd >= %d", *offset, sizeof(d), buflen);
+      return R_BAD_DATA;
+   }
+
+   memcpy(&buf[*offset], &d, sizeof(d));
+   *offset += sizeof(d);
+
+   return 0;
+}
+
+int
+nr_stun_encode_htonl(UINT4 data, size_t buflen, UCHAR *buf, size_t *offset)
+{
+   UINT4 d = htonl(data);
+
+   if (*offset + sizeof(d) > buflen) {
+      r_log(NR_LOG_STUN, LOG_WARNING, "Attempted buffer overrun: %d + %zd > %d", *offset, sizeof(d), buflen);
+      return R_BAD_DATA;
+   }
+
+   memcpy(&buf[*offset], &d, sizeof(d));
+   *offset += sizeof(d);
+
+   return 0;
+}
+
+int
+nr_stun_encode_htonll(UINT8 data, size_t buflen, UCHAR *buf, size_t *offset)
+{
+   UINT8 d = nr_htonll(data);
+
+   if (*offset + sizeof(d) > buflen) {
+      r_log(NR_LOG_STUN, LOG_WARNING, "Attempted buffer overrun: %d + %zd > %d", *offset, sizeof(d), buflen);
+      return R_BAD_DATA;
+   }
+
+   memcpy(&buf[*offset], &d, sizeof(d));
+   *offset += sizeof(d);
+
+   return 0;
+}
+
+int
+nr_stun_encode(UCHAR *data, size_t length, size_t buflen, UCHAR *buf, size_t *offset)
+{
+   if (*offset + length > buflen) {
+      r_log(NR_LOG_STUN, LOG_WARNING, "Attempted buffer overrun: %d + %d > %d", *offset, length, buflen);
+      return R_BAD_DATA;
+   }
+
+   memcpy(&buf[*offset], data, length);
+   *offset += length;
+
+   return 0;
+}
+
+
+int
+nr_stun_decode_htons(UCHAR *buf, size_t buflen, size_t *offset, UINT2 *data)
+{
+   UINT2 d;
+
+   if (*offset + sizeof(d) > buflen) {
+      r_log(NR_LOG_STUN, LOG_WARNING, "Attempted buffer overrun: %d + %zd > %d", *offset, sizeof(d), buflen);
+      return R_BAD_DATA;
+   }
+
+   memcpy(&d, &buf[*offset], sizeof(d));
+   *offset += sizeof(d);
+   *data = htons(d);
+
+   return 0;
+}
+
+int
+nr_stun_decode_htonl(UCHAR *buf, size_t buflen, size_t *offset, UINT4 *data)
+{
+   UINT4 d;
+
+   if (*offset + sizeof(d) > buflen) {
+      r_log(NR_LOG_STUN, LOG_WARNING, "Attempted buffer overrun: %d + %zd > %d", *offset, sizeof(d), buflen);
+      return R_BAD_DATA;
+   }
+
+   memcpy(&d, &buf[*offset], sizeof(d));
+   *offset += sizeof(d);
+   *data = htonl(d);
+
+   return 0;
+}
+
+int
+nr_stun_decode_htonll(UCHAR *buf, size_t buflen, size_t *offset, UINT8 *data)
+{
+   UINT8 d;
+
+   if (*offset + sizeof(d) > buflen) {
+      r_log(NR_LOG_STUN, LOG_WARNING, "Attempted buffer overrun: %d + %zd > %d", *offset, sizeof(d), buflen);
+      return R_BAD_DATA;
+   }
+
+   memcpy(&d, &buf[*offset], sizeof(d));
+   *offset += sizeof(d);
+   *data = nr_htonll(d);
+
+   return 0;
+}
+
+int
+nr_stun_decode(size_t length, UCHAR *buf, size_t buflen, size_t *offset, UCHAR *data)
+{
+   if (*offset + length > buflen) {
+      r_log(NR_LOG_STUN, LOG_WARNING, "Attempted buffer overrun: %d + %d > %d", *offset, length, buflen);
+      return R_BAD_DATA;
+   }
+
+   memcpy(data, &buf[*offset], length);
+   *offset += length;
+
+   return 0;
+}
+
+/**
+ * The argument must be a non-null pointer to a zero-terminated string.
+ *
+ * If the argument is valid UTF-8, returns the number of code points in the
+ * string excluding the zero-terminator.
+ *
+ * If the argument is invalid UTF-8, returns a lower bound for the number of
+ * code points in the string. (If UTF-8 error handling was performed on the
+ * string, new REPLACEMENT CHARACTER code points could be introduced in
+ * a way that would increase the total number of code points compared to
+ * what this function counts.)
+ */
+size_t
+nr_count_utf8_code_points_without_validation(const char *s) {
+    size_t nchars = 0;
+    char c;
+    while ((c = *s)) {
+        if ((c & 0xC0) != 0x80) {
+            ++nchars;
+        }
+        ++s;
+    }
+    return nchars;
+}
+
+int
+nr_stun_attr_string_illegal(nr_stun_attr_info *attr_info, size_t len, void *data, size_t max_bytes, size_t max_chars)
+{
+    int _status;
+    char *s = data;
+    size_t nchars;
+
+    if (len > max_bytes) {
+        r_log(NR_LOG_STUN, LOG_WARNING, "%s is too large: %d bytes", attr_info->name, len);
+        ABORT(R_FAILED);
+    }
+
+    nchars = nr_count_utf8_code_points_without_validation(s);
+    if (nchars > max_chars) {
+        r_log(NR_LOG_STUN, LOG_WARNING, "%s is too large: %zd characters", attr_info->name, nchars);
+        ABORT(R_FAILED);
+    }
+
+    _status = 0;
+  abort:
+    return _status;
+}
+
+int
+nr_stun_attr_error_code_illegal(nr_stun_attr_info *attr_info, size_t attrlen, void *data)
+{
+    int r,_status;
+    nr_stun_attr_error_code *ec = data;
+
+    if (ec->number < 300 || ec->number > 699)
+        ABORT(R_FAILED);
+
+    if ((r=nr_stun_attr_string_illegal(attr_info, strlen(ec->reason), ec->reason, NR_STUN_MAX_ERROR_CODE_REASON_BYTES, NR_STUN_MAX_ERROR_CODE_REASON_CHARS)))
+        ABORT(r);
+
+    _status = 0;
+  abort:
+    return _status;
+}
+
+int
+nr_stun_attr_nonce_illegal(nr_stun_attr_info *attr_info, size_t attrlen, void *data)
+{
+    return nr_stun_attr_string_illegal(attr_info, attrlen, data, NR_STUN_MAX_NONCE_BYTES, NR_STUN_MAX_NONCE_CHARS);
+}
+
+int
+nr_stun_attr_realm_illegal(nr_stun_attr_info *attr_info, size_t attrlen, void *data)
+{
+    return nr_stun_attr_string_illegal(attr_info, attrlen, data, NR_STUN_MAX_REALM_BYTES, NR_STUN_MAX_REALM_CHARS);
+}
+
+int
+nr_stun_attr_server_illegal(nr_stun_attr_info *attr_info, size_t attrlen, void *data)
+{
+    return nr_stun_attr_string_illegal(attr_info, attrlen, data, NR_STUN_MAX_SERVER_BYTES, NR_STUN_MAX_SERVER_CHARS);
+}
+
+int
+nr_stun_attr_username_illegal(nr_stun_attr_info *attr_info, size_t attrlen, void *data)
+{
+    return nr_stun_attr_string_illegal(attr_info, attrlen, data, NR_STUN_MAX_USERNAME_BYTES, -1);
+}
+
+static int
+nr_stun_attr_codec_UCHAR_print(nr_stun_attr_info *attr_info, char *msg, void *data)
+{
+    r_log(NR_LOG_STUN, LOG_DEBUG, "%s %s: %u", msg, attr_info->name, *(UCHAR*)data);
+    return 0;
+}
+
+static int
+nr_stun_attr_codec_UCHAR_encode(nr_stun_attr_info *attr_info, void *data, size_t offset, size_t buflen, UCHAR *buf, size_t *attrlen)
+{
+    int start = offset;
+    UINT4 tmp = *((UCHAR *)data);
+    tmp <<= 24;
+
+    if (nr_stun_encode_htons(attr_info->type    , buflen, buf, &offset)
+     || nr_stun_encode_htons(sizeof(UINT4)      , buflen, buf, &offset)
+     || nr_stun_encode_htonl(tmp                , buflen, buf, &offset))
+        return R_FAILED;
+
+    *attrlen = offset - start;
+
+    return 0;
+}
+
+static int
+nr_stun_attr_codec_UCHAR_decode(nr_stun_attr_info *attr_info, size_t attrlen, UCHAR *buf, size_t offset, size_t buflen, void *data)
+{
+    UINT4 tmp;
+
+    if (attrlen != sizeof(UINT4)) {
+        r_log(NR_LOG_STUN, LOG_WARNING, "Integer is illegal size: %d", attrlen);
+        return R_FAILED;
+    }
+
+    if (nr_stun_decode_htonl(buf, buflen, &offset, &tmp))
+        return R_FAILED;
+
+    *((UCHAR *)data) = (tmp >> 24) & 0xff;
+
+    return 0;
+}
+
+nr_stun_attr_codec nr_stun_attr_codec_UCHAR = {
+    "UCHAR",
+    nr_stun_attr_codec_UCHAR_print,
+    nr_stun_attr_codec_UCHAR_encode,
+    nr_stun_attr_codec_UCHAR_decode
+};
+
+static int
+nr_stun_attr_codec_UINT4_print(nr_stun_attr_info *attr_info, char *msg, void *data)
+{
+    r_log(NR_LOG_STUN, LOG_DEBUG, "%s %s: %u", msg, attr_info->name, *(UINT4*)data);
+    return 0;
+}
+
+static int
+nr_stun_attr_codec_UINT4_encode(nr_stun_attr_info *attr_info, void *data, size_t offset, size_t buflen, UCHAR *buf, size_t *attrlen)
+{
+    int start = offset;
+
+    if (nr_stun_encode_htons(attr_info->type    , buflen, buf, &offset)
+     || nr_stun_encode_htons(sizeof(UINT4)      , buflen, buf, &offset)
+     || nr_stun_encode_htonl(*(UINT4*)data      , buflen, buf, &offset))
+        return R_FAILED;
+
+    *attrlen = offset - start;
+
+    return 0;
+}
+
+static int
+nr_stun_attr_codec_UINT4_decode(nr_stun_attr_info *attr_info, size_t attrlen, UCHAR *buf, size_t offset, size_t buflen, void *data)
+{
+    if (attrlen != sizeof(UINT4)) {
+        r_log(NR_LOG_STUN, LOG_WARNING, "Integer is illegal size: %d", attrlen);
+        return R_FAILED;
+    }
+
+    if (nr_stun_decode_htonl(buf, buflen, &offset, (UINT4*)data))
+        return R_FAILED;
+
+    return 0;
+}
+
+nr_stun_attr_codec nr_stun_attr_codec_UINT4 = {
+    "UINT4",
+    nr_stun_attr_codec_UINT4_print,
+    nr_stun_attr_codec_UINT4_encode,
+    nr_stun_attr_codec_UINT4_decode
+};
+
+static int
+nr_stun_attr_codec_UINT8_print(nr_stun_attr_info *attr_info, char *msg, void *data)
+{
+    r_log(NR_LOG_STUN, LOG_DEBUG, "%s %s: %llu", msg, attr_info->name, *(UINT8*)data);
+    return 0;
+}
+
+static int
+nr_stun_attr_codec_UINT8_encode(nr_stun_attr_info *attr_info, void *data, size_t offset, size_t buflen, UCHAR *buf, size_t *attrlen)
+{
+    int start = offset;
+
+    if (nr_stun_encode_htons(attr_info->type   , buflen, buf, &offset)
+     || nr_stun_encode_htons(sizeof(UINT8)     , buflen, buf, &offset)
+     || nr_stun_encode_htonll(*(UINT8*)data    , buflen, buf, &offset))
+        return R_FAILED;
+
+    *attrlen = offset - start;
+
+    return 0;
+}
+
+static int
+nr_stun_attr_codec_UINT8_decode(nr_stun_attr_info *attr_info, size_t attrlen, UCHAR *buf, size_t offset, size_t buflen, void *data)
+{
+    if (attrlen != sizeof(UINT8)) {
+        r_log(NR_LOG_STUN, LOG_WARNING, "Integer is illegal size: %d", attrlen);
+        return R_FAILED;
+    }
+
+    if (nr_stun_decode_htonll(buf, buflen, &offset, (UINT8*)data))
+        return R_FAILED;
+
+    return 0;
+}
+
+nr_stun_attr_codec nr_stun_attr_codec_UINT8 = {
+    "UINT8",
+    nr_stun_attr_codec_UINT8_print,
+    nr_stun_attr_codec_UINT8_encode,
+    nr_stun_attr_codec_UINT8_decode
+};
+
+static int
+nr_stun_attr_codec_addr_print(nr_stun_attr_info *attr_info, char *msg, void *data)
+{
+    r_log(NR_LOG_STUN, LOG_DEBUG, "%s %s: %s", msg, attr_info->name, ((nr_transport_addr*)data)->as_string);
+    return 0;
+}
+
+static int
+nr_stun_attr_codec_addr_encode(nr_stun_attr_info *attr_info, void *data, size_t offset, size_t buflen, UCHAR *buf, size_t *attrlen)
+{
+    int r,_status;
+    int start = offset;
+    nr_transport_addr *addr = data;
+    UCHAR pad = '\0';
+    UCHAR family;
+
+    if ((r=nr_stun_encode_htons(attr_info->type, buflen, buf, &offset)))
+        ABORT(r);
+
+    switch (addr->ip_version) {
+    case NR_IPV4:
+        family = NR_STUN_IPV4_FAMILY;
+        if (nr_stun_encode_htons(8               , buflen, buf, &offset)
+         || nr_stun_encode(&pad, 1               , buflen, buf, &offset)
+         || nr_stun_encode(&family, 1            , buflen, buf, &offset)
+         || nr_stun_encode_htons(ntohs(addr->u.addr4.sin_port), buflen, buf, &offset)
+         || nr_stun_encode_htonl(ntohl(addr->u.addr4.sin_addr.s_addr), buflen, buf, &offset))
+            ABORT(R_FAILED);
+        break;
+
+    case NR_IPV6:
+        family = NR_STUN_IPV6_FAMILY;
+        if (nr_stun_encode_htons(20              , buflen, buf, &offset)
+         || nr_stun_encode(&pad, 1               , buflen, buf, &offset)
+         || nr_stun_encode(&family, 1            , buflen, buf, &offset)
+         || nr_stun_encode_htons(ntohs(addr->u.addr6.sin6_port), buflen, buf, &offset)
+         || nr_stun_encode(addr->u.addr6.sin6_addr.s6_addr, 16, buflen, buf, &offset))
+            ABORT(R_FAILED);
+        break;
+
+    default:
+        assert(0);
+        ABORT(R_INTERNAL);
+        break;
+    }
+
+    *attrlen = offset - start;
+
+    _status = 0;
+  abort:
+    return _status;
+}
+
+static int
+nr_stun_attr_codec_addr_decode(nr_stun_attr_info *attr_info, size_t attrlen, UCHAR *buf, size_t offset, size_t buflen, void *data)
+{
+    int _status;
+    UCHAR pad;
+    UCHAR family;
+    UINT2 port;
+    UINT4 addr4;
+    struct in6_addr addr6;
+    nr_transport_addr *result = data;
+
+    if (nr_stun_decode(1, buf, buflen, &offset, &pad)
+     || nr_stun_decode(1, buf, buflen, &offset, &family))
+        ABORT(R_FAILED);
+
+    switch (family) {
+    case NR_STUN_IPV4_FAMILY:
+        if (attrlen != 8) {
+            r_log(NR_LOG_STUN, LOG_WARNING, "Illegal attribute length: %d", attrlen);
+            ABORT(R_FAILED);
+        }
+
+        if (nr_stun_decode_htons(buf, buflen, &offset, &port)
+         || nr_stun_decode_htonl(buf, buflen, &offset, &addr4))
+            ABORT(R_FAILED);
+
+        if (nr_ip4_port_to_transport_addr(addr4, port, IPPROTO_UDP, result))
+            ABORT(R_FAILED);
+        break;
+
+    case NR_STUN_IPV6_FAMILY:
+        if (attrlen != 20) {
+            r_log(NR_LOG_STUN, LOG_WARNING, "Illegal attribute length: %d", attrlen);
+            ABORT(R_FAILED);
+        }
+
+        if (nr_stun_decode_htons(buf, buflen, &offset, &port)
+         || nr_stun_decode(16, buf, buflen, &offset, addr6.s6_addr))
+            ABORT(R_FAILED);
+
+        if (nr_ip6_port_to_transport_addr(&addr6, port, IPPROTO_UDP, result))
+            ABORT(R_FAILED);
+        break;
+
+    default:
+        r_log(NR_LOG_STUN, LOG_WARNING, "Illegal address family: %d", family);
+        ABORT(R_FAILED);
+        break;
+    }
+
+    _status = 0;
+  abort:
+    return _status;
+}
+
+nr_stun_attr_codec nr_stun_attr_codec_addr = {
+    "addr",
+    nr_stun_attr_codec_addr_print,
+    nr_stun_attr_codec_addr_encode,
+    nr_stun_attr_codec_addr_decode
+};
+
+static int
+nr_stun_attr_codec_data_print(nr_stun_attr_info *attr_info, char *msg, void *data)
+{
+    nr_stun_attr_data *d = data;
+    r_dump(NR_LOG_STUN, LOG_DEBUG, attr_info->name, (char*)d->data, d->length);
+    return 0;
+}
+
+static int
+nr_stun_attr_codec_data_encode(nr_stun_attr_info *attr_info, void *data, size_t offset, size_t buflen, UCHAR *buf, size_t *attrlen)
+{
+    nr_stun_attr_data *d = data;
+    int start = offset;
+
+    if (nr_stun_encode_htons(attr_info->type     , buflen, buf, &offset)
+     || nr_stun_encode_htons(d->length           , buflen, buf, &offset)
+     || nr_stun_encode(d->data, d->length        , buflen, buf, &offset))
+        return R_FAILED;
+
+    *attrlen = offset - start;
+
+    return 0;
+}
+
+static int
+nr_stun_attr_codec_data_decode(nr_stun_attr_info *attr_info, size_t attrlen, UCHAR *buf, size_t offset, size_t buflen, void *data)
+{
+    int _status;
+    nr_stun_attr_data *result = data;
+
+    /* -1 because it is going to be null terminated just to be safe */
+    if (attrlen >= (sizeof(result->data) - 1)) {
+        r_log(NR_LOG_STUN, LOG_WARNING, "Too much data: %d bytes", attrlen);
+        ABORT(R_FAILED);
+    }
+
+    if (nr_stun_decode(attrlen, buf, buflen, &offset, result->data))
+        ABORT(R_FAILED);
+
+    result->length = attrlen;
+    result->data[attrlen] = '\0'; /* just to be nice */
+
+    _status=0;
+  abort:
+    return _status;
+}
+
+nr_stun_attr_codec nr_stun_attr_codec_data = {
+    "data",
+    nr_stun_attr_codec_data_print,
+    nr_stun_attr_codec_data_encode,
+    nr_stun_attr_codec_data_decode
+};
+
+static int
+nr_stun_attr_codec_error_code_print(nr_stun_attr_info *attr_info, char *msg, void *data)
+{
+    nr_stun_attr_error_code *error_code = data;
+    r_log(NR_LOG_STUN, LOG_DEBUG, "%s %s: %d %s",
+                          msg, attr_info->name, error_code->number,
+                          error_code->reason);
+    return 0;
+}
+
+static int
+nr_stun_attr_codec_error_code_encode(nr_stun_attr_info *attr_info, void *data, size_t offset, size_t buflen, UCHAR *buf, size_t *attrlen)
+{
+    nr_stun_attr_error_code *error_code = data;
+    int start = offset;
+    int length = strlen(error_code->reason);
+    UCHAR pad[2] = { 0 };
+    UCHAR class = error_code->number / 100;
+    UCHAR number = error_code->number % 100;
+
+    if (nr_stun_encode_htons(attr_info->type   , buflen, buf, &offset)
+     || nr_stun_encode_htons(4 + length        , buflen, buf, &offset)
+     || nr_stun_encode(pad, 2                  , buflen, buf, &offset)
+     || nr_stun_encode(&class, 1               , buflen, buf, &offset)
+     || nr_stun_encode(&number, 1              , buflen, buf, &offset)
+     || nr_stun_encode((UCHAR*)error_code->reason, length, buflen, buf, &offset))
+        return R_FAILED;
+
+    *attrlen = offset - start;
+
+    return 0;
+}
+
+static int
+nr_stun_attr_codec_error_code_decode(nr_stun_attr_info *attr_info, size_t attrlen, UCHAR *buf, size_t offset, size_t buflen, void *data)
+{
+    int _status;
+    nr_stun_attr_error_code *result = data;
+    UCHAR pad[2];
+    UCHAR class;
+    UCHAR number;
+    size_t size_reason;
+
+    if (nr_stun_decode(2, buf, buflen, &offset, pad)
+     || nr_stun_decode(1, buf, buflen, &offset, &class)
+     || nr_stun_decode(1, buf, buflen, &offset, &number))
+        ABORT(R_FAILED);
+
+    result->number = (class * 100) + number;
+
+    size_reason = attrlen - 4;
+
+    /* -1 because the string will be null terminated */
+    if (size_reason > (sizeof(result->reason) - 1)) {
+        r_log(NR_LOG_STUN, LOG_WARNING, "Reason is too large, truncating");
+        /* don't fail, but instead truncate the reason */
+        size_reason = sizeof(result->reason) - 1;
+    }
+
+    if (nr_stun_decode(size_reason, buf, buflen, &offset, (UCHAR*)result->reason))
+        ABORT(R_FAILED);
+    result->reason[size_reason] = '\0';
+
+    _status=0;
+  abort:
+    return _status;
+}
+
+nr_stun_attr_codec nr_stun_attr_codec_error_code = {
+    "error_code",
+    nr_stun_attr_codec_error_code_print,
+    nr_stun_attr_codec_error_code_encode,
+    nr_stun_attr_codec_error_code_decode
+};
+
+static int
+nr_stun_attr_codec_fingerprint_print(nr_stun_attr_info *attr_info, char *msg, void *data)
+{
+    nr_stun_attr_fingerprint *fingerprint = data;
+    r_log(NR_LOG_STUN, LOG_DEBUG, "%s %s: %08x", msg, attr_info->name, fingerprint->checksum);
+    return 0;
+}
+
+static int
+nr_stun_attr_codec_fingerprint_encode(nr_stun_attr_info *attr_info, void *data, size_t offset, size_t buflen, UCHAR *buf, size_t *attrlen)
+{
+    UINT4 checksum;
+    nr_stun_attr_fingerprint *fingerprint = data;
+    nr_stun_message_header *header = (nr_stun_message_header*)buf;
+
+    /* the length must include the FINGERPRINT attribute when computing
+     * the fingerprint */
+    header->length = ntohs(header->length);
+    header->length += 8;  /* Fingerprint */
+    header->length = htons(header->length);
+
+    if (r_crc32((char*)buf, offset, &checksum)) {
+        r_log(NR_LOG_STUN, LOG_WARNING, "Unable to compute fingerprint");
+        return R_FAILED;
+    }
+
+    fingerprint->checksum = checksum ^ 0x5354554e;
+
+    r_log(NR_LOG_STUN, LOG_DEBUG, "Computed FINGERPRINT %08x", fingerprint->checksum);
+
+    fingerprint->valid = 1;
+    return nr_stun_attr_codec_UINT4.encode(attr_info, &fingerprint->checksum, offset, buflen, buf, attrlen);
+}
+
+static int
+nr_stun_attr_codec_fingerprint_decode(nr_stun_attr_info *attr_info, size_t attrlen, UCHAR *buf, size_t offset, size_t buflen, void *data)
+{
+    int r,_status;
+    nr_stun_attr_fingerprint *fingerprint = data;
+    nr_stun_message_header *header = (nr_stun_message_header*)buf;
+    size_t length;
+    UINT4 checksum;
+
+    if ((r=nr_stun_attr_codec_UINT4.decode(attr_info, attrlen, buf, offset, buflen, &fingerprint->checksum)))
+        ABORT(r);
+
+    offset -= 4; /* rewind to before the length and type fields */
+
+    /* the length must include the FINGERPRINT attribute when computing
+     * the fingerprint */
+    length = offset;  /* right before FINGERPRINT */
+    length -= sizeof(*header); /* remove header length */
+    length += 8;  /* add length of Fingerprint */
+    header->length = htons(length);
+
+    /* make sure FINGERPRINT is final attribute in message */
+    if (length + sizeof(*header) != buflen) {
+        r_log(NR_LOG_STUN, LOG_WARNING, "Fingerprint is not final attribute in message");
+        ABORT(R_FAILED);
+    }
+
+    if (r_crc32((char*)buf, offset, &checksum)) {
+        r_log(NR_LOG_STUN, LOG_WARNING, "Unable to compute fingerprint");
+        ABORT(R_FAILED);
+    }
+
+    fingerprint->valid = (fingerprint->checksum == (checksum ^ 0x5354554e));
+
+    r_log(NR_LOG_STUN, LOG_DEBUG, "Computed FINGERPRINT %08x", (checksum ^ 0x5354554e));
+    if (! fingerprint->valid)
+        r_log(NR_LOG_STUN, LOG_WARNING, "Invalid FINGERPRINT %08x", fingerprint->checksum);
+
+    _status=0;
+  abort:
+    return _status;
+}
+
+nr_stun_attr_codec nr_stun_attr_codec_fingerprint = {
+    "fingerprint",
+    nr_stun_attr_codec_fingerprint_print,
+    nr_stun_attr_codec_fingerprint_encode,
+    nr_stun_attr_codec_fingerprint_decode
+};
+
+static int
+nr_stun_attr_codec_flag_print(nr_stun_attr_info *attr_info, char *msg, void *data)
+{
+    r_log(NR_LOG_STUN, LOG_DEBUG, "%s %s: on", msg, attr_info->name);
+    return 0;
+}
+
+static int
+nr_stun_attr_codec_flag_encode(nr_stun_attr_info *attr_info, void *data, size_t offset, size_t buflen, UCHAR *buf, size_t *attrlen)
+{
+    int start = offset;
+
+    if (nr_stun_encode_htons(attr_info->type  , buflen, buf, &offset)
+     || nr_stun_encode_htons(0                , buflen, buf, &offset))
+        return R_FAILED;
+
+    *attrlen = offset - start;
+
+    return 0;
+}
+
+static int
+nr_stun_attr_codec_flag_decode(nr_stun_attr_info *attr_info, size_t attrlen, UCHAR *buf, size_t offset, size_t buflen, void *data)
+{
+    if (attrlen != 0) {
+        r_log(NR_LOG_STUN, LOG_WARNING, "Illegal flag length: %d", attrlen);
+        return R_FAILED;
+    }
+
+    return 0;
+}
+
+nr_stun_attr_codec nr_stun_attr_codec_flag = {
+    "flag",
+    nr_stun_attr_codec_flag_print,
+    nr_stun_attr_codec_flag_encode,
+    nr_stun_attr_codec_flag_decode
+};
+
+static int
+nr_stun_attr_codec_message_integrity_print(nr_stun_attr_info *attr_info, char *msg, void *data)
+{
+    nr_stun_attr_message_integrity *integrity = data;
+    r_dump(NR_LOG_STUN, LOG_DEBUG, attr_info->name, (char*)integrity->hash, sizeof(integrity->hash));
+    return 0;
+}
+
+static int
+nr_stun_compute_message_integrity(UCHAR *buf, int offset, UCHAR *password, int passwordlen, UCHAR *computedHMAC)
+{
+    int r,_status;
+    UINT2 hold;
+    UINT2 length;
+    nr_stun_message_header *header;
+
+    r_log(NR_LOG_STUN, LOG_DEBUG, "Computing MESSAGE-INTEGRITY");
+
+    header = (nr_stun_message_header*)buf;
+    hold = header->length;
+
+    /* adjust the length of the message */
+    length = offset;
+    length -= sizeof(*header);
+    length += 24; /* for MESSAGE-INTEGRITY attribute */
+    header->length = htons(length);
+
+    if ((r=nr_crypto_hmac_sha1((UCHAR*)password, passwordlen,
+                               buf, offset, computedHMAC)))
+        ABORT(r);
+
+    r_dump(NR_LOG_STUN, LOG_DEBUG, "Computed MESSAGE-INTEGRITY ", (char*)computedHMAC, 20);
+
+    _status=0;
+ abort:
+    header->length = hold;
+    return _status;
+}
+
+static int
+nr_stun_attr_codec_message_integrity_encode(nr_stun_attr_info *attr_info, void *data, size_t offset, size_t buflen, UCHAR *buf, size_t *attrlen)
+{
+    int start = offset;
+    nr_stun_attr_message_integrity *integrity = data;
+
+    if (nr_stun_compute_message_integrity(buf, offset, integrity->password, integrity->passwordlen, integrity->hash))
+        return R_FAILED;
+
+    if (nr_stun_encode_htons(attr_info->type                         , buflen, buf, &offset)
+     || nr_stun_encode_htons(sizeof(integrity->hash)                 , buflen, buf, &offset)
+     || nr_stun_encode(integrity->hash, sizeof(integrity->hash)      , buflen, buf, &offset))
+        return R_FAILED;
+
+    *attrlen = offset - start;
+
+    return 0;
+}
+
+static int
+nr_stun_attr_codec_message_integrity_decode(nr_stun_attr_info *attr_info, size_t attrlen, UCHAR *buf, size_t offset, size_t buflen, void *data)
+{
+    int _status;
+    int start;
+    nr_stun_attr_message_integrity *result = data;
+    UCHAR computedHMAC[20];
+
+    result->valid = 0;
+
+    if (attrlen != 20) {
+        r_log(NR_LOG_STUN, LOG_WARNING, "%s must be 20 bytes, not %d", attr_info->name, attrlen);
+        ABORT(R_FAILED);
+    }
+
+    start = offset - 4; /* rewind to before the length and type fields */
+    if (start < 0)
+        ABORT(R_INTERNAL);
+
+    if (nr_stun_decode(attrlen, buf, buflen, &offset, result->hash))
+        ABORT(R_FAILED);
+
+    if (result->unknown_user) {
+        result->valid = 0;
+    }
+    else {
+        if (nr_stun_compute_message_integrity(buf, start, result->password, result->passwordlen, computedHMAC))
+            ABORT(R_FAILED);
+
+        assert(sizeof(computedHMAC) == sizeof(result->hash));
+
+        result->valid = (memcmp(computedHMAC, result->hash, 20) == 0);
+    }
+
+   _status=0;
+ abort:
+   return _status;
+}
+
+nr_stun_attr_codec nr_stun_attr_codec_message_integrity = {
+    "message_integrity",
+    nr_stun_attr_codec_message_integrity_print,
+    nr_stun_attr_codec_message_integrity_encode,
+    nr_stun_attr_codec_message_integrity_decode
+};
+
+static int
+nr_stun_attr_codec_noop_decode(nr_stun_attr_info *attr_info, size_t attrlen, UCHAR *buf, size_t offset, size_t buflen, void *data)
+{
+    return SKIP_ATTRIBUTE_DECODE;
+}
+
+nr_stun_attr_codec nr_stun_attr_codec_noop = {
+    "NOOP",
+    0,  /* ignore, never print these attributes */
+    0,  /* ignore, never encode these attributes */
+    nr_stun_attr_codec_noop_decode
+};
+
+static int
+nr_stun_attr_codec_quoted_string_print(nr_stun_attr_info *attr_info, char *msg, void *data)
+{
+    r_log(NR_LOG_STUN, LOG_DEBUG, "%s %s: %s",
+                          msg, attr_info->name, (char*)data);
+    return 0;
+}
+
+static int
+nr_stun_attr_codec_quoted_string_encode(nr_stun_attr_info *attr_info, void *data, size_t offset, size_t buflen, UCHAR *buf, size_t *attrlen)
+{
+//TODO: !nn! syntax check, conversion if not quoted already?
+//We'll just restrict this in the API -- EKR
+    return nr_stun_attr_codec_string.encode(attr_info, data, offset, buflen, buf, attrlen);
+}
+
+static int
+nr_stun_attr_codec_quoted_string_decode(nr_stun_attr_info *attr_info, size_t attrlen, UCHAR *buf, size_t offset, size_t buflen, void *data)
+{
+//TODO: !nn! I don't see any need to unquote this but we may
+//find one later -- EKR
+    return nr_stun_attr_codec_string.decode(attr_info, attrlen, buf, offset, buflen, data);
+}
+
+nr_stun_attr_codec nr_stun_attr_codec_quoted_string = {
+    "quoted_string",
+    nr_stun_attr_codec_quoted_string_print,
+    nr_stun_attr_codec_quoted_string_encode,
+    nr_stun_attr_codec_quoted_string_decode
+};
+
+static int
+nr_stun_attr_codec_string_print(nr_stun_attr_info *attr_info, char *msg, void *data)
+{
+    r_log(NR_LOG_STUN, LOG_DEBUG, "%s %s: %s",
+                          msg, attr_info->name, (char*)data);
+    return 0;
+}
+
+static int
+nr_stun_attr_codec_string_encode(nr_stun_attr_info *attr_info, void *data, size_t offset, size_t buflen, UCHAR *buf, size_t *attrlen)
+{
+    int start = offset;
+    char *str = data;
+    int length = strlen(str);
+
+    if (nr_stun_encode_htons(attr_info->type  , buflen, buf, &offset)
+     || nr_stun_encode_htons(length           , buflen, buf, &offset)
+     || nr_stun_encode((UCHAR*)str, length    , buflen, buf, &offset))
+        return R_FAILED;
+
+    *attrlen = offset - start;
+
+    return 0;
+}
+
+static int
+nr_stun_attr_codec_string_decode(nr_stun_attr_info *attr_info, size_t attrlen, UCHAR *buf, size_t offset, size_t buflen, void *data)
+{
+    int _status;
+    char *result = data;
+
+    /* actual enforcement of the specific string size happens elsewhere */
+    if (attrlen >= NR_STUN_MAX_STRING_SIZE) {
+        r_log(NR_LOG_STUN, LOG_WARNING, "String is too large: %d bytes", attrlen);
+        ABORT(R_FAILED);
+    }
+
+    if (nr_stun_decode(attrlen, buf, buflen, &offset, (UCHAR*)result))
+        ABORT(R_FAILED);
+    result[attrlen] = '\0'; /* just to be nice */
+
+    if (strlen(result) != attrlen) {
+        /* stund 0.96 sends a final null in the Server attribute, so
+         * only error if the null appears anywhere else in a string */
+        if (strlen(result) != attrlen-1) {
+            r_log(NR_LOG_STUN, LOG_WARNING, "Error in string: %zd/%d", strlen(result), attrlen);
+            ABORT(R_FAILED);
+        }
+    }
+
+    _status = 0;
+  abort:
+    return _status;
+}
+
+nr_stun_attr_codec nr_stun_attr_codec_string = {
+    "string",
+    nr_stun_attr_codec_string_print,
+    nr_stun_attr_codec_string_encode,
+    nr_stun_attr_codec_string_decode
+};
+
+static int
+nr_stun_attr_codec_unknown_attributes_print(nr_stun_attr_info *attr_info, char *msg, void *data)
+{
+    nr_stun_attr_unknown_attributes *unknown_attributes = data;
+    char type[9];
+    char str[64 + (NR_STUN_MAX_UNKNOWN_ATTRIBUTES * sizeof(type))];
+    int i;
+
+    snprintf(str, sizeof(str), "%s %s:", msg, attr_info->name);
+    for (i = 0; i < unknown_attributes->num_attributes; ++i) {
+        snprintf(type, sizeof(type), "%s 0x%04x", ((i>0)?",":""), unknown_attributes->attribute[i]);
+        strlcat(str, type, sizeof(str));
+    }
+
+    r_log(NR_LOG_STUN, LOG_DEBUG, "%s", str);
+    return 0;
+}
+
+static int
+nr_stun_attr_codec_unknown_attributes_encode(nr_stun_attr_info *attr_info, void *data, size_t offset, size_t buflen, UCHAR *buf, size_t *attrlen)
+{
+    int _status;
+    int start = offset;
+    nr_stun_attr_unknown_attributes *unknown_attributes = data;
+    int length = (2 * unknown_attributes->num_attributes);
+    int i;
+
+    if (unknown_attributes->num_attributes > NR_STUN_MAX_UNKNOWN_ATTRIBUTES) {
+        r_log(NR_LOG_STUN, LOG_WARNING, "Too many UNKNOWN-ATTRIBUTES: %d", unknown_attributes->num_attributes);
+        ABORT(R_FAILED);
+    }
+
+    if (nr_stun_encode_htons(attr_info->type  , buflen, buf, &offset)
+     || nr_stun_encode_htons(length           , buflen, buf, &offset))
+        ABORT(R_FAILED);
+
+    for (i = 0; i < unknown_attributes->num_attributes; ++i) {
+        if (nr_stun_encode_htons(unknown_attributes->attribute[i], buflen, buf, &offset))
+            ABORT(R_FAILED);
+    }
+
+    *attrlen = offset - start;
+
+    _status = 0;
+  abort:
+    return _status;
+}
+
+static int
+nr_stun_attr_codec_unknown_attributes_decode(nr_stun_attr_info *attr_info, size_t attrlen, UCHAR *buf, size_t offset, size_t buflen, void *data)
+{
+    int _status;
+    nr_stun_attr_unknown_attributes *unknown_attributes = data;
+    int i;
+    UINT2 *a;
+
+    if ((attrlen % 4) != 0) {
+        r_log(NR_LOG_STUN, LOG_WARNING, "Attribute is illegal size: %d", attrlen);
+        ABORT(R_REJECTED);
+    }
+
+    unknown_attributes->num_attributes = attrlen / 2;
+
+    if (unknown_attributes->num_attributes > NR_STUN_MAX_UNKNOWN_ATTRIBUTES) {
+        r_log(NR_LOG_STUN, LOG_WARNING, "Too many UNKNOWN-ATTRIBUTES: %d", unknown_attributes->num_attributes);
+        ABORT(R_REJECTED);
+    }
+
+    for (i = 0; i < unknown_attributes->num_attributes; ++i) {
+        a = &(unknown_attributes->attribute[i]);
+        if (nr_stun_decode_htons(buf, buflen, &offset, a))
+            return R_FAILED;
+    }
+
+    _status = 0;
+  abort:
+    return _status;
+}
+
+nr_stun_attr_codec nr_stun_attr_codec_unknown_attributes = {
+    "unknown_attributes",
+    nr_stun_attr_codec_unknown_attributes_print,
+    nr_stun_attr_codec_unknown_attributes_encode,
+    nr_stun_attr_codec_unknown_attributes_decode
+};
+
+static int
+nr_stun_attr_codec_xor_mapped_address_print(nr_stun_attr_info *attr_info, char *msg, void *data)
+{
+    nr_stun_attr_xor_mapped_address *xor_mapped_address = data;
+    r_log(NR_LOG_STUN, LOG_DEBUG, "%s %s: %s (unmasked) %s (masked)",
+                          msg, attr_info->name,
+                          xor_mapped_address->unmasked.as_string,
+                          xor_mapped_address->masked.as_string);
+    return 0;
+}
+
+static int
+nr_stun_attr_codec_xor_mapped_address_encode(nr_stun_attr_info *attr_info, void *data, size_t offset, size_t buflen, UCHAR *buf, size_t *attrlen)
+{
+    nr_stun_attr_xor_mapped_address *xor_mapped_address = data;
+    nr_stun_message_header *header = (nr_stun_message_header*)buf;
+    UINT4 magic_cookie;
+
+    r_log(NR_LOG_STUN, LOG_DEBUG, "Unmasked XOR-MAPPED-ADDRESS = %s", xor_mapped_address->unmasked.as_string);
+
+    /* this needs to be the magic cookie in the header and not
+     * the MAGIC_COOKIE constant because if we're talking to
+     * older servers (that don't have a magic cookie) they use
+     * message ID for this */
+    magic_cookie = ntohl(header->magic_cookie);
+
+    nr_stun_xor_mapped_address(magic_cookie, header->id, &xor_mapped_address->unmasked, &xor_mapped_address->masked);
+
+    r_log(NR_LOG_STUN, LOG_DEBUG, "Masked XOR-MAPPED-ADDRESS = %s", xor_mapped_address->masked.as_string);
+
+    if (nr_stun_attr_codec_addr.encode(attr_info, &xor_mapped_address->masked, offset, buflen, buf, attrlen))
+        return R_FAILED;
+
+    return 0;
+}
+
+static int
+nr_stun_attr_codec_xor_mapped_address_decode(nr_stun_attr_info *attr_info, size_t attrlen, UCHAR *buf, size_t offset, size_t buflen, void *data)
+{
+    int r,_status;
+    nr_stun_attr_xor_mapped_address *xor_mapped_address = data;
+    nr_stun_message_header *header = (nr_stun_message_header*)buf;
+    UINT4 magic_cookie;
+
+    if ((r=nr_stun_attr_codec_addr.decode(attr_info, attrlen, buf, offset, buflen, &xor_mapped_address->masked)))
+        ABORT(r);
+
+    r_log(NR_LOG_STUN, LOG_DEBUG, "Masked XOR-MAPPED-ADDRESS = %s", xor_mapped_address->masked.as_string);
+
+    /* this needs to be the magic cookie in the header and not
+     * the MAGIC_COOKIE constant because if we're talking to
+     * older servers (that don't have a magic cookie) they use
+     * message ID for this */
+    magic_cookie = ntohl(header->magic_cookie);
+
+    nr_stun_xor_mapped_address(magic_cookie, header->id, &xor_mapped_address->masked, &xor_mapped_address->unmasked);
+
+    r_log(NR_LOG_STUN, LOG_DEBUG, "Unmasked XOR-MAPPED-ADDRESS = %s", xor_mapped_address->unmasked.as_string);
+
+    _status = 0;
+  abort:
+    return _status;
+}
+
+nr_stun_attr_codec nr_stun_attr_codec_xor_mapped_address = {
+    "xor_mapped_address",
+    nr_stun_attr_codec_xor_mapped_address_print,
+    nr_stun_attr_codec_xor_mapped_address_encode,
+    nr_stun_attr_codec_xor_mapped_address_decode
+};
+
+nr_stun_attr_codec nr_stun_attr_codec_old_xor_mapped_address = {
+    "xor_mapped_address",
+    nr_stun_attr_codec_xor_mapped_address_print,
+    0, /* never encode this type */
+    nr_stun_attr_codec_xor_mapped_address_decode
+};
+
+nr_stun_attr_codec nr_stun_attr_codec_xor_peer_address = {
+    "xor_peer_address",
+    nr_stun_attr_codec_xor_mapped_address_print,
+    nr_stun_attr_codec_xor_mapped_address_encode,
+    nr_stun_attr_codec_xor_mapped_address_decode
+};
+
+#define NR_ADD_STUN_ATTRIBUTE(type, name, codec, illegal) \
+ { (type), (name), &(codec), illegal },
+
+#define NR_ADD_STUN_ATTRIBUTE_IGNORE(type, name) \
+ { (type), (name), &nr_stun_attr_codec_noop, 0 },
+
+
+static nr_stun_attr_info attrs[] = {
+   NR_ADD_STUN_ATTRIBUTE(NR_STUN_ATTR_ALTERNATE_SERVER, "ALTERNATE-SERVER", nr_stun_attr_codec_addr, 0)
+#ifdef USE_STUND_0_96
+   NR_ADD_STUN_ATTRIBUTE(NR_STUN_ATTR_OLD_CHANGE_REQUEST, "CHANGE-REQUEST", nr_stun_attr_codec_UINT4, 0)
+#endif
+   NR_ADD_STUN_ATTRIBUTE(NR_STUN_ATTR_ERROR_CODE, "ERROR-CODE", nr_stun_attr_codec_error_code, nr_stun_attr_error_code_illegal)
+   NR_ADD_STUN_ATTRIBUTE(NR_STUN_ATTR_FINGERPRINT, "FINGERPRINT", nr_stun_attr_codec_fingerprint, 0)
+   NR_ADD_STUN_ATTRIBUTE(NR_STUN_ATTR_MAPPED_ADDRESS, "MAPPED-ADDRESS", nr_stun_attr_codec_addr, 0)
+   NR_ADD_STUN_ATTRIBUTE(NR_STUN_ATTR_MESSAGE_INTEGRITY, "MESSAGE-INTEGRITY", nr_stun_attr_codec_message_integrity, 0)
+   NR_ADD_STUN_ATTRIBUTE(NR_STUN_ATTR_NONCE, "NONCE", nr_stun_attr_codec_quoted_string, nr_stun_attr_nonce_illegal)
+   NR_ADD_STUN_ATTRIBUTE(NR_STUN_ATTR_REALM, "REALM", nr_stun_attr_codec_quoted_string, nr_stun_attr_realm_illegal)
+   NR_ADD_STUN_ATTRIBUTE(NR_STUN_ATTR_SERVER, "SERVER", nr_stun_attr_codec_string, nr_stun_attr_server_illegal)
+   NR_ADD_STUN_ATTRIBUTE(NR_STUN_ATTR_UNKNOWN_ATTRIBUTES, "UNKNOWN-ATTRIBUTES", nr_stun_attr_codec_unknown_attributes, 0)
+   NR_ADD_STUN_ATTRIBUTE(NR_STUN_ATTR_USERNAME, "USERNAME", nr_stun_attr_codec_string, nr_stun_attr_username_illegal)
+   NR_ADD_STUN_ATTRIBUTE(NR_STUN_ATTR_XOR_MAPPED_ADDRESS, "XOR-MAPPED-ADDRESS", nr_stun_attr_codec_xor_mapped_address, 0)
+
+#ifdef USE_ICE
+   NR_ADD_STUN_ATTRIBUTE(NR_STUN_ATTR_ICE_CONTROLLED, "ICE-CONTROLLED", nr_stun_attr_codec_UINT8, 0)
+   NR_ADD_STUN_ATTRIBUTE(NR_STUN_ATTR_ICE_CONTROLLING, "ICE-CONTROLLING", nr_stun_attr_codec_UINT8, 0)
+   NR_ADD_STUN_ATTRIBUTE(NR_STUN_ATTR_PRIORITY, "PRIORITY", nr_stun_attr_codec_UINT4, 0)
+   NR_ADD_STUN_ATTRIBUTE(NR_STUN_ATTR_USE_CANDIDATE, "USE-CANDIDATE", nr_stun_attr_codec_flag, 0)
+#endif
+
+#ifdef USE_TURN
+   NR_ADD_STUN_ATTRIBUTE(NR_STUN_ATTR_DATA, "DATA", nr_stun_attr_codec_data, 0)
+   NR_ADD_STUN_ATTRIBUTE(NR_STUN_ATTR_LIFETIME, "LIFETIME", nr_stun_attr_codec_UINT4, 0)
+   NR_ADD_STUN_ATTRIBUTE(NR_STUN_ATTR_XOR_RELAY_ADDRESS, "XOR-RELAY-ADDRESS", nr_stun_attr_codec_xor_mapped_address, 0)
+   NR_ADD_STUN_ATTRIBUTE(NR_STUN_ATTR_XOR_PEER_ADDRESS, "XOR-PEER-ADDRESS", nr_stun_attr_codec_xor_peer_address, 0)
+   NR_ADD_STUN_ATTRIBUTE(NR_STUN_ATTR_REQUESTED_TRANSPORT, "REQUESTED-TRANSPORT", nr_stun_attr_codec_UCHAR, 0)
+   NR_ADD_STUN_ATTRIBUTE(NR_STUN_ATTR_BANDWIDTH, "BANDWIDTH", nr_stun_attr_codec_UINT4, 0)
+#endif /* USE_TURN */
+
+   /* for backwards compatibilty */
+   NR_ADD_STUN_ATTRIBUTE(NR_STUN_ATTR_OLD_XOR_MAPPED_ADDRESS, "Old XOR-MAPPED-ADDRESS", nr_stun_attr_codec_old_xor_mapped_address, 0)
+#ifdef USE_RFC_3489_BACKWARDS_COMPATIBLE
+   NR_ADD_STUN_ATTRIBUTE_IGNORE(NR_STUN_ATTR_OLD_RESPONSE_ADDRESS, "RESPONSE-ADDRESS")
+   NR_ADD_STUN_ATTRIBUTE_IGNORE(NR_STUN_ATTR_OLD_SOURCE_ADDRESS, "SOURCE-ADDRESS")
+   NR_ADD_STUN_ATTRIBUTE_IGNORE(NR_STUN_ATTR_OLD_CHANGED_ADDRESS, "CHANGED-ADDRESS")
+   NR_ADD_STUN_ATTRIBUTE_IGNORE(NR_STUN_ATTR_OLD_PASSWORD, "PASSWORD")
+#endif /* USE_RFC_3489_BACKWARDS_COMPATIBLE */
+};
+
+
+int
+nr_stun_find_attr_info(UINT2 type, nr_stun_attr_info **info)
+{
+    int _status;
+    size_t i;
+
+    *info = 0;
+    for (i = 0; i < sizeof(attrs)/sizeof(*attrs); ++i) {
+        if (type == attrs[i].type) {
+            *info = &attrs[i];
+            break;
+        }
+    }
+
+    if (*info == 0)
+        ABORT(R_NOT_FOUND);
+
+    _status=0;
+  abort:
+    return(_status);
+}
+
+int
+nr_stun_fix_attribute_ordering(nr_stun_message *msg)
+{
+    nr_stun_message_attribute *message_integrity;
+    nr_stun_message_attribute *fingerprint;
+
+    /* 2nd to the last */
+    if (nr_stun_message_has_attribute(msg, NR_STUN_ATTR_MESSAGE_INTEGRITY, &message_integrity)) {
+        TAILQ_REMOVE(&msg->attributes, message_integrity, entry);
+        TAILQ_INSERT_TAIL(&msg->attributes, message_integrity, entry);
+    }
+
+    /* last */
+    if (nr_stun_message_has_attribute(msg, NR_STUN_ATTR_FINGERPRINT, &fingerprint)) {
+        TAILQ_REMOVE(&msg->attributes, fingerprint, entry);
+        TAILQ_INSERT_TAIL(&msg->attributes, fingerprint, entry);
+    }
+
+    return 0;
+}
+
+// Since this sanity check is only a collection of assert statements and those
+// assert statements are compiled out in non-debug builds, undef SANITY_CHECKS
+// so we can avoid the warning that padding_bytes is never used in opt builds.
+#ifdef NDEBUG
+#undef SANITY_CHECKS
+#endif
+
+#ifdef SANITY_CHECKS
+static void sanity_check_encoding_stuff(nr_stun_message *msg)
+{
+    nr_stun_message_attribute *attr = 0;
+    int padding_bytes;
+    int l;
+
+    r_log(NR_LOG_STUN, LOG_DEBUG, "Starting to sanity check encoding");
+
+    l = 0;
+    TAILQ_FOREACH(attr, &msg->attributes, entry) {
+        padding_bytes = 0;
+        if ((attr->length % 4) != 0) {
+            padding_bytes = 4 - (attr->length % 4);
+        }
+        assert(attr->length == (attr->encoding_length - (4 + padding_bytes)));
+        assert(((void*)attr->encoding) == (msg->buffer + 20 + l));
+        l += attr->encoding_length;
+        assert((l % 4) == 0);
+    }
+    assert(l == msg->header.length);
+}
+#endif /* SANITY_CHECKS */
+
+
+int
+nr_stun_encode_message(nr_stun_message *msg)
+{
+    int r,_status;
+    size_t length_offset;
+    size_t length_offset_hold;
+    nr_stun_attr_info *attr_info;
+    nr_stun_message_attribute *attr;
+    int padding_bytes;
+
+    r_log(NR_LOG_STUN, LOG_DEBUG, "Encoding STUN message");
+
+    nr_stun_fix_attribute_ordering(msg);
+
+    msg->name = nr_stun_msg_type(msg->header.type);
+    msg->length = 0;
+    msg->header.length = 0;
+
+    if ((r=nr_stun_encode_htons(msg->header.type, sizeof(msg->buffer), msg->buffer, &msg->length)))
+        ABORT(r);
+    if (msg->name)
+        r_log(NR_LOG_STUN, LOG_DEBUG, "Encoded MsgType: %s", msg->name);
+    else
+        r_log(NR_LOG_STUN, LOG_DEBUG, "Encoded MsgType: 0x%03x", msg->header.type);
+
+    /* grab the offset to be used later to re-write the header length field */
+    length_offset_hold = msg->length;
+
+    if ((r=nr_stun_encode_htons(msg->header.length, sizeof(msg->buffer), msg->buffer, &msg->length)))
+        ABORT(r);
+
+    if ((r=nr_stun_encode_htonl(msg->header.magic_cookie, sizeof(msg->buffer), msg->buffer, &msg->length)))
+        ABORT(r);
+    r_log(NR_LOG_STUN, LOG_DEBUG, "Encoded Cookie: %08x", msg->header.magic_cookie);
+
+    if ((r=nr_stun_encode((UCHAR*)(&msg->header.id), sizeof(msg->header.id), sizeof(msg->buffer), msg->buffer, &msg->length)))
+        ABORT(r);
+    r_dump(NR_LOG_STUN, LOG_DEBUG, "Encoded ID", (void*)&msg->header.id, sizeof(msg->header.id));
+
+    TAILQ_FOREACH(attr, &msg->attributes, entry) {
+        if ((r=nr_stun_find_attr_info(attr->type, &attr_info))) {
+            r_log(NR_LOG_STUN, LOG_WARNING, "Unrecognized attribute: 0x%04x", attr->type);
+            ABORT(R_INTERNAL);
+        }
+
+        attr->name = attr_info->name;
+        attr->type_name = attr_info->codec->name;
+        attr->encoding = (nr_stun_encoded_attribute*)&msg->buffer[msg->length];
+
+        if (attr_info->codec->encode != 0) {
+            if ((r=attr_info->codec->encode(attr_info, &attr->u, msg->length, sizeof(msg->buffer), msg->buffer, &attr->encoding_length))) {
+                r_log(NR_LOG_STUN, LOG_WARNING, "Unable to encode %s", attr_info->name);
+                ABORT(r);
+            }
+
+            msg->length += attr->encoding_length;
+            attr->length = attr->encoding_length - 4;  /* -4 for type and length fields */
+
+            if (attr_info->illegal) {
+                if ((r=attr_info->illegal(attr_info, attr->length, &attr->u)))
+                    ABORT(r);
+            }
+
+            attr_info->codec->print(attr_info, "Encoded", &attr->u);
+
+            if ((attr->length % 4) == 0) {
+                padding_bytes = 0;
+            }
+            else {
+                padding_bytes = 4 - (attr->length % 4);
+                nr_stun_encode((UCHAR*)"\0\0\0\0", padding_bytes, sizeof(msg->buffer), msg->buffer, &msg->length);
+                attr->encoding_length += padding_bytes;
+            }
+
+            msg->header.length += attr->encoding_length;
+            length_offset = length_offset_hold;
+            (void)nr_stun_encode_htons(msg->header.length, sizeof(msg->buffer), msg->buffer, &length_offset);
+        }
+        else {
+            r_log(NR_LOG_STUN, LOG_WARNING, "Missing encode function for attribute: %s", attr_info->name);
+        }
+    }
+
+    r_log(NR_LOG_STUN, LOG_DEBUG, "Encoded Length: %d", msg->header.length);
+
+    assert(msg->length < NR_STUN_MAX_MESSAGE_SIZE);
+
+#ifdef SANITY_CHECKS
+    sanity_check_encoding_stuff(msg);
+#endif /* SANITY_CHECKS */
+
+    _status=0;
+abort:
+    return _status;
+}
+
+int
+nr_stun_decode_message(nr_stun_message *msg, int (*get_password)(void *arg, nr_stun_message *msg, Data **password), void *arg)
+{
+    int r,_status;
+    int offset;
+    int size;
+    int padding_bytes;
+    nr_stun_message_attribute *attr;
+    nr_stun_attr_info *attr_info;
+    Data *password;
+
+    r_log(NR_LOG_STUN, LOG_DEBUG, "Parsing STUN message of %d bytes", msg->length);
+
+    if (!TAILQ_EMPTY(&msg->attributes))
+        ABORT(R_BAD_ARGS);
+
+    if (sizeof(nr_stun_message_header) > msg->length) {
+       r_log(NR_LOG_STUN, LOG_WARNING, "Message too small");
+       ABORT(R_FAILED);
+    }
+
+    memcpy(&msg->header, msg->buffer, sizeof(msg->header));
+    msg->header.type = ntohs(msg->header.type);
+    msg->header.length = ntohs(msg->header.length);
+    msg->header.magic_cookie = ntohl(msg->header.magic_cookie);
+
+    msg->name = nr_stun_msg_type(msg->header.type);
+
+    if (msg->name)
+        r_log(NR_LOG_STUN, LOG_DEBUG, "Parsed MsgType: %s", msg->name);
+    else
+        r_log(NR_LOG_STUN, LOG_DEBUG, "Parsed MsgType: 0x%03x", msg->header.type);
+    r_log(NR_LOG_STUN, LOG_DEBUG, "Parsed Length: %d", msg->header.length);
+    r_log(NR_LOG_STUN, LOG_DEBUG, "Parsed Cookie: %08x", msg->header.magic_cookie);
+    r_dump(NR_LOG_STUN, LOG_DEBUG, "Parsed ID", (void*)&msg->header.id, sizeof(msg->header.id));
+
+    if (msg->header.length + sizeof(msg->header) != msg->length) {
+       r_log(NR_LOG_STUN, LOG_WARNING, "Inconsistent message header length: %d/%d",
+                                        msg->header.length, msg->length);
+       ABORT(R_FAILED);
+    }
+
+    size = msg->header.length;
+
+    if ((size % 4) != 0) {
+       r_log(NR_LOG_STUN, LOG_WARNING, "Illegal message size: %d", msg->header.length);
+       ABORT(R_FAILED);
+    }
+
+    offset = sizeof(msg->header);
+
+    while (size > 0) {
+        r_log(NR_LOG_STUN, LOG_DEBUG, "size = %d", size);
+
+        if (size < 4) {
+           r_log(NR_LOG_STUN, LOG_WARNING, "Illegal message length: %d", size);
+           ABORT(R_FAILED);
+        }
+
+        if ((r=nr_stun_message_attribute_create(msg, &attr)))
+            ABORT(R_NO_MEMORY);
+
+        attr->encoding          = (nr_stun_encoded_attribute*)&msg->buffer[offset];
+        attr->type              = ntohs(attr->encoding->type);
+        attr->length            = ntohs(attr->encoding->length);
+        attr->encoding_length   = attr->length + 4;
+
+        if ((attr->length % 4) != 0) {
+            padding_bytes = 4 - (attr->length % 4);
+            attr->encoding_length += padding_bytes;
+        }
+
+        if ((attr->encoding_length) > (size_t)size) {
+           r_log(NR_LOG_STUN, LOG_WARNING, "Attribute length larger than remaining message size: %d/%d", attr->encoding_length, size);
+           ABORT(R_FAILED);
+        }
+
+        if ((r=nr_stun_find_attr_info(attr->type, &attr_info))) {
+            if (attr->type <= 0x7FFF)
+                ++msg->comprehension_required_unknown_attributes;
+            else
+                ++msg->comprehension_optional_unknown_attributes;
+            r_log(NR_LOG_STUN, LOG_INFO, "Unrecognized attribute: 0x%04x", attr->type);
+        }
+        else {
+            attr_info->name = attr_info->name;
+            attr->type_name = attr_info->codec->name;
+
+            if (attr->type == NR_STUN_ATTR_MESSAGE_INTEGRITY) {
+                if (get_password && get_password(arg, msg, &password) == 0) {
+                    if (password->len > sizeof(attr->u.message_integrity.password)) {
+                        r_log(NR_LOG_STUN, LOG_WARNING, "Password too long: %d bytes", password->len);
+                        ABORT(R_FAILED);
+                    }
+
+                    memcpy(attr->u.message_integrity.password, password->data, password->len);
+                    attr->u.message_integrity.passwordlen = password->len;
+                }
+                else {
+                    /* set to user "not found" */
+                    attr->u.message_integrity.unknown_user = 1;
+                }
+            }
+            else if (attr->type == NR_STUN_ATTR_OLD_XOR_MAPPED_ADDRESS) {
+                attr->type = NR_STUN_ATTR_XOR_MAPPED_ADDRESS;
+                r_log(NR_LOG_STUN, LOG_INFO, "Translating obsolete XOR-MAPPED-ADDRESS type");
+            }
+
+            if ((r=attr_info->codec->decode(attr_info, attr->length, msg->buffer, offset+4, msg->length, &attr->u))) {
+                if (r == SKIP_ATTRIBUTE_DECODE) {
+                    r_log(NR_LOG_STUN, LOG_INFO, "Skipping %s", attr_info->name);
+                }
+                else {
+                    r_log(NR_LOG_STUN, LOG_WARNING, "Unable to parse %s", attr_info->name);
+                }
+
+                attr->invalid = 1;
+            }
+            else {
+                attr_info->codec->print(attr_info, "Parsed", &attr->u);
+
+#ifdef USE_STUN_PEDANTIC
+                r_log(NR_LOG_STUN, LOG_DEBUG, "Before pedantic attr_info checks");
+                if (attr_info->illegal) {
+                    if ((r=attr_info->illegal(attr_info, attr->length, &attr->u))) {
+                        r_log(NR_LOG_STUN, LOG_WARNING, "Failed pedantic attr_info checks");
+                        ABORT(r);
+                    }
+                }
+                r_log(NR_LOG_STUN, LOG_DEBUG, "After pedantic attr_info checks");
+#endif /* USE_STUN_PEDANTIC */
+            }
+        }
+
+        offset += attr->encoding_length;
+        size -= attr->encoding_length;
+    }
+
+#ifdef SANITY_CHECKS
+    sanity_check_encoding_stuff(msg);
+#endif /* SANITY_CHECKS */
+
+    _status=0;
+  abort:
+    return _status;
+}
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_codec.h b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_codec.h
new file mode 100644
index 0000000000..4e4ff60e0c
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_codec.h
@@ -0,0 +1,78 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _stun_codec_h
+#define _stun_codec_h
+
+#include "stun_msg.h"
+
+typedef struct nr_stun_attr_info_  nr_stun_attr_info;
+
+typedef struct nr_stun_attr_codec_ {
+    char     *name;
+    int     (*print)(nr_stun_attr_info *attr_info, char *msg, void *data);
+    int     (*encode)(nr_stun_attr_info *attr_info, void *data, size_t offset, size_t buflen, UCHAR *buf, size_t *attrlen);
+    int     (*decode)(nr_stun_attr_info *attr_info, size_t attrlen, UCHAR *buf, size_t offset, size_t buflen, void *data);
+} nr_stun_attr_codec;
+
+struct nr_stun_attr_info_ {
+     UINT2                 type;
+     char                 *name;
+     nr_stun_attr_codec   *codec;
+     int                 (*illegal)(nr_stun_attr_info *attr_info, size_t attrlen, void *data);
+};
+
+extern nr_stun_attr_codec nr_stun_attr_codec_UINT4;
+extern nr_stun_attr_codec nr_stun_attr_codec_UINT8;
+extern nr_stun_attr_codec nr_stun_attr_codec_addr;
+extern nr_stun_attr_codec nr_stun_attr_codec_bytes;
+extern nr_stun_attr_codec nr_stun_attr_codec_data;
+extern nr_stun_attr_codec nr_stun_attr_codec_error_code;
+extern nr_stun_attr_codec nr_stun_attr_codec_fingerprint;
+extern nr_stun_attr_codec nr_stun_attr_codec_flag;
+extern nr_stun_attr_codec nr_stun_attr_codec_message_integrity;
+extern nr_stun_attr_codec nr_stun_attr_codec_noop;
+extern nr_stun_attr_codec nr_stun_attr_codec_quoted_string;
+extern nr_stun_attr_codec nr_stun_attr_codec_string;
+extern nr_stun_attr_codec nr_stun_attr_codec_unknown_attributes;
+extern nr_stun_attr_codec nr_stun_attr_codec_xor_mapped_address;
+extern nr_stun_attr_codec nr_stun_attr_codec_xor_peer_address;
+extern nr_stun_attr_codec nr_stun_attr_codec_old_xor_mapped_address;
+
+size_t nr_count_utf8_code_points_without_validation(const char *s);
+int nr_stun_encode_message(nr_stun_message *msg);
+int nr_stun_decode_message(nr_stun_message *msg, int (*get_password)(void *arg, nr_stun_message *msg, Data **password), void *arg);
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_hint.c b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_hint.c
new file mode 100644
index 0000000000..8f118e8942
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_hint.c
@@ -0,0 +1,245 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <errno.h>
+#include <csi_platform.h>
+
+#ifdef WIN32
+#include <winsock2.h>
+#include <stdlib.h>
+#include <io.h>
+#include <time.h>
+#else   /* UNIX */
+#include <string.h>
+#endif  /* end UNIX */
+#include <assert.h>
+
+#include "stun.h"
+
+
+/* returns 0 if it's not a STUN message
+ *         1 if it's likely to be a STUN message
+ *         2 if it's super likely to be a STUN message
+ *         3 if it really is a STUN message */
+int
+nr_is_stun_message(UCHAR *buf, size_t len)
+{
+   const UINT4 cookie = htonl(NR_STUN_MAGIC_COOKIE);
+   const UINT4 cookie2 = htonl(NR_STUN_MAGIC_COOKIE2);
+#if 0
+   nr_stun_message msg;
+#endif
+   UINT2 type;
+   nr_stun_encoded_attribute* attr;
+   unsigned int attrLen;
+   int atrType;
+
+   if (sizeof(nr_stun_message_header) > len)
+       return 0;
+
+   if ((buf[0] & (0x80|0x40)) != 0)
+       return 0;
+
+   memcpy(&type, buf, 2);
+   type = ntohs(type);
+
+   switch (type) {
+   case NR_STUN_MSG_BINDING_REQUEST:
+   case NR_STUN_MSG_BINDING_INDICATION:
+   case NR_STUN_MSG_BINDING_RESPONSE:
+   case NR_STUN_MSG_BINDING_ERROR_RESPONSE:
+
+#ifdef USE_TURN
+    case NR_STUN_MSG_ALLOCATE_REQUEST:
+    case NR_STUN_MSG_ALLOCATE_RESPONSE:
+    case NR_STUN_MSG_ALLOCATE_ERROR_RESPONSE:
+    case NR_STUN_MSG_REFRESH_REQUEST:
+    case NR_STUN_MSG_REFRESH_RESPONSE:
+    case NR_STUN_MSG_REFRESH_ERROR_RESPONSE:
+    case NR_STUN_MSG_PERMISSION_REQUEST:
+    case NR_STUN_MSG_PERMISSION_RESPONSE:
+    case NR_STUN_MSG_PERMISSION_ERROR_RESPONSE:
+    case NR_STUN_MSG_CHANNEL_BIND_REQUEST:
+    case NR_STUN_MSG_CHANNEL_BIND_RESPONSE:
+    case NR_STUN_MSG_CHANNEL_BIND_ERROR_RESPONSE:
+    case NR_STUN_MSG_SEND_INDICATION:
+    case NR_STUN_MSG_DATA_INDICATION:
+#ifdef NR_STUN_MSG_CONNECT_REQUEST
+    case NR_STUN_MSG_CONNECT_REQUEST:
+#endif
+#ifdef NR_STUN_MSG_CONNECT_RESPONSE
+    case NR_STUN_MSG_CONNECT_RESPONSE:
+#endif
+#ifdef NR_STUN_MSG_CONNECT_ERROR_RESPONSE
+    case NR_STUN_MSG_CONNECT_ERROR_RESPONSE:
+#endif
+#ifdef NR_STUN_MSG_CONNECT_STATUS_INDICATION
+    case NR_STUN_MSG_CONNECT_STATUS_INDICATION:
+#endif
+#endif /* USE_TURN */
+
+        /* ok so far, continue */
+        break;
+   default:
+        return 0;
+        break;
+   }
+
+   if (!memcmp(&cookie2, &buf[4], sizeof(UINT4))) {
+       /* return here because if it's an old-style message then there will
+        * not be a fingerprint in the message */
+       return 1;
+   }
+
+   if (memcmp(&cookie, &buf[4], sizeof(UINT4)))
+       return 0;
+
+   /* the magic cookie was right, so it's pretty darn likely that what we've
+    * got here is a STUN message */
+
+   attr = (nr_stun_encoded_attribute*)(buf + (len - 8));
+   attrLen = ntohs(attr->length);
+   atrType = ntohs(attr->type);
+
+   if (atrType != NR_STUN_ATTR_FINGERPRINT || attrLen != 4)
+       return 1;
+
+   /* the fingerprint is in the right place and looks sane, so we can be quite
+    * sure we've got a STUN message */
+
+#if 0
+/* nevermind this check ... there's a reasonable chance that a NAT has modified
+ * the message (and thus the fingerprint check will fail), but it's still an
+ * otherwise-perfectly-good STUN message, so skip the check since we're going
+ * to return "true" whether the check succeeds or fails */
+
+   if (nr_stun_parse_attr_UINT4(buf + (len - 4), attrLen, &msg.fingerprint))
+       return 2;
+
+
+   if (nr_stun_compute_fingerprint(buf, len - 8, &computedFingerprint))
+       return 2;
+
+   if (msg.fingerprint.number != computedFingerprint)
+       return 2;
+
+   /* and the fingerprint is good, so it's gotta be a STUN message */
+#endif
+
+   return 3;
+}
+
+int
+nr_is_stun_request_message(UCHAR *buf, size_t len)
+{
+   UINT2 type;
+
+   if (sizeof(nr_stun_message_header) > len)
+       return 0;
+
+   if (!nr_is_stun_message(buf, len))
+       return 0;
+
+   memcpy(&type, buf, 2);
+   type = ntohs(type);
+
+   return NR_STUN_GET_TYPE_CLASS(type) == NR_CLASS_REQUEST;
+}
+
+int
+nr_is_stun_indication_message(UCHAR *buf, size_t len)
+{
+   UINT2 type;
+
+   if (sizeof(nr_stun_message_header) > len)
+       return 0;
+
+   if (!nr_is_stun_message(buf, len))
+       return 0;
+
+   memcpy(&type, buf, 2);
+   type = ntohs(type);
+
+   return NR_STUN_GET_TYPE_CLASS(type) == NR_CLASS_INDICATION;
+}
+
+int
+nr_is_stun_response_message(UCHAR *buf, size_t len)
+{
+   UINT2 type;
+
+   if (sizeof(nr_stun_message_header) > len)
+       return 0;
+
+   if (!nr_is_stun_message(buf, len))
+       return 0;
+
+   memcpy(&type, buf, 2);
+   type = ntohs(type);
+
+   return NR_STUN_GET_TYPE_CLASS(type) == NR_CLASS_RESPONSE
+       || NR_STUN_GET_TYPE_CLASS(type) == NR_CLASS_ERROR_RESPONSE;
+}
+
+int
+nr_has_stun_cookie(UCHAR *buf, size_t len)
+{
+   static UINT4 cookie;
+
+   cookie = htonl(NR_STUN_MAGIC_COOKIE);
+
+   if (sizeof(nr_stun_message_header) > len)
+       return 0;
+
+   if (memcmp(&cookie, &buf[4], sizeof(UINT4)))
+       return 0;
+
+   return 1;
+}
+
+int
+nr_stun_message_length(UCHAR *buf, int buf_len, int *msg_len)
+{
+  nr_stun_message_header *hdr;
+
+  if (!nr_is_stun_message(buf, buf_len))
+    return(R_BAD_DATA);
+
+  hdr = (nr_stun_message_header *)buf;
+
+  *msg_len = ntohs(hdr->length);
+
+  return(0);
+}
+
+
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_hint.h b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_hint.h
new file mode 100644
index 0000000000..c2badc1d2b
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_hint.h
@@ -0,0 +1,44 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+#ifndef _stun_hint_h
+#define _stun_hint_h
+
+int nr_is_stun_message(UCHAR *buf, size_t len);
+int nr_is_stun_request_message(UCHAR *buf, size_t len);
+int nr_is_stun_response_message(UCHAR *buf, size_t len);
+int nr_is_stun_indication_message(UCHAR *buf, size_t len);
+int nr_has_stun_cookie(UCHAR *buf, size_t len);
+int nr_stun_message_length(UCHAR *buf, int len, int *length);
+
+#endif
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_msg.c b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_msg.c
new file mode 100644
index 0000000000..7e01686109
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_msg.c
@@ -0,0 +1,364 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <errno.h>
+#include <csi_platform.h>
+
+#ifdef WIN32
+#include <winsock2.h>
+#include <stdlib.h>
+#include <io.h>
+#include <time.h>
+#else   /* UNIX */
+#include <string.h>
+#endif  /* end UNIX */
+#include <assert.h>
+
+#include "stun.h"
+
+
+int
+nr_stun_message_create(nr_stun_message **msg)
+{
+    int _status;
+    nr_stun_message *m = 0;
+
+    m = RCALLOC(sizeof(*m));
+    if (!m)
+        ABORT(R_NO_MEMORY);
+
+    TAILQ_INIT(&m->attributes);
+
+    *msg = m;
+
+    _status=0;
+  abort:
+    return(_status);
+}
+
+int
+nr_stun_message_create2(nr_stun_message **msg, UCHAR *buffer, size_t length)
+{
+    int r,_status;
+    nr_stun_message *m = 0;
+
+    if (length > sizeof(m->buffer)) {
+        ABORT(R_BAD_DATA);
+    }
+
+    if ((r=nr_stun_message_create(&m)))
+        ABORT(r);
+
+    memcpy(m->buffer, buffer, length);
+    m->length = length;
+
+    *msg = m;
+
+    _status=0;
+  abort:
+    return(_status);
+}
+
+int
+nr_stun_message_destroy(nr_stun_message **msg)
+{
+    int _status;
+    nr_stun_message_attribute_head *attrs;
+    nr_stun_message_attribute *attr;
+
+    if (msg && *msg) {
+        attrs = &(*msg)->attributes;
+        while (!TAILQ_EMPTY(attrs)) {
+            attr = TAILQ_FIRST(attrs);
+            nr_stun_message_attribute_destroy(*msg, &attr);
+        }
+
+        RFREE(*msg);
+
+        *msg = 0;
+    }
+
+    _status=0;
+/*  abort: */
+    return(_status);
+}
+
+int
+nr_stun_message_attribute_create(nr_stun_message *msg, nr_stun_message_attribute **attr)
+{
+    int _status;
+    nr_stun_message_attribute *a = 0;
+
+    a = RCALLOC(sizeof(*a));
+    if (!a)
+        ABORT(R_NO_MEMORY);
+
+    TAILQ_INSERT_TAIL(&msg->attributes, a, entry);
+
+    *attr = a;
+
+    _status=0;
+  abort:
+    return(_status);
+}
+
+int
+nr_stun_message_attribute_destroy(nr_stun_message *msg, nr_stun_message_attribute **attr)
+{
+    int _status;
+    nr_stun_message_attribute *a = 0;
+
+    if (attr && *attr) {
+        a = *attr;
+        TAILQ_REMOVE(&msg->attributes, a, entry);
+
+        RFREE(a);
+
+        *attr = 0;
+    }
+
+    _status=0;
+/*  abort: */
+    return(_status);
+}
+
+int
+nr_stun_message_has_attribute(nr_stun_message *msg, UINT2 type, nr_stun_message_attribute **attribute)
+{
+  nr_stun_message_attribute *attr = 0;
+  nr_stun_message_get_attribute(msg, type, 0, &attr);
+
+  if (attribute)
+    *attribute = attr;
+
+  return attr ? 1 : 0;
+}
+
+int
+nr_stun_message_get_attribute(nr_stun_message *msg, UINT2 type, UINT2 index, nr_stun_message_attribute **attribute)
+{
+  nr_stun_message_attribute *attr;
+  TAILQ_FOREACH(attr, &msg->attributes, entry) {
+    if (attr->type == type && !attr->invalid) {
+      if (!index) {
+        *attribute = attr;
+        return 0;
+      }
+      --index;
+    }
+  }
+  *attribute = 0;
+  return R_NOT_FOUND;
+}
+
+#define NR_STUN_MESSAGE_ADD_ATTRIBUTE(__type, __code) \
+  { \
+    int r,_status; \
+    nr_stun_message_attribute *attr = 0; \
+    if ((r=nr_stun_message_attribute_create(msg, &attr))) \
+        ABORT(r); \
+    attr->type = (__type); \
+    { __code } \
+    _status=0; \
+  abort: \
+    if (_status){ \
+      nr_stun_message_attribute_destroy(msg, &attr); \
+    } \
+    return(_status); \
+  }
+
+
+int
+nr_stun_message_add_alternate_server_attribute(nr_stun_message *msg, nr_transport_addr *alternate_server)
+NR_STUN_MESSAGE_ADD_ATTRIBUTE(
+    NR_STUN_ATTR_ALTERNATE_SERVER,
+    {
+        if ((r=nr_transport_addr_copy(&attr->u.alternate_server, alternate_server)))
+            ABORT(r);
+    }
+)
+
+int
+nr_stun_message_add_error_code_attribute(nr_stun_message *msg, UINT2 number, char *reason)
+NR_STUN_MESSAGE_ADD_ATTRIBUTE(
+    NR_STUN_ATTR_ERROR_CODE,
+    {
+        attr->u.error_code.number = number;
+        (void)strlcpy(attr->u.error_code.reason, reason, sizeof(attr->u.error_code.reason));
+    }
+)
+
+int
+nr_stun_message_add_fingerprint_attribute(nr_stun_message *msg)
+NR_STUN_MESSAGE_ADD_ATTRIBUTE(
+    NR_STUN_ATTR_FINGERPRINT,
+    {}
+)
+
+int
+nr_stun_message_add_message_integrity_attribute(nr_stun_message *msg, Data *password)
+NR_STUN_MESSAGE_ADD_ATTRIBUTE(
+    NR_STUN_ATTR_MESSAGE_INTEGRITY,
+    {
+        if (sizeof(attr->u.message_integrity.password) < password->len)
+            ABORT(R_BAD_DATA);
+
+        memcpy(attr->u.message_integrity.password, password->data, password->len);
+        attr->u.message_integrity.passwordlen = password->len;
+    }
+)
+
+int
+nr_stun_message_add_nonce_attribute(nr_stun_message *msg, char *nonce)
+NR_STUN_MESSAGE_ADD_ATTRIBUTE(
+    NR_STUN_ATTR_NONCE,
+    { (void)strlcpy(attr->u.nonce, nonce, sizeof(attr->u.nonce)); }
+)
+
+int
+nr_stun_message_add_realm_attribute(nr_stun_message *msg, char *realm)
+NR_STUN_MESSAGE_ADD_ATTRIBUTE(
+    NR_STUN_ATTR_REALM,
+    { (void)strlcpy(attr->u.realm, realm, sizeof(attr->u.realm)); }
+)
+
+int
+nr_stun_message_add_server_attribute(nr_stun_message *msg, char *server_name)
+NR_STUN_MESSAGE_ADD_ATTRIBUTE(
+    NR_STUN_ATTR_SERVER,
+    { (void)strlcpy(attr->u.server_name, server_name, sizeof(attr->u.server_name)); }
+)
+
+int
+nr_stun_message_add_unknown_attributes_attribute(nr_stun_message *msg, nr_stun_attr_unknown_attributes *unknown_attributes)
+NR_STUN_MESSAGE_ADD_ATTRIBUTE(
+    NR_STUN_ATTR_UNKNOWN_ATTRIBUTES,
+    { memcpy(&attr->u.unknown_attributes, unknown_attributes, sizeof(attr->u.unknown_attributes)); }
+)
+
+int
+nr_stun_message_add_username_attribute(nr_stun_message *msg, char *username)
+NR_STUN_MESSAGE_ADD_ATTRIBUTE(
+    NR_STUN_ATTR_USERNAME,
+    { (void)strlcpy(attr->u.username, username, sizeof(attr->u.username)); }
+)
+
+int
+nr_stun_message_add_requested_transport_attribute(nr_stun_message *msg, UCHAR protocol)
+NR_STUN_MESSAGE_ADD_ATTRIBUTE(
+    NR_STUN_ATTR_REQUESTED_TRANSPORT,
+    { attr->u.requested_transport = protocol; }
+)
+
+int
+nr_stun_message_add_xor_mapped_address_attribute(nr_stun_message *msg, nr_transport_addr *mapped_address)
+NR_STUN_MESSAGE_ADD_ATTRIBUTE(
+    NR_STUN_ATTR_XOR_MAPPED_ADDRESS,
+    {
+        if ((r=nr_transport_addr_copy(&attr->u.xor_mapped_address.unmasked, mapped_address)))
+            ABORT(r);
+    }
+)
+
+int
+nr_stun_message_add_xor_peer_address_attribute(nr_stun_message *msg, nr_transport_addr *peer_address)
+NR_STUN_MESSAGE_ADD_ATTRIBUTE(
+    NR_STUN_ATTR_XOR_PEER_ADDRESS,
+    {
+        if ((r=nr_transport_addr_copy(&attr->u.xor_mapped_address.unmasked, peer_address)))
+            ABORT(r);
+    }
+)
+
+#ifdef USE_ICE
+int
+nr_stun_message_add_ice_controlled_attribute(nr_stun_message *msg, UINT8 ice_controlled)
+NR_STUN_MESSAGE_ADD_ATTRIBUTE(
+    NR_STUN_ATTR_ICE_CONTROLLED,
+    { attr->u.ice_controlled = ice_controlled; }
+)
+
+int
+nr_stun_message_add_ice_controlling_attribute(nr_stun_message *msg, UINT8 ice_controlling)
+NR_STUN_MESSAGE_ADD_ATTRIBUTE(
+    NR_STUN_ATTR_ICE_CONTROLLING,
+    { attr->u.ice_controlling = ice_controlling; }
+)
+
+int
+nr_stun_message_add_priority_attribute(nr_stun_message *msg, UINT4 priority)
+NR_STUN_MESSAGE_ADD_ATTRIBUTE(
+    NR_STUN_ATTR_PRIORITY,
+    { attr->u.priority = priority; }
+)
+
+int
+nr_stun_message_add_use_candidate_attribute(nr_stun_message *msg)
+NR_STUN_MESSAGE_ADD_ATTRIBUTE(
+    NR_STUN_ATTR_USE_CANDIDATE,
+    {}
+)
+#endif /* USE_ICE */
+
+#ifdef USE_TURN
+int
+nr_stun_message_add_data_attribute(nr_stun_message *msg, UCHAR *data, int length)
+
+NR_STUN_MESSAGE_ADD_ATTRIBUTE(
+    NR_STUN_ATTR_DATA,
+    {
+      if (length > NR_STUN_MAX_MESSAGE_SIZE)
+        ABORT(R_BAD_ARGS);
+
+      memcpy(attr->u.data.data, data, length);
+      attr->u.data.length=length;
+    }
+)
+
+int
+nr_stun_message_add_lifetime_attribute(nr_stun_message *msg, UINT4 lifetime_secs)
+NR_STUN_MESSAGE_ADD_ATTRIBUTE(
+    NR_STUN_ATTR_LIFETIME,
+    { attr->u.lifetime_secs = lifetime_secs; }
+)
+
+#endif /* USE_TURN */
+
+#ifdef USE_STUND_0_96
+int
+nr_stun_message_add_change_request_attribute(nr_stun_message *msg, UINT4 change_request)
+NR_STUN_MESSAGE_ADD_ATTRIBUTE(
+    NR_STUN_ATTR_OLD_CHANGE_REQUEST,
+    { attr->u.change_request = change_request; }
+)
+#endif /* USE_STUND_0_96 */
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_msg.h b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_msg.h
new file mode 100644
index 0000000000..ffd68d3eee
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_msg.h
@@ -0,0 +1,208 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _stun_msg_h
+#define _stun_msg_h
+
+#include "csi_platform.h"
+#include "nr_api.h"
+#include "transport_addr.h"
+
+#define NR_STUN_MAX_USERNAME_BYTES                 513
+#define NR_STUN_MAX_ERROR_CODE_REASON_BYTES        763
+#define NR_STUN_MAX_ERROR_CODE_REASON_CHARS        128
+#define NR_STUN_MAX_REALM_BYTES                    763
+#define NR_STUN_MAX_REALM_CHARS                    128
+#define NR_STUN_MAX_NONCE_BYTES                    763
+#define NR_STUN_MAX_NONCE_CHARS                    128
+#define NR_STUN_MAX_SERVER_BYTES                   763
+#define NR_STUN_MAX_SERVER_CHARS                   128
+#define NR_STUN_MAX_STRING_SIZE                    763  /* any possible string */
+#define NR_STUN_MAX_UNKNOWN_ATTRIBUTES             16
+#define NR_STUN_MAX_MESSAGE_SIZE                   2048
+
+#define NR_STUN_MAGIC_COOKIE            0x2112A442
+#define NR_STUN_MAGIC_COOKIE2           0xc5cb4e1d  /* used recognize old stun messages */
+
+typedef struct { UCHAR octet[12]; }  UINT12;
+
+typedef struct nr_stun_attr_error_code_ {
+    UINT2     number;
+    char      reason[NR_STUN_MAX_ERROR_CODE_REASON_BYTES+1];  /* +1 for \0 */
+} nr_stun_attr_error_code;
+
+typedef struct nr_stun_attr_fingerprint_ {
+    UINT4     checksum;
+    int       valid;
+} nr_stun_attr_fingerprint;
+
+typedef struct nr_stun_attr_message_integrity_ {
+    UCHAR     hash[20];
+    int       unknown_user;
+    UCHAR     password[1024];
+    int       passwordlen;
+    int       valid;
+} nr_stun_attr_message_integrity;
+
+typedef struct nr_stun_attr_unknown_attributes_ {
+    UINT2     attribute[NR_STUN_MAX_UNKNOWN_ATTRIBUTES];
+    int       num_attributes;
+} nr_stun_attr_unknown_attributes;
+
+typedef struct nr_stun_attr_xor_mapped_address_ {
+        nr_transport_addr               masked;
+        nr_transport_addr               unmasked;
+} nr_stun_attr_xor_mapped_address;
+
+typedef struct nr_stun_attr_data_ {
+    UCHAR  data[NR_STUN_MAX_MESSAGE_SIZE];
+    size_t length;
+} nr_stun_attr_data;
+
+
+typedef struct nr_stun_encoded_attribute_ {
+    UINT2                               type;
+    UINT2                               length;
+    UCHAR                               value[NR_STUN_MAX_MESSAGE_SIZE];
+} nr_stun_encoded_attribute;
+
+typedef struct nr_stun_message_attribute_ {
+    UINT2                               type;
+    UINT2                               length;
+    union {
+        nr_transport_addr               address;
+        nr_transport_addr               alternate_server;
+        nr_stun_attr_error_code         error_code;
+        nr_stun_attr_fingerprint        fingerprint;
+        nr_transport_addr               mapped_address;
+        nr_stun_attr_message_integrity  message_integrity;
+        char                            nonce[NR_STUN_MAX_NONCE_BYTES+1];  /* +1 for \0 */
+        char                            realm[NR_STUN_MAX_REALM_BYTES+1];  /* +1 for \0 */
+        nr_stun_attr_xor_mapped_address relay_address;
+        char                            server_name[NR_STUN_MAX_SERVER_BYTES+1];  /* +1 for \0 */
+        nr_stun_attr_unknown_attributes unknown_attributes;
+        char                            username[NR_STUN_MAX_USERNAME_BYTES+1];  /* +1 for \0 */
+        nr_stun_attr_xor_mapped_address xor_mapped_address;
+
+#ifdef USE_ICE
+        UINT4                           priority;
+        UINT8                           ice_controlled;
+        UINT8                           ice_controlling;
+#endif /* USE_ICE */
+
+#ifdef USE_TURN
+        UINT4                           lifetime_secs;
+        nr_transport_addr               remote_address;
+        UCHAR                           requested_transport;
+        nr_stun_attr_data               data;
+#endif /* USE_TURN */
+
+#ifdef USE_STUND_0_96
+        UINT4                           change_request;
+#endif /* USE_STUND_0_96 */
+
+        /* make sure there's enough room here to place any possible
+         * attribute */
+        UCHAR                           largest_possible_attribute[NR_STUN_MAX_MESSAGE_SIZE];
+    } u;
+    nr_stun_encoded_attribute          *encoding;
+    size_t                              encoding_length;
+    char                               *name;
+    char                               *type_name;
+    int                                 invalid;
+    TAILQ_ENTRY(nr_stun_message_attribute_)  entry;
+} nr_stun_message_attribute;
+
+typedef TAILQ_HEAD(nr_stun_message_attribute_head_,nr_stun_message_attribute_) nr_stun_message_attribute_head;
+
+typedef struct nr_stun_message_header_ {
+    UINT2                               type;
+    UINT2                               length;
+    UINT4                               magic_cookie;
+    UINT12                              id;
+} nr_stun_message_header;
+
+typedef struct nr_stun_message_ {
+    char                               *name;
+    UCHAR                               buffer[NR_STUN_MAX_MESSAGE_SIZE];
+    size_t                              length;
+    nr_stun_message_header              header;
+    int                                 comprehension_required_unknown_attributes;
+    int                                 comprehension_optional_unknown_attributes;
+    nr_stun_message_attribute_head      attributes;
+} nr_stun_message;
+
+int nr_stun_message_create(nr_stun_message **msg);
+int nr_stun_message_create2(nr_stun_message **msg, UCHAR *buffer, size_t length);
+int nr_stun_message_destroy(nr_stun_message **msg);
+
+int nr_stun_message_attribute_create(nr_stun_message *msg, nr_stun_message_attribute **attr);
+int nr_stun_message_attribute_destroy(nr_stun_message *msg, nr_stun_message_attribute **attr);
+
+int nr_stun_message_has_attribute(nr_stun_message *msg, UINT2 type, nr_stun_message_attribute **attribute);
+
+int nr_stun_message_get_attribute(nr_stun_message *msg, UINT2 type, UINT2 index, nr_stun_message_attribute **attribute);
+
+int nr_stun_message_add_alternate_server_attribute(nr_stun_message *msg, nr_transport_addr *alternate_server);
+int nr_stun_message_add_error_code_attribute(nr_stun_message *msg, UINT2 number, char *reason);
+int nr_stun_message_add_fingerprint_attribute(nr_stun_message *msg);
+int nr_stun_message_add_message_integrity_attribute(nr_stun_message *msg, Data *password);
+int nr_stun_message_add_nonce_attribute(nr_stun_message *msg, char *nonce);
+int nr_stun_message_add_realm_attribute(nr_stun_message *msg, char *realm);
+int nr_stun_message_add_server_attribute(nr_stun_message *msg, char *server_name);
+int nr_stun_message_add_unknown_attributes_attribute(nr_stun_message *msg, nr_stun_attr_unknown_attributes *unknown_attributes);
+int nr_stun_message_add_username_attribute(nr_stun_message *msg, char *username);
+int nr_stun_message_add_xor_mapped_address_attribute(nr_stun_message *msg, nr_transport_addr *mapped_address);
+
+#ifdef USE_ICE
+int nr_stun_message_add_ice_controlled_attribute(nr_stun_message *msg, UINT8 ice_controlled);
+int nr_stun_message_add_ice_controlling_attribute(nr_stun_message *msg, UINT8 ice_controlling);
+int nr_stun_message_add_priority_attribute(nr_stun_message *msg, UINT4 priority);
+int nr_stun_message_add_use_candidate_attribute(nr_stun_message *msg);
+#endif /* USE_ICE */
+
+#ifdef USE_TURN
+int nr_stun_message_add_data_attribute(nr_stun_message *msg, UCHAR *data, int length);
+int nr_stun_message_add_lifetime_attribute(nr_stun_message *msg, UINT4 lifetime_secs);
+int nr_stun_message_add_requested_transport_attribute(nr_stun_message *msg, UCHAR transport);
+int
+nr_stun_message_add_xor_peer_address_attribute(nr_stun_message *msg, nr_transport_addr *peer_address);
+#endif /* USE_TURN */
+
+#ifdef USE_STUND_0_96
+int nr_stun_message_add_change_request_attribute(nr_stun_message *msg, UINT4 change_request);
+#endif /* USE_STUND_0_96 */
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_proc.c b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_proc.c
new file mode 100644
index 0000000000..13366e265d
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_proc.c
@@ -0,0 +1,554 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <errno.h>
+#include <csi_platform.h>
+
+#ifdef WIN32
+#include <winsock2.h>
+#include <stdlib.h>
+#include <io.h>
+#include <time.h>
+#else   /* UNIX */
+#include <string.h>
+#endif  /* end UNIX */
+#include <assert.h>
+
+#include "stun.h"
+#include "stun_reg.h"
+#include "registry.h"
+
+static int
+nr_stun_add_realm_and_nonce(int new_nonce, nr_stun_server_client *clnt, nr_stun_message *res);
+
+/* draft-ietf-behave-rfc3489bis-10.txt S 7.3 */
+int
+nr_stun_receive_message(nr_stun_message *req, nr_stun_message *msg)
+{
+    int _status;
+    nr_stun_message_attribute *attr;
+
+#ifdef USE_RFC_3489_BACKWARDS_COMPATIBLE
+    /* if this message was generated by an RFC 3489 impementation,
+     * the call to nr_is_stun_message will fail, so skip that
+     * check and puke elsewhere if the message can't be decoded */
+    if (msg->header.magic_cookie == NR_STUN_MAGIC_COOKIE
+     || msg->header.magic_cookie == NR_STUN_MAGIC_COOKIE2) {
+#endif /* USE_RFC_3489_BACKWARDS_COMPATIBLE */
+    if (!nr_is_stun_message(msg->buffer, msg->length)) {
+        r_log(NR_LOG_STUN, LOG_WARNING, "Not a STUN message");
+        ABORT(R_REJECTED);
+    }
+#ifdef USE_RFC_3489_BACKWARDS_COMPATIBLE
+    }
+#endif /* USE_RFC_3489_BACKWARDS_COMPATIBLE */
+
+    if (req == 0) {
+        if (NR_STUN_GET_TYPE_CLASS(msg->header.type) != NR_CLASS_REQUEST) {
+            r_log(NR_LOG_STUN,LOG_WARNING,"Illegal message type: %03x", msg->header.type);
+            ABORT(R_REJECTED);
+        }
+    }
+    else {
+        if (NR_STUN_GET_TYPE_CLASS(msg->header.type) != NR_CLASS_RESPONSE
+         && NR_STUN_GET_TYPE_CLASS(msg->header.type) != NR_CLASS_ERROR_RESPONSE) {
+            r_log(NR_LOG_STUN,LOG_WARNING,"Illegal message class: %03x", msg->header.type);
+            ABORT(R_REJECTED);
+        }
+
+        if (NR_STUN_GET_TYPE_METHOD(req->header.type) != NR_STUN_GET_TYPE_METHOD(msg->header.type)) {
+            r_log(NR_LOG_STUN,LOG_WARNING,"Inconsistent message method: %03x expected %03x", msg->header.type, req->header.type);
+            ABORT(R_REJECTED);
+        }
+
+        if (nr_stun_different_transaction(msg->buffer, msg->length, req)) {
+            r_log(NR_LOG_STUN, LOG_DEBUG, "Unrecognized STUN transaction");
+            ABORT(R_REJECTED);
+        }
+    }
+
+    switch (msg->header.magic_cookie) {
+    case NR_STUN_MAGIC_COOKIE:
+        /* basically draft-ietf-behave-rfc3489bis-10.txt S 6 rules */
+
+        if (nr_stun_message_has_attribute(msg, NR_STUN_ATTR_FINGERPRINT, &attr)
+         && !attr->u.fingerprint.valid) {
+            r_log(NR_LOG_STUN, LOG_WARNING, "Invalid fingerprint");
+            ABORT(R_REJECTED);
+        }
+
+        break;
+
+#ifdef USE_STUND_0_96
+    case NR_STUN_MAGIC_COOKIE2:
+        /* nothing to check in this case */
+        break;
+#endif /* USE_STUND_0_96 */
+
+    default:
+#ifdef USE_RFC_3489_BACKWARDS_COMPATIBLE
+        /* in RFC 3489 there is no magic cookie, it's part of the transaction ID */
+#else
+#ifdef NDEBUG
+        /* in deployment builds we should always see a recognized magic cookie */
+        r_log(NR_LOG_STUN, LOG_WARNING, "Missing Magic Cookie");
+        ABORT(R_REJECTED);
+#else
+        /* ignore this condition because sometimes we like to pretend we're
+         * a server talking to old clients and their messages don't contain
+         * a magic cookie at all but rather the magic cookie field is part
+         * of their ID and therefore random */
+#endif /* NDEBUG */
+#endif /* USE_RFC_3489_BACKWARDS_COMPATIBLE */
+        break;
+    }
+
+    _status=0;
+  abort:
+    return _status;
+}
+
+/* draft-ietf-behave-rfc3489bis-10.txt S 7.3.1 */
+int
+nr_stun_process_request(nr_stun_message *req, nr_stun_message *res)
+{
+    int _status;
+#ifdef USE_STUN_PEDANTIC
+    int r;
+    nr_stun_attr_unknown_attributes unknown_attributes = { { 0 } };
+    nr_stun_message_attribute *attr;
+
+    if (req->comprehension_required_unknown_attributes > 0) {
+        nr_stun_form_error_response(req, res, 420, "Unknown Attributes");
+        r_log(NR_LOG_STUN, LOG_WARNING, "Request contains comprehension required but unknown attributes");
+
+        TAILQ_FOREACH(attr, &req->attributes, entry) {
+            if (attr->name == 0) {
+                /* unrecognized attribute */
+
+                /* should never happen, but truncate if it ever were to occur */
+                if (unknown_attributes.num_attributes > NR_STUN_MAX_UNKNOWN_ATTRIBUTES)
+                    break;
+
+                unknown_attributes.attribute[unknown_attributes.num_attributes++] = attr->type;
+            }
+        }
+
+        assert(req->comprehension_required_unknown_attributes + req->comprehension_optional_unknown_attributes == unknown_attributes.num_attributes);
+
+        if ((r=nr_stun_message_add_unknown_attributes_attribute(res, &unknown_attributes)))
+            ABORT(R_ALREADY);
+
+        ABORT(R_ALREADY);
+    }
+#endif /* USE_STUN_PEDANTIC */
+
+    _status=0;
+#ifdef USE_STUN_PEDANTIC
+  abort:
+#endif /* USE_STUN_PEDANTIC */
+    return _status;
+}
+
+/* draft-ietf-behave-rfc3489bis-10.txt S 7.3.2 */
+int
+nr_stun_process_indication(nr_stun_message *ind)
+{
+    int _status;
+#ifdef USE_STUN_PEDANTIC
+
+    if (ind->comprehension_required_unknown_attributes > 0) {
+        r_log(NR_LOG_STUN, LOG_WARNING, "Indication contains comprehension required but unknown attributes");
+        ABORT(R_REJECTED);
+    }
+#endif /* USE_STUN_PEDANTIC */
+
+    _status=0;
+#ifdef USE_STUN_PEDANTIC
+  abort:
+#endif /* USE_STUN_PEDANTIC */
+    return _status;
+}
+
+/* RFC5389 S 7.3.3, except that we *also* allow a MAPPED_ADDRESS
+   to compensate for a bug in Google's STUN server where it
+   always returns MAPPED_ADDRESS.
+
+   Mozilla bug: 888274.
+ */
+int
+nr_stun_process_success_response(nr_stun_message *res)
+{
+    int _status;
+
+#ifdef USE_STUN_PEDANTIC
+    if (res->comprehension_required_unknown_attributes > 0) {
+        r_log(NR_LOG_STUN, LOG_WARNING, "Response contains comprehension required but unknown attributes");
+        ABORT(R_REJECTED);
+    }
+#endif /* USE_STUN_PEDANTIC */
+
+    if (NR_STUN_GET_TYPE_METHOD(res->header.type) == NR_METHOD_BINDING) {
+        if (! nr_stun_message_has_attribute(res, NR_STUN_ATTR_XOR_MAPPED_ADDRESS, 0) &&
+            ! nr_stun_message_has_attribute(res, NR_STUN_ATTR_MAPPED_ADDRESS, 0)) {
+            r_log(NR_LOG_STUN, LOG_WARNING, "Missing XOR-MAPPED-ADDRESS and MAPPED_ADDRESS");
+            ABORT(R_REJECTED);
+        }
+    }
+
+    _status=0;
+ abort:
+    return _status;
+}
+
+/* draft-ietf-behave-rfc3489bis-10.txt S 7.3.4 */
+int
+nr_stun_process_error_response(nr_stun_message *res, UINT2 *error_code)
+{
+    int _status;
+    nr_stun_message_attribute *attr;
+
+    if (res->comprehension_required_unknown_attributes > 0) {
+        r_log(NR_LOG_STUN, LOG_WARNING, "Error response contains comprehension required but unknown attributes");
+        ABORT(R_REJECTED);
+    }
+
+    if (! nr_stun_message_has_attribute(res, NR_STUN_ATTR_ERROR_CODE, &attr)) {
+        r_log(NR_LOG_STUN, LOG_WARNING, "Missing ERROR-CODE");
+        ABORT(R_REJECTED);
+    }
+
+    *error_code = attr->u.error_code.number;
+
+    switch (attr->u.error_code.number / 100) {
+    case 3:
+      /* We do not treat STUN/300 as retryable. The TURN Allocate handling
+       * code will reset the ctx if appropriate. */
+      ABORT(R_REJECTED);
+      break;
+
+    case 4:
+        /* If the error code is 400 through 499, the client declares the
+         * transaction failed; in the case of 420 (Unknown Attribute), the
+         * response should contain a UNKNOWN-ATTRIBUTES attribute that gives
+         * additional information. */
+        if (attr->u.error_code.number == 420)
+            ABORT(R_REJECTED);
+
+        /* it may be possible to restart given the info that was received in
+         * this response, so retry */
+        ABORT(R_RETRY);
+        break;
+
+    case 5:
+        /* If the error code is 500 through 599, the client MAY resend the
+         * request; clients that do so MUST limit the number of times they do
+         * this. */
+        /* let the retransmit mechanism handle resending the request */
+        break;
+
+    default:
+        ABORT(R_REJECTED);
+        break;
+    }
+
+    /* the spec says: "The client then does any processing specified by the authentication
+     * mechanism (see Section 10).  This may result in a new transaction
+     * attempt." -- but this is handled already elsewhere, so needn't be repeated
+     * in this function */
+
+    _status=0;
+ abort:
+    return _status;
+}
+
+/* draft-ietf-behave-rfc3489bis-10.txt S 10.1.2 */
+int
+nr_stun_receive_request_or_indication_short_term_auth(nr_stun_message *msg,
+                                                      nr_stun_message *res)
+{
+    int _status;
+    nr_stun_message_attribute *attr;
+
+    switch (msg->header.magic_cookie) {
+    default:
+        /* in RFC 3489 there is no magic cookie, it's part of the transaction ID */
+        /* drop thru */
+    case NR_STUN_MAGIC_COOKIE:
+        if (!nr_stun_message_has_attribute(msg, NR_STUN_ATTR_MESSAGE_INTEGRITY, &attr)) {
+            nr_stun_form_error_response(msg, res, 400, "Missing MESSAGE-INTEGRITY");
+            ABORT(R_ALREADY);
+        }
+
+        if (!nr_stun_message_has_attribute(msg, NR_STUN_ATTR_USERNAME, 0)) {
+            nr_stun_form_error_response(msg, res, 400, "Missing USERNAME");
+            ABORT(R_ALREADY);
+        }
+
+        if (attr->u.message_integrity.unknown_user) {
+            nr_stun_form_error_response(msg, res, 401, "Unrecognized USERNAME");
+            ABORT(R_ALREADY);
+        }
+
+        if (!attr->u.message_integrity.valid) {
+            nr_stun_form_error_response(msg, res, 401, "Bad MESSAGE-INTEGRITY");
+            ABORT(R_ALREADY);
+        }
+
+        break;
+
+#ifdef USE_STUND_0_96
+    case NR_STUN_MAGIC_COOKIE2:
+        /* nothing to check in this case */
+        break;
+#endif /* USE_STUND_0_96 */
+    }
+
+    _status=0;
+ abort:
+    return _status;
+}
+
+/* draft-ietf-behave-rfc3489bis-10.txt S 10.1.3 */
+int
+nr_stun_receive_response_short_term_auth(nr_stun_message *res)
+{
+    int _status;
+    nr_stun_message_attribute *attr;
+
+    switch (res->header.magic_cookie) {
+    default:
+        /* in RFC 3489 there is no magic cookie, it's part of the transaction ID */
+        /* drop thru */
+    case NR_STUN_MAGIC_COOKIE:
+        if (!nr_stun_message_has_attribute(res, NR_STUN_ATTR_MESSAGE_INTEGRITY, &attr)) {
+            r_log(NR_LOG_STUN, LOG_WARNING, "Missing MESSAGE-INTEGRITY");
+            ABORT(R_REJECTED);
+        }
+
+        if (!attr->u.message_integrity.valid) {
+            r_log(NR_LOG_STUN, LOG_WARNING, "Bad MESSAGE-INTEGRITY");
+            ABORT(R_REJECTED);
+        }
+
+        break;
+
+#ifdef USE_STUND_0_96
+    case NR_STUN_MAGIC_COOKIE2:
+        /* nothing to check in this case */
+        break;
+#endif /* USE_STUND_0_96 */
+    }
+
+   _status=0;
+ abort:
+     return _status;
+}
+
+static int
+nr_stun_add_realm_and_nonce(int new_nonce, nr_stun_server_client *clnt, nr_stun_message *res)
+{
+    int r,_status;
+    char *realm = 0;
+    char *nonce;
+    UINT2 size;
+
+    if ((r=NR_reg_alloc_string(NR_STUN_REG_PREF_SERVER_REALM, &realm)))
+        ABORT(r);
+
+    if ((r=nr_stun_message_add_realm_attribute(res, realm)))
+        ABORT(r);
+
+    if (clnt) {
+        if (strlen(clnt->nonce) < 1)
+            new_nonce = 1;
+
+        if (new_nonce) {
+            if (NR_reg_get_uint2(NR_STUN_REG_PREF_SERVER_NONCE_SIZE, &size))
+                size = 48;
+
+            if (size > (sizeof(clnt->nonce) - 1))
+                size = sizeof(clnt->nonce) - 1;
+
+            nr_random_alphanum(clnt->nonce, size);
+            clnt->nonce[size] = '\0';
+        }
+
+        nonce = clnt->nonce;
+    }
+    else {
+        /* user is not known, so use a bogus nonce since there's no way to
+         * store a good nonce with the client-specific data -- this nonce
+         * will be recognized as stale if the client attempts another
+         * request */
+        nonce = "STALE";
+    }
+
+    if ((r=nr_stun_message_add_nonce_attribute(res, nonce)))
+        ABORT(r);
+
+    _status=0;
+ abort:
+#ifdef USE_TURN
+assert(_status == 0); /* TODO: !nn! cleanup after I reimplmement TURN */
+#endif
+    RFREE(realm);
+    return _status;
+}
+
+/* draft-ietf-behave-rfc3489bis-10.txt S 10.2.1 - 10.2.2 */
+int
+nr_stun_receive_request_long_term_auth(nr_stun_message *req, nr_stun_server_ctx *ctx, nr_stun_message *res)
+{
+    int r,_status;
+    nr_stun_message_attribute *mi;
+    nr_stun_message_attribute *n;
+    nr_stun_server_client *clnt = 0;
+
+    switch (req->header.magic_cookie) {
+    default:
+        /* in RFC 3489 there is no magic cookie, it's part of the transaction ID */
+        /* drop thru */
+    case NR_STUN_MAGIC_COOKIE:
+        if (!nr_stun_message_has_attribute(req, NR_STUN_ATTR_USERNAME, 0)) {
+            nr_stun_form_error_response(req, res, 400, "Missing USERNAME");
+            nr_stun_add_realm_and_nonce(0, 0, res);
+            ABORT(R_ALREADY);
+        }
+
+        if ((r=nr_stun_get_message_client(ctx, req, &clnt))) {
+            nr_stun_form_error_response(req, res, 401, "Unrecognized USERNAME");
+            nr_stun_add_realm_and_nonce(0, 0, res);
+            ABORT(R_ALREADY);
+        }
+
+        if (!nr_stun_message_has_attribute(req, NR_STUN_ATTR_MESSAGE_INTEGRITY, &mi)) {
+            nr_stun_form_error_response(req, res, 401, "Missing MESSAGE-INTEGRITY");
+            nr_stun_add_realm_and_nonce(0, clnt, res);
+            ABORT(R_ALREADY);
+        }
+
+        assert(!mi->u.message_integrity.unknown_user);
+
+        if (!nr_stun_message_has_attribute(req, NR_STUN_ATTR_REALM, 0)) {
+            nr_stun_form_error_response(req, res, 400, "Missing REALM");
+            ABORT(R_ALREADY);
+        }
+
+        if (!nr_stun_message_has_attribute(req, NR_STUN_ATTR_NONCE, &n)) {
+            nr_stun_form_error_response(req, res, 400, "Missing NONCE");
+            ABORT(R_ALREADY);
+        }
+
+        assert(sizeof(clnt->nonce) == sizeof(n->u.nonce));
+        if (strncmp(clnt->nonce, n->u.nonce, sizeof(n->u.nonce))) {
+            nr_stun_form_error_response(req, res, 438, "Stale NONCE");
+            nr_stun_add_realm_and_nonce(1, clnt, res);
+            ABORT(R_ALREADY);
+        }
+
+        if (!mi->u.message_integrity.valid) {
+            nr_stun_form_error_response(req, res, 401, "Bad MESSAGE-INTEGRITY");
+            nr_stun_add_realm_and_nonce(0, clnt, res);
+            ABORT(R_ALREADY);
+        }
+
+        break;
+
+#ifdef USE_STUND_0_96
+    case NR_STUN_MAGIC_COOKIE2:
+        /* nothing to do in this case */
+        break;
+#endif /* USE_STUND_0_96 */
+    }
+
+    _status=0;
+ abort:
+
+    return _status;
+}
+
+/* draft-ietf-behave-rfc3489bis-10.txt S 10.2.3 */
+int
+nr_stun_receive_response_long_term_auth(nr_stun_message *res, nr_stun_client_ctx *ctx)
+{
+    int _status;
+    nr_stun_message_attribute *attr;
+
+    switch (res->header.magic_cookie) {
+    default:
+        /* in RFC 3489 there is no magic cookie, it's part of the transaction ID */
+        /* drop thru */
+    case NR_STUN_MAGIC_COOKIE:
+        if (nr_stun_message_has_attribute(res, NR_STUN_ATTR_REALM, &attr)) {
+            RFREE(ctx->realm);
+            ctx->realm = r_strdup(attr->u.realm);
+            if (!ctx->realm)
+                ABORT(R_NO_MEMORY);
+        }
+        else {
+            r_log(NR_LOG_STUN, LOG_WARNING, "Missing REALM");
+            ABORT(R_REJECTED);
+        }
+
+        if (nr_stun_message_has_attribute(res, NR_STUN_ATTR_NONCE, &attr)) {
+            RFREE(ctx->nonce);
+            ctx->nonce = r_strdup(attr->u.nonce);
+            if (!ctx->nonce)
+                ABORT(R_NO_MEMORY);
+        }
+        else {
+            r_log(NR_LOG_STUN, LOG_WARNING, "Missing NONCE");
+            ABORT(R_REJECTED);
+        }
+
+        if (nr_stun_message_has_attribute(res, NR_STUN_ATTR_MESSAGE_INTEGRITY, &attr)) {
+            if (!attr->u.message_integrity.valid) {
+                r_log(NR_LOG_STUN, LOG_WARNING, "Bad MESSAGE-INTEGRITY");
+                ABORT(R_REJECTED);
+            }
+        }
+
+        break;
+
+#ifdef USE_STUND_0_96
+    case NR_STUN_MAGIC_COOKIE2:
+        /* nothing to check in this case */
+        break;
+#endif /* USE_STUND_0_96 */
+    }
+
+    _status=0;
+ abort:
+    return _status;
+}
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_proc.h b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_proc.h
new file mode 100644
index 0000000000..5975670779
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_proc.h
@@ -0,0 +1,53 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _stun_proc_h
+#define _stun_proc_h
+
+#include "stun.h"
+
+int nr_stun_receive_message(nr_stun_message *req, nr_stun_message *msg);
+int nr_stun_process_request(nr_stun_message *req, nr_stun_message *res);
+int nr_stun_process_indication(nr_stun_message *ind);
+int nr_stun_process_success_response(nr_stun_message *res);
+int nr_stun_process_error_response(nr_stun_message *res, UINT2 *error_code);
+
+int nr_stun_receive_request_or_indication_short_term_auth(nr_stun_message *msg, nr_stun_message *res);
+int nr_stun_receive_response_short_term_auth(nr_stun_message *res);
+
+int nr_stun_receive_request_long_term_auth(nr_stun_message *req, nr_stun_server_ctx *ctx, nr_stun_message *res);
+int nr_stun_receive_response_long_term_auth(nr_stun_message *res, nr_stun_client_ctx *ctx);
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_reg.h b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_reg.h
new file mode 100644
index 0000000000..2167fcd91b
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_reg.h
@@ -0,0 +1,58 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _stun_reg_h
+#define _stun_reg_h
+#ifdef __cplusplus
+using namespace std;
+extern "C" {
+#endif /* __cplusplus */
+
+#define NR_STUN_REG_PREF_CLNT_RETRANSMIT_TIMEOUT    "stun.client.retransmission_timeout"
+#define NR_STUN_REG_PREF_CLNT_RETRANSMIT_BACKOFF    "stun.client.retransmission_backoff_factor"
+#define NR_STUN_REG_PREF_CLNT_MAXIMUM_TRANSMITS     "stun.client.maximum_transmits"
+#define NR_STUN_REG_PREF_CLNT_FINAL_RETRANSMIT_BACKOFF   "stun.client.final_retransmit_backoff"
+
+#define NR_STUN_REG_PREF_ALLOW_LOOPBACK_ADDRS            "stun.allow_loopback"
+#define NR_STUN_REG_PREF_ALLOW_LINK_LOCAL_ADDRS     "stun.allow_link_local"
+#define NR_STUN_REG_PREF_ADDRESS_PRFX               "stun.address"
+#define NR_STUN_REG_PREF_SERVER_NAME                "stun.server.name"
+#define NR_STUN_REG_PREF_SERVER_NONCE_SIZE          "stun.server.nonce_size"
+#define NR_STUN_REG_PREF_SERVER_REALM               "stun.server.realm"
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_server_ctx.c b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_server_ctx.c
new file mode 100644
index 0000000000..b92b6b5ab6
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_server_ctx.c
@@ -0,0 +1,468 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <string.h>
+#include <assert.h>
+
+#include "nr_api.h"
+#include "stun.h"
+
+static int nr_stun_server_destroy_client(nr_stun_server_client *clnt);
+static int nr_stun_server_send_response(nr_stun_server_ctx *ctx, nr_socket *sock, nr_transport_addr *peer_addr, nr_stun_message *res, nr_stun_server_client *clnt);
+static int nr_stun_server_process_request_auth_checks(nr_stun_server_ctx *ctx, nr_stun_message *req, int auth_rule, nr_stun_message *res);
+
+
+int nr_stun_server_ctx_create(char *label, nr_stun_server_ctx **ctxp)
+  {
+    int r,_status;
+    nr_stun_server_ctx *ctx=0;
+
+    if ((r=nr_stun_startup()))
+      ABORT(r);
+
+    if(!(ctx=RCALLOC(sizeof(nr_stun_server_ctx))))
+      ABORT(R_NO_MEMORY);
+
+    if(!(ctx->label=r_strdup(label)))
+      ABORT(R_NO_MEMORY);
+
+    STAILQ_INIT(&ctx->clients);
+
+    *ctxp=ctx;
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_stun_server_ctx_destroy(nr_stun_server_ctx **ctxp)
+  {
+    nr_stun_server_ctx *ctx;
+    nr_stun_server_client *clnt1,*clnt2;
+
+    if(!ctxp || !*ctxp)
+      return(0);
+
+    ctx=*ctxp;
+
+    STAILQ_FOREACH_SAFE(clnt1, &ctx->clients, entry, clnt2) {
+        nr_stun_server_destroy_client(clnt1);
+    }
+
+    nr_stun_server_destroy_client(ctx->default_client);
+
+    RFREE(ctx->label);
+    RFREE(ctx);
+
+    return(0);
+  }
+
+static int nr_stun_server_client_create(nr_stun_server_ctx *ctx, char *client_label, char *user, Data *pass, nr_stun_server_cb cb, void *cb_arg, nr_stun_server_client **clntp)
+  {
+    nr_stun_server_client *clnt=0;
+    int r,_status;
+
+    if(!(clnt=RCALLOC(sizeof(nr_stun_server_client))))
+      ABORT(R_NO_MEMORY);
+
+    if(!(clnt->label=r_strdup(client_label)))
+      ABORT(R_NO_MEMORY);
+
+    if(!(clnt->username=r_strdup(user)))
+      ABORT(R_NO_MEMORY);
+
+    if(r=r_data_copy(&clnt->password,pass))
+      ABORT(r);
+
+    r_log(NR_LOG_STUN,LOG_DEBUG,"STUN-SERVER(%s)/CLIENT(%s): Adding client for %s",ctx->label, client_label, user);
+    clnt->stun_server_cb=cb;
+    clnt->cb_arg=cb_arg;
+
+    *clntp = clnt;
+    _status=0;
+ abort:
+    if(_status){
+      nr_stun_server_destroy_client(clnt);
+    }
+    return(_status);
+  }
+
+int nr_stun_server_add_client(nr_stun_server_ctx *ctx, char *client_label, char *user, Data *pass, nr_stun_server_cb cb, void *cb_arg)
+  {
+   int r,_status;
+   nr_stun_server_client *clnt;
+
+    if (r=nr_stun_server_client_create(ctx, client_label, user, pass, cb, cb_arg, &clnt))
+      ABORT(r);
+
+    STAILQ_INSERT_TAIL(&ctx->clients,clnt,entry);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_stun_server_add_default_client(nr_stun_server_ctx *ctx, char *ufrag, Data *pass, nr_stun_server_cb cb, void *cb_arg)
+  {
+    int r,_status;
+    nr_stun_server_client *clnt;
+
+    assert(!ctx->default_client);
+    if (ctx->default_client)
+      ABORT(R_INTERNAL);
+
+    if (r=nr_stun_server_client_create(ctx, "default_client", ufrag, pass, cb, cb_arg, &clnt))
+      ABORT(r);
+
+    ctx->default_client = clnt;
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_stun_server_remove_client(nr_stun_server_ctx *ctx, void *cb_arg)
+  {
+    nr_stun_server_client *clnt1,*clnt2;
+    int found = 0;
+
+    STAILQ_FOREACH_SAFE(clnt1, &ctx->clients, entry, clnt2) {
+      if(clnt1->cb_arg == cb_arg) {
+        STAILQ_REMOVE(&ctx->clients, clnt1, nr_stun_server_client_, entry);
+        nr_stun_server_destroy_client(clnt1);
+        found++;
+      }
+    }
+
+    if (!found)
+      ERETURN(R_NOT_FOUND);
+
+    return 0;
+  }
+
+static int nr_stun_server_get_password(void *arg, nr_stun_message *msg, Data **password)
+  {
+    int _status;
+    nr_stun_server_ctx *ctx = (nr_stun_server_ctx*)arg;
+    nr_stun_server_client *clnt = 0;
+    nr_stun_message_attribute *username_attribute;
+
+    if ((nr_stun_get_message_client(ctx, msg, &clnt))) {
+        if (! nr_stun_message_has_attribute(msg, NR_STUN_ATTR_USERNAME, &username_attribute)) {
+           r_log(NR_LOG_STUN,LOG_WARNING,"STUN-SERVER(%s): Missing Username",ctx->label);
+           ABORT(R_NOT_FOUND);
+        }
+
+        /* Although this is an exceptional condition, we'll already have seen a
+         * NOTICE-level log message about the unknown user, so additional log
+         * messages at any level higher than DEBUG are unnecessary. */
+
+        r_log(NR_LOG_STUN,LOG_DEBUG,"STUN-SERVER(%s): Unable to find password for unknown user: %s",ctx->label,username_attribute->u.username);
+        ABORT(R_NOT_FOUND);
+    }
+
+    *password = &clnt->password;
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_stun_server_process_request_auth_checks(nr_stun_server_ctx *ctx, nr_stun_message *req, int auth_rule, nr_stun_message *res)
+  {
+    int r,_status;
+
+    if (nr_stun_message_has_attribute(req, NR_STUN_ATTR_MESSAGE_INTEGRITY, 0)
+     || !(auth_rule & NR_STUN_AUTH_RULE_OPTIONAL)) {
+        /* favor long term credentials over short term, if both are supported */
+
+        if (auth_rule & NR_STUN_AUTH_RULE_LONG_TERM) {
+            if ((r=nr_stun_receive_request_long_term_auth(req, ctx, res)))
+                ABORT(r);
+        }
+        else if (auth_rule & NR_STUN_AUTH_RULE_SHORT_TERM) {
+            if ((r=nr_stun_receive_request_or_indication_short_term_auth(req, res)))
+                ABORT(r);
+        }
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_stun_server_process_request(nr_stun_server_ctx *ctx, nr_socket *sock, char *msg, int len, nr_transport_addr *peer_addr, int auth_rule)
+  {
+    int r,_status;
+    char string[256];
+    nr_stun_message *req = 0;
+    nr_stun_message *res = 0;
+    nr_stun_server_client *clnt = 0;
+    nr_stun_server_request info;
+    int error;
+    int dont_free = 0;
+    nr_transport_addr my_addr;
+
+    if ((r=nr_socket_getaddr(sock, &my_addr))) {
+      ABORT(r);
+    }
+
+    r_log(NR_LOG_STUN,LOG_DEBUG,"STUN-SERVER(%s): Received(my_addr=%s,peer_addr=%s)",ctx->label,my_addr.as_string,peer_addr->as_string);
+
+    snprintf(string, sizeof(string)-1, "STUN-SERVER(%s): Received ", ctx->label);
+    r_dump(NR_LOG_STUN, LOG_DEBUG, string, (char*)msg, len);
+
+    memset(&info,0,sizeof(info));
+
+    if ((r=nr_stun_message_create2(&req, (UCHAR*)msg, len)))
+        ABORT(r);
+
+    if ((r=nr_stun_message_create(&res)))
+        ABORT(r);
+
+    if ((r=nr_stun_decode_message(req, nr_stun_server_get_password, ctx))) {
+        /* RFC5389 S 7.3 says "If any errors are detected, the message is
+         * silently discarded."  */
+#ifndef USE_STUN_PEDANTIC
+        /* ... but that seems like a bad idea, at least return a 400 so
+         * that the server isn't a black hole to the client */
+        nr_stun_form_error_response(req, res, 400, "Bad Request - Failed to decode request");
+        ABORT(R_ALREADY);
+#endif /* USE_STUN_PEDANTIC */
+        ABORT(R_REJECTED);
+    }
+
+    if ((r=nr_stun_receive_message(0, req))) {
+        /* RFC5389 S 7.3 says "If any errors are detected, the message is
+         * silently discarded."  */
+#ifndef USE_STUN_PEDANTIC
+        /* ... but that seems like a bad idea, at least return a 400 so
+         * that the server isn't a black hole to the client */
+        nr_stun_form_error_response(req, res, 400, "Bad Request - Section 7.3 check failed");
+        ABORT(R_ALREADY);
+#endif /* USE_STUN_PEDANTIC */
+        ABORT(R_REJECTED);
+    }
+
+    if (NR_STUN_GET_TYPE_CLASS(req->header.type) != NR_CLASS_REQUEST
+     && NR_STUN_GET_TYPE_CLASS(req->header.type) != NR_CLASS_INDICATION) {
+         r_log(NR_LOG_STUN,LOG_WARNING,"STUN-SERVER(%s): Illegal message type: %04x",ctx->label,req->header.type);
+        /* RFC5389 S 7.3 says "If any errors are detected, the message is
+         * silently discarded."  */
+#ifndef USE_STUN_PEDANTIC
+        /* ... but that seems like a bad idea, at least return a 400 so
+         * that the server isn't a black hole to the client */
+        nr_stun_form_error_response(req, res, 400, "Bad Request - Unsupported message type");
+        ABORT(R_ALREADY);
+#endif /* USE_STUN_PEDANTIC */
+        ABORT(R_REJECTED);
+    }
+
+    /* "The STUN agent then does any checks that are required by a
+     * authentication mechanism that the usage has specified" */
+    if ((r=nr_stun_server_process_request_auth_checks(ctx, req, auth_rule, res)))
+        ABORT(r);
+
+    if (NR_STUN_GET_TYPE_CLASS(req->header.type) == NR_CLASS_INDICATION) {
+        if ((r=nr_stun_process_indication(req)))
+            ABORT(r);
+    }
+    else {
+        if ((r=nr_stun_process_request(req, res)))
+            ABORT(r);
+    }
+
+    assert(res->header.type == 0);
+
+    clnt = 0;
+    if (NR_STUN_GET_TYPE_CLASS(req->header.type) == NR_CLASS_REQUEST) {
+        if ((nr_stun_get_message_client(ctx, req, &clnt))) {
+            if ((r=nr_stun_form_success_response(req, peer_addr, 0, res)))
+                ABORT(r);
+        }
+        else {
+            if ((r=nr_stun_form_success_response(req, peer_addr, &clnt->password, res)))
+                ABORT(r);
+        }
+    }
+
+    if(clnt && clnt->stun_server_cb){
+        r_log(NR_LOG_STUN,LOG_DEBUG,"Entering STUN server callback");
+
+        /* Set up the info */
+        if(r=nr_transport_addr_copy(&info.src_addr,peer_addr))
+            ABORT(r);
+
+        info.request = req;
+        info.response = res;
+
+        error = 0;
+        dont_free = 0;
+        if (clnt->stun_server_cb(clnt->cb_arg,ctx,sock,&info,&dont_free,&error)) {
+            if (error == 0)
+                error = 500;
+
+            nr_stun_form_error_response(req, res, error, "ICE Failure");
+            ABORT(R_ALREADY);
+        }
+    }
+
+    _status=0;
+  abort:
+    if (!res)
+        goto skip_response;
+
+    if (NR_STUN_GET_TYPE_CLASS(req->header.type) == NR_CLASS_INDICATION)
+        goto skip_response;
+
+    /* Now respond */
+
+    if (_status != 0 && ! nr_stun_message_has_attribute(res, NR_STUN_ATTR_ERROR_CODE, 0))
+         nr_stun_form_error_response(req, res, 500, "Failed to specify error");
+
+    if ((r=nr_stun_server_send_response(ctx, sock, peer_addr, res, clnt))) {
+        r_log(NR_LOG_STUN,LOG_ERR,"STUN-SERVER(label=%s): Failed sending response (my_addr=%s,peer_addr=%s)",ctx->label,my_addr.as_string,peer_addr->as_string);
+        _status = R_FAILED;
+    }
+
+#if 0
+    /* EKR: suppressed these checks because if you have an error when
+       you are sending an error, things go wonky */
+#ifdef SANITY_CHECKS
+    if (_status == R_ALREADY) {
+        assert(NR_STUN_GET_TYPE_CLASS(res->header.type) == NR_CLASS_ERROR_RESPONSE);
+        assert(nr_stun_message_has_attribute(res, NR_STUN_ATTR_ERROR_CODE, 0));
+    }
+    else {
+        assert(NR_STUN_GET_TYPE_CLASS(res->header.type) == NR_CLASS_RESPONSE);
+        assert(!nr_stun_message_has_attribute(res, NR_STUN_ATTR_ERROR_CODE, 0));
+    }
+#endif /* SANITY_CHECKS */
+#endif
+
+    if (0) {
+  skip_response:
+        _status = 0;
+    }
+
+    if (!dont_free) {
+      nr_stun_message_destroy(&res);
+      nr_stun_message_destroy(&req);
+    }
+
+    return(_status);
+  }
+
+static int nr_stun_server_send_response(nr_stun_server_ctx *ctx, nr_socket *sock, nr_transport_addr *peer_addr, nr_stun_message *res, nr_stun_server_client *clnt)
+  {
+    int r,_status;
+    char string[256];
+    nr_transport_addr my_addr;
+
+    if ((r=nr_socket_getaddr(sock, &my_addr))) {
+      ABORT(r);
+    }
+
+    r_log(NR_LOG_STUN,LOG_DEBUG,"STUN-SERVER(label=%s): Sending(my_addr=%s,peer_addr=%s)",ctx->label,my_addr.as_string,peer_addr->as_string);
+
+    if ((r=nr_stun_encode_message(res))) {
+        /* should never happen */
+        r_log(NR_LOG_STUN,LOG_ERR,"STUN-SERVER(label=%s): Unable to encode message", ctx->label);
+    }
+    else {
+        snprintf(string, sizeof(string)-1, "STUN(%s): Sending to %s ", ctx->label, peer_addr->as_string);
+        r_dump(NR_LOG_STUN, LOG_DEBUG, string, (char*)res->buffer, res->length);
+
+        if(r=nr_socket_sendto(sock,res->buffer,res->length,0,peer_addr))
+          ABORT(r);
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+static int nr_stun_server_destroy_client(nr_stun_server_client *clnt)
+  {
+    if (!clnt)
+      return 0;
+
+    RFREE(clnt->label);
+    RFREE(clnt->username);
+    r_data_zfree(&clnt->password);
+
+    RFREE(clnt);
+    return(0);
+  }
+
+int nr_stun_get_message_client(nr_stun_server_ctx *ctx, nr_stun_message *req, nr_stun_server_client **out)
+  {
+    int _status;
+    nr_stun_message_attribute *attr;
+    nr_stun_server_client *clnt=0;
+
+    if (! nr_stun_message_has_attribute(req, NR_STUN_ATTR_USERNAME, &attr)) {
+       r_log(NR_LOG_STUN,LOG_WARNING,"STUN-SERVER(%s): Missing Username",ctx->label);
+       ABORT(R_NOT_FOUND);
+    }
+
+    STAILQ_FOREACH(clnt, &ctx->clients, entry) {
+        if (!strncmp(clnt->username, attr->u.username,
+                     sizeof(attr->u.username)))
+            break;
+    }
+
+    if (!clnt && ctx->default_client) {
+      /* If we can't find a specific client see if this matches the default,
+         which means that the username starts with our ufrag.
+       */
+      char *colon = strchr(attr->u.username, ':');
+      if (colon && !strncmp(ctx->default_client->username,
+                            attr->u.username,
+                            colon - attr->u.username)) {
+        clnt = ctx->default_client;
+        r_log(NR_LOG_STUN,LOG_NOTICE,"STUN-SERVER(%s): Falling back to default client, username=: %s",ctx->label,attr->u.username);
+      }
+    }
+
+    if (!clnt) {
+        r_log(NR_LOG_STUN,LOG_WARNING,"STUN-SERVER(%s): Request from unknown user: %s",ctx->label,attr->u.username);
+        ABORT(R_NOT_FOUND);
+    }
+
+    *out = clnt;
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_server_ctx.h b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_server_ctx.h
new file mode 100644
index 0000000000..328675ee22
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_server_ctx.h
@@ -0,0 +1,80 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _stun_server_ctx_h
+#define _stun_server_ctx_h
+
+typedef struct nr_stun_server_ctx_ nr_stun_server_ctx;
+typedef struct nr_stun_server_client_  nr_stun_server_client;
+
+#include "stun_util.h"
+
+
+typedef struct nr_stun_server_request_{
+  nr_transport_addr src_addr;
+  nr_stun_message *request;
+  nr_stun_message *response;
+} nr_stun_server_request;
+
+typedef int (*nr_stun_server_cb)(void *cb_arg, struct nr_stun_server_ctx_ *ctx,nr_socket *sock,nr_stun_server_request *req, int *dont_free, int *error);
+
+struct nr_stun_server_client_ {
+  char *label;
+  char *username;
+  Data password;
+  nr_stun_server_cb stun_server_cb;
+  void *cb_arg;
+  char nonce[NR_STUN_MAX_NONCE_BYTES+1];  /* +1 for \0 */
+  STAILQ_ENTRY(nr_stun_server_client_) entry;
+};
+
+typedef STAILQ_HEAD(nr_stun_server_client_head_, nr_stun_server_client_) nr_stun_server_client_head;
+
+struct nr_stun_server_ctx_ {
+  char *label;
+  nr_stun_server_client_head clients;
+  nr_stun_server_client *default_client;
+};
+
+
+int nr_stun_server_ctx_create(char *label, nr_stun_server_ctx **ctxp);
+int nr_stun_server_ctx_destroy(nr_stun_server_ctx **ctxp);
+int nr_stun_server_add_client(nr_stun_server_ctx *ctx, char *client_label, char *user, Data *pass, nr_stun_server_cb cb, void *cb_arg);
+int nr_stun_server_remove_client(nr_stun_server_ctx *ctx, void *cb_arg);
+int nr_stun_server_add_default_client(nr_stun_server_ctx *ctx, char *ufrag, Data *pass, nr_stun_server_cb cb, void *cb_arg);
+int nr_stun_server_process_request(nr_stun_server_ctx *ctx, nr_socket *sock, char *msg, int len, nr_transport_addr *peer_addr, int auth_rule);
+int nr_stun_get_message_client(nr_stun_server_ctx *ctx, nr_stun_message *req, nr_stun_server_client **clnt);
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_util.c b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_util.c
new file mode 100644
index 0000000000..cbb0128924
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_util.c
@@ -0,0 +1,352 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <errno.h>
+#include <csi_platform.h>
+
+#ifdef WIN32
+#include <winsock2.h>
+#include <stdlib.h>
+#include <io.h>
+#include <time.h>
+#else   /* UNIX */
+#include <string.h>
+#endif  /* end UNIX */
+#include <assert.h>
+
+#include "stun.h"
+#include "stun_reg.h"
+#include "registry.h"
+#include "addrs.h"
+#include "transport_addr_reg.h"
+#include "nr_crypto.h"
+#include "hex.h"
+
+
+int NR_LOG_STUN = 0;
+
+int
+nr_stun_startup(void)
+{
+   int r,_status;
+
+    if ((r=r_log_register("stun", &NR_LOG_STUN)))
+      ABORT(r);
+
+   _status=0;
+ abort:
+   return _status;
+}
+
+int
+nr_stun_xor_mapped_address(UINT4 magicCookie, UINT12 transactionId, nr_transport_addr *from, nr_transport_addr *to)
+{
+    int _status;
+
+    switch (from->ip_version) {
+    case NR_IPV4:
+        nr_ip4_port_to_transport_addr(
+            (ntohl(from->u.addr4.sin_addr.s_addr) ^ magicCookie),
+            (ntohs(from->u.addr4.sin_port) ^ (magicCookie>>16)),
+            from->protocol, to);
+        break;
+    case NR_IPV6:
+        {
+          union {
+            unsigned char addr[16];
+            UINT4 addr32[4];
+          } maskedAddr;
+
+          maskedAddr.addr32[0] = htonl(magicCookie); /* Passed in host byte order */
+          memcpy(&maskedAddr.addr32[1], transactionId.octet, sizeof(transactionId));
+
+          /* We now have the mask in network byte order */
+          /* Xor the address in network byte order */
+          for (size_t i = 0; i < sizeof(maskedAddr); ++i) {
+            maskedAddr.addr[i] ^= from->u.addr6.sin6_addr.s6_addr[i];
+          }
+
+          nr_ip6_port_to_transport_addr(
+              (struct in6_addr*)&maskedAddr,
+              (ntohs(from->u.addr6.sin6_port) ^ (magicCookie>>16)),
+              from->protocol, to);
+        }
+        break;
+    default:
+        assert(0);
+        ABORT(R_INTERNAL);
+        break;
+    }
+
+    _status = 0;
+  abort:
+    return _status;
+}
+
+int
+nr_stun_filter_local_addresses(nr_local_addr addrs[], int *count)
+{
+    int r,_status;
+    char allow_loopback = 0;
+    char allow_link_local = 0;
+
+    if ((r=NR_reg_get_char(NR_STUN_REG_PREF_ALLOW_LOOPBACK_ADDRS,
+                           &allow_loopback))) {
+        if (r != R_NOT_FOUND) {
+            ABORT(r);
+        }
+    }
+
+    if ((r=NR_reg_get_char(NR_STUN_REG_PREF_ALLOW_LINK_LOCAL_ADDRS,
+                           &allow_link_local))) {
+        if (r != R_NOT_FOUND) {
+            ABORT(r);
+        }
+    }
+
+    if ((r=nr_stun_filter_addrs(addrs,
+                                !allow_loopback,
+                                !allow_link_local,
+                                count))) {
+        ABORT(r);
+    }
+
+    _status=0;
+ abort:
+    return _status;
+}
+
+int
+nr_stun_find_local_addresses(nr_local_addr addrs[], int maxaddrs, int *count)
+{
+    int r,_status;
+    //NR_registry *children = 0;
+
+    *count = 0;
+
+#if 0
+    // this really goes with the code commented out below. (mjf)
+    if ((r=NR_reg_get_child_count(NR_STUN_REG_PREF_ADDRESS_PRFX, (unsigned int*)count)))
+        if (r != R_NOT_FOUND)
+            ABORT(r);
+#endif
+
+    if (*count == 0) {
+        if ((r=nr_stun_get_addrs(addrs, maxaddrs, count)))
+            ABORT(r);
+
+        goto done;
+    }
+
+    if (*count >= maxaddrs) {
+        r_log(NR_LOG_STUN, LOG_INFO, "Address list truncated from %d to %d", *count, maxaddrs);
+       *count = maxaddrs;
+    }
+
+#if 0
+    if (*count > 0) {
+      /* TODO(ekr@rtfm.com): Commented out 2012-07-26.
+
+         This code is currently not used in Firefox and needs to be
+         ported to 64-bit */
+        children = RCALLOC((*count + 10) * sizeof(*children));
+        if (!children)
+            ABORT(R_NO_MEMORY);
+
+        assert(sizeof(size_t) == sizeof(*count));
+
+        if ((r=NR_reg_get_children(NR_STUN_REG_PREF_ADDRESS_PRFX, children, (size_t)(*count + 10), (size_t*)count)))
+            ABORT(r);
+
+        for (i = 0; i < *count; ++i) {
+            if ((r=nr_reg_get_transport_addr(children[i], 0, &addrs[i].addr)))
+                ABORT(r);
+        }
+    }
+#endif
+
+  done:
+
+     _status=0;
+ abort:
+     //RFREE(children);
+     return _status;
+}
+
+int
+nr_stun_different_transaction(UCHAR *msg, size_t len, nr_stun_message *req)
+{
+    int _status;
+    nr_stun_message_header header;
+    char reqid[44];
+    char msgid[44];
+    size_t unused;
+
+    if (sizeof(header) > len)
+        ABORT(R_FAILED);
+
+    assert(sizeof(header.id) == sizeof(UINT12));
+
+    memcpy(&header, msg, sizeof(header));
+
+    if (memcmp(&req->header.id, &header.id, sizeof(header.id))) {
+        nr_nbin2hex((UCHAR*)&req->header.id, sizeof(req->header.id), reqid, sizeof(reqid), &unused);
+        nr_nbin2hex((UCHAR*)&header.id, sizeof(header.id), msgid, sizeof(msgid), &unused);
+        r_log(NR_LOG_STUN, LOG_DEBUG, "Mismatched message IDs %s/%s", reqid, msgid);
+        ABORT(R_NOT_FOUND);
+    }
+
+   _status=0;
+ abort:
+   return _status;
+}
+
+char*
+nr_stun_msg_type(int type)
+{
+    char *ret = 0;
+
+    switch (type) {
+    case NR_STUN_MSG_BINDING_REQUEST:
+         ret = "BINDING-REQUEST";
+         break;
+    case NR_STUN_MSG_BINDING_INDICATION:
+         ret = "BINDING-INDICATION";
+         break;
+    case NR_STUN_MSG_BINDING_RESPONSE:
+         ret = "BINDING-RESPONSE";
+         break;
+    case NR_STUN_MSG_BINDING_ERROR_RESPONSE:
+         ret = "BINDING-ERROR-RESPONSE";
+         break;
+
+#ifdef USE_TURN
+    case NR_STUN_MSG_ALLOCATE_REQUEST:
+         ret = "ALLOCATE-REQUEST";
+         break;
+    case NR_STUN_MSG_ALLOCATE_RESPONSE:
+         ret = "ALLOCATE-RESPONSE";
+         break;
+    case NR_STUN_MSG_ALLOCATE_ERROR_RESPONSE:
+         ret = "ALLOCATE-ERROR-RESPONSE";
+         break;
+    case NR_STUN_MSG_REFRESH_REQUEST:
+         ret = "REFRESH-REQUEST";
+         break;
+    case NR_STUN_MSG_REFRESH_RESPONSE:
+         ret = "REFRESH-RESPONSE";
+         break;
+    case NR_STUN_MSG_REFRESH_ERROR_RESPONSE:
+         ret = "REFRESH-ERROR-RESPONSE";
+         break;
+    case NR_STUN_MSG_SEND_INDICATION:
+         ret = "SEND-INDICATION";
+         break;
+    case NR_STUN_MSG_DATA_INDICATION:
+         ret = "DATA-INDICATION";
+         break;
+    case NR_STUN_MSG_PERMISSION_REQUEST:
+         ret = "PERMISSION-REQUEST";
+         break;
+    case NR_STUN_MSG_PERMISSION_RESPONSE:
+         ret = "PERMISSION-RESPONSE";
+         break;
+    case NR_STUN_MSG_PERMISSION_ERROR_RESPONSE:
+         ret = "PERMISSION-ERROR-RESPONSE";
+         break;
+#endif /* USE_TURN */
+
+    default:
+         /* ret remains 0 */
+         break;
+    }
+
+    return ret;
+}
+
+int
+nr_random_alphanum(char *alphanum, int size)
+{
+    static char alphanums[256] = {
+        'A', 'B', 'C', 'D', 'E', 'F', 'G', 'H', 'I', 'J',
+        'K', 'L', 'M', 'N', 'O', 'P', 'Q', 'R', 'S', 'T',
+        'U', 'V', 'W', 'X', 'Y', 'Z',
+        'a', 'b', 'c', 'd', 'e', 'f', 'g', 'h', 'i', 'j',
+        'k', 'l', 'm', 'n', 'o', 'p', 'q', 'r', 's', 't',
+        'u', 'v', 'w', 'x', 'y', 'z',
+        '0', '1', '2', '3', '4', '5', '6', '7', '8', '9',
+        'A', 'B', 'C', 'D', 'E', 'F', 'G', 'H', 'I', 'J',
+        'K', 'L', 'M', 'N', 'O', 'P', 'Q', 'R', 'S', 'T',
+        'U', 'V', 'W', 'X', 'Y', 'Z',
+        'a', 'b', 'c', 'd', 'e', 'f', 'g', 'h', 'i', 'j',
+        'k', 'l', 'm', 'n', 'o', 'p', 'q', 'r', 's', 't',
+        'u', 'v', 'w', 'x', 'y', 'z',
+        '0', '1', '2', '3', '4', '5', '6', '7', '8', '9',
+        'A', 'B', 'C', 'D', 'E', 'F', 'G', 'H', 'I', 'J',
+        'K', 'L', 'M', 'N', 'O', 'P', 'Q', 'R', 'S', 'T',
+        'U', 'V', 'W', 'X', 'Y', 'Z',
+        'a', 'b', 'c', 'd', 'e', 'f', 'g', 'h', 'i', 'j',
+        'k', 'l', 'm', 'n', 'o', 'p', 'q', 'r', 's', 't',
+        'u', 'v', 'w', 'x', 'y', 'z',
+        '0', '1', '2', '3', '4', '5', '6', '7', '8', '9',
+        'A', 'B', 'C', 'D', 'E', 'F', 'G', 'H', 'I', 'J',
+        'K', 'L', 'M', 'N', 'O', 'P', 'Q', 'R', 'S', 'T',
+        'U', 'V', 'W', 'X', 'Y', 'Z',
+        'a', 'b', 'c', 'd', 'e', 'f', 'g', 'h', 'i', 'j',
+        'k', 'l', 'm', 'n', 'o', 'p', 'q', 'r', 's', 't',
+        'u', 'v', 'w', 'x', 'y', 'z',
+        '0', '1', '2', '3', '4', '5', '6', '7', '8', '9',
+        'A', 'B', 'C', 'D', 'E', 'F', 'G', 'H' };
+    int i;
+
+    nr_crypto_random_bytes((UCHAR*)alphanum, size);
+
+    /* now convert from binary to alphanumeric */
+    for (i = 0; i < size; ++i)
+        alphanum[i] = alphanums[(UCHAR)alphanum[i]];
+
+    return 0;
+}
+
+#ifndef UINT2_MAX
+#define UINT2_MAX ((UINT2)(65535U))
+#endif
+
+void nr_accumulate_count(UINT2* orig_count, UINT2 add_count)
+  {
+    if (UINT2_MAX - add_count < *orig_count) {
+      // don't rollover, just stop accumulating at MAX value
+      *orig_count = UINT2_MAX;
+    } else {
+      *orig_count += add_count;
+    }
+  }
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_util.h b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_util.h
new file mode 100644
index 0000000000..379b4fdd3d
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/stun_util.h
@@ -0,0 +1,62 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _stun_util_h
+#define _stun_util_h
+
+#include "stun.h"
+#include "local_addr.h"
+
+extern int NR_LOG_STUN;
+
+int nr_stun_startup(void);
+
+int nr_stun_xor_mapped_address(UINT4 magicCookie, UINT12 transactionId, nr_transport_addr *from, nr_transport_addr *to);
+
+// removes duplicates, and based on prefs, loopback and link_local addresses
+int nr_stun_filter_local_addresses(nr_local_addr addrs[], int *count);
+
+int nr_stun_find_local_addresses(nr_local_addr addrs[], int maxaddrs, int *count);
+
+int nr_stun_different_transaction(UCHAR *msg, size_t len, nr_stun_message *req);
+
+char* nr_stun_msg_type(int type);
+
+int nr_random_alphanum(char *alphanum, int size);
+
+// accumulate a count without worrying about rollover
+void nr_accumulate_count(UINT2* orig_count, UINT2 add_count);
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/turn_client_ctx.c b/dom/media/webrtc/transport/third_party/nICEr/src/stun/turn_client_ctx.c
new file mode 100644
index 0000000000..707d43a4a9
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/turn_client_ctx.c
@@ -0,0 +1,1277 @@
+/*
+  Copyright (c) 2007, Adobe Systems, Incorporated
+  Copyright (c) 2013, Mozilla
+
+  All rights reserved.
+
+  Redistribution and use in source and binary forms, with or without
+  modification, are permitted provided that the following conditions are
+  met:
+
+  * Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+  * Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+  * Neither the name of Adobe Systems, Network Resonance, Mozilla nor
+  the names of its contributors may be used to endorse or promote
+  products derived from this software without specific prior written
+  permission.
+
+  THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+  "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+  LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+  A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+  OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+  SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+  LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+  DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+  THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+  (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+  OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#ifdef USE_TURN
+
+#include <assert.h>
+#include <string.h>
+
+#include "nr_api.h"
+#include "r_time.h"
+#include "async_timer.h"
+#include "nr_socket_buffered_stun.h"
+#include "stun.h"
+#include "turn_client_ctx.h"
+#include "ice_ctx.h"
+
+int NR_LOG_TURN = 0;
+
+#define TURN_MAX_PENDING_BYTES 32000
+
+#define TURN_RTO 100  /* Hardcoded RTO estimate */
+#define TURN_LIFETIME_REQUEST_SECONDS    3600  /* One hour */
+#define TURN_USECS_PER_S                 1000000
+#define TURN_REFRESH_SLACK_SECONDS       10    /* How long before expiry to refresh
+                                                  allocations/permissions. The RFC 5766
+                                                  Section 7 recommendation if 60 seconds,
+                                                  but this is silly since the transaction
+                                                  times out after about 5. */
+#define TURN_PERMISSION_LIFETIME_SECONDS 300   /* 5 minutes. From RFC 5766 2.3 */
+
+// Set to enable a temporary fix that will run the TURN reservation keep-alive
+// logic when data is received via a TURN relayed path: a DATA_INDICATION packet is received.
+// TODO(pkerr@mozilla.com) This should be replace/removed when bug 935806 is implemented.
+#define REFRESH_RESERVATION_ON_RECV 1
+
+static int nr_turn_stun_ctx_create(nr_turn_client_ctx *tctx, int type,
+                                   NR_async_cb success_cb,
+                                   NR_async_cb failure_cb,
+                                   nr_turn_stun_ctx **ctxp);
+static int nr_turn_stun_ctx_destroy(nr_turn_stun_ctx **ctxp);
+static void nr_turn_stun_ctx_cb(NR_SOCKET s, int how, void *arg);
+static int nr_turn_stun_set_auth_params(nr_turn_stun_ctx *ctx,
+                                        char *realm, char *nonce);
+static void nr_turn_client_refresh_timer_cb(NR_SOCKET s, int how, void *arg);
+static int nr_turn_client_refresh_setup(nr_turn_client_ctx *ctx,
+                                        nr_turn_stun_ctx **sctx);
+static int nr_turn_client_start_refresh_timer(nr_turn_client_ctx *ctx,
+                                              nr_turn_stun_ctx *sctx,
+                                              UINT4 lifetime);
+static int nr_turn_permission_create(nr_turn_client_ctx *ctx,
+                                     const nr_transport_addr *addr,
+                                     nr_turn_permission **permp);
+static int nr_turn_permission_find(nr_turn_client_ctx *ctx,
+                                   const nr_transport_addr *addr,
+                                   nr_turn_permission **permp);
+static int nr_turn_permission_destroy(nr_turn_permission **permp);
+static void nr_turn_client_refresh_cb(NR_SOCKET s, int how, void *arg);
+static void nr_turn_client_permissions_cb(NR_SOCKET s, int how, void *cb);
+static int nr_turn_client_send_stun_request(nr_turn_client_ctx *ctx,
+                                            nr_stun_message *req,
+                                            int flags);
+
+int nr_transport_addr_listnode_create(const nr_transport_addr *addr, nr_transport_addr_listnode **listnodep)
+{
+  nr_transport_addr_listnode *listnode = 0;
+  int r,_status;
+
+  if (!(listnode=RCALLOC(sizeof(nr_transport_addr_listnode)))) {
+    ABORT(R_NO_MEMORY);
+  }
+
+  if ((r = nr_transport_addr_copy(&listnode->value, addr))) {
+    ABORT(r);
+  }
+
+  *listnodep = listnode;
+  listnode = 0;
+  _status = 0;
+
+abort:
+  nr_transport_addr_listnode_destroy(&listnode);
+  return(_status);
+}
+
+void nr_transport_addr_listnode_destroy(nr_transport_addr_listnode **listnode)
+{
+  RFREE(*listnode);
+  *listnode = 0;
+}
+
+/* nr_turn_stun_ctx functions */
+static int nr_turn_stun_ctx_create(nr_turn_client_ctx *tctx, int mode,
+                                   NR_async_cb success_cb,
+                                   NR_async_cb error_cb,
+                                   nr_turn_stun_ctx **ctxp)
+{
+  nr_turn_stun_ctx *sctx = 0;
+  int r,_status;
+  char label[256];
+
+  if (!(sctx=RCALLOC(sizeof(nr_turn_stun_ctx))))
+    ABORT(R_NO_MEMORY);
+
+  /* TODO(ekr@rtfm.com): label by phase */
+  snprintf(label, sizeof(label), "%s:%s", tctx->label, ":TURN");
+
+  if ((r=nr_stun_client_ctx_create(label, tctx->sock, &tctx->turn_server_addr,
+                                   TURN_RTO, &sctx->stun))) {
+    ABORT(r);
+  }
+
+  /* Set the STUN auth parameters, but don't set authentication on.
+     For that we need the nonce, set in nr_turn_stun_set_auth_params.
+  */
+  sctx->stun->auth_params.username=tctx->username;
+  INIT_DATA(sctx->stun->auth_params.password,
+            tctx->password->data, tctx->password->len);
+
+  sctx->tctx=tctx;
+  sctx->success_cb=success_cb;
+  sctx->error_cb=error_cb;
+  sctx->mode=mode;
+  sctx->last_error_code=0;
+  STAILQ_INIT(&sctx->addresses_tried);
+
+  /* Add ourselves to the tctx's list */
+  STAILQ_INSERT_TAIL(&tctx->stun_ctxs, sctx, entry);
+  *ctxp=sctx;
+
+  _status=0;
+abort:
+  if (_status) {
+    nr_turn_stun_ctx_destroy(&sctx);
+  }
+  return(_status);
+}
+
+/* Note: this function does not pull us off the tctx's list. */
+static int nr_turn_stun_ctx_destroy(nr_turn_stun_ctx **ctxp)
+{
+  nr_turn_stun_ctx *ctx;
+
+  if (!ctxp || !*ctxp)
+    return 0;
+
+  ctx = *ctxp;
+  *ctxp = 0;
+
+  nr_stun_client_ctx_destroy(&ctx->stun);
+  RFREE(ctx->realm);
+  RFREE(ctx->nonce);
+
+  while (!STAILQ_EMPTY(&ctx->addresses_tried)) {
+    nr_transport_addr_listnode *listnode = STAILQ_FIRST(&ctx->addresses_tried);
+    STAILQ_REMOVE_HEAD(&ctx->addresses_tried, entry);
+    nr_transport_addr_listnode_destroy(&listnode);
+  }
+
+  RFREE(ctx);
+
+  return 0;
+}
+
+static int nr_turn_stun_set_auth_params(nr_turn_stun_ctx *ctx,
+                                        char *realm, char *nonce)
+{
+  int _status;
+
+  RFREE(ctx->realm);
+  RFREE(ctx->nonce);
+
+  assert(realm);
+  if (!realm)
+    ABORT(R_BAD_ARGS);
+  ctx->realm=r_strdup(realm);
+  if (!ctx->realm)
+    ABORT(R_NO_MEMORY);
+
+  assert(nonce);
+  if (!nonce)
+    ABORT(R_BAD_ARGS);
+  ctx->nonce=r_strdup(nonce);
+  if (!ctx->nonce)
+    ABORT(R_NO_MEMORY);
+
+  RFREE(ctx->stun->realm);
+  ctx->stun->realm = r_strdup(ctx->realm);
+  if (!ctx->stun->realm)
+    ABORT(R_NO_MEMORY);
+
+  ctx->stun->auth_params.realm = ctx->realm;
+  ctx->stun->auth_params.nonce = ctx->nonce;
+  ctx->stun->auth_params.authenticate = 1;  /* May already be 1 */
+
+  _status=0;
+abort:
+  return(_status);
+}
+
+static int nr_turn_stun_ctx_start(nr_turn_stun_ctx *ctx)
+{
+  int r, _status;
+  nr_turn_client_ctx *tctx = ctx->tctx;
+  nr_transport_addr_listnode *address_tried = 0;
+
+  if ((r=nr_stun_client_reset(ctx->stun))) {
+    r_log(NR_LOG_TURN, LOG_ERR, "TURN(%s): Couldn't reset STUN",
+          tctx->label);
+    ABORT(r);
+  }
+
+  if ((r=nr_stun_client_start(ctx->stun, ctx->mode, nr_turn_stun_ctx_cb, ctx))) {
+    r_log(NR_LOG_TURN, LOG_ERR, "TURN(%s): Couldn't start STUN",
+          tctx->label);
+    ABORT(r);
+  }
+
+  if ((r=nr_transport_addr_listnode_create(&ctx->stun->peer_addr, &address_tried))) {
+    ABORT(r);
+  }
+
+  STAILQ_INSERT_TAIL(&ctx->addresses_tried, address_tried, entry);
+
+  _status=0;
+abort:
+  return _status;
+}
+
+static int nr_turn_stun_ctx_handle_redirect(nr_turn_stun_ctx *ctx)
+{
+  int r, _status;
+  nr_turn_client_ctx *tctx = ctx->tctx;
+  nr_stun_message_attribute *ec;
+  nr_stun_message_attribute *attr;
+  nr_transport_addr *alternate_addr = 0;
+  int index = 0;
+
+  if (!tctx->ctx) {
+    /* If we were to require TCP nr_sockets to allow multiple connect calls by
+     * disconnecting and re-connecting, we could avoid this requirement. */
+    r_log(NR_LOG_TURN, LOG_ERR,
+          "TURN(%s): nr_turn_stun_ctx_handle_redirect is not supported when "
+          "there is no ICE ctx, and by extension no socket factory, since we "
+          "need that to create new sockets in the TCP case",
+          tctx->label);
+    ABORT(R_BAD_ARGS);
+  }
+
+  if (ctx->stun->response->header.type != NR_STUN_MSG_ALLOCATE_ERROR_RESPONSE) {
+    r_log(NR_LOG_TURN, LOG_ERR,
+          "TURN(%s): nr_turn_stun_ctx_handle_redirect called for something "
+          "other than an Allocate error response (type=%d)",
+          tctx->label, ctx->stun->response->header.type);
+    ABORT(R_BAD_ARGS);
+  }
+
+  if (!nr_stun_message_has_attribute(ctx->stun->response,
+                                     NR_STUN_ATTR_ERROR_CODE, &ec) ||
+      (ec->u.error_code.number != 300)) {
+    r_log(NR_LOG_TURN, LOG_ERR,
+          "TURN(%s): nr_turn_stun_ctx_handle_redirect called without a "
+          "300 response",
+          tctx->label);
+    ABORT(R_BAD_ARGS);
+  }
+
+  while (!alternate_addr && !nr_stun_message_get_attribute(
+      ctx->stun->response, NR_STUN_ATTR_ALTERNATE_SERVER, index++, &attr)) {
+    alternate_addr = &attr->u.alternate_server;
+
+    // TODO: Someday we may need to handle IP version switching, but it is
+    // unclear how that is supposed to work with ICE when the IP version of
+    // the candidate's base address is fixed...
+    if (alternate_addr->ip_version != tctx->turn_server_addr.ip_version) {
+      r_log(NR_LOG_TURN, LOG_INFO,
+            "TURN(%s): nr_turn_stun_ctx_handle_redirect not trying %s, since it is a different IP version",
+            tctx->label, alternate_addr->as_string);
+      alternate_addr = 0;
+      continue;
+    }
+
+    /* Check if we've already tried this, and ignore if we have */
+    nr_transport_addr_listnode *address_tried = 0;
+    STAILQ_FOREACH(address_tried, &ctx->addresses_tried, entry) {
+      /* Ignore protocol */
+      alternate_addr->protocol = address_tried->value.protocol;
+      if (!nr_transport_addr_cmp(alternate_addr, &address_tried->value, NR_TRANSPORT_ADDR_CMP_MODE_ALL)) {
+        r_log(NR_LOG_TURN, LOG_INFO,
+              "TURN(%s): nr_turn_stun_ctx_handle_redirect already tried %s, ignoring",
+              tctx->label, alternate_addr->as_string);
+        alternate_addr = 0;
+        break;
+      }
+    }
+  }
+
+  if (!alternate_addr) {
+    /* Should we use a different error code depending on why we didn't find
+     * one? (eg; no ALTERNATE-SERVERS at all, none that we have not tried
+     * already, none that are of a compatible IP version?) */
+    r_log(NR_LOG_TURN, LOG_ERR,
+          "TURN(%s): nr_turn_stun_ctx_handle_redirect did not find a viable "
+          "ALTERNATE-SERVER",
+          tctx->label);
+    ABORT(R_FAILED);
+  }
+
+  r_log(NR_LOG_TURN, LOG_INFO,
+      "TURN(%s): nr_turn_stun_ctx_handle_redirect trying %s",
+      tctx->label, alternate_addr->as_string);
+
+  /* This also handles the call to nr_transport_addr_fmt_addr_string */
+  if ((r = nr_transport_addr_copy_addrport(&tctx->turn_server_addr,
+                                           alternate_addr))) {
+    r_log(NR_LOG_TURN, LOG_ERR,
+          "TURN(%s): nr_turn_stun_ctx_handle_redirect copying ALTERNATE-SERVER "
+          "failed(%d)!",
+          tctx->label, r);
+    assert(0);
+    ABORT(r);
+  }
+
+  /* TURN server address is now updated. Restart the STUN Allocate ctx. Note
+   * that we do not attempt to update the local address; if the TURN server
+   * redirects to something that is not best reached from the already-selected
+   * local address, oh well. */
+
+  if (tctx->turn_server_addr.protocol == IPPROTO_TCP) {
+    /* For TCP, we need to replace the underlying nr_socket, since we cannot
+     * un-connect it from the old server. */
+    /* If we were to require TCP nr_sockets to allow multiple connect calls by
+     * disconnecting and re-connecting, we could avoid this stuff, and just
+     * call nr_socket_connect. */
+    nr_transport_addr old_local_addr;
+    nr_socket* new_socket;
+    if ((r = nr_socket_getaddr(tctx->sock, &old_local_addr))) {
+      r_log(NR_LOG_TURN, LOG_ERR,
+            "TURN(%s): nr_turn_stun_ctx_handle_redirect "
+            "failed to get old local address (%d)!",
+            tctx->label, r);
+      assert(0);
+      ABORT(r);
+    }
+
+    if ((r = nr_socket_factory_create_socket(tctx->ctx->socket_factory,
+            &old_local_addr, &new_socket))) {
+      r_log(NR_LOG_TURN, LOG_ERR,
+            "TURN(%s): nr_turn_stun_ctx_handle_redirect "
+            "failed to create new raw TCP socket for redirect (%d)!",
+            tctx->label, r);
+      assert(0);
+      ABORT(r);
+    }
+
+    if ((r = nr_socket_buffered_stun_reset(tctx->sock, new_socket))) {
+      /* nr_socket_buffered_stun_reset always takes ownership of |new_socket| */
+      r_log(NR_LOG_TURN, LOG_ERR,
+            "TURN(%s): nr_turn_stun_ctx_handle_redirect "
+            "failed to update raw TCP socket (%d)!",
+            tctx->label, r);
+      assert(0);
+      ABORT(r);
+    }
+
+    if ((r = nr_socket_connect(tctx->sock, &tctx->turn_server_addr))) {
+      if (r != R_WOULDBLOCK) {
+        r_log(NR_LOG_TURN, LOG_ERR,
+              "TURN(%s): nr_turn_stun_ctx_handle_redirect nr_socket_connect "
+              "failed(%d)!",
+              tctx->label, r);
+        assert(0);
+        ABORT(r);
+      }
+    }
+  }
+
+  nr_transport_addr_copy(&ctx->stun->peer_addr, &tctx->turn_server_addr);
+
+  if ((r = nr_turn_stun_ctx_start(ctx))) {
+    r_log(NR_LOG_TURN, LOG_ERR,
+          "TURN(%s): nr_turn_stun_ctx_handle_redirect nr_turn_stun_ctx_start "
+          "failed(%d)!",
+          tctx->label, r);
+    assert(0);
+    ABORT(r);
+  }
+
+  _status = 0;
+abort:
+  return _status;
+}
+
+static void nr_turn_stun_ctx_cb(NR_SOCKET s, int how, void *arg)
+{
+  int r, _status;
+  nr_turn_stun_ctx *ctx = (nr_turn_stun_ctx *)arg;
+
+  ctx->last_error_code = ctx->stun->error_code;
+
+  switch (ctx->stun->state) {
+    case NR_STUN_CLIENT_STATE_DONE:
+      /* Save the realm and nonce */
+      if (ctx->stun->realm && (!ctx->tctx->realm || strcmp(ctx->stun->realm,
+                                                           ctx->tctx->realm))) {
+        RFREE(ctx->tctx->realm);
+        ctx->tctx->realm = r_strdup(ctx->stun->realm);
+        if (!ctx->tctx->realm)
+          ABORT(R_NO_MEMORY);
+      }
+      if (ctx->stun->nonce && (!ctx->tctx->nonce || strcmp(ctx->stun->nonce,
+                                                           ctx->tctx->nonce))) {
+        RFREE(ctx->tctx->nonce);
+        ctx->tctx->nonce = r_strdup(ctx->stun->nonce);
+        if (!ctx->tctx->nonce)
+          ABORT(R_NO_MEMORY);
+      }
+
+      ctx->retry_ct=0;
+      ctx->success_cb(0, 0, ctx);
+      break;
+
+    case NR_STUN_CLIENT_STATE_FAILED:
+      /* Special case: if this is an authentication error,
+         we retry once. This allows the 401/438 nonce retry
+         paradigm. After that, we fail */
+      /* TODO(ekr@rtfm.com): 401 needs a #define */
+      /* TODO(ekr@rtfm.com): Add alternate-server (Mozilla bug 857688) */
+      if (ctx->stun->error_code == 438) {
+        // track 438s for ice telemetry
+        nr_accumulate_count(&(ctx->tctx->cnt_438s), 1);
+      }
+      if (ctx->stun->error_code == 401 || ctx->stun->error_code == 438) {
+        if (ctx->retry_ct > 0) {
+          if (ctx->stun->error_code == 401) {
+            // track 401s for ice telemetry
+            nr_accumulate_count(&(ctx->tctx->cnt_401s), 1);
+          }
+          r_log(NR_LOG_TURN, LOG_WARNING, "TURN(%s): Exceeded the number of retries", ctx->tctx->label);
+          ABORT(R_FAILED);
+        }
+
+        if (!ctx->stun->nonce) {
+          r_log(NR_LOG_TURN, LOG_WARNING, "TURN(%s): 401 but no nonce", ctx->tctx->label);
+          ABORT(R_FAILED);
+        }
+        if (!ctx->stun->realm) {
+          r_log(NR_LOG_TURN, LOG_WARNING, "TURN(%s): 401 but no realm", ctx->tctx->label);
+          ABORT(R_FAILED);
+        }
+
+        /* Try to retry */
+        if ((r=nr_turn_stun_set_auth_params(ctx, ctx->stun->realm,
+                                            ctx->stun->nonce)))
+          ABORT(r);
+
+        ctx->stun->error_code = 0;  /* Reset to avoid inf-looping */
+
+        if ((r=nr_turn_stun_ctx_start(ctx))) {
+          r_log(NR_LOG_TURN, LOG_ERR, "TURN(%s): Couldn't start STUN", ctx->tctx->label);
+          ABORT(r);
+        }
+
+        ctx->retry_ct++;
+      } else if (ctx->stun->error_code == 300) {
+        r_log(NR_LOG_TURN, LOG_INFO,
+              "TURN(%s): Redirect received, restarting TURN", ctx->tctx->label);
+        /* We don't limit this with a retry counter, we limit redirects by
+         * checking whether we've tried the ALTERNATE-SERVER yet. */
+        ctx->retry_ct = 0;
+        if ((r = nr_turn_stun_ctx_handle_redirect(ctx))) {
+          ABORT(r);
+        }
+      } else {
+        ABORT(R_FAILED);
+      }
+      break;
+
+    case NR_STUN_CLIENT_STATE_TIMED_OUT:
+      ABORT(R_FAILED);
+      break;
+
+    case NR_STUN_CLIENT_STATE_CANCELLED:
+      assert(0);  /* Shouldn't happen */
+      return;
+      break;
+
+    default:
+      assert(0);  /* Shouldn't happen */
+      return;
+  }
+
+  _status=0;
+abort:
+  if (_status) {
+    ctx->error_cb(0, 0, ctx);
+  }
+}
+
+/* nr_turn_client_ctx functions */
+int nr_turn_client_ctx_create(const char* label, nr_socket* sock,
+                              const char* username, Data* password,
+                              nr_transport_addr* addr, nr_ice_ctx* ice_ctx,
+                              nr_turn_client_ctx** ctxp) {
+  nr_turn_client_ctx *ctx=0;
+  int r,_status;
+
+  if ((r=r_log_register("turn", &NR_LOG_TURN)))
+    ABORT(r);
+
+  if(!(ctx=RCALLOC(sizeof(nr_turn_client_ctx))))
+    ABORT(R_NO_MEMORY);
+
+  STAILQ_INIT(&ctx->stun_ctxs);
+  STAILQ_INIT(&ctx->permissions);
+
+  if(!(ctx->label=r_strdup(label)))
+    ABORT(R_NO_MEMORY);
+
+  ctx->sock=sock;
+  ctx->username = r_strdup(username);
+  if (!ctx->username)
+    ABORT(R_NO_MEMORY);
+
+  if ((r=r_data_create(&ctx->password, password->data, password->len)))
+    ABORT(r);
+  if ((r=nr_transport_addr_copy(&ctx->turn_server_addr, addr)))
+    ABORT(r);
+
+  ctx->state = NR_TURN_CLIENT_STATE_INITTED;
+  if (addr->protocol == IPPROTO_TCP) {
+    if ((r=nr_socket_connect(ctx->sock, &ctx->turn_server_addr))) {
+      if (r != R_WOULDBLOCK)
+        ABORT(r);
+    }
+  }
+
+  ctx->ctx = ice_ctx;
+
+  *ctxp=ctx;
+
+  _status=0;
+abort:
+  if(_status){
+    nr_turn_client_ctx_destroy(&ctx);
+  }
+  return(_status);
+}
+
+int
+nr_turn_client_ctx_destroy(nr_turn_client_ctx **ctxp)
+{
+  nr_turn_client_ctx *ctx;
+
+  if(!ctxp || !*ctxp)
+    return(0);
+
+  ctx=*ctxp;
+  *ctxp = 0;
+
+  if (ctx->label)
+    r_log(NR_LOG_TURN, LOG_DEBUG, "TURN(%s): destroy", ctx->label);
+
+  nr_turn_client_deallocate(ctx);
+
+  /* Cancel frees the rest of our data */
+  RFREE(ctx->label);
+  ctx->label = 0;
+
+  nr_turn_client_cancel(ctx);
+
+  RFREE(ctx->username);
+  ctx->username = 0;
+  r_data_destroy(&ctx->password);
+  RFREE(ctx->nonce);
+  ctx->nonce = 0;
+  RFREE(ctx->realm);
+  ctx->realm = 0;
+
+  /* Destroy the STUN client ctxs */
+  while (!STAILQ_EMPTY(&ctx->stun_ctxs)) {
+    nr_turn_stun_ctx *stun = STAILQ_FIRST(&ctx->stun_ctxs);
+    STAILQ_REMOVE_HEAD(&ctx->stun_ctxs, entry);
+    nr_turn_stun_ctx_destroy(&stun);
+  }
+
+  /* Destroy the permissions */
+  while (!STAILQ_EMPTY(&ctx->permissions)) {
+    nr_turn_permission *perm = STAILQ_FIRST(&ctx->permissions);
+    STAILQ_REMOVE_HEAD(&ctx->permissions, entry);
+    nr_turn_permission_destroy(&perm);
+  }
+
+  RFREE(ctx);
+
+  return(0);
+}
+
+int nr_turn_client_cancel(nr_turn_client_ctx *ctx)
+{
+  nr_turn_stun_ctx *stun = 0;
+
+  if (ctx->state == NR_TURN_CLIENT_STATE_CANCELLED ||
+      ctx->state == NR_TURN_CLIENT_STATE_FAILED)
+    return(0);
+
+  if (ctx->label)
+    r_log(NR_LOG_TURN, LOG_INFO, "TURN(%s): cancelling", ctx->label);
+
+  /* Cancel the STUN client ctxs */
+  stun = STAILQ_FIRST(&ctx->stun_ctxs);
+  while (stun) {
+    nr_stun_client_cancel(stun->stun);
+    stun = STAILQ_NEXT(stun, entry);
+  }
+
+  /* Cancel the timers, if not already cancelled */
+  NR_async_timer_cancel(ctx->connected_timer_handle);
+  NR_async_timer_cancel(ctx->refresh_timer_handle);
+
+  ctx->state = NR_TURN_CLIENT_STATE_CANCELLED;
+
+  return(0);
+}
+
+int nr_turn_client_send_stun_request(nr_turn_client_ctx *ctx,
+                                     nr_stun_message *req,
+                                     int flags)
+{
+  int r,_status;
+
+  if ((r=nr_stun_encode_message(req)))
+    ABORT(r);
+
+  if ((r=nr_socket_sendto(ctx->sock,
+                          req->buffer, req->length, flags,
+                          &ctx->turn_server_addr))) {
+    r_log(NR_LOG_TURN, LOG_WARNING, "TURN(%s): Failed sending request",
+          ctx->label);
+    ABORT(r);
+  }
+
+  _status=0;
+abort:
+  return(_status);
+}
+
+int nr_turn_client_deallocate(nr_turn_client_ctx *ctx)
+{
+  int r,_status;
+  nr_stun_message *aloc = 0;
+  nr_stun_client_auth_params auth;
+  nr_stun_client_refresh_request_params refresh;
+
+  if (ctx->state != NR_TURN_CLIENT_STATE_ALLOCATED)
+    return(0);
+
+  r_log(NR_LOG_TURN, LOG_INFO, "TURN(%s): deallocating", ctx->label);
+
+  refresh.lifetime_secs = 0;
+
+  auth.username = ctx->username;
+  INIT_DATA(auth.password, ctx->password->data, ctx->password->len);
+
+  auth.realm = ctx->realm;
+  auth.nonce = ctx->nonce;
+
+  auth.authenticate = 1;
+
+  if ((r=nr_stun_build_refresh_request(&auth, &refresh, &aloc)))
+    ABORT(r);
+
+  // We are only sending a single request here because we are in the process of
+  // shutting everything down. Theoretically we should probably start a seperate
+  // STUN transaction which outlives the TURN context.
+  if ((r=nr_turn_client_send_stun_request(ctx, aloc, 0)))
+    ABORT(r);
+
+  ctx->state = NR_TURN_CLIENT_STATE_DEALLOCATING;
+
+  _status=0;
+abort:
+  nr_stun_message_destroy(&aloc);
+  return(_status);
+}
+
+static void nr_turn_client_fire_finished_cb(nr_turn_client_ctx *ctx)
+  {
+    if (ctx->finished_cb) {
+      NR_async_cb finished_cb=ctx->finished_cb;
+      ctx->finished_cb=0;
+      finished_cb(0, 0, ctx->cb_arg);
+    }
+  }
+
+int nr_turn_client_failed(nr_turn_client_ctx *ctx)
+{
+  if (ctx->state == NR_TURN_CLIENT_STATE_FAILED ||
+      ctx->state == NR_TURN_CLIENT_STATE_CANCELLED)
+    return(0);
+
+  r_log(NR_LOG_TURN, LOG_WARNING, "TURN(%s) failed", ctx->label);
+  nr_turn_client_cancel(ctx);
+  ctx->state = NR_TURN_CLIENT_STATE_FAILED;
+  nr_turn_client_fire_finished_cb(ctx);
+
+  return(0);
+}
+
+int nr_turn_client_get_relayed_address(nr_turn_client_ctx *ctx,
+                                       nr_transport_addr *relayed_address)
+{
+  int r, _status;
+
+  if (ctx->state != NR_TURN_CLIENT_STATE_ALLOCATED)
+    ABORT(R_FAILED);
+
+  if (r=nr_transport_addr_copy(relayed_address, &ctx->relay_addr))
+    ABORT(r);
+
+  _status=0;
+abort:
+  return(_status);
+}
+
+int nr_turn_client_get_mapped_address(nr_turn_client_ctx *ctx,
+                                       nr_transport_addr *mapped_address)
+{
+  int r, _status;
+
+  if (ctx->state != NR_TURN_CLIENT_STATE_ALLOCATED)
+    ABORT(R_FAILED);
+
+  if (r=nr_transport_addr_copy(mapped_address, &ctx->mapped_addr))
+    ABORT(r);
+
+  _status=0;
+abort:
+  return(_status);
+}
+
+static void nr_turn_client_allocate_cb(NR_SOCKET s, int how, void *arg)
+{
+  nr_turn_stun_ctx *ctx = (nr_turn_stun_ctx *)arg;
+  nr_turn_stun_ctx *refresh_ctx;
+  int r,_status;
+
+  ctx->tctx->state = NR_TURN_CLIENT_STATE_ALLOCATED;
+
+  if ((r=nr_transport_addr_copy(
+          &ctx->tctx->relay_addr,
+          &ctx->stun->results.allocate_response.relay_addr)))
+    ABORT(r);
+
+  if ((r=nr_transport_addr_copy(
+          &ctx->tctx->mapped_addr,
+          &ctx->stun->results.allocate_response.mapped_addr)))
+    ABORT(r);
+
+  if ((r=nr_turn_client_refresh_setup(ctx->tctx, &refresh_ctx)))
+    ABORT(r);
+
+  if ((r=nr_turn_client_start_refresh_timer(
+          ctx->tctx, refresh_ctx,
+          ctx->stun->results.allocate_response.lifetime_secs)))
+    ABORT(r);
+
+  r_log(NR_LOG_TURN, LOG_INFO,
+        "TURN(%s): Succesfully allocated addr %s lifetime=%u",
+        ctx->tctx->label,
+        ctx->tctx->relay_addr.as_string,
+        ctx->stun->results.allocate_response.lifetime_secs);
+
+  nr_turn_client_fire_finished_cb(ctx->tctx);
+  _status=0;
+abort:
+  if (_status) {
+    nr_turn_client_failed(ctx->tctx);
+  }
+}
+
+static void nr_turn_client_error_cb(NR_SOCKET s, int how, void *arg)
+{
+  nr_turn_stun_ctx *ctx = (nr_turn_stun_ctx *)arg;
+
+  r_log(NR_LOG_TURN, LOG_WARNING, "TURN(%s): mode %d, %s",
+        ctx->tctx->label, ctx->mode, __FUNCTION__);
+
+  nr_turn_client_failed(ctx->tctx);
+}
+
+static void nr_turn_client_permission_error_cb(NR_SOCKET s, int how, void *arg)
+{
+  nr_turn_stun_ctx *ctx = (nr_turn_stun_ctx *)arg;
+
+  if (ctx->last_error_code == 403) {
+    // track 403s for ice telemetry
+    nr_accumulate_count(&(ctx->tctx->cnt_403s), 1);
+    r_log(NR_LOG_TURN, LOG_WARNING, "TURN(%s): mode %d, permission denied",
+          ctx->tctx->label, ctx->mode);
+
+  } else{
+    nr_turn_client_error_cb(0, 0, ctx);
+  }
+}
+
+int nr_turn_client_allocate(nr_turn_client_ctx *ctx,
+                            NR_async_cb finished_cb, void *cb_arg)
+{
+  nr_turn_stun_ctx *stun = 0;
+  int r,_status;
+
+  if(ctx->state == NR_TURN_CLIENT_STATE_FAILED ||
+     ctx->state == NR_TURN_CLIENT_STATE_CANCELLED){
+    /* TURN TCP contexts can fail before we ever try to form an allocation,
+     * since the TCP connection can fail. It is also conceivable that a TURN
+     * TCP context could be cancelled before we are done forming all
+     * allocations (although we do not do this at the time this code was
+     * written) */
+    assert(ctx->turn_server_addr.protocol == IPPROTO_TCP);
+    ABORT(R_NOT_FOUND);
+  }
+
+  assert(ctx->state == NR_TURN_CLIENT_STATE_INITTED);
+
+  ctx->finished_cb=finished_cb;
+  ctx->cb_arg=cb_arg;
+
+  if ((r=nr_turn_stun_ctx_create(ctx, NR_TURN_CLIENT_MODE_ALLOCATE_REQUEST,
+                                 nr_turn_client_allocate_cb,
+                                 nr_turn_client_error_cb,
+                                 &stun)))
+    ABORT(r);
+  stun->stun->params.allocate_request.lifetime_secs =
+      TURN_LIFETIME_REQUEST_SECONDS;
+
+  if (ctx->state == NR_TURN_CLIENT_STATE_INITTED) {
+      if ((r=nr_turn_stun_ctx_start(stun)))
+        ABORT(r);
+      ctx->state = NR_TURN_CLIENT_STATE_ALLOCATING;
+  } else {
+      ABORT(R_ALREADY);
+  }
+
+  _status=0;
+abort:
+  if (_status) {
+    nr_turn_client_failed(ctx);
+  }
+
+  return(_status);
+}
+
+int nr_turn_client_process_response(nr_turn_client_ctx *ctx,
+                                    UCHAR *msg, int len,
+                                    nr_transport_addr *turn_server_addr)
+{
+  int r, _status;
+  nr_turn_stun_ctx *sc1;
+
+  switch (ctx->state) {
+    case NR_TURN_CLIENT_STATE_ALLOCATING:
+    case NR_TURN_CLIENT_STATE_ALLOCATED:
+      break;
+    default:
+      ABORT(R_FAILED);
+  }
+
+  sc1 = STAILQ_FIRST(&ctx->stun_ctxs);
+  while (sc1) {
+    r = nr_stun_client_process_response(sc1->stun, msg, len, turn_server_addr);
+    if (!r)
+      break;
+    if (r==R_RETRY)  /* Likely a 401 and we will retry */
+      break;
+    if (r != R_REJECTED)
+      ABORT(r);
+    sc1 = STAILQ_NEXT(sc1, entry);
+  }
+  if (!sc1)
+    ABORT(R_REJECTED);
+
+  _status=0;
+abort:
+  return(_status);
+}
+
+static int nr_turn_client_refresh_setup(nr_turn_client_ctx *ctx,
+                                        nr_turn_stun_ctx **sctx)
+{
+  nr_turn_stun_ctx *stun = 0;
+  int r,_status;
+
+  assert(ctx->state == NR_TURN_CLIENT_STATE_ALLOCATED);
+  if (ctx->state != NR_TURN_CLIENT_STATE_ALLOCATED)
+    ABORT(R_NOT_PERMITTED);
+
+  if ((r=nr_turn_stun_ctx_create(ctx, NR_TURN_CLIENT_MODE_REFRESH_REQUEST,
+                                 nr_turn_client_refresh_cb,
+                                 nr_turn_client_error_cb,
+                                 &stun)))
+    ABORT(r);
+
+  if ((r=nr_turn_stun_set_auth_params(stun, ctx->realm, ctx->nonce)))
+    ABORT(r);
+
+  stun->stun->params.refresh_request.lifetime_secs =
+      TURN_LIFETIME_REQUEST_SECONDS;
+
+
+  *sctx=stun;
+
+  _status=0;
+abort:
+  return(_status);
+}
+
+static int nr_turn_client_start_refresh_timer(nr_turn_client_ctx *tctx,
+                                              nr_turn_stun_ctx *sctx,
+                                              UINT4 lifetime)
+{
+  int _status;
+
+  assert(!tctx->refresh_timer_handle);
+
+  if (lifetime <= TURN_REFRESH_SLACK_SECONDS) {
+    r_log(NR_LOG_TURN, LOG_ERR, "Too short lifetime specified for turn %u", lifetime);
+    ABORT(R_BAD_DATA);
+  }
+
+  if (lifetime > 3600)
+    lifetime = 3600;
+
+  lifetime -= TURN_REFRESH_SLACK_SECONDS;
+
+  r_log(NR_LOG_TURN, LOG_DEBUG, "TURN(%s): Setting refresh timer for %u seconds",
+        tctx->label, lifetime);
+  NR_ASYNC_TIMER_SET(lifetime * 1000, nr_turn_client_refresh_timer_cb, sctx,
+                     &tctx->refresh_timer_handle);
+
+  _status=0;
+abort:
+  if (_status) {
+    nr_turn_client_failed(tctx);
+  }
+  return _status;
+}
+
+static void nr_turn_client_refresh_timer_cb(NR_SOCKET s, int how, void *arg)
+{
+  nr_turn_stun_ctx *ctx = (nr_turn_stun_ctx *)arg;
+  int r,_status;
+
+  r_log(NR_LOG_TURN, LOG_DEBUG, "TURN(%s): Refresh timer fired",
+        ctx->tctx->label);
+
+  ctx->tctx->refresh_timer_handle=0;
+  if ((r=nr_turn_stun_ctx_start(ctx))) {
+    ABORT(r);
+  }
+
+  _status=0;
+abort:
+  if (_status) {
+    nr_turn_client_failed(ctx->tctx);
+  }
+  return;
+}
+
+static void nr_turn_client_refresh_cb(NR_SOCKET s, int how, void *arg)
+{
+  int r, _status;
+  nr_turn_stun_ctx *ctx = (nr_turn_stun_ctx *)arg;
+  /* Save lifetime from the reset */
+  UINT4 lifetime = ctx->stun->results.refresh_response.lifetime_secs;
+
+  r_log(NR_LOG_TURN, LOG_DEBUG, "TURN(%s): Refresh succeeded. lifetime=%u",
+        ctx->tctx->label, lifetime);
+
+  if ((r=nr_turn_client_start_refresh_timer(
+          ctx->tctx, ctx, lifetime)))
+    ABORT(r);
+
+  _status=0;
+
+abort:
+  if (_status) {
+    nr_turn_client_failed(ctx->tctx);
+  }
+}
+
+/* TODO(ekr@rtfm.com): We currently don't support channels.
+   We might in the future. Mozilla bug 857736 */
+int nr_turn_client_send_indication(nr_turn_client_ctx *ctx,
+                                   const UCHAR *msg, size_t len,
+                                   int flags, const nr_transport_addr *remote_addr)
+{
+  int r,_status;
+  nr_stun_client_send_indication_params params = { { 0 } };
+  nr_stun_message *ind = 0;
+
+  if (ctx->state != NR_TURN_CLIENT_STATE_ALLOCATED)
+    ABORT(R_FAILED);
+
+  r_log(NR_LOG_TURN, LOG_DEBUG, "TURN(%s): Send indication len=%zu",
+        ctx->label, len);
+
+  if ((r=nr_turn_client_ensure_perm(ctx, remote_addr)))
+    ABORT(r);
+
+  if ((r=nr_transport_addr_copy(&params.remote_addr, remote_addr)))
+    ABORT(r);
+
+  params.data.data = (UCHAR*)msg;
+  params.data.len = len;
+
+  if ((r=nr_stun_build_send_indication(&params, &ind)))
+    ABORT(r);
+
+  if ((r=nr_turn_client_send_stun_request(ctx, ind, flags)))
+    ABORT(r);
+
+  _status=0;
+abort:
+  nr_stun_message_destroy(&ind);
+  return(_status);
+}
+
+int nr_turn_client_parse_data_indication(nr_turn_client_ctx *ctx,
+                                         nr_transport_addr *source_addr,
+                                         UCHAR *msg, size_t len,
+                                         UCHAR *newmsg, size_t *newlen,
+                                         size_t newsize,
+                                         nr_transport_addr *remote_addr)
+{
+  int r,_status;
+  nr_stun_message *ind=0;
+  nr_stun_message_attribute *attr;
+  nr_turn_permission *perm;
+
+  if (nr_transport_addr_cmp(&ctx->turn_server_addr, source_addr,
+                            NR_TRANSPORT_ADDR_CMP_MODE_ALL)) {
+    r_log(NR_LOG_TURN, LOG_WARNING,
+          "TURN(%s): Indication from unexpected source addr %s (expected %s)",
+          ctx->label, source_addr->as_string, ctx->turn_server_addr.as_string);
+    ABORT(R_REJECTED);
+  }
+
+  if ((r=nr_stun_message_create2(&ind, msg, len)))
+    ABORT(r);
+  if ((r=nr_stun_decode_message(ind, 0, 0)))
+    ABORT(r);
+
+  if (ind->header.type != NR_STUN_MSG_DATA_INDICATION)
+    ABORT(R_BAD_ARGS);
+
+  if (!nr_stun_message_has_attribute(ind, NR_STUN_ATTR_XOR_PEER_ADDRESS, &attr))
+    ABORT(R_BAD_ARGS);
+
+  if ((r=nr_turn_permission_find(ctx, &attr->u.xor_mapped_address.unmasked,
+                                 &perm))) {
+    if (r == R_NOT_FOUND) {
+      r_log(NR_LOG_TURN, LOG_WARNING,
+            "TURN(%s): Indication from peer addr %s with no permission",
+            ctx->label, attr->u.xor_mapped_address.unmasked.as_string);
+    }
+    ABORT(r);
+  }
+
+  if ((r=nr_transport_addr_copy(remote_addr,
+                                &attr->u.xor_mapped_address.unmasked)))
+    ABORT(r);
+
+#if REFRESH_RESERVATION_ON_RECV
+  if ((r=nr_turn_client_ensure_perm(ctx, remote_addr))) {
+    ABORT(r);
+  }
+#endif
+
+  if (!nr_stun_message_has_attribute(ind, NR_STUN_ATTR_DATA, &attr)) {
+    ABORT(R_BAD_DATA);
+  }
+
+  assert(newsize >= attr->u.data.length);
+  if (newsize < attr->u.data.length)
+    ABORT(R_BAD_ARGS);
+
+  memcpy(newmsg, attr->u.data.data, attr->u.data.length);
+  *newlen = attr->u.data.length;
+
+  _status=0;
+abort:
+  nr_stun_message_destroy(&ind);
+  return(_status);
+}
+
+
+
+/* The permissions model is as follows:
+
+   - We keep a list of all the permissions we have ever requested
+     along with when they were last established.
+   - Whenever someone sends a packet, we automatically create/
+     refresh the permission.
+
+   This means that permissions automatically time out if
+   unused.
+
+*/
+int nr_turn_client_ensure_perm(nr_turn_client_ctx *ctx, const nr_transport_addr *addr)
+{
+  int r, _status;
+  nr_turn_permission *perm = 0;
+  UINT8 now;
+  UINT8 turn_permission_refresh = (TURN_PERMISSION_LIFETIME_SECONDS -
+                                   TURN_REFRESH_SLACK_SECONDS) * TURN_USECS_PER_S;
+
+  if ((r=nr_turn_permission_find(ctx, addr, &perm))) {
+    if (r == R_NOT_FOUND) {
+      if ((r=nr_turn_permission_create(ctx, addr, &perm)))
+        ABORT(r);
+    }
+    else {
+      ABORT(r);
+    }
+  }
+
+  assert(perm);
+
+  /* Now check that the permission is up-to-date */
+  now = r_gettimeint();
+
+  if ((now - perm->last_used) > turn_permission_refresh) {
+    r_log(NR_LOG_TURN, LOG_DEBUG, "TURN(%s): Permission for %s requires refresh",
+          ctx->label, perm->addr.as_string);
+
+    if ((r=nr_turn_stun_ctx_start(perm->stun)))
+      ABORT(r);
+
+    perm->last_used = now;  /* Update the time now so we don't retry on
+                               next packet */
+  }
+
+  _status=0;
+abort:
+  return(_status);
+}
+
+static int nr_turn_permission_create(nr_turn_client_ctx *ctx, const nr_transport_addr *addr,
+                                     nr_turn_permission **permp)
+{
+  int r, _status;
+  nr_turn_permission *perm = 0;
+
+  assert(ctx->state == NR_TURN_CLIENT_STATE_ALLOCATED);
+
+  r_log(NR_LOG_TURN, LOG_INFO, "TURN(%s): Creating permission for %s",
+        ctx->label, addr->as_string);
+
+  if (!(perm = RCALLOC(sizeof(nr_turn_permission))))
+    ABORT(R_NO_MEMORY);
+
+  if ((r=nr_transport_addr_copy(&perm->addr, addr)))
+    ABORT(r);
+
+  perm->last_used = 0;
+
+  if ((r=nr_turn_stun_ctx_create(ctx, NR_TURN_CLIENT_MODE_PERMISSION_REQUEST,
+                                 nr_turn_client_permissions_cb,
+                                 nr_turn_client_permission_error_cb,
+                                 &perm->stun)))
+    ABORT(r);
+
+  /* We want to authenticate on the first packet */
+  if ((r=nr_turn_stun_set_auth_params(perm->stun, ctx->realm, ctx->nonce)))
+    ABORT(r);
+
+  if ((r=nr_transport_addr_copy(
+          &perm->stun->stun->params.permission_request.remote_addr, addr)))
+    ABORT(r);
+  STAILQ_INSERT_TAIL(&ctx->permissions, perm, entry);
+
+  *permp = perm;
+
+  _status=0;
+abort:
+  if (_status) {
+    nr_turn_permission_destroy(&perm);
+  }
+  return(_status);
+}
+
+
+static int nr_turn_permission_find(nr_turn_client_ctx *ctx, const nr_transport_addr *addr,
+                                   nr_turn_permission **permp)
+{
+  nr_turn_permission *perm;
+  int _status;
+
+  perm = STAILQ_FIRST(&ctx->permissions);
+  while (perm) {
+    if (!nr_transport_addr_cmp(&perm->addr, addr,
+                               NR_TRANSPORT_ADDR_CMP_MODE_ADDR))
+      break;
+
+    perm = STAILQ_NEXT(perm, entry);
+  }
+
+  if (!perm) {
+    ABORT(R_NOT_FOUND);
+  }
+  if (perm->stun->last_error_code == 403) {
+    ABORT(R_NOT_PERMITTED);
+  }
+  *permp = perm;
+
+  _status=0;
+abort:
+  return(_status);
+}
+
+static void nr_turn_client_permissions_cb(NR_SOCKET s, int how, void *arg)
+{
+  nr_turn_stun_ctx *ctx = (nr_turn_stun_ctx *)arg;
+  r_log(NR_LOG_TURN, LOG_DEBUG, "TURN(%s): Successfully refreshed permission",
+        ctx->tctx->label);
+}
+
+/* Note that we don't destroy the nr_turn_stun_ctx. That is owned by the
+   nr_turn_client_ctx. */
+static int nr_turn_permission_destroy(nr_turn_permission **permp)
+{
+  nr_turn_permission *perm;
+
+  if (!permp || !*permp)
+    return(0);
+
+  perm = *permp;
+  *permp = 0;
+
+  RFREE(perm);
+
+  return(0);
+}
+
+#endif /* USE_TURN */
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/stun/turn_client_ctx.h b/dom/media/webrtc/transport/third_party/nICEr/src/stun/turn_client_ctx.h
new file mode 100644
index 0000000000..2049f1bdfb
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/stun/turn_client_ctx.h
@@ -0,0 +1,161 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+Copyright (c) 2013, Mozilla
+
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance, Mozilla nor
+  the names of its contributors may be used to endorse or promote
+  products derived from this software without specific prior written
+  permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _turn_client_ctx_h
+#define _turn_client_ctx_h
+
+struct nr_ice_ctx_;
+
+typedef struct nr_transport_addr_listnode_ {
+  nr_transport_addr value;
+  STAILQ_ENTRY(nr_transport_addr_listnode_) entry;
+} nr_transport_addr_listnode;
+typedef STAILQ_HEAD(nr_transport_addr_listnode_head_, nr_transport_addr_listnode_) nr_transport_addr_listnode_head;
+
+/*
+   Represents a single set of STUN transactions, i.e.,
+   Allocate, Refresh, Permission. It automatically handles
+   permissions and restarts.
+ */
+typedef struct nr_turn_stun_ctx_ {
+  struct nr_turn_client_ctx_ *tctx;
+  int mode;  /* From stun_client_ctx.h, NR_TURN_CLIENT_MODE_* */
+  int retry_ct;
+  nr_stun_client_ctx *stun;
+  char *nonce;
+  char *realm;
+  NR_async_cb success_cb;
+  NR_async_cb error_cb;
+  int last_error_code;
+
+  nr_transport_addr_listnode_head addresses_tried;
+
+  STAILQ_ENTRY(nr_turn_stun_ctx_) entry;
+} nr_turn_stun_ctx;
+typedef STAILQ_HEAD(nr_turn_stun_ctx_head_, nr_turn_stun_ctx_)
+    nr_turn_stun_ctx_head;
+
+/* Represents a single TURN permission */
+typedef struct nr_turn_permission_ {
+  nr_transport_addr addr;
+  nr_turn_stun_ctx *stun;
+  UINT8 last_used;
+
+  STAILQ_ENTRY(nr_turn_permission_) entry;
+} nr_turn_permission;
+typedef STAILQ_HEAD(nr_turn_permission_head_, nr_turn_permission_)
+    nr_turn_permission_head;
+
+/* A single connection to a TURN server. Use one
+   turn_client_ctx per socket/server pair. */
+typedef struct nr_turn_client_ctx_ {
+  int state;
+#define NR_TURN_CLIENT_STATE_INITTED         1
+#define NR_TURN_CLIENT_STATE_ALLOCATING      2
+#define NR_TURN_CLIENT_STATE_ALLOCATED       3
+#define NR_TURN_CLIENT_STATE_FAILED          4
+#define NR_TURN_CLIENT_STATE_CANCELLED       5
+#define NR_TURN_CLIENT_STATE_DEALLOCATING    6
+
+  char *label;
+  nr_socket *sock;
+
+  char *username;
+  Data *password;
+  char *nonce;
+  char *realm;
+
+  nr_transport_addr turn_server_addr;
+  nr_transport_addr relay_addr;
+  nr_transport_addr mapped_addr;
+
+  nr_turn_stun_ctx_head stun_ctxs;
+  nr_turn_permission_head permissions;
+
+  /* We need access to the socket factory to create new TCP sockets for handling
+   * STUN/300 responses. */
+  /* If we were to require TCP nr_sockets to allow multiple connect calls by
+   * disconnecting and re-connecting, we could avoid this requirement. */
+  struct nr_ice_ctx_* ctx;
+
+  NR_async_cb finished_cb;
+  void *cb_arg;
+
+  void *connected_timer_handle;
+  void *refresh_timer_handle;
+
+  // ice telemetry
+  UINT2 cnt_401s;
+  UINT2 cnt_403s;
+  UINT2 cnt_438s;
+} nr_turn_client_ctx;
+
+extern int NR_LOG_TURN;
+
+int nr_transport_addr_listnode_create(const nr_transport_addr *addr, nr_transport_addr_listnode **listnodep);
+void nr_transport_addr_listnode_destroy(nr_transport_addr_listnode **listnode);
+int nr_turn_client_ctx_create(const char* label, nr_socket* sock,
+                              const char* username, Data* password,
+                              nr_transport_addr* addr,
+                              struct nr_ice_ctx_* ice_ctx,
+                              nr_turn_client_ctx** ctxp);
+int nr_turn_client_ctx_destroy(nr_turn_client_ctx **ctxp);
+int nr_turn_client_allocate(nr_turn_client_ctx *ctx,
+                            NR_async_cb finished_cb, void *cb_arg);
+int nr_turn_client_get_relayed_address(nr_turn_client_ctx *ctx,
+                                       nr_transport_addr *relayed_address);
+int nr_turn_client_get_mapped_address(nr_turn_client_ctx *ctx,
+                                      nr_transport_addr *mapped_address);
+int nr_turn_client_process_response(nr_turn_client_ctx *ctx,
+                                    UCHAR *msg, int len,
+                                    nr_transport_addr *turn_server_addr);
+int nr_turn_client_cancel(nr_turn_client_ctx *ctx);
+int nr_turn_client_failed(nr_turn_client_ctx *ctx);
+int nr_turn_client_deallocate(nr_turn_client_ctx *ctx);
+int nr_turn_client_send_indication(nr_turn_client_ctx *ctx,
+                                   const UCHAR *msg, size_t len,
+                                   int flags, const nr_transport_addr *remote_addr);
+int nr_turn_client_parse_data_indication(nr_turn_client_ctx *ctx,
+                                         nr_transport_addr *source_addr,
+                                         UCHAR *msg, size_t len,
+                                         UCHAR *newmsg, size_t *newlen,
+                                         size_t newsize,
+                                         nr_transport_addr *remote_addr);
+int nr_turn_client_ensure_perm(nr_turn_client_ctx *ctx,
+                               const nr_transport_addr *addr);
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/util/cb_args.c b/dom/media/webrtc/transport/third_party/nICEr/src/util/cb_args.c
new file mode 100644
index 0000000000..81f7731d74
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/util/cb_args.c
@@ -0,0 +1,57 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <stdarg.h>
+#include "nr_api.h"
+#include "cb_args.h"
+
+void **nr_pack_cb_args(int ct,...)
+  {
+    void **vlist;
+    va_list ap;
+    int i;
+
+    va_start(ap,ct);
+    if(!(vlist=RCALLOC(sizeof(void *)*ct+1)))
+      abort();
+
+    for(i=0;i<ct;i++){
+      vlist[i]=va_arg(ap, void *);
+    }
+
+    va_end(ap);
+
+    return(vlist);
+  }
+
+
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/util/cb_args.h b/dom/media/webrtc/transport/third_party/nICEr/src/util/cb_args.h
new file mode 100644
index 0000000000..83f7831ccc
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/util/cb_args.h
@@ -0,0 +1,41 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _cb_args_h
+#define _cb_args_h
+
+void **nr_pack_cb_args(int ct,...);
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/util/ice_util.c b/dom/media/webrtc/transport/third_party/nICEr/src/util/ice_util.c
new file mode 100644
index 0000000000..249cfe350f
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/util/ice_util.c
@@ -0,0 +1,71 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <stdarg.h>
+#include <string.h>
+#include "nr_api.h"
+#include "ice_util.h"
+
+int nr_concat_strings(char **outp,...)
+  {
+    va_list ap;
+    char *s,*out=0;
+    int len=0;
+    int _status;
+
+    va_start(ap,outp);
+    while(s=va_arg(ap,char *)){
+      len+=strlen(s);
+    }
+    va_end(ap);
+
+
+    if(!(out=RMALLOC(len+1)))
+      ABORT(R_NO_MEMORY);
+
+    *outp=out;
+
+    va_start(ap,outp);
+    while(s=va_arg(ap,char *)){
+      len=strlen(s);
+      memcpy(out,s,len);
+      out+=len;
+    }
+    va_end(ap);
+
+    *out=0;
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
diff --git a/dom/media/webrtc/transport/third_party/nICEr/src/util/ice_util.h b/dom/media/webrtc/transport/third_party/nICEr/src/util/ice_util.h
new file mode 100644
index 0000000000..44751edaf8
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nICEr/src/util/ice_util.h
@@ -0,0 +1,41 @@
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+
+
+#ifndef _ice_util_h
+#define _ice_util_h
+
+int nr_concat_strings(char **outp,...);
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/COPYRIGHT b/dom/media/webrtc/transport/third_party/nrappkit/COPYRIGHT
new file mode 100644
index 0000000000..b0bb25595e
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/COPYRIGHT
@@ -0,0 +1,159 @@
+
+Copyright (C) 2006, Network Resonance, Inc.
+All Rights Reserved
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+
+1. Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+2. Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+3. Neither the name of Network Resonance, Inc. nor the name of any
+   contributors to this software may be used to endorse or promote 
+   products derived from this software without specific prior written
+   permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE 
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR 
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF 
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN 
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+
+
+This distribution also contains material from ssldump, tcpdump, and
+FreeBSD. The licenses are on the individual source files but follow
+here as well.
+
+SSLDUMP LICENSE
+Copyright (C) 1999-2001 RTFM, Inc.
+All Rights Reserved
+
+This package is a SSLv3/TLS protocol analyzer written by Eric Rescorla
+<ekr@rtfm.com> and licensed by RTFM, Inc.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+1. Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+2. Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+3. All advertising materials mentioning features or use of this software
+   must display the following acknowledgement:
+
+   This product includes software developed by Eric Rescorla for
+   RTFM, Inc.
+
+4. Neither the name of RTFM, Inc. nor the name of Eric Rescorla may be
+   used to endorse or promote products derived from this
+   software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE ERIC RESCORLA AND RTFM ``AS IS'' AND
+ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+SUCH DAMAGE.
+
+TCPDUMP LICENSE
+The manual page for this software is partially excerpted from
+the tcpdump manual page, which is subject to the following license:
+Copyright (c) 1987, 1988, 1989, 1990, 1991, 1992, 1994, 1995, 1996, 1997
+	The Regents of the University of California.  All rights reserved.
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that: (1) source code distributions
+retain the above copyright notice and this paragraph in its entirety, (2)
+distributions including binary code include the above copyright notice and
+this paragraph in its entirety in the documentation or other materials
+provided with the distribution, and (3) all advertising materials mentioning
+features or use of this software display the following acknowledgement:
+``This product includes software developed by the University of California,
+Lawrence Berkeley Laboratory and its contributors.'' Neither the name of
+the University nor the names of its contributors may be used to endorse
+or promote products derived from this software without specific prior
+written permission.
+THIS SOFTWARE IS PROVIDED ``AS IS'' AND WITHOUT ANY EXPRESS OR IMPLIED
+WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED WARRANTIES OF
+MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE.
+
+
+The compilation of software known as FreeBSD is distributed under the
+following terms:
+
+Copyright (C) 1992-2004 The FreeBSD Project. All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+1. Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+2. Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+THIS SOFTWARE IS PROVIDED BY AUTHOR AND CONTRIBUTORS ``AS IS'' AND
+ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED.  IN NO EVENT SHALL AUTHOR OR CONTRIBUTORS BE LIABLE
+FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+SUCH DAMAGE.
+
+The 4.4BSD and 4.4BSD-Lite software is distributed under the following
+terms:
+
+All of the documentation and software included in the 4.4BSD and 4.4BSD-Lite
+Releases is copyrighted by The Regents of the University of California.
+
+Copyright 1979, 1980, 1983, 1986, 1988, 1989, 1991, 1992, 1993, 1994
+	The Regents of the University of California.  All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+1. Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+2. Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+3. All advertising materials mentioning features or use of this software
+   must display the following acknowledgement:
+This product includes software developed by the University of
+California, Berkeley and its contributors.
+4. Neither the name of the University nor the names of its contributors
+   may be used to endorse or promote products derived from this software
+   without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND
+ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+SUCH DAMAGE.
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/README b/dom/media/webrtc/transport/third_party/nrappkit/README
new file mode 100644
index 0000000000..9cf43319b3
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/README
@@ -0,0 +1,133 @@
+$Id: README,v 1.3 2007/11/21 00:09:10 adamcain Exp $ 
+
+nrappkit 1.0b2
+Copyright (C) 2006 Network Resonance, Inc.
+
+
+nrappkit is a toolkit for building standalone applications and
+appliances. It provides:
+
+- registry-based configuration (with change callbacks)
+- extensible command and configuration shell
+- extensible statistics system
+- configurable logging system
+- event and timer handling
+- generic plugin system
+- launcher daemon
+
+The contents of nrappkit were extracted from Network Resonance's
+product on the theory that they were generally useful for 
+application developers. 
+
+THIS PACKAGE DOES NOT GRANT A LICENSE OR RIGHT TO ANY OTHER NETWORK
+RESONANCE TECHNOLOGY OR SOFTWARE.
+
+
+
+BUILDING
+
+Builds are done semi-manually with port directories for each
+platform. There are pre-existing ports to FreeBSD, Linux (Ubuntu
+and Fedora Core), and Darwin (MacOSX). To build the system:
+
+	cd src/make/<platform>
+	gmake
+
+Some of the platforms come in several variants. Most notably,
+if a platform exists in "regular" and "-appliance" variant,
+this means that the regular variant just builds binaries intended
+to be run out of the make directory (for development) and the
+appliance variant is intended to be installed in a real system.
+
+By default we want to install things owned as user "pcecap".
+Either make this user or edit the Makefile to be a user you
+like (e.g., nobody).
+
+If you want to include the 'nrsh' command-line configuration
+tool in your build, you will need to make sure the line
+  BUILD_NRSH=yes
+appears (uncommented-out) in your platform Makefile.  You will 
+also need to to build OpenSSL and libedit and point your nrappkit
+Makefile to the correct paths. You can obtain these packages at:
+    openssl-0.9.7l
+    http://www.openssl.org/source/openssl-0.9.7l.tar.gz
+
+    libedit-20060829-2.9
+    http://freshmeat.net/redir/editline/53029/url_tgz/libedit-20060829-2.9.tar.gz
+
+
+INSTALLING
+If you're doing an appliance as opposed to a development build,
+you'll want to install it. This is easy:
+
+	su
+ 	gmake install
+
+Most binaries and libraries ends up in /usr/local/pcecap while
+data files are in /var/pcecap. However, you can tweak
+this in the Makefile. By default it's all owned by pcecap.
+
+To ensure that dynamic libraries are loaded correctly at runtime,
+you'd want to make sure the right directory is included in your
+LD_LIBRARY_PATH or via ldconfig.
+
+
+QUICK TOUR
+The build makes the following binaries that you may find useful:
+
+- captured    -- the launcher (the name is historical)
+- registryd   -- the registry daemon
+- nrregctl    -- a registry control program
+- nrsh        -- the command shell (when included in build)
+- nrstatsctl  -- the stats control program
+
+Using the nrcapctl script is the easiest way to interact with
+the applications. It is run as "nrcapctl <command>" with the
+following commands recognized:
+
+  startup            -- fires up captured, which in turn runs and
+                        initializes the registry
+
+  shutdown           -- kills captured and its child processes
+
+  status             -- prints the running status of captured in
+                        human-readable form
+
+  stat               -- prints the running status of captured in
+                        a form easily parsed by scripts
+
+  enable             -- alters the mode.txt file so that captured
+                        starts 
+
+  disable            -- alters the mode.txt file so that captured
+                        does not start
+
+  clear-statistics   -- equivalent to "nrstatsctl -z" (requires
+                        that captured be running)
+
+Note: the "start" and "stop" nrcapctl commands do nothing as they
+use components not included in nrappkit. However the associated
+script logic in nrcapctl demonstrates how additional applications
+might be launched using nrcapctl and particular registry settings.
+
+
+EXTENDING
+When things come up, they're pretty dumb. You'll probably want to
+write your own applications, otherwise it's not clear why you're doing
+this. The general idea is that you write your application using the
+facilities that nrappkit provides and then write plugins to the
+nrappkit components as necessary. So, for example, say you want
+to write a network daemon. You would:
+
+ - configure the launcher to launch your daemon (using the registry,
+   naturally).
+ - make calls to the registry to get configuration data
+ - make calls to the logging system to log data
+ - implement a stats module to record statistics
+ - write a plugin to nrsh to let people configure your parameters
+
+Examples of some of this stuff can be found in examples/demo_plugin.
+Otherwise, read the source. More documentation will be on the way,
+hopefully.
+
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/README_MOZILLA b/dom/media/webrtc/transport/third_party/nrappkit/README_MOZILLA
new file mode 100644
index 0000000000..f7cdb4cabb
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/README_MOZILLA
@@ -0,0 +1,21 @@
+This import of nrappkit is a subset of the distribution at:
+
+  nrappkit.sourceforge.net
+
+
+The last revision included in this import was on Nov 25, 2008, version 1.0b2.
+
+The upstream project is now dead (no update since 2013), we can conisder this
+version as a fork.
+
+Out of the list in the README, we use:
+
+- registry-based configuration (with change callbacks)
+  [but without the registry daemon]
+- configurable logging system
+- event and timer handling
+  [though partly reimplemented]
+
+Also, we use a bunch of the generic utilities such as string handling,
+generic hash tables in C, yet another concrete type mapping, etc.
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/nrappkit.gyp b/dom/media/webrtc/transport/third_party/nrappkit/nrappkit.gyp
new file mode 100644
index 0000000000..c3f88af4bc
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/nrappkit.gyp
@@ -0,0 +1,251 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+#
+# nrappkit.gyp
+#
+#
+{
+  'targets' : [
+      {
+          'target_name' : 'nrappkit',
+          'type' : 'static_library',
+
+          'include_dirs' : [
+              # EXTERNAL
+              # INTERNAL
+              'src/event',
+              'src/log',
+              'src/port/generic/include',
+              'src/registry',
+              'src/share',
+              'src/stats',
+              'src/util',
+              'src/util/libekr',
+          ],
+
+          'sources' : [
+              # Shared
+#              './src/share/nr_api.h',
+              './src/share/nr_common.h',
+#              './src/share/nr_dynlib.h',
+              './src/share/nr_reg_keys.h',
+#              './src/share/nr_startup.c',
+#              './src/share/nr_startup.h',
+#              './src/share/nrappkit_static_plugins.c',
+               './src/port/generic/include'
+
+              # libekr
+              './src/util/libekr/assoc.h',
+#              './src/util/libekr/debug.c',
+#              './src/util/libekr/debug.h',
+              './src/util/libekr/r_assoc.c',
+              './src/util/libekr/r_assoc.h',
+#              './src/util/libekr/r_assoc_test.c',
+              './src/util/libekr/r_common.h',
+              './src/util/libekr/r_crc32.c',
+              './src/util/libekr/r_crc32.h',
+              './src/util/libekr/r_data.c',
+              './src/util/libekr/r_data.h',
+              './src/util/libekr/r_defaults.h',
+              './src/util/libekr/r_errors.c',
+              './src/util/libekr/r_errors.h',
+              './src/util/libekr/r_includes.h',
+#              './src/util/libekr/r_list.c',
+#              './src/util/libekr/r_list.h',
+              './src/util/libekr/r_macros.h',
+              './src/util/libekr/r_memory.c',
+              './src/util/libekr/r_memory.h',
+              './src/util/libekr/r_replace.c',
+              './src/util/libekr/r_thread.h',
+              './src/util/libekr/r_time.c',
+              './src/util/libekr/r_time.h',
+              './src/util/libekr/r_types.h',
+
+              # Utilities
+              './src/util/byteorder.c',
+              './src/util/byteorder.h',
+              #'./src/util/escape.c',
+              #'./src/util/escape.h',
+              #'./src/util/filename.c',
+              #'./src/util/getopt.c',
+              #'./src/util/getopt.h',
+              './src/util/hex.c',
+              './src/util/hex.h',
+              #'./src/util/mem_util.c',
+              #'./src/util/mem_util.h',
+              #'./src/util/mutex.c',
+              #'./src/util/mutex.h',
+              './src/util/p_buf.c',
+              './src/util/p_buf.h',
+              #'./src/util/ssl_util.c',
+              #'./src/util/ssl_util.h',
+              './src/util/util.c',
+              './src/util/util.h',
+              #'./src/util/util_db.c',
+              #'./src/util/util_db.h',
+
+              # Events
+#              './src/event/async_timer.c',
+              './src/event/async_timer.h',
+#              './src/event/async_wait.c',
+              './src/event/async_wait.h',
+              './src/event/async_wait_int.h',
+
+              # Logging
+              './src/log/r_log.c',
+              './src/log/r_log.h',
+              #'./src/log/r_log_plugin.c',
+
+              # Registry
+              './src/registry/c2ru.c',
+              './src/registry/c2ru.h',
+              #'./src/registry/mod_registry/mod_registry.c',
+              #'./src/registry/nrregctl.c',
+              #'./src/registry/nrregistryctl.c',
+              './src/registry/registry.c',
+              './src/registry/registry.h',
+              './src/registry/registry_int.h',
+              './src/registry/registry_local.c',
+              #'./src/registry/registry_plugin.c',
+              './src/registry/registry_vtbl.h',
+              './src/registry/registrycb.c',
+              #'./src/registry/registryd.c',
+              #'./src/registry/regrpc.h',
+              #'./src/registry/regrpc_client.c',
+              #'./src/registry/regrpc_client.h',
+              #'./src/registry/regrpc_client_cb.c',
+              #'./src/registry/regrpc_clnt.c',
+              #'./src/registry/regrpc_server.c',
+              #'./src/registry/regrpc_svc.c',
+              #'./src/registry/regrpc_xdr.c',
+
+              # Statistics
+              #'./src/stats/nrstats.c',
+              #'./src/stats/nrstats.h',
+              #'./src/stats/nrstats_app.c',
+              #'./src/stats/nrstats_int.h',
+              #'./src/stats/nrstats_memory.c',
+          ],
+
+          'defines' : [
+              'SANITY_CHECKS',
+              'R_PLATFORM_INT_TYPES=<stdint.h>',
+              'R_DEFINED_INT2=int16_t',
+              'R_DEFINED_UINT2=uint16_t',
+              'R_DEFINED_INT4=int32_t',
+              'R_DEFINED_UINT4=uint32_t',
+              'R_DEFINED_INT8=int64_t',
+              'R_DEFINED_UINT8=uint64_t',
+          ],
+
+          'conditions' : [
+              ## Mac and BSDs
+              [ 'OS == "mac"', {
+                'defines' : [
+                    'DARWIN',
+                ],
+              }],
+              [ 'os_bsd == 1', {
+                'defines' : [
+                    'BSD',
+                ],
+              }],
+              [ 'OS == "mac" or OS == "ios" or os_bsd == 1', {
+                'cflags_mozilla': [
+                    '-Wall',
+                    '-Wno-parentheses',
+                    '-Wno-strict-prototypes',
+                    '-Wmissing-prototypes',
+                    '-Wno-format',
+                    '-Wno-format-security',
+                 ],
+                 'defines' : [
+                     'HAVE_LIBM=1',
+                     'HAVE_STRDUP=1',
+                     'HAVE_STRLCPY=1',
+                     'HAVE_SYS_TIME_H=1',
+                     'HAVE_VFPRINTF=1',
+                     'NEW_STDIO'
+                     'RETSIGTYPE=void',
+                     'TIME_WITH_SYS_TIME_H=1',
+                     '__UNUSED__=__attribute__((unused))',
+                 ],
+
+                 'include_dirs': [
+                     'src/port/darwin/include'
+                 ],
+
+                 'sources': [
+                      './src/port/darwin/include/csi_platform.h',
+                 ],
+              }],
+
+              ## Win
+              [ 'OS == "win"', {
+                 'defines' : [
+                     'WIN',
+                     '__UNUSED__=',
+                     'HAVE_STRDUP=1',
+                     'NO_REG_RPC'
+                 ],
+
+                 'include_dirs': [
+                     'src/port/win32/include'
+                 ],
+
+                 'sources': [
+                      './src/port/win32/include/csi_platform.h',
+                 ],
+              }],
+
+              # Windows, clang-cl build
+              [ 'clang_cl == 1', {
+                'cflags_mozilla': [
+                    '-Xclang',
+                    '-Wall',
+                    '-Xclang',
+                    '-Wno-parentheses',
+                    '-Wno-strict-prototypes',
+                    '-Wmissing-prototypes',
+                    '-Wno-format',
+                    '-Wno-format-security',
+                 ],
+              }],
+
+              ## Linux/Android
+              [ '(OS == "linux") or (OS == "android")', {
+                 'cflags_mozilla': [
+                     '-Wall',
+                     '-Wno-parentheses',
+                     '-Wno-strict-prototypes',
+                     '-Wmissing-prototypes',
+                     '-Wno-format',
+                     '-Wno-format-security',
+                 ],
+                 'defines' : [
+                     'LINUX',
+                     'HAVE_LIBM=1',
+                     'HAVE_STRDUP=1',
+                     'HAVE_STRLCPY=1',
+                     'HAVE_SYS_TIME_H=1',
+                     'HAVE_VFPRINTF=1',
+                     'NEW_STDIO'
+                     'RETSIGTYPE=void',
+                     'TIME_WITH_SYS_TIME_H=1',
+                     'NO_REG_RPC=1',
+                     '__UNUSED__=__attribute__((unused))',
+                 ],
+
+                 'include_dirs': [
+                     'src/port/linux/include'
+                 ],
+                 'sources': [
+                      './src/port/linux/include/csi_platform.h',
+                 ],
+              }]
+          ]
+      }]
+}
+
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/event/async_timer.h b/dom/media/webrtc/transport/third_party/nrappkit/src/event/async_timer.h
new file mode 100644
index 0000000000..544a3d44fd
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/event/async_timer.h
@@ -0,0 +1,54 @@
+/**
+   async_timer.h
+
+
+   Copyright (C) 2004, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   ekr@rtfm.com  Sun Feb 22 19:35:24 2004
+ */
+
+
+#ifndef _async_timer_h
+#define _async_timer_h
+
+
+int NR_async_timer_init(void);
+int NR_async_timer_set(int delay_ms,NR_async_cb cb,void *cb_arg,char *function,int line,void **handle);
+int NR_async_timer_cancel(void *handle);
+int NR_async_timer_update_time(struct timeval *tv);
+int NR_async_timer_next_timeout(int *delay_ms);
+int NR_async_timer_sanity_check_for_cb_deleted(NR_async_cb cb,void *cb_arg);
+
+#define NR_ASYNC_TIMER_SET(d,c,a,hp) NR_async_timer_set(d,c,a,(char *)__FUNCTION__,__LINE__,hp)
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/event/async_wait.h b/dom/media/webrtc/transport/third_party/nrappkit/src/event/async_wait.h
new file mode 100644
index 0000000000..58bcd435eb
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/event/async_wait.h
@@ -0,0 +1,83 @@
+/**
+   async_wait.h
+
+
+   Copyright (C) 2001-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   ekr@rtfm.com  Thu Dec 20 20:14:49 2001
+ */
+
+
+#ifndef _async_wait_h
+#define _async_wait_h
+
+#ifdef HAVE_SYS_TIME_H
+#include <sys/time.h>
+#endif
+#include <csi_platform.h>
+
+typedef void (*NR_async_cb)(NR_SOCKET resource,int how,void *arg);
+
+#define NR_ASYNC_WAIT_READ 0
+#define NR_ASYNC_WAIT_WRITE 1
+
+
+int NR_async_wait_init(void);
+int NR_async_wait(NR_SOCKET sock, int how, NR_async_cb cb,void *cb_arg,
+                            char *function,int line);
+int NR_async_cancel(NR_SOCKET sock,int how);
+int NR_async_schedule(NR_async_cb cb,void *arg,char *function,int line);
+int NR_async_event_wait(int *eventsp);
+int NR_async_event_wait2(int *eventsp,struct timeval *tv);
+
+
+#ifdef NR_DEBUG_ASYNC
+#define NR_ASYNC_WAIT(s,h,cb,arg) do { \
+fprintf(stderr,"NR_async_wait(%d,%s,%s) at %s(%d)\n",s,#h,#cb,__FUNCTION__,__LINE__); \
+       NR_async_wait(s,h,cb,arg,(char *)__FUNCTION__,__LINE__);            \
+} while (0)
+#define NR_ASYNC_SCHEDULE(cb,arg) do { \
+fprintf(stderr,"NR_async_schedule(%s) at %s(%d)\n",#cb,__FUNCTION__,__LINE__);\
+       NR_async_schedule(cb,arg,(char *)__FUNCTION__,__LINE__);            \
+} while (0)
+#define NR_ASYNC_CANCEL(s,h) do { \
+       fprintf(stderr,"NR_async_cancel(%d,%s) at %s(%d)\n",s,#h,(char *)__FUNCTION__,__LINE__); \
+NR_async_cancel(s,h); \
+} while (0)
+#else
+#define NR_ASYNC_WAIT(a,b,c,d) NR_async_wait(a,b,c,d,(char *)__FUNCTION__,__LINE__)
+#define NR_ASYNC_SCHEDULE(a,b) NR_async_schedule(a,b,(char *)__FUNCTION__,__LINE__)
+#define NR_ASYNC_CANCEL NR_async_cancel
+#endif
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/event/async_wait_int.h b/dom/media/webrtc/transport/third_party/nrappkit/src/event/async_wait_int.h
new file mode 100644
index 0000000000..d17372d183
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/event/async_wait_int.h
@@ -0,0 +1,62 @@
+/**
+   async_wait_int.h
+
+
+   Copyright (C) 2004, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   ekr@rtfm.com  Sun Feb 22 19:16:01 2004
+ */
+
+
+#ifndef _async_wait_int_h
+#define _async_wait_int_h
+
+
+typedef struct callback_ {
+     NR_async_cb cb;
+     void *arg;
+     int how;
+     int resource;
+     void *backptr;
+
+     char *func;
+     int free_func;
+     int line;
+     TAILQ_ENTRY(callback_) entry;
+} callback;
+
+int nr_async_create_cb(NR_async_cb cb,void *arg,int how,
+  int resource,char *func,int line,callback **cbp);
+int nr_async_destroy_cb(callback **cbp);
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/log/r_log.c b/dom/media/webrtc/transport/third_party/nrappkit/src/log/r_log.c
new file mode 100644
index 0000000000..09bb24749f
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/log/r_log.c
@@ -0,0 +1,696 @@
+/**
+   r_log.c
+
+
+   Copyright (C) 2001, RTFM, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   ekr@rtfm.com  Mon Dec  3 15:24:38 2001
+ */
+
+#ifdef LINUX
+#define _BSD_SOURCE
+#define _DEFAULT_SOURCE
+#endif
+
+#include "r_log.h"
+#include "hex.h"
+
+#include <string.h>
+#include <errno.h>
+#ifndef _MSC_VER
+#include <strings.h>
+#include <syslog.h>
+#endif
+#include <registry.h>
+#include <time.h>
+
+
+#include "nr_common.h"
+#include "nr_reg_keys.h"
+
+
+#define LOGGING_DEFAULT_LEVEL  5
+
+int NR_LOG_LOGGING = 0;
+
+static char *log_level_strings[]={
+     "EMERG",
+     "ALERT",
+     "CRIT",
+     "ERR",
+     "WARNING",
+     "NOTICE",
+     "INFO",
+     "DEBUG"
+};
+
+static char *log_level_reg_strings[]={
+     "emergency",
+     "alert",
+     "critical",
+     "error",
+     "warning",
+     "notice",
+     "info",
+     "debug"
+};
+
+#define LOGGING_REG_PREFIX "logging"
+
+#define MAX_ERROR_STRING_SIZE   512
+
+#define R_LOG_INITTED1   1
+#define R_LOG_INITTED2   2
+
+#define LOG_NUM_DESTINATIONS 3
+
+typedef struct log_type_ {
+     char *facility_name;
+     int level[LOG_NUM_DESTINATIONS];
+     NR_registry dest_facility_key[LOG_NUM_DESTINATIONS];
+} log_type;
+
+#define MAX_LOG_TYPES 16
+
+static log_type log_types[MAX_LOG_TYPES];
+static int log_type_ct;
+
+
+typedef struct log_destination_ {
+     char *dest_name;
+     int enabled;
+     int default_level;
+     r_dest_vlog *dest_vlog;
+} log_destination;
+
+
+#define LOG_LEVEL_UNDEFINED        -1
+#define LOG_LEVEL_NONE             -2
+#define LOG_LEVEL_USE_DEST_DEFAULT -3
+
+static int stderr_vlog(int facility,int level,const char *format,va_list ap);
+static int syslog_vlog(int facility,int level,const char *format,va_list ap);
+static int noop_vlog(int facility,int level,const char *format,va_list ap);
+
+static log_destination log_destinations[LOG_NUM_DESTINATIONS]={
+     {
+          "stderr",
+          0,
+          LOGGING_DEFAULT_LEVEL,
+          stderr_vlog,
+     },
+     {
+          "syslog",
+#ifndef WIN32
+          1,
+#else
+          0,
+#endif
+          LOGGING_DEFAULT_LEVEL,
+          syslog_vlog,
+     },
+     {
+          "extra",
+          0,
+          LOGGING_DEFAULT_LEVEL,
+          noop_vlog,
+     },
+};
+
+static int r_log_level=LOGGING_DEFAULT_LEVEL;
+static int r_log_level_environment=0;
+static int r_log_initted=0;
+static int r_log_env_verbose=0;
+
+static void r_log_facility_change_cb(void *cb_arg, char action, NR_registry name);
+static void r_log_facility_delete_cb(void *cb_arg, char action, NR_registry name);
+static void r_log_destination_change_cb(void *cb_arg, char action, NR_registry name);
+static void r_log_default_level_change_cb(void *cb_arg, char action, NR_registry name);
+static int r_log_get_default_level(void);
+static int r_log_get_destinations(int usereg);
+static int r_logging_dest(int dest_index, int facility, int level);
+static int _r_log_init(int usereg);
+static int r_log_get_reg_level(NR_registry name, int *level);
+
+int r_log_register(char *facility_name,int *log_facility)
+  {
+    int i,j;
+    int level;
+    int r,_status;
+    char *buf=0;
+    NR_registry dest_prefix, dest_facility_prefix;
+
+    for(i=0;i<log_type_ct;i++){
+      if(!strcmp(facility_name,log_types[i].facility_name)){
+        *log_facility=i;
+        return(0);
+      }
+    }
+
+    if(log_type_ct==MAX_LOG_TYPES){
+      ABORT(R_INTERNAL);
+    }
+
+    i=log_type_ct;
+
+    /* Initial registration completed, increment log_type_ct */
+    log_types[i].facility_name=r_strdup(facility_name);
+    *log_facility=log_type_ct;
+    log_type_ct++;
+
+    for(j=0; j<LOG_NUM_DESTINATIONS; j++){
+      log_types[i].level[j]=LOG_LEVEL_UNDEFINED;
+
+      if(NR_reg_initted()){
+
+        if((size_t)snprintf(dest_prefix,sizeof(NR_registry),
+          "logging.%s.facility",log_destinations[j].dest_name)>=sizeof(NR_registry))
+          ABORT(R_INTERNAL);
+
+        if (r=NR_reg_make_registry(dest_prefix,facility_name,dest_facility_prefix))
+          ABORT(r);
+
+        if((size_t)snprintf(log_types[i].dest_facility_key[j],sizeof(NR_registry),
+          "%s.level",dest_facility_prefix)>=sizeof(NR_registry))
+          ABORT(R_INTERNAL);
+
+        if(!r_log_get_reg_level(log_types[i].dest_facility_key[j],&level)){
+          log_types[i].level[j]=level;
+        }
+
+        /* Set a callback for the facility's level */
+        if(r=NR_reg_register_callback(log_types[i].dest_facility_key[j],
+          NR_REG_CB_ACTION_ADD|NR_REG_CB_ACTION_CHANGE,
+          r_log_facility_change_cb,(void *)&(log_types[i].level[j])))
+          ABORT(r);
+        if(r=NR_reg_register_callback(log_types[i].dest_facility_key[j],
+          NR_REG_CB_ACTION_DELETE,
+          r_log_facility_delete_cb,(void *)&(log_types[i].level[j])))
+          ABORT(r);
+
+      }
+    }
+
+    _status=0;
+  abort:
+    if(_status)
+      RFREE(buf);
+    return(_status);
+  }
+
+int r_log_facility(int facility,char **typename)
+  {
+    if(facility >= 0 && facility < log_type_ct){
+      *typename=log_types[facility].facility_name;
+      return(0);
+    }
+    return(R_NOT_FOUND);
+  }
+
+static int r_log_get_reg_level(NR_registry name, int *out)
+  {
+    char level[32];
+    int r,_status;
+    int i;
+
+    if(r=NR_reg_get_string(name,level,sizeof(level)))
+      ABORT(r);
+
+    if(!strcasecmp(level,"none")){
+      *out=LOG_LEVEL_NONE;
+      return(0);
+    }
+
+    for(i=0;i<=LOG_DEBUG;i++){
+      if(!strcasecmp(level,log_level_reg_strings[i])){
+        *out=(int)i;
+        return(0);
+      }
+    }
+
+    if(i>LOG_DEBUG){
+      *out=LOG_LEVEL_UNDEFINED;
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+/* Handle the case where a value changes */
+static void r_log_facility_change_cb(void *cb_arg, char action, NR_registry name)
+  {
+    int *lt_level=(int *)cb_arg;
+    int level;
+    int r,_status;
+
+    if(r=r_log_get_reg_level(name,&level))
+      ABORT(r);
+
+    *lt_level=level;
+
+    _status=0;
+  abort:
+    (void)_status; // to avoid unused variable warning and still conform to
+                   // pattern of using ABORT
+    return;
+  }
+
+/* Handle the case where a value is deleted */
+static void r_log_facility_delete_cb(void *cb_arg, char action, NR_registry name)
+  {
+    int *lt_level=(int *)cb_arg;
+
+    *lt_level=LOG_LEVEL_UNDEFINED;
+  }
+
+int r_log(int facility,int level,const char *format,...)
+  {
+    va_list ap;
+
+    va_start(ap,format);
+
+    r_vlog(facility,level,format,ap);
+    va_end(ap);
+
+    return(0);
+  }
+
+int r_dump(int facility,int level,char *name,char *data,int len)
+  {
+    char *hex = 0;
+    size_t unused;
+
+    if(!r_logging(facility,level))
+      return(0);
+
+    hex=RMALLOC((len*2)+1);
+    if (!hex)
+      return(R_FAILED);
+
+    if (nr_nbin2hex((UCHAR*)data, len, hex, len*2+1, &unused))
+      strcpy(hex, "?");
+
+    if(name)
+      r_log(facility,level,"%s[%d]=%s",name,len,hex);
+    else
+      r_log(facility,level,"%s",hex);
+
+    RFREE(hex);
+    return(0);
+  }
+
+// Some platforms (notably WIN32) do not have this
+#ifndef va_copy
+  #ifdef WIN32
+    #define va_copy(dest, src) ( (dest) = (src) )
+  #else  // WIN32
+    #error va_copy undefined, and semantics of assignment on va_list unknown
+  #endif //WIN32
+#endif //va_copy
+
+int r_vlog(int facility,int level,const char *format,va_list ap)
+  {
+    char log_fmt_buf[MAX_ERROR_STRING_SIZE];
+    char *level_str="unknown";
+    char *facility_str="unknown";
+    char *fmt_str=(char *)format;
+    int i;
+
+    if(r_log_env_verbose){
+      if((level>=LOG_EMERG) && (level<=LOG_DEBUG))
+        level_str=log_level_strings[level];
+
+      if(facility >= 0 && facility < log_type_ct)
+        facility_str=log_types[facility].facility_name;
+
+      snprintf(log_fmt_buf, MAX_ERROR_STRING_SIZE, "(%s/%s) %s",
+        facility_str,level_str,format);
+
+      log_fmt_buf[MAX_ERROR_STRING_SIZE-1]=0;
+      fmt_str=log_fmt_buf;
+    }
+
+    for(i=0; i<LOG_NUM_DESTINATIONS; i++){
+      if(r_logging_dest(i,facility,level)){
+        // Some platforms do not react well when you use a va_list more than
+        // once
+        va_list copy;
+        va_copy(copy, ap);
+        log_destinations[i].dest_vlog(facility,level,fmt_str,copy);
+        va_end(copy);
+      }
+    }
+    return(0);
+  }
+
+int stderr_vlog(int facility,int level,const char *format,va_list ap)
+  {
+#if 0 /* remove time stamping, for now */
+    char cbuf[30];
+    time_t tt;
+
+    tt=time(0);
+
+    ctime_r(&tt,cbuf);
+    cbuf[strlen(cbuf)-1]=0;
+
+    fprintf(stderr,"%s: ",cbuf);
+#endif
+
+    vfprintf(stderr,format,ap);
+      fprintf(stderr,"\n");
+    return(0);
+    }
+
+int syslog_vlog(int facility,int level,const char *format,va_list ap)
+  {
+#ifndef WIN32
+    vsyslog(level|LOG_LOCAL0,format,ap);
+#endif
+    return(0);
+  }
+
+int noop_vlog(int facility,int level,const char *format,va_list ap)
+  {
+    return(0);
+  }
+
+int r_log_e(int facility,int level,const char *format,...)
+  {
+    va_list ap;
+
+    va_start(ap,format);
+    r_vlog_e(facility,level,format,ap);
+    va_end(ap);
+
+    return(0);
+  }
+
+int r_vlog_e(int facility,int level,const char *format,va_list ap)
+  {
+    char log_fmt_buf[MAX_ERROR_STRING_SIZE];
+    if(r_logging(facility,level)) {
+      int formatlen = strlen(format);
+
+      if(formatlen+2 > MAX_ERROR_STRING_SIZE)
+        return(1);
+
+      strncpy(log_fmt_buf, format, formatlen);
+      strcpy(&log_fmt_buf[formatlen], ": ");
+      snprintf(&log_fmt_buf[formatlen+2], MAX_ERROR_STRING_SIZE - formatlen - 2, "%s",
+#ifdef WIN32
+               strerror(WSAGetLastError()));
+#else
+               strerror(errno));
+#endif
+      log_fmt_buf[MAX_ERROR_STRING_SIZE-1]=0;
+
+      r_vlog(facility,level,log_fmt_buf,ap);
+    }
+    return(0);
+  }
+
+int r_log_nr(int facility,int level,int r,const char *format,...)
+  {
+    va_list ap;
+
+    va_start(ap,format);
+    r_vlog_nr(facility,level,r,format,ap);
+    va_end(ap);
+
+    return(0);
+  }
+
+int r_vlog_nr(int facility,int level,int r,const char *format,va_list ap)
+  {
+    char log_fmt_buf[MAX_ERROR_STRING_SIZE];
+    if(r_logging(facility,level)) {
+      int formatlen = strlen(format);
+
+      if(formatlen+2 > MAX_ERROR_STRING_SIZE)
+        return(1);
+      strncpy(log_fmt_buf, format, formatlen);
+      strcpy(&log_fmt_buf[formatlen], ": ");
+      snprintf(&log_fmt_buf[formatlen+2], MAX_ERROR_STRING_SIZE - formatlen - 2, "%s",
+               nr_strerror(r));
+
+      log_fmt_buf[MAX_ERROR_STRING_SIZE-1]=0;
+
+      r_vlog(facility,level,log_fmt_buf,ap);
+    }
+    return(0);
+  }
+
+static int r_logging_dest(int dest_index, int facility, int level)
+  {
+    int thresh;
+
+    _r_log_init(0);
+
+    if(!log_destinations[dest_index].enabled)
+      return(0);
+
+    if(level <= r_log_level_environment)
+      return(1);
+
+    if(r_log_initted<R_LOG_INITTED2)
+      return(level<=r_log_level);
+
+    if(facility < 0 || facility > log_type_ct)
+      thresh=r_log_level;
+    else{
+      if(log_types[facility].level[dest_index]==LOG_LEVEL_NONE)
+        return(0);
+
+      if(log_types[facility].level[dest_index]>=0)
+        thresh=log_types[facility].level[dest_index];
+      else if(log_destinations[dest_index].default_level!=LOG_LEVEL_UNDEFINED)
+        thresh=log_destinations[dest_index].default_level;
+      else
+        thresh=r_log_level;
+    }
+
+    if(level<=thresh)
+      return(1);
+
+    return(0);
+  }
+
+int r_logging(int facility, int level)
+  {
+    int i;
+
+    _r_log_init(0);
+
+    /* return 1 if logging is on for any dest */
+
+    for(i=0; i<LOG_NUM_DESTINATIONS; i++){
+      if(r_logging_dest(i,facility,level))
+        return(1);
+    }
+
+    return(0);
+  }
+
+
+static int r_log_get_default_level(void)
+  {
+    char *log;
+    int _status;
+
+    log=getenv("R_LOG_LEVEL");
+
+    if(log){
+      r_log_level=atoi(log);
+      r_log_level_environment=atoi(log);
+    }
+    else{
+      r_log_level=LOGGING_DEFAULT_LEVEL;
+    }
+
+    _status=0;
+  //abort:
+    return(_status);
+  }
+
+
+static int r_log_get_destinations(int usereg)
+  {
+    char *log;
+    int i;
+    int r,_status;
+
+    log=getenv("R_LOG_DESTINATION");
+    if(log){
+      for(i=0; i<LOG_NUM_DESTINATIONS; i++)
+        log_destinations[i].enabled=!strcmp(log,log_destinations[i].dest_name);
+    }
+    else if(usereg){
+      NR_registry reg_key;
+      int i;
+      int value;
+      char c;
+
+      /* Get the data out of the registry */
+      for(i=0; i<LOG_NUM_DESTINATIONS; i++){
+        /* set callback for default level */
+        if((size_t)snprintf(reg_key,sizeof(reg_key),"%s.%s.level",LOGGING_REG_PREFIX,
+          log_destinations[i].dest_name)>=sizeof(reg_key))
+          ABORT(R_INTERNAL);
+
+        NR_reg_register_callback(reg_key,
+          NR_REG_CB_ACTION_ADD|NR_REG_CB_ACTION_CHANGE|NR_REG_CB_ACTION_DELETE,
+          r_log_default_level_change_cb,0);
+
+        if(r=r_log_get_reg_level(reg_key,&value)){
+          if(r==R_NOT_FOUND)
+            log_destinations[i].default_level=LOG_LEVEL_UNDEFINED;
+          else
+            ABORT(R_INTERNAL);
+        }
+        else
+          log_destinations[i].default_level=value;
+
+        /* set callback for the enabled key for this logging dest */
+        if((size_t)snprintf(reg_key,sizeof(reg_key),"%s.%s.enabled",LOGGING_REG_PREFIX,
+          log_destinations[i].dest_name)>=sizeof(reg_key))
+          ABORT(R_INTERNAL);
+
+        NR_reg_register_callback(reg_key,
+          NR_REG_CB_ACTION_ADD|NR_REG_CB_ACTION_CHANGE|NR_REG_CB_ACTION_DELETE,
+          r_log_destination_change_cb,0);
+
+        if(r=NR_reg_get_char(reg_key,&c)){
+          if(r==R_NOT_FOUND)
+            log_destinations[i].enabled=0;
+          else
+            ABORT(r);
+        }
+        else
+          log_destinations[i].enabled=c;
+      }
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+static void r_log_destination_change_cb(void *cb_arg, char action, NR_registry name)
+  {
+    r_log_get_destinations(1);
+  }
+
+static void r_log_default_level_change_cb(void *cb_arg, char action, NR_registry name)
+  {
+    r_log_get_destinations(1);
+  }
+
+
+int r_log_init()
+  {
+    _r_log_init(1);
+
+    return 0;
+  }
+
+int _r_log_init(int use_reg)
+  {
+#ifndef WIN32
+    char *log;
+#endif
+
+    if(r_log_initted==0) {
+#ifdef WIN32
+      r_log_env_verbose=1;
+#else
+      log=getenv("R_LOG_VERBOSE");
+      if(log)
+        r_log_env_verbose=atoi(log);
+#endif
+
+    }
+
+    if(!use_reg){
+      if(r_log_initted<R_LOG_INITTED1){
+        r_log_get_default_level();
+        r_log_get_destinations(0);
+
+        r_log_initted=R_LOG_INITTED1;
+      }
+    }
+    else{
+      if(r_log_initted<R_LOG_INITTED2){
+        int facility;
+
+        r_log_get_default_level();
+        r_log_get_destinations(1);
+
+        r_log_register("generic",&facility);
+        r_log_register("logging",&NR_LOG_LOGGING);
+
+        r_log_initted=R_LOG_INITTED2;
+      }
+    }
+
+    return(0);
+  }
+
+int r_log_set_extra_destination(int default_level, r_dest_vlog *dest_vlog)
+  {
+    int i;
+    log_destination *dest = 0;
+
+    for(i=0; i<LOG_NUM_DESTINATIONS; i++){
+      if(!strcmp("extra",log_destinations[i].dest_name)){
+        dest=&log_destinations[i];
+        break;
+      }
+    }
+
+    if(!dest)
+      return(R_INTERNAL);
+
+    if (dest_vlog==0){
+      dest->enabled=0;
+      dest->dest_vlog=noop_vlog;
+    }
+    else{
+      dest->enabled=1;
+      dest->default_level=default_level;
+      dest->dest_vlog=dest_vlog;
+    }
+
+    return(0);
+  }
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/log/r_log.h b/dom/media/webrtc/transport/third_party/nrappkit/src/log/r_log.h
new file mode 100644
index 0000000000..a72dfa0667
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/log/r_log.h
@@ -0,0 +1,85 @@
+/**
+   r_log.h
+
+
+   Copyright (C) 2001, RTFM, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   ekr@rtfm.com  Mon Dec  3 15:14:45 2001
+ */
+
+
+#ifndef _r_log_h
+#define _r_log_h
+
+#ifndef WIN32
+#include <syslog.h>
+#endif
+#include <stdarg.h>
+#include <r_common.h>
+
+int r_log(int facility,int level,const char *fmt,...)
+#ifdef __GNUC__
+  __attribute__ ((format (printf, 3, 4)))
+#endif
+;
+
+int r_vlog(int facility,int level,const char *fmt,va_list ap);
+int r_dump(int facility,int level,char *name,char *data,int len);
+
+int r_log_e(int facility,int level,const char *fmt,...)
+#ifdef __GNUC__
+  __attribute__ ((format (printf, 3, 4)))
+#endif
+;
+
+int r_vlog_e(int facility,int level,const char *fmt,va_list ap);
+int r_log_nr(int facility,int level,int r,const char *fmt,...)
+#ifdef __GNUC__
+  __attribute__ ((format (printf, 4, 5)))
+#endif
+;
+
+int r_vlog_nr(int facility,int level,int r,const char *fmt,va_list ap);
+
+int r_log_register(char *tipename,int *facility);
+int r_log_facility(int facility,char **tipename);
+int r_logging(int facility, int level);
+int r_log_init(void);
+
+#define LOG_GENERIC 0
+#define LOG_COMMON 0
+
+typedef int r_dest_vlog(int facility,int level,const char *format,va_list ap);
+int r_log_set_extra_destination(int default_level, r_dest_vlog *dest_vlog);
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/plugin/nr_plugin.h b/dom/media/webrtc/transport/third_party/nrappkit/src/plugin/nr_plugin.h
new file mode 100644
index 0000000000..56da0624a6
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/plugin/nr_plugin.h
@@ -0,0 +1,57 @@
+/**
+   nr_plugin.h
+
+
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   ekr@networkresonance.com  Mon Jun 19 18:18:54 2006
+ */
+
+
+#ifndef _nr_plugin_h
+#define _nr_plugin_h
+
+typedef int (NR_plugin_hook)(void);
+
+typedef struct NR_plugin_hook_def_ {
+     char *type;
+     NR_plugin_hook *func;
+} NR_plugin_hook_def;
+
+typedef struct NR_plugin_def_ {
+     int api_version; // Should be 1
+     char *name;
+     char *version;
+     NR_plugin_hook_def *hooks;
+} NR_plugin_def;
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/port/android/include/android_funcs.h b/dom/media/webrtc/transport/third_party/nrappkit/src/port/android/include/android_funcs.h
new file mode 100644
index 0000000000..d48bbe1373
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/port/android/include/android_funcs.h
@@ -0,0 +1,62 @@
+/**
+   linux_funcs.h
+
+
+   Copyright (C) 2004, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   ekr@rtfm.com  Mon Dec 13 16:28:22 2004
+ */
+
+
+#ifndef _android_funcs_h
+#define _android_funcs_h
+
+#include <sys/types.h>
+#include <netinet/in.h>
+#include <arpa/inet.h>
+#include <string.h>
+#include <unistd.h>
+
+#define ETHERTYPE_VLAN 0x8100
+
+#define STDIO_BYTES_BUFFERED(fp) (fp->_IO_read_end - fp->_IO_read_ptr)
+
+size_t strlcat(char *dst, const char *src, size_t siz);
+#ifndef strlcpy
+#define strlcpy(a,b,c) \
+	(strncpy((a),(b),(c)), \
+	 ((c)<= 0 ? 0 : ((a)[(c)-1]='\0')), \
+	 strlen((b)))
+#endif
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/port/android/include/csi_platform.h b/dom/media/webrtc/transport/third_party/nrappkit/src/port/android/include/csi_platform.h
new file mode 100644
index 0000000000..bcfb2bce77
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/port/android/include/csi_platform.h
@@ -0,0 +1,55 @@
+/**
+   platform.h
+
+
+   Copyright (C) 2004, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   ekr@rtfm.com  Mon Dec 13 17:26:51 2004
+ */
+
+
+#ifndef _platform_h
+#define _platform_h
+
+#include <android_funcs.h>
+
+#ifdef NR_SOCKET_IS_VOID_PTR
+typedef void* NR_SOCKET;
+#else
+typedef int NR_SOCKET;
+#define NR_SOCKET_READ(sock,buf,count)   read((sock),(buf),(count))
+#define NR_SOCKET_WRITE(sock,buf,count)  write((sock),(buf),(count))
+#define NR_SOCKET_CLOSE(sock)            close(sock)
+#endif
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/port/android/include/sys/ttycom.h b/dom/media/webrtc/transport/third_party/nrappkit/src/port/android/include/sys/ttycom.h
new file mode 100644
index 0000000000..852bf9103b
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/port/android/include/sys/ttycom.h
@@ -0,0 +1,38 @@
+/*
+ *
+ *    Copyright (C) 2006, Network Resonance, Inc.
+ *    All Rights Reserved
+ *
+ *    Redistribution and use in source and binary forms, with or without
+ *    modification, are permitted provided that the following conditions
+ *    are met:
+ *
+ *    1. Redistributions of source code must retain the above copyright
+ *       notice, this list of conditions and the following disclaimer.
+ *    2. Redistributions in binary form must reproduce the above copyright
+ *       notice, this list of conditions and the following disclaimer in the
+ *       documentation and/or other materials provided with the distribution.
+ *    3. Neither the name of Network Resonance, Inc. nor the name of any
+ *       contributors to this software may be used to endorse or promote
+ *       products derived from this software without specific prior written
+ *       permission.
+ *
+ *    THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+ *    AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ *    IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ *    ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+ *    LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ *    CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ *    SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+ *    INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+ *    CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ *    ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+ *    POSSIBILITY OF SUCH DAMAGE.
+ *
+ */
+#ifndef __ANDROID_TTYCOM_H
+#define __ANDROID_TTYCOM_H
+
+#include <asm/ioctls.h>
+
+#endif
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/port/android/port-impl.mk b/dom/media/webrtc/transport/third_party/nrappkit/src/port/android/port-impl.mk
new file mode 100644
index 0000000000..6704cfedf9
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/port/android/port-impl.mk
@@ -0,0 +1,31 @@
+#    
+#    Copyright (C) 2006, Network Resonance, Inc.
+#    All Rights Reserved
+#    
+#    Redistribution and use in source and binary forms, with or without
+#    modification, are permitted provided that the following conditions
+#    are met:
+#    
+#    1. Redistributions of source code must retain the above copyright
+#       notice, this list of conditions and the following disclaimer.
+#    2. Redistributions in binary form must reproduce the above copyright
+#       notice, this list of conditions and the following disclaimer in the
+#       documentation and/or other materials provided with the distribution.
+#    3. Neither the name of Network Resonance, Inc. nor the name of any
+#       contributors to this software may be used to endorse or promote 
+#       products derived from this software without specific prior written
+#       permission.
+#    
+#    THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+#    AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+#    IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+#    ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE 
+#    LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR 
+#    CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF 
+#    SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+#    INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN 
+#    CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+#    ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+#    POSSIBILITY OF SUCH DAMAGE.
+#    
+#EXTATTR_IMPL=xattr
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/port/darwin/include/csi_platform.h b/dom/media/webrtc/transport/third_party/nrappkit/src/port/darwin/include/csi_platform.h
new file mode 100644
index 0000000000..41f1a5b097
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/port/darwin/include/csi_platform.h
@@ -0,0 +1,57 @@
+/**
+   platform.h
+
+
+   Copyright (C) 2005, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   briank@network-resonance.com  Tue Mar 15 14:28:12 PST 2005
+ */
+
+
+#ifndef _platform_h
+#define _platform_h
+
+#include <unistd.h>
+
+#define STDIO_BYTES_BUFFERED(fp) (fp->_r)
+
+#ifdef NR_SOCKET_IS_VOID_PTR
+typedef void* NR_SOCKET;
+#else
+typedef int NR_SOCKET;
+#define NR_SOCKET_READ(sock,buf,count)   read((sock),(buf),(count))
+#define NR_SOCKET_WRITE(sock,buf,count)  write((sock),(buf),(count))
+#define NR_SOCKET_CLOSE(sock)            close(sock)
+#endif
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/port/generic/include/sys/queue.h b/dom/media/webrtc/transport/third_party/nrappkit/src/port/generic/include/sys/queue.h
new file mode 100644
index 0000000000..4d7b998a34
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/port/generic/include/sys/queue.h
@@ -0,0 +1,562 @@
+/*
+ * Copyright (c) 1991, 1993
+ *	The Regents of the University of California.  All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ *    notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ *    notice, this list of conditions and the following disclaimer in the
+ *    documentation and/or other materials provided with the distribution.
+ * 4. Neither the name of the University nor the names of its contributors
+ *    may be used to endorse or promote products derived from this software
+ *    without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND
+ * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ * ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+ * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+ * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ *
+ *	@(#)queue.h	8.5 (Berkeley) 8/20/94
+ * $FreeBSD: src/sys/sys/queue.h,v 1.58 2004/04/07 04:19:49 imp Exp $
+ */
+
+#ifndef _SYS_QUEUE_H_
+#define	_SYS_QUEUE_H_
+
+#include <stddef.h>
+
+#ifndef offsetof
+#define offsetof(type, field) ((size_t)(&((type *)0)->field))
+#endif
+
+#define STAILQ_FOREACH_SAFE(var, head, field, tvar)                     \
+         for ((var) = STAILQ_FIRST((head));                              \
+             (var) && ((tvar) = STAILQ_NEXT((var), field), 1);           \
+             (var) = (tvar))
+
+/*
+ * This file defines four types of data structures: singly-linked lists,
+ * singly-linked tail queues, lists and tail queues.
+ *
+ * A singly-linked list is headed by a single forward pointer. The elements
+ * are singly linked for minimum space and pointer manipulation overhead at
+ * the expense of O(n) removal for arbitrary elements. New elements can be
+ * added to the list after an existing element or at the head of the list.
+ * Elements being removed from the head of the list should use the explicit
+ * macro for this purpose for optimum efficiency. A singly-linked list may
+ * only be traversed in the forward direction.  Singly-linked lists are ideal
+ * for applications with large datasets and few or no removals or for
+ * implementing a LIFO queue.
+ *
+ * A singly-linked tail queue is headed by a pair of pointers, one to the
+ * head of the list and the other to the tail of the list. The elements are
+ * singly linked for minimum space and pointer manipulation overhead at the
+ * expense of O(n) removal for arbitrary elements. New elements can be added
+ * to the list after an existing element, at the head of the list, or at the
+ * end of the list. Elements being removed from the head of the tail queue
+ * should use the explicit macro for this purpose for optimum efficiency.
+ * A singly-linked tail queue may only be traversed in the forward direction.
+ * Singly-linked tail queues are ideal for applications with large datasets
+ * and few or no removals or for implementing a FIFO queue.
+ *
+ * A list is headed by a single forward pointer (or an array of forward
+ * pointers for a hash table header). The elements are doubly linked
+ * so that an arbitrary element can be removed without a need to
+ * traverse the list. New elements can be added to the list before
+ * or after an existing element or at the head of the list. A list
+ * may only be traversed in the forward direction.
+ *
+ * A tail queue is headed by a pair of pointers, one to the head of the
+ * list and the other to the tail of the list. The elements are doubly
+ * linked so that an arbitrary element can be removed without a need to
+ * traverse the list. New elements can be added to the list before or
+ * after an existing element, at the head of the list, or at the end of
+ * the list. A tail queue may be traversed in either direction.
+ *
+ * For details on the use of these macros, see the queue(3) manual page.
+ *
+ *
+ *				SLIST	LIST	STAILQ	TAILQ
+ * _HEAD			+	+	+	+
+ * _HEAD_INITIALIZER		+	+	+	+
+ * _ENTRY			+	+	+	+
+ * _INIT			+	+	+	+
+ * _EMPTY			+	+	+	+
+ * _FIRST			+	+	+	+
+ * _NEXT			+	+	+	+
+ * _PREV			-	-	-	+
+ * _LAST			-	-	+	+
+ * _FOREACH			+	+	+	+
+ * _FOREACH_SAFE		+	+	+	+
+ * _FOREACH_REVERSE		-	-	-	+
+ * _FOREACH_REVERSE_SAFE	-	-	-	+
+ * _INSERT_HEAD			+	+	+	+
+ * _INSERT_BEFORE		-	+	-	+
+ * _INSERT_AFTER		+	+	+	+
+ * _INSERT_TAIL			-	-	+	+
+ * _CONCAT			-	-	+	+
+ * _REMOVE_HEAD			+	-	+	-
+ * _REMOVE			+	+	+	+
+ *
+ */
+#define	QUEUE_MACRO_DEBUG 0
+#if QUEUE_MACRO_DEBUG
+/* Store the last 2 places the queue element or head was altered */
+struct qm_trace {
+	char * lastfile;
+	int lastline;
+	char * prevfile;
+	int prevline;
+};
+
+#define	TRACEBUF	struct qm_trace trace;
+#define	TRASHIT(x)	do {(x) = (void *)-1;} while (0)
+
+#define	QMD_TRACE_HEAD(head) do {					\
+	(head)->trace.prevline = (head)->trace.lastline;		\
+	(head)->trace.prevfile = (head)->trace.lastfile;		\
+	(head)->trace.lastline = __LINE__;				\
+	(head)->trace.lastfile = __FILE__;				\
+} while (0)
+
+#define	QMD_TRACE_ELEM(elem) do {					\
+	(elem)->trace.prevline = (elem)->trace.lastline;		\
+	(elem)->trace.prevfile = (elem)->trace.lastfile;		\
+	(elem)->trace.lastline = __LINE__;				\
+	(elem)->trace.lastfile = __FILE__;				\
+} while (0)
+
+#else
+#define	QMD_TRACE_ELEM(elem)
+#define	QMD_TRACE_HEAD(head)
+#define	TRACEBUF
+#define	TRASHIT(x)
+#endif	/* QUEUE_MACRO_DEBUG */
+
+/*
+ * Singly-linked List declarations.
+ */
+#define	SLIST_HEAD(name, type)						\
+struct name {								\
+	struct type *slh_first;	/* first element */			\
+}
+
+#define	SLIST_HEAD_INITIALIZER(head)					\
+	{ NULL }
+
+#define	SLIST_ENTRY(type)						\
+struct {								\
+	struct type *sle_next;	/* next element */			\
+}
+
+/*
+ * Singly-linked List functions.
+ */
+#define	SLIST_EMPTY(head)	((head)->slh_first == NULL)
+
+#define	SLIST_FIRST(head)	((head)->slh_first)
+
+#define	SLIST_FOREACH(var, head, field)					\
+	for ((var) = SLIST_FIRST((head));				\
+	    (var);							\
+	    (var) = SLIST_NEXT((var), field))
+
+#define	SLIST_FOREACH_SAFE(var, head, field, tvar)			\
+	for ((var) = SLIST_FIRST((head));				\
+	    (var) && ((tvar) = SLIST_NEXT((var), field), 1);		\
+	    (var) = (tvar))
+
+#define	SLIST_FOREACH_PREVPTR(var, varp, head, field)			\
+	for ((varp) = &SLIST_FIRST((head));				\
+	    ((var) = *(varp)) != NULL;					\
+	    (varp) = &SLIST_NEXT((var), field))
+
+#define	SLIST_INIT(head) do {						\
+	SLIST_FIRST((head)) = NULL;					\
+} while (0)
+
+#define	SLIST_INSERT_AFTER(slistelm, elm, field) do {			\
+	SLIST_NEXT((elm), field) = SLIST_NEXT((slistelm), field);	\
+	SLIST_NEXT((slistelm), field) = (elm);				\
+} while (0)
+
+#define	SLIST_INSERT_HEAD(head, elm, field) do {			\
+	SLIST_NEXT((elm), field) = SLIST_FIRST((head));			\
+	SLIST_FIRST((head)) = (elm);					\
+} while (0)
+
+#define	SLIST_NEXT(elm, field)	((elm)->field.sle_next)
+
+#define	SLIST_REMOVE(head, elm, type, field) do {			\
+	if (SLIST_FIRST((head)) == (elm)) {				\
+		SLIST_REMOVE_HEAD((head), field);			\
+	}								\
+	else {								\
+		struct type *curelm = SLIST_FIRST((head));		\
+		while (SLIST_NEXT(curelm, field) != (elm))		\
+			curelm = SLIST_NEXT(curelm, field);		\
+		SLIST_NEXT(curelm, field) =				\
+		    SLIST_NEXT(SLIST_NEXT(curelm, field), field);	\
+	}								\
+} while (0)
+
+#define	SLIST_REMOVE_HEAD(head, field) do {				\
+	SLIST_FIRST((head)) = SLIST_NEXT(SLIST_FIRST((head)), field);	\
+} while (0)
+
+/*
+ * Singly-linked Tail queue declarations.
+ */
+#define	STAILQ_HEAD(name, type)						\
+struct name {								\
+	struct type *stqh_first;/* first element */			\
+	struct type **stqh_last;/* addr of last next element */		\
+}
+
+#define	STAILQ_HEAD_INITIALIZER(head)					\
+	{ NULL, &(head).stqh_first }
+
+#define	STAILQ_ENTRY(type)						\
+struct {								\
+	struct type *stqe_next;	/* next element */			\
+}
+
+/*
+ * Singly-linked Tail queue functions.
+ */
+#define	STAILQ_CONCAT(head1, head2) do {				\
+	if (!STAILQ_EMPTY((head2))) {					\
+		*(head1)->stqh_last = (head2)->stqh_first;		\
+		(head1)->stqh_last = (head2)->stqh_last;		\
+		STAILQ_INIT((head2));					\
+	}								\
+} while (0)
+
+#define	STAILQ_EMPTY(head)	((head)->stqh_first == NULL)
+
+#define	STAILQ_FIRST(head)	((head)->stqh_first)
+
+#define	STAILQ_FOREACH(var, head, field)				\
+	for((var) = STAILQ_FIRST((head));				\
+	   (var);							\
+	   (var) = STAILQ_NEXT((var), field))
+
+
+#define	STAILQ_FOREACH_SAFE(var, head, field, tvar)			\
+	for ((var) = STAILQ_FIRST((head));				\
+	    (var) && ((tvar) = STAILQ_NEXT((var), field), 1);		\
+	    (var) = (tvar))
+
+#define	STAILQ_INIT(head) do {						\
+	STAILQ_FIRST((head)) = NULL;					\
+	(head)->stqh_last = &STAILQ_FIRST((head));			\
+} while (0)
+
+#define	STAILQ_INSERT_AFTER(head, tqelm, elm, field) do {		\
+	if ((STAILQ_NEXT((elm), field) = STAILQ_NEXT((tqelm), field)) == NULL)\
+		(head)->stqh_last = &STAILQ_NEXT((elm), field);		\
+	STAILQ_NEXT((tqelm), field) = (elm);				\
+} while (0)
+
+#define	STAILQ_INSERT_HEAD(head, elm, field) do {			\
+	if ((STAILQ_NEXT((elm), field) = STAILQ_FIRST((head))) == NULL)	\
+		(head)->stqh_last = &STAILQ_NEXT((elm), field);		\
+	STAILQ_FIRST((head)) = (elm);					\
+} while (0)
+
+#define	STAILQ_INSERT_TAIL(head, elm, field) do {			\
+	STAILQ_NEXT((elm), field) = NULL;				\
+	*(head)->stqh_last = (elm);					\
+	(head)->stqh_last = &STAILQ_NEXT((elm), field);			\
+} while (0)
+
+#define	STAILQ_LAST(head, type, field)					\
+	(STAILQ_EMPTY((head)) ?						\
+		NULL :							\
+	        ((struct type *)					\
+		((char *)((head)->stqh_last) - offsetof(struct type, field))))
+
+#define	STAILQ_NEXT(elm, field)	((elm)->field.stqe_next)
+
+#define	STAILQ_REMOVE(head, elm, type, field) do {			\
+	if (STAILQ_FIRST((head)) == (elm)) {				\
+		STAILQ_REMOVE_HEAD((head), field);			\
+	}								\
+	else {								\
+		struct type *curelm = STAILQ_FIRST((head));		\
+		while (STAILQ_NEXT(curelm, field) != (elm))		\
+			curelm = STAILQ_NEXT(curelm, field);		\
+		if ((STAILQ_NEXT(curelm, field) =			\
+		     STAILQ_NEXT(STAILQ_NEXT(curelm, field), field)) == NULL)\
+			(head)->stqh_last = &STAILQ_NEXT((curelm), field);\
+	}								\
+} while (0)
+
+#define	STAILQ_REMOVE_HEAD(head, field) do {				\
+	if ((STAILQ_FIRST((head)) =					\
+	     STAILQ_NEXT(STAILQ_FIRST((head)), field)) == NULL)		\
+		(head)->stqh_last = &STAILQ_FIRST((head));		\
+} while (0)
+
+#define	STAILQ_REMOVE_HEAD_UNTIL(head, elm, field) do {			\
+	if ((STAILQ_FIRST((head)) = STAILQ_NEXT((elm), field)) == NULL)	\
+		(head)->stqh_last = &STAILQ_FIRST((head));		\
+} while (0)
+
+/*
+ * List declarations.
+ */
+#define	LIST_HEAD(name, type)						\
+struct name {								\
+	struct type *lh_first;	/* first element */			\
+}
+
+#define	LIST_HEAD_INITIALIZER(head)					\
+	{ NULL }
+
+#define	LIST_ENTRY(type)						\
+struct {								\
+	struct type *le_next;	/* next element */			\
+	struct type **le_prev;	/* address of previous next element */	\
+}
+
+/*
+ * List functions.
+ */
+
+#define	LIST_EMPTY(head)	((head)->lh_first == NULL)
+
+#define	LIST_FIRST(head)	((head)->lh_first)
+
+#define	LIST_FOREACH(var, head, field)					\
+	for ((var) = LIST_FIRST((head));				\
+	    (var);							\
+	    (var) = LIST_NEXT((var), field))
+
+#define	LIST_FOREACH_SAFE(var, head, field, tvar)			\
+	for ((var) = LIST_FIRST((head));				\
+	    (var) && ((tvar) = LIST_NEXT((var), field), 1);		\
+	    (var) = (tvar))
+
+#define	LIST_INIT(head) do {						\
+	LIST_FIRST((head)) = NULL;					\
+} while (0)
+
+#define	LIST_INSERT_AFTER(listelm, elm, field) do {			\
+	if ((LIST_NEXT((elm), field) = LIST_NEXT((listelm), field)) != NULL)\
+		LIST_NEXT((listelm), field)->field.le_prev =		\
+		    &LIST_NEXT((elm), field);				\
+	LIST_NEXT((listelm), field) = (elm);				\
+	(elm)->field.le_prev = &LIST_NEXT((listelm), field);		\
+} while (0)
+
+#define	LIST_INSERT_BEFORE(listelm, elm, field) do {			\
+	(elm)->field.le_prev = (listelm)->field.le_prev;		\
+	LIST_NEXT((elm), field) = (listelm);				\
+	*(listelm)->field.le_prev = (elm);				\
+	(listelm)->field.le_prev = &LIST_NEXT((elm), field);		\
+} while (0)
+
+#define	LIST_INSERT_HEAD(head, elm, field) do {				\
+	if ((LIST_NEXT((elm), field) = LIST_FIRST((head))) != NULL)	\
+		LIST_FIRST((head))->field.le_prev = &LIST_NEXT((elm), field);\
+	LIST_FIRST((head)) = (elm);					\
+	(elm)->field.le_prev = &LIST_FIRST((head));			\
+} while (0)
+
+#define	LIST_NEXT(elm, field)	((elm)->field.le_next)
+
+#define	LIST_REMOVE(elm, field) do {					\
+	if (LIST_NEXT((elm), field) != NULL)				\
+		LIST_NEXT((elm), field)->field.le_prev = 		\
+		    (elm)->field.le_prev;				\
+	*(elm)->field.le_prev = LIST_NEXT((elm), field);		\
+} while (0)
+
+/*
+ * Tail queue declarations.
+ */
+#define	TAILQ_HEAD(name, type)						\
+struct name {								\
+	struct type *tqh_first;	/* first element */			\
+	struct type **tqh_last;	/* addr of last next element */		\
+	TRACEBUF							\
+}
+
+#define	TAILQ_HEAD_INITIALIZER(head)					\
+	{ NULL, &(head).tqh_first }
+
+#define	TAILQ_ENTRY(type)						\
+struct {								\
+	struct type *tqe_next;	/* next element */			\
+	struct type **tqe_prev;	/* address of previous next element */	\
+	TRACEBUF							\
+}
+
+/*
+ * Tail queue functions.
+ */
+#define	TAILQ_CONCAT(head1, head2, field) do {				\
+	if (!TAILQ_EMPTY(head2)) {					\
+		*(head1)->tqh_last = (head2)->tqh_first;		\
+		(head2)->tqh_first->field.tqe_prev = (head1)->tqh_last;	\
+		(head1)->tqh_last = (head2)->tqh_last;			\
+		TAILQ_INIT((head2));					\
+		QMD_TRACE_HEAD(head);					\
+		QMD_TRACE_HEAD(head2);					\
+	}								\
+} while (0)
+
+#define	TAILQ_EMPTY(head)	((head)->tqh_first == NULL)
+
+#define	TAILQ_FIRST(head)	((head)->tqh_first)
+
+#define	TAILQ_FOREACH(var, head, field)					\
+	for ((var) = TAILQ_FIRST((head));				\
+	    (var);							\
+	    (var) = TAILQ_NEXT((var), field))
+
+#define	TAILQ_FOREACH_SAFE(var, head, field, tvar)			\
+	for ((var) = TAILQ_FIRST((head));				\
+	    (var) && ((tvar) = TAILQ_NEXT((var), field), 1);		\
+	    (var) = (tvar))
+
+#define	TAILQ_FOREACH_REVERSE(var, head, headname, field)		\
+	for ((var) = TAILQ_LAST((head), headname);			\
+	    (var);							\
+	    (var) = TAILQ_PREV((var), headname, field))
+
+#define	TAILQ_FOREACH_REVERSE_SAFE(var, head, headname, field, tvar)	\
+	for ((var) = TAILQ_LAST((head), headname);			\
+	    (var) && ((tvar) = TAILQ_PREV((var), headname, field), 1);	\
+	    (var) = (tvar))
+
+#define	TAILQ_INIT(head) do {						\
+	TAILQ_FIRST((head)) = NULL;					\
+	(head)->tqh_last = &TAILQ_FIRST((head));			\
+	QMD_TRACE_HEAD(head);						\
+} while (0)
+
+#define	TAILQ_INSERT_AFTER(head, listelm, elm, field) do {		\
+	if ((TAILQ_NEXT((elm), field) = TAILQ_NEXT((listelm), field)) != NULL)\
+		TAILQ_NEXT((elm), field)->field.tqe_prev = 		\
+		    &TAILQ_NEXT((elm), field);				\
+	else {								\
+		(head)->tqh_last = &TAILQ_NEXT((elm), field);		\
+		QMD_TRACE_HEAD(head);					\
+	}								\
+	TAILQ_NEXT((listelm), field) = (elm);				\
+	(elm)->field.tqe_prev = &TAILQ_NEXT((listelm), field);		\
+	QMD_TRACE_ELEM(&(elm)->field);					\
+	QMD_TRACE_ELEM(&listelm->field);				\
+} while (0)
+
+#define	TAILQ_INSERT_BEFORE(listelm, elm, field) do {			\
+	(elm)->field.tqe_prev = (listelm)->field.tqe_prev;		\
+	TAILQ_NEXT((elm), field) = (listelm);				\
+	*(listelm)->field.tqe_prev = (elm);				\
+	(listelm)->field.tqe_prev = &TAILQ_NEXT((elm), field);		\
+	QMD_TRACE_ELEM(&(elm)->field);					\
+	QMD_TRACE_ELEM(&listelm->field);				\
+} while (0)
+
+#define	TAILQ_INSERT_HEAD(head, elm, field) do {			\
+	if ((TAILQ_NEXT((elm), field) = TAILQ_FIRST((head))) != NULL)	\
+		TAILQ_FIRST((head))->field.tqe_prev =			\
+		    &TAILQ_NEXT((elm), field);				\
+	else								\
+		(head)->tqh_last = &TAILQ_NEXT((elm), field);		\
+	TAILQ_FIRST((head)) = (elm);					\
+	(elm)->field.tqe_prev = &TAILQ_FIRST((head));			\
+	QMD_TRACE_HEAD(head);						\
+	QMD_TRACE_ELEM(&(elm)->field);					\
+} while (0)
+
+#define	TAILQ_INSERT_TAIL(head, elm, field) do {			\
+	TAILQ_NEXT((elm), field) = NULL;				\
+	(elm)->field.tqe_prev = (head)->tqh_last;			\
+	*(head)->tqh_last = (elm);					\
+	(head)->tqh_last = &TAILQ_NEXT((elm), field);			\
+	QMD_TRACE_HEAD(head);						\
+	QMD_TRACE_ELEM(&(elm)->field);					\
+} while (0)
+
+#define	TAILQ_LAST(head, headname)					\
+	(*(((struct headname *)((head)->tqh_last))->tqh_last))
+
+#define	TAILQ_NEXT(elm, field) ((elm)->field.tqe_next)
+
+#define	TAILQ_PREV(elm, headname, field)				\
+	(*(((struct headname *)((elm)->field.tqe_prev))->tqh_last))
+
+#define	TAILQ_REMOVE(head, elm, field) do {				\
+	if ((TAILQ_NEXT((elm), field)) != NULL)				\
+		TAILQ_NEXT((elm), field)->field.tqe_prev = 		\
+		    (elm)->field.tqe_prev;				\
+	else {								\
+		(head)->tqh_last = (elm)->field.tqe_prev;		\
+		QMD_TRACE_HEAD(head);					\
+	}								\
+	*(elm)->field.tqe_prev = TAILQ_NEXT((elm), field);		\
+	TRASHIT((elm)->field.tqe_next);					\
+	TRASHIT((elm)->field.tqe_prev);					\
+	QMD_TRACE_ELEM(&(elm)->field);					\
+} while (0)
+
+
+#ifdef _KERNEL
+
+/*
+ * XXX insque() and remque() are an old way of handling certain queues.
+ * They bogusly assumes that all queue heads look alike.
+ */
+
+struct quehead {
+	struct quehead *qh_link;
+	struct quehead *qh_rlink;
+};
+
+#if defined(__GNUC__) || defined(__INTEL_COMPILER)
+
+static __inline void
+insque(void *a, void *b)
+{
+	struct quehead *element = (struct quehead *)a,
+		 *head = (struct quehead *)b;
+
+	element->qh_link = head->qh_link;
+	element->qh_rlink = head;
+	head->qh_link = element;
+	element->qh_link->qh_rlink = element;
+}
+
+static __inline void
+remque(void *a)
+{
+	struct quehead *element = (struct quehead *)a;
+
+	element->qh_link->qh_rlink = element->qh_rlink;
+	element->qh_rlink->qh_link = element->qh_link;
+	element->qh_rlink = 0;
+}
+
+#else /* !(__GNUC__ || __INTEL_COMPILER) */
+
+void	insque(void *a, void *b);
+void	remque(void *a);
+
+#endif /* __GNUC__ || __INTEL_COMPILER */
+
+#endif /* _KERNEL */
+
+#endif /* !_SYS_QUEUE_H_ */
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/port/linux/include/csi_platform.h b/dom/media/webrtc/transport/third_party/nrappkit/src/port/linux/include/csi_platform.h
new file mode 100644
index 0000000000..8aefaf9249
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/port/linux/include/csi_platform.h
@@ -0,0 +1,55 @@
+/**
+   platform.h
+
+
+   Copyright (C) 2004, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   ekr@rtfm.com  Mon Dec 13 17:26:51 2004
+ */
+
+
+#ifndef _platform_h
+#define _platform_h
+
+#include <linux_funcs.h>
+
+#ifdef NR_SOCKET_IS_VOID_PTR
+typedef void* NR_SOCKET;
+#else
+typedef int NR_SOCKET;
+#define NR_SOCKET_READ(sock,buf,count)   read((sock),(buf),(count))
+#define NR_SOCKET_WRITE(sock,buf,count)  write((sock),(buf),(count))
+#define NR_SOCKET_CLOSE(sock)            close(sock)
+#endif
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/port/linux/include/linux_funcs.h b/dom/media/webrtc/transport/third_party/nrappkit/src/port/linux/include/linux_funcs.h
new file mode 100644
index 0000000000..1619136306
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/port/linux/include/linux_funcs.h
@@ -0,0 +1,62 @@
+/**
+   linux_funcs.h
+
+
+   Copyright (C) 2004, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   ekr@rtfm.com  Mon Dec 13 16:28:22 2004
+ */
+
+
+#ifndef _linux_funcs_h
+#define _linux_funcs_h
+
+#include <sys/types.h>
+#include <netinet/in.h>
+#include <arpa/inet.h>
+#include <string.h>
+#include <unistd.h>
+
+#define ETHERTYPE_VLAN 0x8100
+
+#define STDIO_BYTES_BUFFERED(fp) (fp->_IO_read_end - fp->_IO_read_ptr)
+
+size_t strlcat(char *dst, const char *src, size_t siz);
+#ifndef strlcpy
+#define strlcpy(a,b,c) \
+	(strncpy((a),(b),(c)), \
+	 ((c)<= 0 ? 0 : ((a)[(c)-1]='\0')), \
+	 strlen((b)))
+#endif
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/port/linux/include/sys/ttycom.h b/dom/media/webrtc/transport/third_party/nrappkit/src/port/linux/include/sys/ttycom.h
new file mode 100644
index 0000000000..6f8a3066d6
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/port/linux/include/sys/ttycom.h
@@ -0,0 +1,38 @@
+/*
+ *
+ *    Copyright (C) 2006, Network Resonance, Inc.
+ *    All Rights Reserved
+ *
+ *    Redistribution and use in source and binary forms, with or without
+ *    modification, are permitted provided that the following conditions
+ *    are met:
+ *
+ *    1. Redistributions of source code must retain the above copyright
+ *       notice, this list of conditions and the following disclaimer.
+ *    2. Redistributions in binary form must reproduce the above copyright
+ *       notice, this list of conditions and the following disclaimer in the
+ *       documentation and/or other materials provided with the distribution.
+ *    3. Neither the name of Network Resonance, Inc. nor the name of any
+ *       contributors to this software may be used to endorse or promote
+ *       products derived from this software without specific prior written
+ *       permission.
+ *
+ *    THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+ *    AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ *    IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ *    ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+ *    LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ *    CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ *    SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+ *    INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+ *    CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ *    ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+ *    POSSIBILITY OF SUCH DAMAGE.
+ *
+ */
+#ifndef __LINUX_TTYCOM_H
+#define __LINUX_TTYCOM_H
+
+#include <asm/ioctls.h>
+
+#endif
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/port/linux/port-impl.mk b/dom/media/webrtc/transport/third_party/nrappkit/src/port/linux/port-impl.mk
new file mode 100644
index 0000000000..6704cfedf9
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/port/linux/port-impl.mk
@@ -0,0 +1,31 @@
+#    
+#    Copyright (C) 2006, Network Resonance, Inc.
+#    All Rights Reserved
+#    
+#    Redistribution and use in source and binary forms, with or without
+#    modification, are permitted provided that the following conditions
+#    are met:
+#    
+#    1. Redistributions of source code must retain the above copyright
+#       notice, this list of conditions and the following disclaimer.
+#    2. Redistributions in binary form must reproduce the above copyright
+#       notice, this list of conditions and the following disclaimer in the
+#       documentation and/or other materials provided with the distribution.
+#    3. Neither the name of Network Resonance, Inc. nor the name of any
+#       contributors to this software may be used to endorse or promote 
+#       products derived from this software without specific prior written
+#       permission.
+#    
+#    THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+#    AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+#    IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+#    ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE 
+#    LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR 
+#    CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF 
+#    SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+#    INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN 
+#    CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+#    ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+#    POSSIBILITY OF SUCH DAMAGE.
+#    
+#EXTATTR_IMPL=xattr
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/port/win32/include/csi_platform.h b/dom/media/webrtc/transport/third_party/nrappkit/src/port/win32/include/csi_platform.h
new file mode 100644
index 0000000000..27c64c53d9
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/port/win32/include/csi_platform.h
@@ -0,0 +1,107 @@
+/**
+   platform.h
+
+
+   Copyright (C) 2005, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   briank@network-resonance.com  Tue Mar 15 14:28:12 PST 2005
+ */
+
+
+#ifndef _csi_platform_h
+#define _csi_platform_h
+
+#ifndef _WIN32_WINNT
+#define _WIN32_WINNT 0x0400  // This prevents weird "'TryEnterCriticalSection': identifier not found"
+                             // compiler errors when poco/win32_mutex.h is included
+#endif
+
+#define UINT8 UBLAH_IGNORE_ME_PLEASE
+#define INT8 BLAH_IGNORE_ME_PLEASE
+#include <winsock2.h>
+#undef UINT8
+#undef INT8
+#include <r_types.h>
+#include <errno.h>
+
+#define strcasecmp _stricmp
+#define strncasecmp _strnicmp
+
+#define strcasestr stristr
+
+/* Hack version of strlcpy (in util/util.c) */
+size_t strlcat(char *dst, const char *src, size_t siz);
+
+/* Hack version of getopt() (in util/getopt.c) */
+int getopt(int argc, char *argv[], char *opstring);
+extern char *optarg;
+extern int optind;
+extern int opterr;
+
+/* Hack version of gettimeofday() (in util/util.c) */
+int gettimeofday(struct timeval *tv, void *tz);
+
+#ifdef NR_SOCKET_IS_VOID_PTR
+typedef void* NR_SOCKET;
+#else
+typedef SOCKET NR_SOCKET;
+#define NR_SOCKET_READ(sock,buf,count)   recv((sock),(buf),(count),0)
+#define NR_SOCKET_WRITE(sock,buf,count)  send((sock),(buf),(count),0)
+#define NR_SOCKET_CLOSE(sock)            closesocket(sock)
+#endif
+
+#ifndef EHOSTUNREACH
+#define EHOSTUNREACH    WSAEHOSTUNREACH
+#endif
+
+#define LOG_EMERG       0
+#define LOG_ALERT       1
+#define LOG_CRIT        2
+#define LOG_ERR         3
+#define LOG_WARNING     4
+#define LOG_NOTICE      5
+#define LOG_INFO        6
+#define LOG_DEBUG       7
+
+// Until we refine the Windows port....
+
+#define in_addr_t   UINT4
+
+#ifndef strlcpy
+#define strlcpy(a,b,c) \
+        (strncpy((a),(b),(c)), \
+         ((c)<= 0 ? 0 : ((a)[(c)-1]='\0')), \
+         strlen((b)))
+#endif
+
+#endif  /* _csi_platform_h */
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/registry/c2ru.c b/dom/media/webrtc/transport/third_party/nrappkit/src/registry/c2ru.c
new file mode 100644
index 0000000000..3e71ca37d9
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/registry/c2ru.c
@@ -0,0 +1,320 @@
+/*
+ *
+ *    c2ru.c
+ *
+ *    $Source: /Users/ekr/tmp/nrappkit-dump/nrappkit/src/registry/c2ru.c,v $
+ *    $Revision: 1.3 $
+ *    $Date: 2007/06/26 22:37:50 $
+ *
+ *    c2r utility methods
+ *
+ *
+ *    Copyright (C) 2005, Network Resonance, Inc.
+ *    Copyright (C) 2006, Network Resonance, Inc.
+ *    All Rights Reserved
+ *
+ *    Redistribution and use in source and binary forms, with or without
+ *    modification, are permitted provided that the following conditions
+ *    are met:
+ *
+ *    1. Redistributions of source code must retain the above copyright
+ *       notice, this list of conditions and the following disclaimer.
+ *    2. Redistributions in binary form must reproduce the above copyright
+ *       notice, this list of conditions and the following disclaimer in the
+ *       documentation and/or other materials provided with the distribution.
+ *    3. Neither the name of Network Resonance, Inc. nor the name of any
+ *       contributors to this software may be used to endorse or promote
+ *       products derived from this software without specific prior written
+ *       permission.
+ *
+ *    THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+ *    AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ *    IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ *    ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+ *    LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ *    CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ *    SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+ *    INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+ *    CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ *    ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+ *    POSSIBILITY OF SUCH DAMAGE.
+ *
+ *
+ */
+
+#include <sys/queue.h>
+#include <string.h>
+#include <registry.h>
+#include "nr_common.h"
+#include <r_errors.h>
+#include <r_macros.h>
+#include <ctype.h>
+#include "c2ru.h"
+
+
+#define NRGET(func, type, get) \
+int                                                                  \
+func(NR_registry parent, char *child, type **out)                    \
+{                                                                    \
+  int r, _status;                                                    \
+  NR_registry registry;                                              \
+  type tmp;                                                          \
+                                                                     \
+  if ((r = nr_c2ru_make_registry(parent, child, registry)))          \
+    ABORT(r);                                                        \
+                                                                     \
+  if ((r = get(registry, &tmp))) {                                   \
+    if (r != R_NOT_FOUND)                                            \
+      ABORT(r);                                                      \
+    *out = 0;                                                        \
+  }                                                                  \
+  else {                                                             \
+    *out = RCALLOC(sizeof(tmp));                                     \
+    if (*out == 0)                                                   \
+      ABORT(R_NO_MEMORY);                                            \
+    **out = tmp;                                                     \
+  }                                                                  \
+                                                                     \
+  _status = 0;                                                       \
+abort:                                                               \
+  return (_status);                                                  \
+}
+
+int
+nr_c2ru_get_char(NR_registry parent, char *child, char **out)
+{
+  int r, _status;
+  NR_registry registry;
+  char tmp;
+
+  if ((r = nr_c2ru_make_registry(parent, child, registry)))
+    ABORT(r);
+
+  if ((r = NR_reg_get_char(registry, &tmp))) {
+    if (r != R_NOT_FOUND)
+      ABORT(r);
+    *out = 0;
+  }
+  else {
+    *out = RCALLOC(sizeof(tmp));
+    if (*out == 0)
+      ABORT(R_NO_MEMORY);
+    **out = tmp;
+  }
+
+  _status = 0;
+abort:
+  return (_status);
+}
+NRGET(nr_c2ru_get_uchar,    UCHAR,   NR_reg_get_uchar)
+NRGET(nr_c2ru_get_int2,     INT2,    NR_reg_get_int2)
+NRGET(nr_c2ru_get_uint2,    UINT2,   NR_reg_get_uint2)
+NRGET(nr_c2ru_get_int4,     INT4,    NR_reg_get_int4)
+NRGET(nr_c2ru_get_uint4,    UINT4,   NR_reg_get_uint4)
+NRGET(nr_c2ru_get_int8,     INT8,    NR_reg_get_int8)
+NRGET(nr_c2ru_get_uint8,    UINT8,   NR_reg_get_uint8)
+NRGET(nr_c2ru_get_double,   double,  NR_reg_get_double)
+NRGET(nr_c2ru_get_string,   char*,   NR_reg_alloc_string)
+NRGET(nr_c2ru_get_data,     Data,    NR_reg_alloc_data)
+
+
+#define NRSET(func, type, set) \
+int                                                                  \
+func(NR_registry parent, char *child, type *in)                      \
+{                                                                    \
+  int r, _status;                                                    \
+  NR_registry registry;                                              \
+                                                                     \
+  if (in == 0)                                                       \
+    return 0;                                                        \
+                                                                     \
+  if ((r = nr_c2ru_make_registry(parent, child, registry)))          \
+    ABORT(r);                                                        \
+                                                                     \
+  if ((r = set(registry, *in)))                                      \
+    ABORT(r);                                                        \
+                                                                     \
+  _status = 0;                                                       \
+abort:                                                               \
+  return (_status);                                                  \
+}
+
+NRSET(nr_c2ru_set_char,      char,       NR_reg_set_char)
+NRSET(nr_c2ru_set_uchar,     UCHAR,      NR_reg_set_uchar)
+NRSET(nr_c2ru_set_int2,      INT2,       NR_reg_set_int2)
+NRSET(nr_c2ru_set_uint2,     UINT2,      NR_reg_set_uint2)
+NRSET(nr_c2ru_set_int4,      INT4,       NR_reg_set_int4)
+NRSET(nr_c2ru_set_uint4,     UINT4,      NR_reg_set_uint4)
+NRSET(nr_c2ru_set_int8,      INT8,       NR_reg_set_int8)
+NRSET(nr_c2ru_set_uint8,     UINT8,      NR_reg_set_uint8)
+NRSET(nr_c2ru_set_double,    double,     NR_reg_set_double)
+NRSET(nr_c2ru_set_string,    char*,      NR_reg_set_string)
+
+int
+nr_c2ru_set_data(NR_registry parent, char *child, Data *in)
+{
+  int r, _status;
+  NR_registry registry;
+
+  if (in == 0)
+    return 0;
+
+  if ((r = nr_c2ru_make_registry(parent, child, registry)))
+    ABORT(r);
+
+  if ((r = NR_reg_set_bytes(registry, in->data, in->len)))
+    ABORT(r);
+
+  _status = 0;
+abort:
+  return (_status);
+}
+
+#define NRFREE(func, type) \
+int                                                                  \
+func(type *in)                                                       \
+{                                                                    \
+  if (in)                                                            \
+    RFREE(in);                                                       \
+  return 0;                                                          \
+}
+
+NRFREE(nr_c2ru_free_char,      char)
+NRFREE(nr_c2ru_free_uchar,     UCHAR)
+NRFREE(nr_c2ru_free_int2,      INT2)
+NRFREE(nr_c2ru_free_uint2,     UINT2)
+NRFREE(nr_c2ru_free_int4,      INT4)
+NRFREE(nr_c2ru_free_uint4,     UINT4)
+NRFREE(nr_c2ru_free_int8,      INT8)
+NRFREE(nr_c2ru_free_uint8,     UINT8)
+NRFREE(nr_c2ru_free_double,    double)
+
+
+int
+nr_c2ru_free_string(char **in)
+{
+  if (*in)
+    RFREE(*in);
+  if (in)
+    RFREE(in);
+  return 0;
+}
+
+int
+nr_c2ru_free_data(Data *in)
+{
+  int r, _status;
+
+  if (in) {
+    if ((r=r_data_destroy(&in)))
+      ABORT(r);
+  }
+
+  _status = 0;
+abort:
+  return (_status);
+}
+
+int
+nr_c2ru_get_children(NR_registry parent, char *child, void *ptr, size_t size, int (*get)(NR_registry, void*))
+{
+  int r, _status;
+  NR_registry registry;
+  unsigned int count;
+  unsigned int i;
+  NR_registry name;
+  struct entry { TAILQ_ENTRY(entry) entries; } *entry;
+  TAILQ_HEAD(, entry) *tailq = (void*)ptr;
+
+  TAILQ_INIT(tailq);
+
+  if ((r=nr_c2ru_make_registry(parent, child, registry)))
+    ABORT(r);
+
+  if ((r=NR_reg_get_child_count(registry, &count))) {
+    if (r != R_NOT_FOUND)
+      ABORT(r);
+  }
+  else {
+    for (i = 0; i < count; ++i) {
+      if ((r=NR_reg_get_child_registry(registry, i, name))) {
+        /* ignore R_NOT_FOUND errors */
+        if (r == R_NOT_FOUND)
+          continue;
+        else
+          ABORT(r);
+      }
+
+      if ((r=get(name, &entry))) {
+        /* ignore R_NOT_FOUND errors */
+        if (r == R_NOT_FOUND)
+          continue;
+        else
+          ABORT(r);
+      }
+
+      TAILQ_INSERT_TAIL(tailq, entry, entries);
+    }
+  }
+
+  _status = 0;
+abort:
+  return (_status);
+}
+
+int
+nr_c2ru_set_children(NR_registry parent, char *child, void *ptr, int (*set)(NR_registry, void*), int (*label)(NR_registry, void*, char[NR_REG_MAX_NR_REGISTRY_LEN]))
+{
+  int r, _status;
+  NR_registry registry;
+  int i;
+  NR_registry name;
+  char buffer[NR_REG_MAX_NR_REGISTRY_LEN];
+  struct entry { TAILQ_ENTRY(entry) entries; } *entry;
+  TAILQ_HEAD(, entry) *tailq = (void*)ptr;
+
+  if ((r=nr_c2ru_make_registry(parent, child, registry)))
+    ABORT(r);
+
+  (void)NR_reg_del(registry);
+
+  i = 0;
+  TAILQ_FOREACH(entry, tailq, entries) {
+    if (label == 0 || (r=label(registry, entry, buffer))) {
+      snprintf(buffer, sizeof(buffer), "%d", i);
+    }
+    if ((r=nr_c2ru_make_registry(registry, buffer, name)))
+      ABORT(r);
+
+    if ((r=set(name, entry)))
+      ABORT(r);
+
+    ++i;
+  }
+
+  _status = 0;
+abort:
+  return (_status);
+}
+
+int
+nr_c2ru_free_children(void *ptr, int (*free)(void*))
+{
+  struct entry { TAILQ_ENTRY(entry) entries; } *entry;
+  TAILQ_HEAD(, entry) *tailq = (void*)ptr;
+
+  while (! TAILQ_EMPTY(tailq)) {
+    entry = TAILQ_FIRST(tailq);
+    TAILQ_REMOVE(tailq, entry, entries);
+    (void)free(entry);
+  }
+
+  return 0;
+}
+
+/* requires parent already in legal form */
+int
+nr_c2ru_make_registry(NR_registry parent, char *child, NR_registry out)
+{
+    return NR_reg_make_registry(parent, child, out);
+}
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/registry/c2ru.h b/dom/media/webrtc/transport/third_party/nrappkit/src/registry/c2ru.h
new file mode 100644
index 0000000000..0f8c38ecc2
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/registry/c2ru.h
@@ -0,0 +1,96 @@
+/*
+ *
+ *    c2ru.h
+ *
+ *    $Source: /Users/ekr/tmp/nrappkit-dump/nrappkit/src/registry/c2ru.h,v $
+ *    $Revision: 1.2 $
+ *    $Date: 2006/08/16 19:39:13 $
+ *
+ *    c2r utility methods
+ *
+ *
+ *    Copyright (C) 2005, Network Resonance, Inc.
+ *    Copyright (C) 2006, Network Resonance, Inc.
+ *    All Rights Reserved
+ *
+ *    Redistribution and use in source and binary forms, with or without
+ *    modification, are permitted provided that the following conditions
+ *    are met:
+ *
+ *    1. Redistributions of source code must retain the above copyright
+ *       notice, this list of conditions and the following disclaimer.
+ *    2. Redistributions in binary form must reproduce the above copyright
+ *       notice, this list of conditions and the following disclaimer in the
+ *       documentation and/or other materials provided with the distribution.
+ *    3. Neither the name of Network Resonance, Inc. nor the name of any
+ *       contributors to this software may be used to endorse or promote
+ *       products derived from this software without specific prior written
+ *       permission.
+ *
+ *    THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+ *    AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ *    IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ *    ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+ *    LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ *    CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ *    SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+ *    INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+ *    CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ *    ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+ *    POSSIBILITY OF SUCH DAMAGE.
+ *
+ *
+ */
+
+#ifndef __C2RU_H__
+#define __C2RU_H__
+
+#include <sys/types.h>
+#include <r_types.h>
+#include <r_data.h>
+#include "registry_int.h"
+
+int nr_c2ru_get_char(NR_registry parent, char *child, char **out);
+int nr_c2ru_get_uchar(NR_registry parent, char *child, UCHAR **out);
+int nr_c2ru_get_int2(NR_registry parent, char *child, INT2 **out);
+int nr_c2ru_get_uint2(NR_registry parent, char *child, UINT2 **out);
+int nr_c2ru_get_int4(NR_registry parent, char *child, INT4 **out);
+int nr_c2ru_get_uint4(NR_registry parent, char *child, UINT4 **out);
+int nr_c2ru_get_int8(NR_registry parent, char *child, INT8 **out);
+int nr_c2ru_get_uint8(NR_registry parent, char *child, UINT8 **out);
+int nr_c2ru_get_double(NR_registry parent, char *child, double **out);
+int nr_c2ru_get_data(NR_registry parent, char *child, Data **out);
+int nr_c2ru_get_string(NR_registry parent, char *child, char ***out);
+
+int nr_c2ru_set_char(NR_registry parent, char *child, char *data);
+int nr_c2ru_set_uchar(NR_registry parent, char *child, UCHAR *data);
+int nr_c2ru_set_int2(NR_registry parent, char *child, INT2 *data);
+int nr_c2ru_set_uint2(NR_registry parent, char *child, UINT2 *data);
+int nr_c2ru_set_int4(NR_registry parent, char *child, INT4 *data);
+int nr_c2ru_set_uint4(NR_registry parent, char *child, UINT4 *data);
+int nr_c2ru_set_int8(NR_registry parent, char *child, INT8 *data);
+int nr_c2ru_set_uint8(NR_registry parent, char *child, UINT8 *data);
+int nr_c2ru_set_double(NR_registry parent, char *child, double *data);
+int nr_c2ru_set_registry(NR_registry parent, char *child);
+int nr_c2ru_set_data(NR_registry parent, char *child, Data *data);
+int nr_c2ru_set_string(NR_registry parent, char *child, char **data);
+
+int nr_c2ru_free_char(char *data);
+int nr_c2ru_free_uchar(UCHAR *data);
+int nr_c2ru_free_int2(INT2 *data);
+int nr_c2ru_free_uint2(UINT2 *data);
+int nr_c2ru_free_int4(INT4 *data);
+int nr_c2ru_free_uint4(UINT4 *data);
+int nr_c2ru_free_int8(INT8 *data);
+int nr_c2ru_free_uint8(UINT8 *data);
+int nr_c2ru_free_double(double *data);
+int nr_c2ru_free_data(Data *data);
+int nr_c2ru_free_string(char **data);
+
+int nr_c2ru_get_children(NR_registry parent, char *child, void *ptr, size_t size, int (*get)(NR_registry, void*));
+int nr_c2ru_set_children(NR_registry parent, char *child, void *ptr, int (*set)(NR_registry, void*), int (*label)(NR_registry, void*, char[NR_REG_MAX_NR_REGISTRY_LEN]));
+int nr_c2ru_free_children(void *ptr, int (*free)(void*));
+
+int nr_c2ru_make_registry(NR_registry parent, char *child, NR_registry out);
+
+#endif
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/registry/registry.c b/dom/media/webrtc/transport/third_party/nrappkit/src/registry/registry.c
new file mode 100644
index 0000000000..709b1c3fb7
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/registry/registry.c
@@ -0,0 +1,604 @@
+/*
+ *
+ *    registry.c
+ *
+ *    $Source: /Users/ekr/tmp/nrappkit-dump/nrappkit/src/registry/registry.c,v $
+ *    $Revision: 1.6 $
+ *    $Date: 2007/11/21 00:09:12 $
+ *
+ *    Datastore for tracking configuration and related info.
+ *
+ *
+ *    Copyright (C) 2005, Network Resonance, Inc.
+ *    Copyright (C) 2006, Network Resonance, Inc.
+ *    All Rights Reserved
+ *
+ *    Redistribution and use in source and binary forms, with or without
+ *    modification, are permitted provided that the following conditions
+ *    are met:
+ *
+ *    1. Redistributions of source code must retain the above copyright
+ *       notice, this list of conditions and the following disclaimer.
+ *    2. Redistributions in binary form must reproduce the above copyright
+ *       notice, this list of conditions and the following disclaimer in the
+ *       documentation and/or other materials provided with the distribution.
+ *    3. Neither the name of Network Resonance, Inc. nor the name of any
+ *       contributors to this software may be used to endorse or promote
+ *       products derived from this software without specific prior written
+ *       permission.
+ *
+ *    THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+ *    AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ *    IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ *    ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+ *    LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ *    CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ *    SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+ *    INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+ *    CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ *    ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+ *    POSSIBILITY OF SUCH DAMAGE.
+ *
+ *
+ */
+
+#include <assert.h>
+#include <string.h>
+#ifndef _MSC_VER
+#include <strings.h>
+#include <sys/param.h>
+#include <netinet/in.h>
+#endif
+#ifdef OPENSSL
+#include <openssl/ssl.h>
+#endif
+#include <ctype.h>
+#include "registry.h"
+#include "registry_int.h"
+#include "registry_vtbl.h"
+#include "r_assoc.h"
+#include "nr_common.h"
+#include "r_log.h"
+#include "r_errors.h"
+#include "r_macros.h"
+#include "c2ru.h"
+
+/* vtbl used to switch hit between local and remote invocations */
+static nr_registry_module *reg_vtbl = 0;
+
+/* must be in the order the types are numbered */
+static char *typenames[] = { "char", "UCHAR", "INT2", "UINT2", "INT4", "UINT4", "INT8", "UINT8", "double", "Data", "string", "registry" };
+
+int NR_LOG_REGISTRY=0;
+
+NR_registry NR_TOP_LEVEL_REGISTRY = "";
+
+int
+NR_reg_init(void *mode)
+{
+    int r, _status;
+    nr_registry_module *module = (nr_registry_module*)mode;
+#ifdef SANITY_CHECKS
+    NR_registry registry;
+#endif
+
+    if (reg_vtbl) {
+        if (reg_vtbl != module) {
+          r_log(LOG_GENERIC,LOG_ERR,"Can't reinitialize registry in different mode");
+          ABORT(R_INTERNAL);
+        }
+
+        return(0);
+    }
+
+    reg_vtbl = module;
+
+    if ((r=reg_vtbl->vtbl->init(mode)))
+        ABORT(r);
+
+#ifdef SANITY_CHECKS
+    if ((r=NR_reg_get_registry(NR_TOP_LEVEL_REGISTRY, registry)))
+        ABORT(r);
+    assert(strcmp(registry, NR_TOP_LEVEL_REGISTRY) == 0);
+#endif
+
+     r_log_init();
+     r_log_register("registry",&NR_LOG_REGISTRY);
+
+    _status=0;
+  abort:
+    r_log(NR_LOG_REGISTRY,
+          (_status ? LOG_ERR                        :  LOG_INFO),
+          (_status ? "Couldn't initialize registry" : "Initialized registry"));
+    return(_status);
+}
+
+int
+NR_reg_initted(void)
+{
+    return reg_vtbl!=0;
+}
+
+#define NRREGGET(func, method, type)                                \
+int                                                                 \
+func(NR_registry name, type *out)                                   \
+{                                                                   \
+    return reg_vtbl->vtbl->method(name, out);                             \
+}
+
+NRREGGET(NR_reg_get_char,     get_char,     char)
+NRREGGET(NR_reg_get_uchar,    get_uchar,    UCHAR)
+NRREGGET(NR_reg_get_int2,     get_int2,     INT2)
+NRREGGET(NR_reg_get_uint2,    get_uint2,    UINT2)
+NRREGGET(NR_reg_get_int4,     get_int4,     INT4)
+NRREGGET(NR_reg_get_uint4,    get_uint4,    UINT4)
+NRREGGET(NR_reg_get_int8,     get_int8,     INT8)
+NRREGGET(NR_reg_get_uint8,    get_uint8,    UINT8)
+NRREGGET(NR_reg_get_double,   get_double,   double)
+
+int
+NR_reg_get_registry(NR_registry name, NR_registry out)
+{
+    return reg_vtbl->vtbl->get_registry(name, out);
+}
+
+int
+NR_reg_get_bytes(NR_registry name, UCHAR *out, size_t size, size_t *length)
+{
+    return reg_vtbl->vtbl->get_bytes(name, out, size, length);
+}
+
+int
+NR_reg_get_string(NR_registry name, char *out, size_t size)
+{
+    return reg_vtbl->vtbl->get_string(name, out, size);
+}
+
+int
+NR_reg_get_length(NR_registry name, size_t *length)
+{
+    return reg_vtbl->vtbl->get_length(name, length);
+}
+
+int
+NR_reg_get_type(NR_registry name, NR_registry_type type)
+{
+    return reg_vtbl->vtbl->get_type(name, type);
+}
+
+#define NRREGSET(func, method, type)                            \
+int                                                             \
+func(NR_registry name, type data)                               \
+{                                                               \
+    return reg_vtbl->vtbl->method(name, data);                        \
+}
+
+NRREGSET(NR_reg_set_char,     set_char,     char)
+NRREGSET(NR_reg_set_uchar,    set_uchar,    UCHAR)
+NRREGSET(NR_reg_set_int2,     set_int2,     INT2)
+NRREGSET(NR_reg_set_uint2,    set_uint2,    UINT2)
+NRREGSET(NR_reg_set_int4,     set_int4,     INT4)
+NRREGSET(NR_reg_set_uint4,    set_uint4,    UINT4)
+NRREGSET(NR_reg_set_int8,     set_int8,     INT8)
+NRREGSET(NR_reg_set_uint8,    set_uint8,    UINT8)
+NRREGSET(NR_reg_set_double,   set_double,   double)
+NRREGSET(NR_reg_set_string,   set_string,   char*)
+
+int
+NR_reg_set_registry(NR_registry name)
+{
+    return reg_vtbl->vtbl->set_registry(name);
+}
+
+int
+NR_reg_set_bytes(NR_registry name, unsigned char *data, size_t length)
+{
+    return reg_vtbl->vtbl->set_bytes(name, data, length);
+}
+
+
+int
+NR_reg_del(NR_registry name)
+{
+    return reg_vtbl->vtbl->del(name);
+}
+
+int
+NR_reg_fin(NR_registry name)
+{
+    return reg_vtbl->vtbl->fin(name);
+}
+
+int
+NR_reg_get_child_count(NR_registry parent, unsigned int *count)
+{
+    assert(sizeof(count) == sizeof(size_t));
+    return reg_vtbl->vtbl->get_child_count(parent, (size_t*)count);
+}
+
+int
+NR_reg_get_child_registry(NR_registry parent, unsigned int i, NR_registry child)
+{
+    int r, _status;
+    size_t count;
+    NR_registry *children=0;
+
+    if ((r=reg_vtbl->vtbl->get_child_count(parent, &count)))
+      ABORT(r);
+
+    if (i >= count)
+        ABORT(R_NOT_FOUND);
+    else {
+        count++;
+        children = (NR_registry *)RCALLOC(count * sizeof(NR_registry));
+        if (!children)
+            ABORT(R_NO_MEMORY);
+
+        if ((r=reg_vtbl->vtbl->get_children(parent, children, count, &count)))
+            ABORT(r);
+
+        if (i >= count)
+            ABORT(R_NOT_FOUND);
+
+        strncpy(child, children[i], sizeof(NR_registry));
+    }
+
+    _status=0;
+  abort:
+    RFREE(children);
+    return(_status);
+}
+
+int
+NR_reg_get_children(NR_registry parent, NR_registry *children, size_t size, size_t *length)
+{
+    return reg_vtbl->vtbl->get_children(parent, children, size, length);
+}
+
+int
+NR_reg_dump()
+{
+    int r, _status;
+
+    if ((r=reg_vtbl->vtbl->dump(0)))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    return(_status);
+}
+
+// convenience methods, call RFREE on the returned data
+int
+NR_reg_alloc_data(NR_registry name, Data *data)
+{
+    int r, _status;
+    size_t length;
+    UCHAR  *tmp = 0;
+    size_t sanity_check;
+
+    if ((r=NR_reg_get_length(name, &length)))
+      ABORT(r);
+
+    if (!(tmp = (void*)RMALLOC(length)))
+      ABORT(R_NO_MEMORY);
+
+    if ((r=NR_reg_get_bytes(name, tmp, length, &sanity_check)))
+      ABORT(r);
+
+    assert(length == sanity_check);
+
+    data->len = length;
+    data->data = tmp;
+
+    _status=0;
+  abort:
+    if (_status) {
+      if (tmp) RFREE(tmp);
+    }
+    return(_status);
+}
+
+int
+NR_reg_alloc_string(NR_registry name, char **data)
+{
+    int r, _status;
+    size_t length;
+    char  *tmp = 0;
+
+    if ((r=NR_reg_get_length(name, &length)))
+      ABORT(r);
+
+    if (!(tmp = (void*)RMALLOC(length+1)))
+      ABORT(R_NO_MEMORY);
+
+    if ((r=NR_reg_get_string(name, tmp, length+1)))
+      ABORT(r);
+
+    assert(length == strlen(tmp));
+
+    *data = tmp;
+
+    _status=0;
+  abort:
+    if (_status) {
+      if (tmp) RFREE(tmp);
+    }
+    return(_status);
+}
+
+
+char *
+nr_reg_type_name(int type)
+{
+    if ((type < NR_REG_TYPE_CHAR) || (type > NR_REG_TYPE_REGISTRY))
+       return(NULL);
+
+    return(typenames[type]);
+}
+
+int
+nr_reg_compute_type(char *typename, int *type)
+{
+    int _status;
+    size_t i;
+
+#ifdef SANITY_CHECKS
+    assert(!strcasecmp(typenames[NR_REG_TYPE_CHAR],     "char"));
+    assert(!strcasecmp(typenames[NR_REG_TYPE_UCHAR],    "UCHAR"));
+    assert(!strcasecmp(typenames[NR_REG_TYPE_INT2],     "INT2"));
+    assert(!strcasecmp(typenames[NR_REG_TYPE_UINT2],    "UINT2"));
+    assert(!strcasecmp(typenames[NR_REG_TYPE_INT4],     "INT4"));
+    assert(!strcasecmp(typenames[NR_REG_TYPE_UINT4],    "UINT4"));
+    assert(!strcasecmp(typenames[NR_REG_TYPE_INT8],     "INT8"));
+    assert(!strcasecmp(typenames[NR_REG_TYPE_UINT8],    "UINT8"));
+    assert(!strcasecmp(typenames[NR_REG_TYPE_DOUBLE],   "double"));
+    assert(!strcasecmp(typenames[NR_REG_TYPE_BYTES],    "Data"));
+    assert(!strcasecmp(typenames[NR_REG_TYPE_STRING],   "string"));
+    assert(!strcasecmp(typenames[NR_REG_TYPE_REGISTRY], "registry"));
+    assert(sizeof(typenames)/sizeof(*typenames) == (NR_REG_TYPE_REGISTRY+1));
+#endif
+
+    for (i = 0; i < sizeof(typenames)/sizeof(*typenames); ++i) {
+      if (!strcasecmp(typenames[i], typename)) {
+        *type = i;
+        return 0;
+      }
+    }
+    ABORT(R_BAD_ARGS);
+
+    _status=0;
+  abort:
+    return(_status);
+}
+
+/* More convenience functions: the same as their parents but they
+   take a prefix and a suffix */
+#define NRGET2(func, type, get) \
+int                                                                  \
+func(NR_registry parent, char *child, type *out)                     \
+{                                                                    \
+  int r, _status;                                                    \
+  NR_registry registry;                                              \
+                                                                     \
+  if ((r = NR_reg_make_registry(parent, child, registry)))           \
+    ABORT(r);                                                        \
+                                                                     \
+  if ((r = get(registry, out))) {                                    \
+    ABORT(r);                                                        \
+  }                                                                  \
+                                                                     \
+  _status = 0;                                                       \
+abort:                                                               \
+  return (_status);                                                  \
+}
+
+NRGET2(NR_reg_get2_char,     char,    NR_reg_get_char)
+NRGET2(NR_reg_get2_uchar,    UCHAR,   NR_reg_get_uchar)
+NRGET2(NR_reg_get2_int2,     INT2,    NR_reg_get_int2)
+NRGET2(NR_reg_get2_uint2,    UINT2,   NR_reg_get_uint2)
+NRGET2(NR_reg_get2_int4,     INT4,    NR_reg_get_int4)
+NRGET2(NR_reg_get2_uint4,    UINT4,   NR_reg_get_uint4)
+NRGET2(NR_reg_get2_int8,     INT8,    NR_reg_get_int8)
+NRGET2(NR_reg_get2_uint8,    UINT8,   NR_reg_get_uint8)
+NRGET2(NR_reg_get2_double,   double,  NR_reg_get_double)
+NRGET2(NR_reg_alloc2_string,   char*,   NR_reg_alloc_string)
+NRGET2(NR_reg_alloc2_data,     Data,    NR_reg_alloc_data)
+
+int
+NR_reg_get2_bytes(NR_registry parent, char *child, UCHAR *out, size_t size, size_t *length)
+{
+    int r, _status;
+    NR_registry registry;
+
+    if ((r=NR_reg_make_registry(parent, child, registry)))
+      ABORT(r);
+
+    if ((r=NR_reg_get_bytes(registry, out, size, length)))
+      ABORT(r);
+
+    _status = 0;
+abort:
+    return (_status);
+}
+
+int
+NR_reg_get2_string(NR_registry parent, char *child, char *out, size_t size)
+{
+    int r, _status;
+    NR_registry registry;
+
+    if ((r=NR_reg_make_registry(parent, child, registry)))
+      ABORT(r);
+
+    if ((r=NR_reg_get_string(registry, out, size)))
+      ABORT(r);
+
+    _status = 0;
+abort:
+    return (_status);
+}
+
+/* More convenience functions: the same as their parents but they
+   take a prefix and a suffix */
+#define NRSET2(func, type, set) \
+int                                                                  \
+func(NR_registry parent, char *child, type in)                       \
+{                                                                    \
+  int r, _status;                                                    \
+  NR_registry registry;                                              \
+                                                                     \
+  if ((r = NR_reg_make_registry(parent, child, registry)))           \
+    ABORT(r);                                                        \
+                                                                     \
+  if ((r = set(registry, in))) {                                     \
+    ABORT(r);                                                        \
+  }                                                                  \
+                                                                     \
+  _status = 0;                                                       \
+abort:                                                               \
+  return (_status);                                                  \
+}
+
+NRSET2(NR_reg_set2_char,     char,    NR_reg_set_char)
+NRSET2(NR_reg_set2_uchar,    UCHAR,   NR_reg_set_uchar)
+NRSET2(NR_reg_set2_int2,     INT2,    NR_reg_set_int2)
+NRSET2(NR_reg_set2_uint2,    UINT2,   NR_reg_set_uint2)
+NRSET2(NR_reg_set2_int4,     INT4,    NR_reg_set_int4)
+NRSET2(NR_reg_set2_uint4,    UINT4,   NR_reg_set_uint4)
+NRSET2(NR_reg_set2_int8,     INT8,    NR_reg_set_int8)
+NRSET2(NR_reg_set2_uint8,    UINT8,   NR_reg_set_uint8)
+NRSET2(NR_reg_set2_double,   double,  NR_reg_set_double)
+NRSET2(NR_reg_set2_string,   char*,   NR_reg_set_string)
+
+int
+NR_reg_set2_bytes(NR_registry prefix, char *name, UCHAR *data, size_t length)
+{
+    int r, _status;
+    NR_registry registry;
+
+    if ((r = NR_reg_make_registry(prefix, name, registry)))
+      ABORT(r);
+
+    if ((r = NR_reg_set_bytes(registry, data, length)))
+      ABORT(r);
+
+    _status = 0;
+abort:
+    return (_status);
+}
+
+
+int
+NR_reg_make_child_registry(NR_registry parent, NR_registry descendant, unsigned int generation, NR_registry child)
+{
+    int _status;
+    size_t length;
+
+    length = strlen(parent);
+
+    if (strncasecmp(parent, descendant, length))
+        ABORT(R_BAD_ARGS);
+
+    while (descendant[length] != '\0') {
+        if (descendant[length] == '.') {
+            if (generation == 0)
+                break;
+
+            --generation;
+        }
+
+        ++length;
+        if (length >= sizeof(NR_registry))
+            ABORT(R_BAD_ARGS);
+    }
+
+    strncpy(child, descendant, length);
+    child[length] = '\0';
+
+    _status=0;
+  abort:
+    return(_status);
+}
+
+int
+NR_reg_get2_child_count(NR_registry base, NR_registry name, unsigned int *count)
+  {
+    int r, _status;
+    NR_registry registry;
+
+    if ((r=nr_c2ru_make_registry(base, name, registry)))
+      ABORT(r);
+
+    if (r=NR_reg_get_child_count(registry,count))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int
+NR_reg_get2_child_registry(NR_registry base, NR_registry name, unsigned int i, NR_registry child)
+  {
+    int r, _status;
+    NR_registry registry;
+
+    if ((r=nr_c2ru_make_registry(base, name, registry)))
+      ABORT(r);
+
+    if (r=NR_reg_get_child_registry(registry, i, child))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+
+/* requires parent already in legal form */
+int
+NR_reg_make_registry(NR_registry parent, char *child, NR_registry out)
+{
+    int r, _status;
+    size_t plen;
+    size_t clen;
+    char *c;
+    size_t i;
+
+    if ((r=nr_reg_is_valid(parent)))
+        ABORT(r);
+
+    if (*child == '.')
+        ABORT(R_BAD_ARGS);
+
+    clen = strlen(child);
+    if (!clen)
+        ABORT(R_BAD_ARGS);
+    plen = strlen(parent);
+    if ((plen + clen + 2) > sizeof(NR_registry))
+        ABORT(R_BAD_ARGS);
+
+    if (out != parent)
+        strcpy(out, parent);
+
+    c = &(out[plen]);
+
+    if (parent[0] != '\0') {
+        *c = '.';
+        ++c;
+    }
+
+    for (i = 0; i < clen; ++i, ++c) {
+        *c = child[i];
+        if (isspace(*c) || *c == '.' || *c == '/' || ! isprint(*c))
+            *c = '_';
+    }
+
+    *c = '\0';
+
+    _status = 0;
+abort:
+    return _status;
+}
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/registry/registry.h b/dom/media/webrtc/transport/third_party/nrappkit/src/registry/registry.h
new file mode 100644
index 0000000000..b48893ba72
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/registry/registry.h
@@ -0,0 +1,154 @@
+/*
+ *
+ *    registry.h
+ *
+ *    $Source: /Users/ekr/tmp/nrappkit-dump/nrappkit/src/registry/registry.h,v $
+ *    $Revision: 1.3 $
+ *    $Date: 2007/07/17 17:58:16 $
+ *
+ *    Datastore for tracking configuration and related info.
+ *
+ *
+ *    Copyright (C) 2005, Network Resonance, Inc.
+ *    Copyright (C) 2006, Network Resonance, Inc.
+ *    All Rights Reserved
+ *
+ *    Redistribution and use in source and binary forms, with or without
+ *    modification, are permitted provided that the following conditions
+ *    are met:
+ *
+ *    1. Redistributions of source code must retain the above copyright
+ *       notice, this list of conditions and the following disclaimer.
+ *    2. Redistributions in binary form must reproduce the above copyright
+ *       notice, this list of conditions and the following disclaimer in the
+ *       documentation and/or other materials provided with the distribution.
+ *    3. Neither the name of Network Resonance, Inc. nor the name of any
+ *       contributors to this software may be used to endorse or promote
+ *       products derived from this software without specific prior written
+ *       permission.
+ *
+ *    THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+ *    AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ *    IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ *    ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+ *    LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ *    CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ *    SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+ *    INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+ *    CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ *    ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+ *    POSSIBILITY OF SUCH DAMAGE.
+ *
+ *
+ */
+
+#ifndef __REGISTRY_H__
+#define __REGISTRY_H__
+
+#include <stdio.h>
+#include <sys/types.h>
+#include <r_types.h>
+#include <r_data.h>
+
+#define NR_REG_MAX_NR_REGISTRY_LEN     128
+#define NR_REG_MAX_TYPE_LEN            32
+
+typedef char NR_registry[NR_REG_MAX_NR_REGISTRY_LEN];
+typedef char NR_registry_type[NR_REG_MAX_TYPE_LEN];
+
+extern NR_registry NR_TOP_LEVEL_REGISTRY;
+
+extern void *NR_REG_MODE_LOCAL;
+extern void *NR_REG_MODE_REMOTE;
+
+int NR_reg_init(void *mode);
+
+int NR_reg_initted(void);
+
+int NR_reg_get_char(NR_registry name, char *out);
+int NR_reg_get_uchar(NR_registry name, UCHAR *out);
+int NR_reg_get_int2(NR_registry name, INT2 *out);
+int NR_reg_get_uint2(NR_registry name, UINT2 *out);
+int NR_reg_get_int4(NR_registry name, INT4 *out);
+int NR_reg_get_uint4(NR_registry name, UINT4 *out);
+int NR_reg_get_int8(NR_registry name, INT8 *out);
+int NR_reg_get_uint8(NR_registry name, UINT8 *out);
+int NR_reg_get_double(NR_registry name, double *out);
+int NR_reg_get_registry(NR_registry name, NR_registry out);
+
+int NR_reg_get_bytes(NR_registry name, UCHAR *out, size_t size, size_t *length);
+int NR_reg_get_string(NR_registry name, char *out, size_t size);
+int NR_reg_get_length(NR_registry name, size_t *length);
+int NR_reg_get_type(NR_registry name, NR_registry_type type);
+
+
+int NR_reg_get2_char(NR_registry prefix, char *name, char *);
+int NR_reg_get2_uchar(NR_registry prefix, char *name, UCHAR *);
+int NR_reg_get2_int2(NR_registry prefix, char *name, INT2 *);
+int NR_reg_get2_uint2(NR_registry prefix, char *name, UINT2 *);
+int NR_reg_get2_int4(NR_registry prefix, char *name, INT4 *);
+int NR_reg_get2_uint4(NR_registry prefix, char *name, UINT4 *);
+int NR_reg_get2_int8(NR_registry prefix, char *name,  INT8 *);
+int NR_reg_get2_uint8(NR_registry prefix, char *name, UINT8 *);
+int NR_reg_get2_double(NR_registry prefix, char *name, double *);
+int NR_reg_get2_bytes(NR_registry prefix, char *name, UCHAR *out, size_t size, size_t *length);
+int NR_reg_get2_string(NR_registry prefix, char *name, char *out, size_t size);
+
+int NR_reg_alloc2_string(NR_registry prefix, char *name, char **);
+int NR_reg_alloc2_data(NR_registry prefix, char *name, Data *);
+
+int NR_reg_set_char(NR_registry name, char data);
+int NR_reg_set_uchar(NR_registry name, UCHAR data);
+int NR_reg_set_int2(NR_registry name, INT2 data);
+int NR_reg_set_uint2(NR_registry name, UINT2 data);
+int NR_reg_set_int4(NR_registry name, INT4 data);
+int NR_reg_set_uint4(NR_registry name, UINT4 data);
+int NR_reg_set_int8(NR_registry name, INT8 data);
+int NR_reg_set_uint8(NR_registry name, UINT8 data);
+int NR_reg_set_double(NR_registry name, double data);
+
+int NR_reg_set_registry(NR_registry name);
+
+int NR_reg_set_bytes(NR_registry name, UCHAR *data, size_t length);
+int NR_reg_set_string(NR_registry name, char *data);
+
+int NR_reg_set2_char(NR_registry prefix, char *name, char data);
+int NR_reg_set2_uchar(NR_registry prefix, char *name, UCHAR data);
+int NR_reg_set2_int2(NR_registry prefix, char *name, INT2 data);
+int NR_reg_set2_uint2(NR_registry prefix, char *name, UINT2 data);
+int NR_reg_set2_int4(NR_registry prefix, char *name, INT4 data);
+int NR_reg_set2_uint4(NR_registry prefix, char *name, UINT4 data);
+int NR_reg_set2_int8(NR_registry prefix, char *name, INT8 data);
+int NR_reg_set2_uint8(NR_registry prefix, char *name, UINT8 data);
+int NR_reg_set2_double(NR_registry prefix, char *name, double data);
+
+int NR_reg_set2_bytes(NR_registry prefix, char *name, UCHAR *data, size_t length);
+int NR_reg_set2_string(NR_registry prefix, char *name, char *data);
+
+int NR_reg_del(NR_registry name);
+
+int NR_reg_fin(NR_registry name);
+
+int NR_reg_get_child_count(NR_registry parent, unsigned int *count);
+int NR_reg_get_child_registry(NR_registry parent, unsigned int i, NR_registry child);
+int NR_reg_get2_child_count(NR_registry base, NR_registry name, unsigned int *count);
+int NR_reg_get2_child_registry(NR_registry base, NR_registry name, unsigned int i, NR_registry child);
+int NR_reg_get_children(NR_registry parent, NR_registry children[], size_t size, size_t *length);
+
+int NR_reg_dump(void);
+
+/* convenience methods, call RFREE on the returned data */
+int NR_reg_alloc_data(NR_registry name, Data *data);
+int NR_reg_alloc_string(NR_registry name, char **data);
+
+#define NR_REG_CB_ACTION_ADD      (1<<0)
+#define NR_REG_CB_ACTION_CHANGE   (1<<1)
+#define NR_REG_CB_ACTION_DELETE   (1<<2)
+#define NR_REG_CB_ACTION_FINAL    (1<<6)
+int NR_reg_register_callback(NR_registry name, char action, void (*cb)(void *cb_arg, char action, NR_registry name), void *cb_arg);
+int NR_reg_unregister_callback(NR_registry name, char action, void (*cb)(void *cb_arg, char action, NR_registry name));
+
+int NR_reg_make_registry(NR_registry parent, char *child, NR_registry out);
+int NR_reg_make_child_registry(NR_registry parent, NR_registry descendant, unsigned int generation, NR_registry child);
+
+#endif
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/registry/registry_int.h b/dom/media/webrtc/transport/third_party/nrappkit/src/registry/registry_int.h
new file mode 100644
index 0000000000..d6d412c543
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/registry/registry_int.h
@@ -0,0 +1,97 @@
+#if 0
+#define NR_LOG_REGISTRY BLAHBLAH()
+#define LOG_REGISTRY BLAHBLAH()
+static int BLAHBLAH() {
+int blahblah;
+r_log_register("registry",&blahblah);
+return blahblah;
+}
+#endif
+
+/*
+ *
+ *    registry_int.h
+ *
+ *    $Source: /Users/ekr/tmp/nrappkit-dump/nrappkit/src/registry/registry_int.h,v $
+ *    $Revision: 1.3 $
+ *    $Date: 2007/06/26 22:37:51 $
+ *
+ *    Callback-related functions
+ *
+ *
+ *    Copyright (C) 2005, Network Resonance, Inc.
+ *    Copyright (C) 2006, Network Resonance, Inc.
+ *    All Rights Reserved
+ *
+ *    Redistribution and use in source and binary forms, with or without
+ *    modification, are permitted provided that the following conditions
+ *    are met:
+ *
+ *    1. Redistributions of source code must retain the above copyright
+ *       notice, this list of conditions and the following disclaimer.
+ *    2. Redistributions in binary form must reproduce the above copyright
+ *       notice, this list of conditions and the following disclaimer in the
+ *       documentation and/or other materials provided with the distribution.
+ *    3. Neither the name of Network Resonance, Inc. nor the name of any
+ *       contributors to this software may be used to endorse or promote
+ *       products derived from this software without specific prior written
+ *       permission.
+ *
+ *    THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+ *    AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ *    IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ *    ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+ *    LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ *    CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ *    SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+ *    INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+ *    CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ *    ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+ *    POSSIBILITY OF SUCH DAMAGE.
+ *
+ *
+ */
+
+#ifndef __REGISTRY_INT_H__
+#define __REGISTRY_INT_H__
+
+#include <sys/types.h>
+#include <r_types.h>
+#ifndef NO_REG_RPC
+#include <rpc/rpc.h>
+#endif
+
+extern int NR_LOG_REGISTRY;
+
+int nr_reg_is_valid(NR_registry name);
+
+#define NR_REG_TYPE_CHAR               0
+#define NR_REG_TYPE_UCHAR              1
+#define NR_REG_TYPE_INT2               2
+#define NR_REG_TYPE_UINT2              3
+#define NR_REG_TYPE_INT4               4
+#define NR_REG_TYPE_UINT4              5
+#define NR_REG_TYPE_INT8               6
+#define NR_REG_TYPE_UINT8              7
+#define NR_REG_TYPE_DOUBLE             8
+#define NR_REG_TYPE_BYTES              9
+#define NR_REG_TYPE_STRING             10
+#define NR_REG_TYPE_REGISTRY           11
+char *nr_reg_type_name(int type);
+int nr_reg_compute_type(char *type_name, int *type);
+
+char *nr_reg_action_name(int action);
+
+int nr_reg_cb_init(void);
+int nr_reg_client_cb_init(void);
+int nr_reg_register_for_callbacks(int fd, int connect_to_port);
+int nr_reg_raise_event(NR_registry name, int action);
+#ifndef NO_REG_RPC
+int nr_reg_get_client(CLIENT **client);
+#endif
+
+#define CALLBACK_SERVER_ADDR     "127.0.0.1"
+#define CALLBACK_SERVER_PORT     8082
+#define CALLBACK_SERVER_BACKLOG  32
+
+#endif
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/registry/registry_local.c b/dom/media/webrtc/transport/third_party/nrappkit/src/registry/registry_local.c
new file mode 100644
index 0000000000..ed6e19aaa0
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/registry/registry_local.c
@@ -0,0 +1,1168 @@
+/*
+ *
+ *    registry.c
+ *
+ *    $Source: /Users/ekr/tmp/nrappkit-dump/nrappkit/src/registry/registry_local.c,v $
+ *    $Revision: 1.4 $
+ *    $Date: 2007/11/21 00:09:13 $
+ *
+ *    Datastore for tracking configuration and related info.
+ *
+ *
+ *    Copyright (C) 2005, Network Resonance, Inc.
+ *    Copyright (C) 2006, Network Resonance, Inc.
+ *    All Rights Reserved
+ *
+ *    Redistribution and use in source and binary forms, with or without
+ *    modification, are permitted provided that the following conditions
+ *    are met:
+ *
+ *    1. Redistributions of source code must retain the above copyright
+ *       notice, this list of conditions and the following disclaimer.
+ *    2. Redistributions in binary form must reproduce the above copyright
+ *       notice, this list of conditions and the following disclaimer in the
+ *       documentation and/or other materials provided with the distribution.
+ *    3. Neither the name of Network Resonance, Inc. nor the name of any
+ *       contributors to this software may be used to endorse or promote
+ *       products derived from this software without specific prior written
+ *       permission.
+ *
+ *    THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+ *    AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ *    IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ *    ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+ *    LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ *    CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ *    SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+ *    INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+ *    CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ *    ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+ *    POSSIBILITY OF SUCH DAMAGE.
+ *
+ *
+ */
+
+#include <assert.h>
+#include <string.h>
+#ifndef WIN32
+#include <strings.h>
+#include <sys/param.h>
+#include <netinet/in.h>
+#endif
+#ifdef OPENSSL
+#include <openssl/ssl.h>
+#endif
+#include <ctype.h>
+#include "registry.h"
+#include "registry_int.h"
+#include "registry_vtbl.h"
+#include "r_assoc.h"
+#include "nr_common.h"
+#include "r_log.h"
+#include "r_errors.h"
+#include "r_macros.h"
+
+/* if C were an object-oriented language, nr_scalar_registry_node and
+ * nr_array_registry_node would subclass nr_registry_node, but it isn't
+ * object-oriented language, so this is used in cases where the pointer
+ * could be of either type */
+typedef struct nr_registry_node_ {
+    unsigned char  type;
+} nr_registry_node;
+
+typedef struct nr_scalar_registry_node_ {
+    unsigned char  type;
+    union {
+        char          _char;
+        UCHAR         _uchar;
+        INT2       _nr_int2;
+        UINT2      _nr_uint2;
+        INT4       _nr_int4;
+        UINT4      _nr_uint4;
+        INT8       _nr_int8;
+        UINT8      _nr_uint8;
+        double        _double;
+    } scalar;
+} nr_scalar_registry_node;
+
+/* string, bytes */
+typedef struct nr_array_registry_node_ {
+    unsigned char    type;
+    struct {
+        unsigned int     length;
+        unsigned char    data[1];
+    } array;
+} nr_array_registry_node;
+
+static int nr_reg_local_init(nr_registry_module *me);
+static int nr_reg_local_get_char(NR_registry name, char *data);
+static int nr_reg_local_get_uchar(NR_registry name, UCHAR *data);
+static int nr_reg_local_get_int2(NR_registry name, INT2 *data);
+static int nr_reg_local_get_uint2(NR_registry name, UINT2 *data);
+static int nr_reg_local_get_int4(NR_registry name, INT4 *data);
+static int nr_reg_local_get_uint4(NR_registry name, UINT4 *data);
+static int nr_reg_local_get_int8(NR_registry name, INT8 *data);
+static int nr_reg_local_get_uint8(NR_registry name, UINT8 *data);
+static int nr_reg_local_get_double(NR_registry name, double *data);
+static int nr_reg_local_get_registry(NR_registry name, NR_registry data);
+static int nr_reg_local_get_bytes(NR_registry name, UCHAR *data, size_t size, size_t *length);
+static int nr_reg_local_get_string(NR_registry name, char *data, size_t size);
+static int nr_reg_local_get_length(NR_registry name, size_t *len);
+static int nr_reg_local_get_type(NR_registry name, NR_registry_type type);
+static int nr_reg_local_set_char(NR_registry name, char data);
+static int nr_reg_local_set_uchar(NR_registry name, UCHAR data);
+static int nr_reg_local_set_int2(NR_registry name, INT2 data);
+static int nr_reg_local_set_uint2(NR_registry name, UINT2 data);
+static int nr_reg_local_set_int4(NR_registry name, INT4 data);
+static int nr_reg_local_set_uint4(NR_registry name, UINT4 data);
+static int nr_reg_local_set_int8(NR_registry name, INT8 data);
+static int nr_reg_local_set_uint8(NR_registry name, UINT8 data);
+static int nr_reg_local_set_double(NR_registry name, double data);
+static int nr_reg_local_set_registry(NR_registry name);
+static int nr_reg_local_set_bytes(NR_registry name, UCHAR *data, size_t length);
+static int nr_reg_local_set_string(NR_registry name, char *data);
+static int nr_reg_local_del(NR_registry name);
+static int nr_reg_local_get_child_count(NR_registry parent, size_t *count);
+static int nr_reg_local_get_children(NR_registry parent, NR_registry *data, size_t size, size_t *length);
+static int nr_reg_local_fin(NR_registry name);
+static int nr_reg_local_dump(int sorted);
+static int nr_reg_insert_node(char *name, void *node);
+static int nr_reg_change_node(char *name, void *node, void *old);
+static int nr_reg_get(char *name, int type, void *out);
+static int nr_reg_get_data(NR_registry name, nr_scalar_registry_node *node, void *out);
+static int nr_reg_get_array(char *name, unsigned char type, UCHAR *out, size_t size, size_t *length);
+static int nr_reg_set(char *name, int type, void *data);
+static int nr_reg_set_array(char *name, unsigned char type, UCHAR *data, size_t length);
+static int nr_reg_set_parent_registries(char *name);
+
+/* make these static OLD_REGISTRY */
+#if 0
+static int nr_reg_fetch_node(char *name, unsigned char type, nr_registry_node **node, int *free_node);
+static char *nr_reg_alloc_node_data(char *name, nr_registry_node *node, int *freeit);
+#else
+int nr_reg_fetch_node(char *name, unsigned char type, nr_registry_node **node, int *free_node);
+char *nr_reg_alloc_node_data(char *name, nr_registry_node *node, int *freeit);
+#endif
+static int nr_reg_rfree(void *ptr);
+#if 0  /* Unused currently */
+static int nr_reg_noop(void *ptr);
+#endif
+static int nr_reg_compute_length(char *name, nr_registry_node *node, size_t *length);
+char *nr_reg_action_name(int action);
+
+/* the registry, containing mappings like "foo.bar.baz" to registry
+ * nodes, which are either of type nr_scalar_registry_node or
+ * nr_array_registry_node */
+static r_assoc     *nr_registry = 0;
+
+#if 0  /* Unused currently */
+static nr_array_registry_node nr_top_level_node;
+#endif
+
+typedef struct nr_reg_find_children_arg_ {
+    size_t         size;
+    NR_registry   *children;
+    size_t         length;
+} nr_reg_find_children_arg;
+
+static int nr_reg_local_iter(NR_registry prefix, int (*action)(void *ptr, r_assoc_iterator *iter, char *prefix, char *name, nr_registry_node *node), void *ptr);
+static int nr_reg_local_iter_delete(void *ptr, r_assoc_iterator *iter, char *prefix, char *name, nr_registry_node *node);
+static int nr_reg_local_find_children(void *ptr, r_assoc_iterator *iter, char *prefix, char *name, nr_registry_node *node);
+static int nr_reg_local_count_children(void *ptr, r_assoc_iterator *iter, char *prefix, char *name, nr_registry_node *node);
+static int nr_reg_local_dump_print(void *ptr, r_assoc_iterator *iter, char *prefix, char *name, nr_registry_node *node);
+
+
+
+int
+nr_reg_local_iter(NR_registry prefix, int (*action)(void *ptr, r_assoc_iterator *iter, char *prefix, char *name, nr_registry_node *node), void *ptr)
+{
+    int r, _status;
+    r_assoc_iterator iter;
+    char *name;
+    int namel;
+    nr_registry_node *node;
+    int prefixl;
+
+    if (prefix == 0)
+        ABORT(R_INTERNAL);
+
+    if ((r=r_assoc_init_iter(nr_registry, &iter)))
+        ABORT(r);
+
+    prefixl = strlen(prefix);
+
+    for (;;) {
+      if ((r=r_assoc_iter(&iter, (void*)&name, &namel, (void*)&node))) {
+        if (r == R_EOD)
+          break;
+        else
+          ABORT(r);
+      }
+
+      /* subtract to remove the '\0' character from the string length */
+      --namel;
+
+      /* sanity check that the name is null-terminated */
+      assert(namel >= 0);
+      assert(name[namel] == '\0');
+
+      if (namel < 0 || name[namel] != '\0' || node == 0)
+          break;
+
+      /* 3 cases where action will be called:
+       *   1) prefix == ""
+       *   2) prefix == name
+       *   3) name == prefix + '.'
+       */
+      if (prefixl == 0
+       || ((namel == prefixl || (namel > prefixl && name[prefixl] == '.'))
+         && !strncmp(prefix, name, prefixl))) {
+        if ((r=action(ptr, &iter, prefix, name, node)))
+          ABORT(r);
+      }
+    }
+
+    _status=0;
+  abort:
+
+    return(_status);
+}
+
+int
+nr_reg_local_iter_delete(void *ptr, r_assoc_iterator *iter, char *prefix, char *name, nr_registry_node *node)
+{
+    int r, _status;
+
+    if ((r=r_assoc_iter_delete(iter)))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    return(_status);
+}
+
+int
+nr_reg_local_find_children(void *ptr, r_assoc_iterator *iter, char *prefix, char *name, nr_registry_node *node)
+{
+  int _status;
+  int prefixl = strlen(prefix);
+  char *dot;
+  nr_reg_find_children_arg *arg = (void*)ptr;
+
+  assert(sizeof(*(arg->children)) == sizeof(NR_registry));
+
+  /* only grovel through immediate children */
+  if (prefixl == 0 || name[prefixl] == '.') {
+    if (name[prefixl] != '\0') {
+      dot = strchr(&name[prefixl+1], '.');
+      if (dot == 0) {
+        strncpy(arg->children[arg->length], name, sizeof(NR_registry)-1);
+        ++arg->length;
+
+        /* only grab as many as there are room for */
+        if (arg->length >= arg->size)
+          ABORT(R_INTERRUPTED);
+      }
+    }
+  }
+
+  _status = 0;
+ abort:
+  return _status;
+}
+
+int
+nr_reg_local_count_children(void *ptr, r_assoc_iterator *iter, char *prefix, char *name, nr_registry_node *node)
+{
+  int prefixl = strlen(prefix);
+  char *dot;
+
+  /* only count children */
+  if (name[prefixl] == '.') {
+    dot = strchr(&name[prefixl+1], '.');
+    if (dot == 0)
+      ++(*(unsigned int *)ptr);
+  }
+  else if (name[0] != '\0') {
+    if (prefixl == 0)
+      ++(*(unsigned int *)ptr);
+  }
+
+  return 0;
+}
+
+int
+nr_reg_local_dump_print(void *ptr, r_assoc_iterator *iter, char *prefix, char *name, nr_registry_node *node)
+{
+    int _status;
+    int freeit = 0;
+    char *data;
+
+    /* only print leaf nodes */
+    if (node->type != NR_REG_TYPE_REGISTRY) {
+      data = nr_reg_alloc_node_data(name, node, &freeit);
+      if (ptr)
+        fprintf((FILE*)ptr, "%s: %s\n", name, data);
+      else
+        r_log(NR_LOG_REGISTRY, LOG_INFO, "%s: %s", name, data);
+      if (freeit)
+        RFREE(data);
+    }
+
+    _status=0;
+  //abort:
+    return(_status);
+}
+
+
+#if 0  /* Unused currently */
+int
+nr_reg_noop(void *ptr)
+{
+    return 0;
+}
+#endif
+
+int
+nr_reg_rfree(void *ptr)
+{
+    RFREE(ptr);
+    return 0;
+}
+
+int
+nr_reg_fetch_node(char *name, unsigned char type, nr_registry_node **node, int *free_node)
+{
+    int r, _status;
+
+    *node = 0;
+    *free_node = 0;
+
+    if ((r=nr_reg_is_valid(name)))
+      ABORT(r);
+
+    if ((r=r_assoc_fetch(nr_registry, name, strlen(name)+1, (void*)node)))
+      ABORT(r);
+
+    if ((*node)->type != type)
+      ABORT(R_FAILED);
+
+    _status=0;
+  abort:
+    if (_status) {
+        if (*node)
+            r_log(NR_LOG_REGISTRY, LOG_DEBUG, "Couldn't fetch node '%s' ('%s'), found '%s' instead",
+              name, nr_reg_type_name(type), nr_reg_type_name((*node)->type));
+        else
+            r_log(NR_LOG_REGISTRY, LOG_DEBUG, "Couldn't fetch node '%s' ('%s')",
+              name, nr_reg_type_name(type));
+    }
+    else {
+        r_log(NR_LOG_REGISTRY, LOG_DEBUG, "Fetched node '%s' ('%s')",
+              name, nr_reg_type_name(type));
+    }
+    return(_status);
+}
+
+int
+nr_reg_insert_node(char *name, void *node)
+{
+    int r, _status;
+
+    if ((r=nr_reg_is_valid(name)))
+      ABORT(r);
+
+    /* since the registry application is not multi-threaded, a node being
+     * inserted should always be a new node because the registry app must
+     * have looked for a node with this key but not found it, so it is
+     * being created/inserted now using R_ASSOC_NEW */
+    if ((r=r_assoc_insert(nr_registry, name, strlen(name)+1, node, 0, nr_reg_rfree, R_ASSOC_NEW)))
+      ABORT(r);
+
+    if ((r=nr_reg_set_parent_registries(name)))
+      ABORT(r);
+
+    if ((r=nr_reg_raise_event(name, NR_REG_CB_ACTION_ADD)))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    if (r_logging(NR_LOG_REGISTRY, LOG_INFO)) {
+      int freeit;
+      char *data = nr_reg_alloc_node_data(name, (void*)node, &freeit);
+      r_log(NR_LOG_REGISTRY, LOG_INFO,
+             "insert '%s' (%s) %s: %s", name,
+             nr_reg_type_name(((nr_registry_node*)node)->type),
+             (_status ? "FAILED" : "succeeded"), data);
+      if (freeit)
+        RFREE(data);
+    }
+    return(_status);
+}
+
+int
+nr_reg_change_node(char *name, void *node, void *old)
+{
+    int r, _status;
+
+    if ((r=nr_reg_is_valid(name)))
+      ABORT(r);
+
+    if (old != node) {
+        if ((r=r_assoc_insert(nr_registry, name, strlen(name)+1, node, 0, nr_reg_rfree, R_ASSOC_REPLACE)))
+          ABORT(r);
+    }
+
+    if ((r=nr_reg_raise_event(name, NR_REG_CB_ACTION_CHANGE)))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    if (r_logging(NR_LOG_REGISTRY, LOG_INFO)) {
+      int freeit;
+      char *data = nr_reg_alloc_node_data(name, (void*)node, &freeit);
+      r_log(NR_LOG_REGISTRY, LOG_INFO,
+             "change '%s' (%s) %s: %s", name,
+             nr_reg_type_name(((nr_registry_node*)node)->type),
+             (_status ? "FAILED" : "succeeded"), data);
+      if (freeit)
+        RFREE(data);
+    }
+    return(_status);
+}
+
+char *
+nr_reg_alloc_node_data(char *name, nr_registry_node *node, int *freeit)
+{
+    char *s = 0;
+    int len;
+    int alloc = 0;
+    unsigned int i;
+
+    *freeit = 0;
+
+    switch (node->type) {
+    default:
+      alloc = 100;    /* plenty of room for any of the scalar types */
+      break;
+    case NR_REG_TYPE_REGISTRY:
+      alloc = strlen(name) + 1;
+      break;
+    case NR_REG_TYPE_BYTES:
+      alloc = (2 * ((nr_array_registry_node*)node)->array.length) + 1;
+      break;
+    case NR_REG_TYPE_STRING:
+      alloc = 0;
+      break;
+    }
+
+    if (alloc > 0) {
+      s = (void*)RMALLOC(alloc);
+      if (!s)
+        return "";
+
+      *freeit = 1;
+    }
+
+    len = alloc;
+
+    switch (node->type) {
+    case NR_REG_TYPE_CHAR:
+      i = ((nr_scalar_registry_node*)node)->scalar._char;
+      if (isprint(i) && ! isspace(i))
+          snprintf(s, len, "%c", (char)i);
+      else
+          snprintf(s, len, "\\%03o", (char)i);
+      break;
+    case NR_REG_TYPE_UCHAR:
+      snprintf(s, len, "0x%02x", ((nr_scalar_registry_node*)node)->scalar._uchar);
+      break;
+    case NR_REG_TYPE_INT2:
+      snprintf(s, len, "%d", ((nr_scalar_registry_node*)node)->scalar._nr_int2);
+      break;
+    case NR_REG_TYPE_UINT2:
+      snprintf(s, len, "%u", ((nr_scalar_registry_node*)node)->scalar._nr_uint2);
+      break;
+    case NR_REG_TYPE_INT4:
+      snprintf(s, len, "%d", ((nr_scalar_registry_node*)node)->scalar._nr_int4);
+      break;
+    case NR_REG_TYPE_UINT4:
+      snprintf(s, len, "%u", ((nr_scalar_registry_node*)node)->scalar._nr_uint4);
+      break;
+    case NR_REG_TYPE_INT8:
+      snprintf(s, len, "%lld", ((nr_scalar_registry_node*)node)->scalar._nr_int8);
+      break;
+    case NR_REG_TYPE_UINT8:
+      snprintf(s, len, "%llu", ((nr_scalar_registry_node*)node)->scalar._nr_uint8);
+      break;
+    case NR_REG_TYPE_DOUBLE:
+      snprintf(s, len, "%#f", ((nr_scalar_registry_node*)node)->scalar._double);
+      break;
+    case NR_REG_TYPE_REGISTRY:
+      snprintf(s, len, "%s", name);
+      break;
+    case NR_REG_TYPE_BYTES:
+      for (i = 0; i < ((nr_array_registry_node*)node)->array.length; ++i) {
+        sprintf(&s[2*i], "%02x", ((nr_array_registry_node*)node)->array.data[i]);
+      }
+      break;
+    case NR_REG_TYPE_STRING:
+      s = (char*)((nr_array_registry_node*)node)->array.data;
+      break;
+    default:
+      assert(0); /* bad value */
+      *freeit = 0;
+      s = "";
+      break;
+    }
+
+    return s;
+}
+
+int
+nr_reg_get(char *name, int type, void *out)
+{
+    int r, _status;
+    nr_scalar_registry_node *node = 0;
+    int free_node = 0;
+
+    if ((r=nr_reg_fetch_node(name, type, (void*)&node, &free_node)))
+      ABORT(r);
+
+    if ((r=nr_reg_get_data(name, node, out)))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    if (free_node) RFREE(node);
+    return(_status);
+}
+
+int
+nr_reg_get_data(NR_registry name, nr_scalar_registry_node *node, void *out)
+{
+    int _status;
+
+    switch (node->type) {
+    case NR_REG_TYPE_CHAR:
+      *(char*)out = node->scalar._char;
+      break;
+    case NR_REG_TYPE_UCHAR:
+      *(UCHAR*)out = node->scalar._uchar;
+      break;
+    case NR_REG_TYPE_INT2:
+      *(INT2*)out = node->scalar._nr_int2;
+      break;
+    case NR_REG_TYPE_UINT2:
+      *(UINT2*)out = node->scalar._nr_uint2;
+      break;
+    case NR_REG_TYPE_INT4:
+      *(INT4*)out = node->scalar._nr_int4;
+      break;
+    case NR_REG_TYPE_UINT4:
+      *(UINT4*)out = node->scalar._nr_uint4;
+      break;
+    case NR_REG_TYPE_INT8:
+      *(INT8*)out = node->scalar._nr_int8;
+      break;
+    case NR_REG_TYPE_UINT8:
+      *(UINT8*)out = node->scalar._nr_uint8;
+      break;
+    case NR_REG_TYPE_DOUBLE:
+      *(double*)out = node->scalar._double;
+      break;
+    default:
+      ABORT(R_INTERNAL);
+      break;
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+}
+
+int
+nr_reg_get_array(char *name, unsigned char type, unsigned char *out, size_t size, size_t *length)
+{
+    int r, _status;
+    nr_array_registry_node *node = 0;
+    int free_node = 0;
+
+    if ((r=nr_reg_fetch_node(name, type, (void*)&node, &free_node)))
+      ABORT(r);
+
+    if (size < node->array.length)
+      ABORT(R_BAD_ARGS);
+
+    if (out != 0)
+        memcpy(out, node->array.data, node->array.length);
+    if (length != 0)
+        *length = node->array.length;
+
+    _status=0;
+  abort:
+    if (node && free_node) RFREE(node);
+    return(_status);
+}
+
+int
+nr_reg_set(char *name, int type, void *data)
+{
+    int r, _status;
+    nr_scalar_registry_node *node = 0;
+    int create_node = 0;
+    int changed = 0;
+    int free_node = 0;
+
+    if ((r=nr_reg_fetch_node(name, type, (void*)&node, &free_node)))
+      if (r == R_NOT_FOUND) {
+        create_node = 1;
+        free_node = 1;
+      }
+      else
+        ABORT(r);
+
+    if (create_node) {
+      if (!(node=(void*)RCALLOC(sizeof(nr_scalar_registry_node))))
+        ABORT(R_NO_MEMORY);
+
+      node->type = type;
+    }
+    else {
+      if (node->type != type)
+        ABORT(R_BAD_ARGS);
+    }
+
+    switch (type) {
+#define CASE(TYPE, _name, type)                   \
+    case TYPE:                                    \
+      if (node->scalar._name != *(type*)data) {   \
+        node->scalar._name = *(type*)data;        \
+        if (! create_node)                        \
+          changed = 1;                            \
+      }                                           \
+      break;
+    CASE(NR_REG_TYPE_CHAR,       _char,       char)
+    CASE(NR_REG_TYPE_UCHAR,      _uchar,      UCHAR)
+    CASE(NR_REG_TYPE_INT2,    _nr_int2,    INT2)
+    CASE(NR_REG_TYPE_UINT2,   _nr_uint2,   UINT2)
+    CASE(NR_REG_TYPE_INT4,    _nr_int4,    INT4)
+    CASE(NR_REG_TYPE_UINT4,   _nr_uint4,   UINT4)
+    CASE(NR_REG_TYPE_INT8,    _nr_int8,    INT8)
+    CASE(NR_REG_TYPE_UINT8,   _nr_uint8,   UINT8)
+    CASE(NR_REG_TYPE_DOUBLE,     _double,     double)
+#undef CASE
+
+    case NR_REG_TYPE_REGISTRY:
+      /* do nothing */
+      break;
+
+    default:
+      ABORT(R_INTERNAL);
+      break;
+    }
+
+    if (create_node) {
+      if ((r=nr_reg_insert_node(name, node)))
+        ABORT(r);
+      free_node = 0;
+    }
+    else {
+      if (changed) {
+        if ((r=nr_reg_change_node(name, node, node)))
+          ABORT(r);
+        free_node = 0;
+      }
+    }
+
+    _status=0;
+  abort:
+    if (_status) {
+      if (node && free_node) RFREE(node);
+    }
+    return(_status);
+}
+
+int
+nr_reg_set_array(char *name, unsigned char type, UCHAR *data, size_t length)
+{
+    int r, _status;
+    nr_array_registry_node *old = 0;
+    nr_array_registry_node *node = 0;
+    int free_node = 0;
+    int added = 0;
+    int changed = 0;
+
+    if ((r=nr_reg_fetch_node(name, type, (void*)&old, &free_node))) {
+        if (r != R_NOT_FOUND)
+            ABORT(r);
+    }
+    else {
+      assert(free_node == 0);
+    }
+
+    if (old) {
+        if (old->type != type)
+            ABORT(R_BAD_ARGS);
+
+        if (old->array.length != length
+         || memcmp(old->array.data, data, length)) {
+            changed = 1;
+
+            if (old->array.length < length) {
+                if (!(node=(void*)RCALLOC(sizeof(nr_array_registry_node)+length)))
+                    ABORT(R_NO_MEMORY);
+            }
+            else {
+                node = old;
+            }
+        }
+    }
+    else {
+        if (!(node=(void*)RCALLOC(sizeof(nr_array_registry_node)+length)))
+            ABORT(R_NO_MEMORY);
+
+        added = 1;
+    }
+
+    if (added || changed) {
+        node->type = type;
+        node->array.length = length;
+        memcpy(node->array.data, data, length);
+    }
+
+    if (added) {
+        if ((r=nr_reg_insert_node(name, node)))
+            ABORT(r);
+    }
+    else if (changed) {
+        if ((r=nr_reg_change_node(name, node, old)))
+          ABORT(r);
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+}
+
+int
+nr_reg_set_parent_registries(char *name)
+{
+    int r, _status;
+    char *parent = 0;
+    char *dot;
+
+    if ((parent = r_strdup(name)) == 0)
+      ABORT(R_NO_MEMORY);
+
+    if ((dot = strrchr(parent, '.')) != 0) {
+      *dot = '\0';
+      if ((r=NR_reg_set_registry(parent)))
+        ABORT(r);
+    }
+
+    _status=0;
+  abort:
+    if (parent) RFREE(parent);
+    return(_status);
+}
+
+
+
+
+
+/* NON-STATIC METHODS */
+
+int
+nr_reg_is_valid(NR_registry name)
+{
+    int _status;
+    unsigned int length;
+    unsigned int i;
+
+    if (name == 0)
+      ABORT(R_BAD_ARGS);
+
+    /* make sure the key is null-terminated */
+    if (memchr(name, '\0', sizeof(NR_registry)) == 0)
+      ABORT(R_BAD_ARGS);
+
+    length = strlen(name);
+
+    /* cannot begin or end with a period */
+    if (name[0] == '.')
+      ABORT(R_BAD_ARGS);
+    if (strlen(name) > 0 && name[length-1] == '.')
+      ABORT(R_BAD_ARGS);
+
+    /* all characters cannot be space, and must be printable and not / */
+    for (i = 0; i < length; ++i) {
+      if (isspace(name[i]) || !isprint(name[i]) || name[i] == '/')
+        ABORT(R_BAD_ARGS);
+    }
+
+    _status=0;
+  abort:
+    if (_status) {
+      r_log(NR_LOG_REGISTRY, LOG_DEBUG, "invalid name '%s'", name);
+    }
+    return(_status);
+}
+
+
+int
+nr_reg_compute_length(char *name, nr_registry_node *in, size_t *length)
+{
+    int _status;
+    nr_array_registry_node *node = (nr_array_registry_node*)in;
+
+    switch (node->type) {
+    case NR_REG_TYPE_STRING:
+      *length = node->array.length - 1;
+      break;
+    case NR_REG_TYPE_BYTES:
+      *length = node->array.length;
+      break;
+    case NR_REG_TYPE_CHAR:
+      *length = sizeof(char);
+      break;
+    case NR_REG_TYPE_UCHAR:
+      *length = sizeof(UCHAR);
+      break;
+    case NR_REG_TYPE_INT2:
+    case NR_REG_TYPE_UINT2:
+      *length = 2;
+      break;
+    case NR_REG_TYPE_INT4:
+    case NR_REG_TYPE_UINT4:
+      *length = 4;
+      break;
+    case NR_REG_TYPE_INT8:
+    case NR_REG_TYPE_UINT8:
+      *length = 8;
+      break;
+    case NR_REG_TYPE_DOUBLE:
+      *length = sizeof(double);
+      break;
+    case NR_REG_TYPE_REGISTRY:
+      *length = strlen(name);
+      break;
+    default:
+      ABORT(R_INTERNAL);
+      break;
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+}
+
+
+/* VTBL METHODS */
+
+int
+nr_reg_local_init(nr_registry_module *me)
+{
+    int r, _status;
+
+    if (nr_registry == 0) {
+      if ((r=r_assoc_create(&nr_registry, r_assoc_crc32_hash_compute, 12)))
+        ABORT(r);
+
+      if ((r=nr_reg_cb_init()))
+            ABORT(r);
+
+      /* make sure NR_TOP_LEVEL_REGISTRY always exists */
+      if ((r=nr_reg_local_set_registry(NR_TOP_LEVEL_REGISTRY)))
+          ABORT(r);
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+}
+
+#define NRREGLOCALGET(func, TYPE, type)                             \
+int                                                                 \
+func(NR_registry name, type *out)                                   \
+{                                                                   \
+    return nr_reg_get(name, TYPE, out);                             \
+}
+
+NRREGLOCALGET(nr_reg_local_get_char,     NR_REG_TYPE_CHAR,     char)
+NRREGLOCALGET(nr_reg_local_get_uchar,    NR_REG_TYPE_UCHAR,    UCHAR)
+NRREGLOCALGET(nr_reg_local_get_int2,     NR_REG_TYPE_INT2,     INT2)
+NRREGLOCALGET(nr_reg_local_get_uint2,    NR_REG_TYPE_UINT2,    UINT2)
+NRREGLOCALGET(nr_reg_local_get_int4,     NR_REG_TYPE_INT4,     INT4)
+NRREGLOCALGET(nr_reg_local_get_uint4,    NR_REG_TYPE_UINT4,    UINT4)
+NRREGLOCALGET(nr_reg_local_get_int8,     NR_REG_TYPE_INT8,     INT8)
+NRREGLOCALGET(nr_reg_local_get_uint8,    NR_REG_TYPE_UINT8,    UINT8)
+NRREGLOCALGET(nr_reg_local_get_double,   NR_REG_TYPE_DOUBLE,   double)
+
+int
+nr_reg_local_get_registry(NR_registry name, NR_registry out)
+{
+    int r, _status;
+    nr_scalar_registry_node *node = 0;
+    int free_node = 0;
+
+    if ((r=nr_reg_fetch_node(name, NR_REG_TYPE_REGISTRY, (void*)&node, &free_node)))
+      ABORT(r);
+
+    strncpy(out, name, sizeof(NR_registry));
+
+    _status=0;
+  abort:
+    if (free_node) RFREE(node);
+    return(_status);
+
+}
+
+int
+nr_reg_local_get_bytes(NR_registry name, UCHAR *out, size_t size, size_t *length)
+{
+    return nr_reg_get_array(name, NR_REG_TYPE_BYTES, out, size, length);
+}
+
+int
+nr_reg_local_get_string(NR_registry name, char *out, size_t size)
+{
+    return nr_reg_get_array(name, NR_REG_TYPE_STRING, (UCHAR*)out, size, 0);
+}
+
+int
+nr_reg_local_get_length(NR_registry name, size_t *length)
+{
+    int r, _status;
+    nr_registry_node *node = 0;
+
+    if ((r=nr_reg_is_valid(name)))
+      ABORT(r);
+
+    if ((r=r_assoc_fetch(nr_registry, name, strlen(name)+1, (void*)&node)))
+      ABORT(r);
+
+    if ((r=nr_reg_compute_length(name, node, length)))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    return(_status);
+}
+
+int
+nr_reg_local_get_type(NR_registry name, NR_registry_type type)
+{
+    int r, _status;
+    nr_registry_node *node = 0;
+    char *str;
+
+    if ((r=nr_reg_is_valid(name)))
+      ABORT(r);
+
+    if ((r=r_assoc_fetch(nr_registry, name, strlen(name)+1, (void*)&node)))
+      ABORT(r);
+
+    str = nr_reg_type_name(node->type);
+    if (! str)
+        ABORT(R_BAD_ARGS);
+
+    strncpy(type, str, sizeof(NR_registry_type));
+
+    _status=0;
+  abort:
+    return(_status);
+}
+
+
+#define NRREGLOCALSET(func, TYPE, type)                         \
+int                                                             \
+func(NR_registry name, type data)                               \
+{                                                               \
+    return nr_reg_set(name, TYPE, &data);                       \
+}
+
+NRREGLOCALSET(nr_reg_local_set_char,     NR_REG_TYPE_CHAR,     char)
+NRREGLOCALSET(nr_reg_local_set_uchar,    NR_REG_TYPE_UCHAR,    UCHAR)
+NRREGLOCALSET(nr_reg_local_set_int2,     NR_REG_TYPE_INT2,     INT2)
+NRREGLOCALSET(nr_reg_local_set_uint2,    NR_REG_TYPE_UINT2,    UINT2)
+NRREGLOCALSET(nr_reg_local_set_int4,     NR_REG_TYPE_INT4,     INT4)
+NRREGLOCALSET(nr_reg_local_set_uint4,    NR_REG_TYPE_UINT4,    UINT4)
+NRREGLOCALSET(nr_reg_local_set_int8,     NR_REG_TYPE_INT8,     INT8)
+NRREGLOCALSET(nr_reg_local_set_uint8,    NR_REG_TYPE_UINT8,    UINT8)
+NRREGLOCALSET(nr_reg_local_set_double,   NR_REG_TYPE_DOUBLE,   double)
+
+int
+nr_reg_local_set_registry(NR_registry name)
+{
+    return nr_reg_set(name, NR_REG_TYPE_REGISTRY, 0);
+}
+
+int
+nr_reg_local_set_bytes(NR_registry name, unsigned char *data, size_t length)
+{
+    return nr_reg_set_array(name, NR_REG_TYPE_BYTES, data, length);
+}
+
+int
+nr_reg_local_set_string(NR_registry name, char *data)
+{
+    return nr_reg_set_array(name, NR_REG_TYPE_STRING, (UCHAR*)data, strlen(data)+1);
+}
+
+int
+nr_reg_local_del(NR_registry name)
+{
+    int r, _status;
+
+    if ((r=nr_reg_is_valid(name)))
+      ABORT(r);
+
+    /* delete from NR_registry */
+    if ((r=nr_reg_local_iter(name, nr_reg_local_iter_delete, 0)))
+      ABORT(r);
+
+    if ((r=nr_reg_raise_event(name, NR_REG_CB_ACTION_DELETE)))
+      ABORT(r);
+
+    /* if deleting from the root, re-insert the root */
+    if (! strcasecmp(name, NR_TOP_LEVEL_REGISTRY)) {
+      if ((r=nr_reg_local_set_registry(NR_TOP_LEVEL_REGISTRY)))
+          ABORT(r);
+    }
+
+    _status=0;
+  abort:
+    r_log(NR_LOG_REGISTRY,
+          (_status ? LOG_INFO               :  LOG_INFO),
+          "delete of '%s' %s", name,
+          (_status ? "FAILED"               : "succeeded"));
+    return(_status);
+}
+
+int
+nr_reg_local_get_child_count(NR_registry parent, size_t *count)
+{
+    int r, _status;
+    nr_registry_node *ignore1;
+    int ignore2;
+
+
+    if ((r=nr_reg_is_valid(parent)))
+      ABORT(r);
+
+    /* test to see whether it is present */
+    if ((r=nr_reg_fetch_node(parent, NR_REG_TYPE_REGISTRY, &ignore1, &ignore2)))
+      ABORT(r);
+
+    /* sanity check that there isn't any memory to free */
+    assert(ignore2 == 0);
+
+    *count = 0;
+
+    if ((r=nr_reg_local_iter(parent, nr_reg_local_count_children, count)))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    return(_status);
+}
+
+int
+nr_reg_local_get_children(NR_registry parent, NR_registry *data, size_t size, size_t *length)
+{
+    int r, _status;
+    nr_reg_find_children_arg arg;
+
+    if ((r=nr_reg_is_valid(parent)))
+      ABORT(r);
+
+    arg.children = data;
+    arg.size = size;
+    arg.length = 0;
+
+    if ((r=nr_reg_local_iter(parent, nr_reg_local_find_children, (void*)&arg))) {
+        if (r == R_INTERRUPTED)
+            ABORT(R_BAD_ARGS);
+        else
+            ABORT(r);
+    }
+
+    assert(sizeof(*arg.children) == sizeof(NR_registry));
+    qsort(arg.children, arg.length, sizeof(*arg.children), (void*)strcasecmp);
+
+    *length = arg.length;
+
+    _status = 0;
+  abort:
+    return(_status);
+}
+
+int
+nr_reg_local_fin(NR_registry name)
+{
+    int r, _status;
+
+    if ((r=nr_reg_raise_event(name, NR_REG_CB_ACTION_FINAL)))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    return(_status);
+}
+
+int
+nr_reg_local_dump(int sorted)
+{
+    int r, _status;
+
+    if ((r=nr_reg_local_iter(NR_TOP_LEVEL_REGISTRY, nr_reg_local_dump_print, 0)))
+      ABORT(r);
+
+    _status=0;
+  abort:
+    return(_status);
+}
+
+
+
+static nr_registry_module_vtbl nr_reg_local_vtbl = {
+    nr_reg_local_init,
+    nr_reg_local_get_char,
+    nr_reg_local_get_uchar,
+    nr_reg_local_get_int2,
+    nr_reg_local_get_uint2,
+    nr_reg_local_get_int4,
+    nr_reg_local_get_uint4,
+    nr_reg_local_get_int8,
+    nr_reg_local_get_uint8,
+    nr_reg_local_get_double,
+    nr_reg_local_get_registry,
+    nr_reg_local_get_bytes,
+    nr_reg_local_get_string,
+    nr_reg_local_get_length,
+    nr_reg_local_get_type,
+    nr_reg_local_set_char,
+    nr_reg_local_set_uchar,
+    nr_reg_local_set_int2,
+    nr_reg_local_set_uint2,
+    nr_reg_local_set_int4,
+    nr_reg_local_set_uint4,
+    nr_reg_local_set_int8,
+    nr_reg_local_set_uint8,
+    nr_reg_local_set_double,
+    nr_reg_local_set_registry,
+    nr_reg_local_set_bytes,
+    nr_reg_local_set_string,
+    nr_reg_local_del,
+    nr_reg_local_get_child_count,
+    nr_reg_local_get_children,
+    nr_reg_local_fin,
+    nr_reg_local_dump
+};
+
+static nr_registry_module nr_reg_local_module = { 0, &nr_reg_local_vtbl };
+
+void *NR_REG_MODE_LOCAL = &nr_reg_local_module;
+
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/registry/registry_vtbl.h b/dom/media/webrtc/transport/third_party/nrappkit/src/registry/registry_vtbl.h
new file mode 100644
index 0000000000..0f75a03b95
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/registry/registry_vtbl.h
@@ -0,0 +1,96 @@
+/*
+ *
+ *    registry_vtbl.h
+ *
+ *    $Source: /Users/ekr/tmp/nrappkit-dump/nrappkit/src/registry/registry_vtbl.h,v $
+ *    $Revision: 1.2 $
+ *    $Date: 2006/08/16 19:39:14 $
+ *
+ *
+ *    Copyright (C) 2006, Network Resonance, Inc.
+ *    All Rights Reserved
+ *
+ *    Redistribution and use in source and binary forms, with or without
+ *    modification, are permitted provided that the following conditions
+ *    are met:
+ *
+ *    1. Redistributions of source code must retain the above copyright
+ *       notice, this list of conditions and the following disclaimer.
+ *    2. Redistributions in binary form must reproduce the above copyright
+ *       notice, this list of conditions and the following disclaimer in the
+ *       documentation and/or other materials provided with the distribution.
+ *    3. Neither the name of Network Resonance, Inc. nor the name of any
+ *       contributors to this software may be used to endorse or promote
+ *       products derived from this software without specific prior written
+ *       permission.
+ *
+ *    THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+ *    AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ *    IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ *    ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+ *    LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ *    CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ *    SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+ *    INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+ *    CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ *    ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+ *    POSSIBILITY OF SUCH DAMAGE.
+ *
+ *
+ */
+
+#ifndef __REGISTRY_VTBL_H__
+#define __REGISTRY_VTBL_H__
+
+typedef struct nr_registry_module_ nr_registry_module;
+
+typedef struct nr_registry_module_vtbl_ {
+    int (*init)(nr_registry_module*);
+
+    int (*get_char)(NR_registry name, char *out);
+    int (*get_uchar)(NR_registry name, UCHAR *out);
+    int (*get_int2)(NR_registry name, INT2 *out);
+    int (*get_uint2)(NR_registry name, UINT2 *out);
+    int (*get_int4)(NR_registry name, INT4 *out);
+    int (*get_uint4)(NR_registry name, UINT4 *out);
+    int (*get_int8)(NR_registry name, INT8 *out);
+    int (*get_uint8)(NR_registry name, UINT8 *out);
+    int (*get_double)(NR_registry name, double *out);
+    int (*get_registry)(NR_registry name, NR_registry out);
+
+    int (*get_bytes)(NR_registry name, UCHAR *out, size_t size, size_t *length);
+    int (*get_string)(NR_registry name, char *out, size_t size);
+    int (*get_length)(NR_registry name, size_t *length);
+    int (*get_type)(NR_registry name, NR_registry_type type);
+
+    int (*set_char)(NR_registry name, char data);
+    int (*set_uchar)(NR_registry name, UCHAR data);
+    int (*set_int2)(NR_registry name, INT2 data);
+    int (*set_uint2)(NR_registry name, UINT2 data);
+    int (*set_int4)(NR_registry name, INT4 data);
+    int (*set_uint4)(NR_registry name, UINT4 data);
+    int (*set_int8)(NR_registry name, INT8 data);
+    int (*set_uint8)(NR_registry name, UINT8 data);
+    int (*set_double)(NR_registry name, double data);
+    int (*set_registry)(NR_registry name);
+
+    int (*set_bytes)(NR_registry name, UCHAR *data, size_t length);
+    int (*set_string)(NR_registry name, char *data);
+
+    int (*del)(NR_registry name);
+
+    int (*get_child_count)(NR_registry parent, size_t *count);
+    int (*get_children)(NR_registry parent, NR_registry *data, size_t size, size_t *length);
+
+    int (*fin)(NR_registry name);
+
+    int (*dump)(int sorted);
+} nr_registry_module_vtbl;
+
+struct nr_registry_module_ {
+    void *handle;
+    nr_registry_module_vtbl *vtbl;
+};
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/registry/registrycb.c b/dom/media/webrtc/transport/third_party/nrappkit/src/registry/registrycb.c
new file mode 100644
index 0000000000..4b326a1ee2
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/registry/registrycb.c
@@ -0,0 +1,440 @@
+/*
+ *
+ *    registrycb.c
+ *
+ *    $Source: /Users/ekr/tmp/nrappkit-dump/nrappkit/src/registry/registrycb.c,v $
+ *    $Revision: 1.3 $
+ *    $Date: 2007/06/26 22:37:51 $
+ *
+ *    Callback-related functions
+ *
+ *
+ *    Copyright (C) 2005, Network Resonance, Inc.
+ *    Copyright (C) 2006, Network Resonance, Inc.
+ *    All Rights Reserved
+ *
+ *    Redistribution and use in source and binary forms, with or without
+ *    modification, are permitted provided that the following conditions
+ *    are met:
+ *
+ *    1. Redistributions of source code must retain the above copyright
+ *       notice, this list of conditions and the following disclaimer.
+ *    2. Redistributions in binary form must reproduce the above copyright
+ *       notice, this list of conditions and the following disclaimer in the
+ *       documentation and/or other materials provided with the distribution.
+ *    3. Neither the name of Network Resonance, Inc. nor the name of any
+ *       contributors to this software may be used to endorse or promote
+ *       products derived from this software without specific prior written
+ *       permission.
+ *
+ *    THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+ *    AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ *    IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ *    ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+ *    LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ *    CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ *    SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+ *    INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+ *    CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ *    ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+ *    POSSIBILITY OF SUCH DAMAGE.
+ *
+ *
+ */
+
+#include <assert.h>
+#include <string.h>
+#include "registry.h"
+#include "registry_int.h"
+#include "r_assoc.h"
+#include "r_errors.h"
+#include "nr_common.h"
+#include "r_log.h"
+#include "r_macros.h"
+
+static char CB_ACTIONS[] = { NR_REG_CB_ACTION_ADD,
+                             NR_REG_CB_ACTION_DELETE,
+                             NR_REG_CB_ACTION_CHANGE,
+                             NR_REG_CB_ACTION_FINAL };
+
+typedef struct nr_reg_cb_info_ {
+     char            action;
+     void          (*cb)(void *cb_arg, char action, NR_registry name);
+     void           *cb_arg;
+     NR_registry     name;
+} nr_reg_cb_info;
+
+/* callbacks that are registered, a mapping from names like "foo.bar.baz"
+ * to an r_assoc which contains possibly several nr_reg_cb_info*'s */
+static r_assoc     *nr_registry_callbacks = 0;
+
+//static size_t SIZEOF_CB_ID = (sizeof(void (*)()) + 1);
+#define SIZEOF_CB_ID  (sizeof(void (*)()) + 1)
+
+static int nr_reg_validate_action(char action);
+static int nr_reg_assoc_destroy(void *ptr);
+static int compute_cb_id(void *cb, char action, unsigned char cb_id[SIZEOF_CB_ID]);
+static int nr_reg_info_free(void *ptr);
+static int nr_reg_raise_event_recurse(char *name, char *tmp, int action);
+static int nr_reg_register_callback(NR_registry name, char action, void (*cb)(void *cb_arg, char action, NR_registry name), void *cb_arg);
+static int nr_reg_unregister_callback(char *name, char action, void (*cb)(void *cb_arg, char action, NR_registry name));
+
+int
+nr_reg_cb_init()
+{
+    int r, _status;
+
+    if (nr_registry_callbacks == 0) {
+        if ((r=r_assoc_create(&nr_registry_callbacks, r_assoc_crc32_hash_compute, 12)))
+            ABORT(r);
+    }
+
+    _status=0;
+  abort:
+    if (_status) {
+       r_log(NR_LOG_REGISTRY, LOG_DEBUG, "Couldn't init notifications: %s", nr_strerror(_status));
+    }
+    return(_status);
+}
+
+int
+nr_reg_validate_action(char action)
+{
+    int _status;
+    size_t i;
+
+    for (i = 0; i < sizeof(CB_ACTIONS); ++i) {
+        if (action == CB_ACTIONS[i])
+            return 0;
+    }
+    ABORT(R_BAD_ARGS);
+
+    _status=0;
+  abort:
+    return(_status);
+}
+
+int
+nr_reg_register_callback(NR_registry name, char action, void (*cb)(void *cb_arg, char action, NR_registry name), void *cb_arg)
+{
+    int r, _status;
+    r_assoc *assoc;
+    int create_assoc = 0;
+    nr_reg_cb_info *info;
+    int create_info = 0;
+    unsigned char cb_id[SIZEOF_CB_ID];
+
+    if (name == 0 || cb == 0)
+      ABORT(R_BAD_ARGS);
+
+    if (nr_registry_callbacks == 0)
+      ABORT(R_FAILED);
+
+    if ((r=nr_reg_is_valid(name)))
+      ABORT(r);
+
+    if ((r=nr_reg_validate_action(action)))
+      ABORT(r);
+
+    if ((r=r_assoc_fetch(nr_registry_callbacks, name, strlen(name)+1, (void*)&assoc))) {
+      if (r == R_NOT_FOUND)
+        create_assoc = 1;
+      else
+        ABORT(r);
+    }
+
+    if (create_assoc) {
+      if ((r=r_assoc_create(&assoc, r_assoc_crc32_hash_compute, 5)))
+        ABORT(r);
+
+      if ((r=r_assoc_insert(nr_registry_callbacks, name, strlen(name)+1, assoc, 0, nr_reg_assoc_destroy, R_ASSOC_NEW)))
+        ABORT(r);
+    }
+
+    if ((r=compute_cb_id(cb, action, cb_id)))
+      ABORT(r);
+
+    if ((r=r_assoc_fetch(assoc, (char*)cb_id, SIZEOF_CB_ID, (void*)&info))) {
+      if (r == R_NOT_FOUND)
+        create_info = 1;
+      else
+        ABORT(r);
+    }
+
+    if (create_info) {
+      if (!(info=(void*)RCALLOC(sizeof(*info))))
+        ABORT(R_NO_MEMORY);
+    }
+
+    strncpy(info->name, name, sizeof(info->name));
+    info->action = action;
+    info->cb = cb;
+    info->cb_arg = cb_arg;
+
+    if (create_info) {
+      if ((r=r_assoc_insert(assoc, (char*)cb_id, SIZEOF_CB_ID, info, 0, nr_reg_info_free, R_ASSOC_NEW)))
+        ABORT(r);
+    }
+
+    _status=0;
+  abort:
+    r_log(NR_LOG_REGISTRY, LOG_DEBUG, "register callback %p on '%s' for '%s' %s", cb, name, nr_reg_action_name(action), (_status ? "FAILED" : "succeeded"));
+
+    if (_status) {
+      if (create_info && info) RFREE(info);
+      if (create_assoc && assoc) nr_reg_assoc_destroy(&assoc);
+    }
+    return(_status);
+}
+
+int
+nr_reg_unregister_callback(char *name, char action, void (*cb)(void *cb_arg, char action, NR_registry name))
+{
+    int r, _status;
+    r_assoc *assoc;
+    int size;
+    unsigned char cb_id[SIZEOF_CB_ID];
+
+    if (name == 0 || cb == 0)
+      ABORT(R_BAD_ARGS);
+
+    if (nr_registry_callbacks == 0)
+      ABORT(R_FAILED);
+
+    if ((r=nr_reg_is_valid(name)))
+      ABORT(r);
+
+    if ((r=nr_reg_validate_action(action)))
+      ABORT(r);
+
+    if ((r=r_assoc_fetch(nr_registry_callbacks, name, strlen(name)+1, (void*)&assoc))) {
+      if (r != R_NOT_FOUND)
+        ABORT(r);
+    }
+    else {
+      if ((r=compute_cb_id(cb, action, cb_id)))
+        ABORT(r);
+
+      if ((r=r_assoc_delete(assoc, (char*)cb_id, SIZEOF_CB_ID))) {
+        if (r != R_NOT_FOUND)
+          ABORT(r);
+      }
+
+      if ((r=r_assoc_num_elements(assoc, &size)))
+        ABORT(r);
+
+      if (size == 0) {
+        if ((r=r_assoc_delete(nr_registry_callbacks, name, strlen(name)+1)))
+          ABORT(r);
+      }
+    }
+
+    _status=0;
+  abort:
+    r_log(NR_LOG_REGISTRY, LOG_DEBUG, "unregister callback %p on '%s' for '%s' %s", cb, name, nr_reg_action_name(action), (_status ? "FAILED" : "succeeded"));
+
+    return(_status);
+}
+
+int
+compute_cb_id(void *cb, char action, unsigned char cb_id[SIZEOF_CB_ID])
+{
+   /* callbacks are identified by the pointer to the cb function plus
+    * the action being watched */
+   assert(sizeof(cb) == sizeof(void (*)()));
+   assert(sizeof(cb) == (SIZEOF_CB_ID - 1));
+
+   memcpy(cb_id, &(cb), sizeof(cb));
+   cb_id[SIZEOF_CB_ID-1] = action;
+
+   return 0;
+}
+
+char *
+nr_reg_action_name(int action)
+{
+    char *name = "*Unknown*";
+
+    switch (action) {
+    case NR_REG_CB_ACTION_ADD:     name = "add";     break;
+    case NR_REG_CB_ACTION_DELETE:  name = "delete";  break;
+    case NR_REG_CB_ACTION_CHANGE:  name = "change";  break;
+    case NR_REG_CB_ACTION_FINAL:   name = "final";  break;
+    }
+
+    return name;
+}
+
+int
+nr_reg_assoc_destroy(void *ptr)
+{
+  return r_assoc_destroy((r_assoc**)&ptr);
+}
+
+int
+nr_reg_info_free(void *ptr)
+{
+  RFREE(ptr);
+  return 0;
+}
+
+/* call with tmp=0 */
+int
+nr_reg_raise_event_recurse(char *name, char *tmp, int action)
+{
+    int r, _status;
+    r_assoc *assoc;
+    nr_reg_cb_info *info;
+    r_assoc_iterator iter;
+    char *key;
+    int keyl;
+    char *c;
+    int free_tmp = 0;
+    int count;
+
+    if (tmp == 0) {
+      if (!(tmp = (char*)r_strdup(name)))
+        ABORT(R_NO_MEMORY);
+      free_tmp = 1;
+    }
+
+    if ((r=r_assoc_fetch(nr_registry_callbacks, tmp, strlen(tmp)+1, (void*)&assoc))) {
+      if (r != R_NOT_FOUND)
+        ABORT(r);
+
+      r_log(NR_LOG_REGISTRY, LOG_DEBUG, "No callbacks found on '%s'", tmp);
+    }
+    else {
+      if (!r_assoc_num_elements(assoc, &count)) {
+          r_log(NR_LOG_REGISTRY, LOG_DEBUG, "%d callback%s found on '%s'",
+                count, ((count == 1) ? "" : "s"), tmp);
+      }
+
+      if ((r=r_assoc_init_iter(assoc, &iter)))
+          ABORT(r);
+
+      for (;;) {
+        if ((r=r_assoc_iter(&iter, (void*)&key, &keyl, (void*)&info))) {
+          if (r == R_EOD)
+            break;
+          else
+            ABORT(r);
+        }
+
+        if (info->action == action) {
+          r_log(NR_LOG_REGISTRY, LOG_DEBUG,
+                "Invoking callback %p for '%s'",
+                info->cb,
+                nr_reg_action_name(info->action));
+
+          (void)info->cb(info->cb_arg, action, name);
+        }
+        else {
+          r_log(NR_LOG_REGISTRY, LOG_DEBUG,
+                "Skipping callback %p for '%s'",
+                info->cb,
+                nr_reg_action_name(info->action));
+        }
+      }
+    }
+
+    if (strlen(tmp) > 0) {
+        c = strrchr(tmp, '.');
+        if (c != 0)
+          *c = '\0';
+        else
+          tmp[0] = '\0';
+
+        if ((r=nr_reg_raise_event_recurse(name, tmp, action)))
+          ABORT(r);
+    }
+
+    _status=0;
+  abort:
+    if (free_tmp && tmp != 0) RFREE(tmp);
+    return(_status);
+}
+
+
+/* NON-STATIC METHODS */
+
+int
+nr_reg_raise_event(NR_registry name, int action)
+{
+    int r, _status;
+    int count;
+    char *event = nr_reg_action_name(action);
+
+    r_log(NR_LOG_REGISTRY, LOG_DEBUG, "raising event '%s' on '%s'", event, name);
+
+    if (name == 0)
+      ABORT(R_BAD_ARGS);
+
+    if ((r=nr_reg_validate_action(action)))
+      ABORT(r);
+
+    if ((r=r_assoc_num_elements(nr_registry_callbacks, &count)))
+      ABORT(r);
+
+    if (count > 0) {
+      if ((r=nr_reg_raise_event_recurse(name, 0, action)))
+        ABORT(r);
+    }
+    else {
+      r_log(NR_LOG_REGISTRY, LOG_DEBUG, "No callbacks found");
+      return 0;
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+}
+
+
+/* PUBLIC METHODS */
+
+int
+NR_reg_register_callback(NR_registry name, char action, void (*cb)(void *cb_arg, char action, NR_registry name), void *cb_arg)
+{
+    int r, _status;
+    size_t i;
+
+    for (i = 0; i < sizeof(CB_ACTIONS); ++i) {
+        if (action & CB_ACTIONS[i]) {
+            if ((r=nr_reg_register_callback(name, CB_ACTIONS[i], cb, cb_arg)))
+                ABORT(r);
+
+            action &= ~(CB_ACTIONS[i]);
+        }
+    }
+
+    if (action)
+        ABORT(R_BAD_ARGS);
+
+    _status=0;
+  abort:
+    return(_status);
+}
+
+int
+NR_reg_unregister_callback(NR_registry name, char action, void (*cb)(void *cb_arg, char action, NR_registry name))
+{
+    int r, _status;
+    size_t i;
+
+    for (i = 0; i < sizeof(CB_ACTIONS); ++i) {
+        if (action & CB_ACTIONS[i]) {
+            if ((r=nr_reg_unregister_callback(name, CB_ACTIONS[i], cb)))
+                ABORT(r);
+
+            action &= ~(CB_ACTIONS[i]);
+        }
+    }
+
+    if (action)
+        ABORT(R_BAD_ARGS);
+
+    _status=0;
+  abort:
+    return(_status);
+}
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/share/nr_api.h b/dom/media/webrtc/transport/third_party/nrappkit/src/share/nr_api.h
new file mode 100644
index 0000000000..ce6055f0d9
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/share/nr_api.h
@@ -0,0 +1,51 @@
+/**
+   nr_pce.h
+
+
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   ekr@networkresonance.com  Wed Jul 19 13:18:39 2006
+ */
+
+
+#ifndef _nr_pce_h
+#define _nr_pce_h
+
+#include <sys/queue.h>
+#include <csi_platform.h>
+#include <r_common.h>
+#include <r_log.h>
+#include <nrstats.h>
+#include <nr_plugin.h>
+#include <async_wait.h>
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/share/nr_common.h b/dom/media/webrtc/transport/third_party/nrappkit/src/share/nr_common.h
new file mode 100644
index 0000000000..ac3ff44004
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/share/nr_common.h
@@ -0,0 +1,108 @@
+/**
+   nr_common.h
+
+
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+ */
+
+
+#ifndef _nr_common_h
+#define _nr_common_h
+
+#include <csi_platform.h>
+
+#ifdef USE_MPATROL
+#define USEDEBUG 1
+#include <mpatrol.h>
+#endif
+
+#ifdef USE_DMALLOC
+#include <dmalloc.h>
+#endif
+
+#include <string.h>
+#include <time.h>
+
+#ifdef WIN32
+#include <winsock2.h>
+#include <errno.h>
+#else
+#include <sys/errno.h>
+#endif
+
+#ifdef HAVE_SYS_TIME_H
+#include <sys/time.h>
+#endif
+
+#include <sys/types.h>
+#include <sys/queue.h>
+#include <r_log.h>
+
+extern int NR_LOG_REASSD;
+
+#include "registry.h"
+#include "nrstats.h"
+
+typedef struct nr_captured_packet_ {
+     UCHAR cap_interface;       /* 1 for primary, 2 for secondary */
+     struct timeval ts;     /* The time this packet was captured */
+     UINT4 len;             /* The length of the packet */
+     UINT8 packet_number;   /* The listener's packet index */
+} nr_captured_packet;
+
+#ifndef NR_ROOT_PATH
+#define NR_ROOT_PATH "/usr/local/ctc/"
+#endif
+
+#define NR_ARCHIVE_DIR NR_ROOT_PATH  "archive/"
+#define NR_TEMP_DIR NR_ROOT_PATH  "tmp/"
+#define NR_ARCHIVE_STATEFILE NR_ROOT_PATH  "archive/state"
+#define NR_CAPTURED_PID_FILENAME  NR_ROOT_PATH "captured.pid"
+#define NR_REASSD_PID_FILENAME  NR_ROOT_PATH "reassd.pid"
+#define NR_MODE_FILENAME  NR_ROOT_PATH "mode.txt"
+
+char *nr_revision_number(void);
+
+
+
+
+/* Memory buckets for CTC memory types */
+#define NR_MEM_TCP       1
+#define NR_MEM_HTTP      2
+#define NR_MEM_DELIVERY  3
+#define NR_MEM_OUT_HM    4
+#define NR_MEM_OUT_SSL   5
+#define NR_MEM_SSL       7
+#define NR_MEM_COMMON    8
+#define NR_MEM_CODEC     9
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/share/nr_reg_keys.h b/dom/media/webrtc/transport/third_party/nrappkit/src/share/nr_reg_keys.h
new file mode 100644
index 0000000000..b051f51d4a
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/share/nr_reg_keys.h
@@ -0,0 +1,167 @@
+/*
+ *
+ *    nr_reg_keys.h
+ *
+ *    $Source: /Users/ekr/tmp/nrappkit-dump/nrappkit/src/share/nr_reg_keys.h,v $
+ *    $Revision: 1.3 $
+ *    $Date: 2008/01/29 00:34:00 $
+ *
+ *
+ *    Copyright (C) 2006, Network Resonance, Inc.
+ *    All Rights Reserved
+ *
+ *    Redistribution and use in source and binary forms, with or without
+ *    modification, are permitted provided that the following conditions
+ *    are met:
+ *
+ *    1. Redistributions of source code must retain the above copyright
+ *       notice, this list of conditions and the following disclaimer.
+ *    2. Redistributions in binary form must reproduce the above copyright
+ *       notice, this list of conditions and the following disclaimer in the
+ *       documentation and/or other materials provided with the distribution.
+ *    3. Neither the name of Network Resonance, Inc. nor the name of any
+ *       contributors to this software may be used to endorse or promote
+ *       products derived from this software without specific prior written
+ *       permission.
+ *
+ *    THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+ *    AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ *    IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ *    ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+ *    LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ *    CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ *    SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+ *    INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+ *    CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ *    ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+ *    POSSIBILITY OF SUCH DAMAGE.
+ *
+ *
+ */
+
+#ifndef __NR_REG_KEYS_H__
+#define __NR_REG_KEYS_H__
+
+#include <stdio.h>
+
+#define NR_REG_NAME_LENGTH_MIN                                        1
+#define NR_REG_NAME_LENGTH_MAX                                        32
+
+#define NR_REG_HOSTNAME            "hostname"
+#define NR_REG_ADDRESS             "address"
+#define NR_REG_NETMASKSIZE         "netmasksize"
+#define NR_REG_ADDRESS_NETMASKSIZE "address/netmasksize"   /* use only in clic files */
+#define NR_REG_PORT                "port"
+
+#define NR_REG_LOGGING_SYSLOG_ENABLED                                 "logging.syslog.enabled"
+#define NR_REG_LOGGING_SYSLOG_SERVERS                                 "logging.syslog.servers"
+#define NR_REG_LOGGING_SYSLOG_FACILITY                                "logging.syslog.facility"
+#define NR_REG_LOGGING_SYSLOG_LEVEL                                   "logging.syslog.level"
+
+#define NR_REG_CAPTURED_DAEMONS                                       "captured.daemons"
+
+#define NR_REG_LISTEND_ENABLED                                        "listend.enabled"
+
+#define NR_REG_LISTEND_MAX_INPUT_BUFFER_SIZE                          "listend.max_input_buffer_size"
+#define NR_REG_LISTEND_MAX_INPUT_BUFFER_SIZE_MIN                      1               // 1 byte?
+#define NR_REG_LISTEND_MAX_INPUT_BUFFER_SIZE_MAX                      (10ULL*1024*1024*1024)     // 10 GB
+
+#define NR_REG_LISTEND_INTERFACE                                      "listend.interface"
+#define NR_REG_LISTEND_INTERFACE_PRIMARY                              "listend.interface.primary"
+#define NR_REG_LISTEND_INTERFACE_SECONDARY                            "listend.interface.secondary"
+#define NR_REG_LISTEND_LISTEN_ON_BOTH_INTERFACES                      "listend.interface.listen_on_both_interfaces"
+#define NR_REG_LISTEND_ENABLE_VLAN                                    "listend.enable_vlan"
+
+#define NR_REG_LISTEND_LISTEN_TO                                      "listend.listen_to"
+#define NR_REG_LISTEND_IGNORE                                         "listend.ignore"
+
+#define NR_REG_LISTEND_PORT_MIN                                       0
+#define NR_REG_LISTEND_PORT_MAX                                       65535
+
+#define NR_REG_REASSD_MAX_MEMORY_CONSUMPTION                          "reassd.max_memory_consumption"
+#define NR_REG_REASSD_MAX_MEMORY_CONSUMPTION_MIN                      (10*1024)    // 100 KB
+#define NR_REG_REASSD_MAX_MEMORY_CONSUMPTION_MAX                      (10ULL*1024*1024*1024)     // 10 GB
+
+#define NR_REG_REASSD_DECODER_TCP_IGNORE_CHECKSUMS                    "reassd.decoder.tcp.ignore_checksums"
+
+#define NR_REG_REASSD_DECODER_TCP_MAX_CONNECTIONS_IN_SYN_STATE        "reassd.decoder.tcp.max_connections_in_syn_state"
+#define NR_REG_REASSD_DECODER_TCP_MAX_CONNECTIONS_IN_SYN_STATE_MIN    1
+#define NR_REG_REASSD_DECODER_TCP_MAX_CONNECTIONS_IN_SYN_STATE_MAX    500000
+
+#define NR_REG_REASSD_DECODER_TCP_MAX_SIMULTANEOUS_CONNECTIONS        "reassd.decoder.tcp.max_simultaneous_connections"
+#define NR_REG_REASSD_DECODER_TCP_MAX_SIMULTANEOUS_CONNECTIONS_MIN    1
+#define NR_REG_REASSD_DECODER_TCP_MAX_SIMULTANEOUS_CONNECTIONS_MAX    1000000  // 1 million
+
+#define NR_REG_REASSD_DECODER_SSL_MAX_SESSION_CACHE_SIZE              "reassd.decoder.ssl.max_session_cache_size"
+#define NR_REG_REASSD_DECODER_SSL_MAX_SESSION_CACHE_SIZE_MIN          0
+#define NR_REG_REASSD_DECODER_SSL_MAX_SESSION_CACHE_SIZE_MAX          (1ULL*1024*1024*1024)   // 1GB
+
+#define NR_REG_REASSD_DECODER_SSL_REVEAL_LOCAL_KEYS                   "reassd.decoder.ssl.reveal.local.keys"
+
+#define NR_REG_REASSD_DECODER_HTTP_HANGING_RESPONSE_TIMEOUT           "reassd.decoder.http.hanging_response_timeout"
+#define NR_REG_REASSD_DECODER_HTTP_HANGING_RESPONSE_TIMEOUT_MIN       1
+#define NR_REG_REASSD_DECODER_HTTP_HANGING_RESPONSE_TIMEOUT_MAX       1023
+
+#define NR_REG_REASSD_DECODER_HTTP_HANGING_TRANSMISSION_TIMEOUT       "reassd.decoder.http.hanging_transmission_timeout"
+#define NR_REG_REASSD_DECODER_HTTP_HANGING_TRANSMISSION_TIMEOUT_MIN   1
+#define NR_REG_REASSD_DECODER_HTTP_HANGING_TRANSMISSION_TIMEOUT_MAX   1023
+
+#define NR_REG_REASSD_DECODER_HTTP_MAX_HTTP_MESSAGE_SIZE              "reassd.decoder.http.max_http_message_size"
+#define NR_REG_REASSD_DECODER_HTTP_MAX_HTTP_MESSAGE_SIZE_MIN          0
+#define NR_REG_REASSD_DECODER_HTTP_MAX_HTTP_MESSAGE_SIZE_MAX          (10ULL*1024*1024*1024)
+
+/* PCE-only: */
+#define NR_REG_LISTEND_ARCHIIVE                                       "listend.archive"
+#define NR_REG_LISTEND_ARCHIVE_MAX_SIZE                               "listend.archive.max_size"
+#define NR_REG_LISTEND_ARCHIVE_MAX_SIZE_MIN                           0
+#define NR_REG_LISTEND_ARCHIVE_MAX_SIZE_MAX                           (10ULL*1024*1024*1024*1024)  // 10 TB
+
+#define NR_REG_LISTEND_ARCHIVE_RECORDING_ENABLED                      "listend.archive.recording_enabled"
+
+#define NR_REG_REASSD_DECODER_NET_DELIVER_BATCH_INTERVAL              "reassd.decoder.net_deliver.batch_interval"
+#define NR_REG_REASSD_DECODER_NET_DELIVER_BATCH_INTERVAL_MIN          0
+#define NR_REG_REASSD_DECODER_NET_DELIVER_BATCH_INTERVAL_MAX          1023
+
+#define NR_REG_REASSD_DECODER_NET_DELIVER_MAX_QUEUE_DEPTH             "reassd.decoder.net_deliver.max_queue_depth"
+#define NR_REG_REASSD_DECODER_NET_DELIVER_MAX_QUEUE_DEPTH_MIN         0
+#define NR_REG_REASSD_DECODER_NET_DELIVER_MAX_QUEUE_DEPTH_MAX         (1ULL*1024*1024*1024)     // 1 GB
+
+#define NR_REG_REASSD_DECODER_NET_DELIVER_MY_KEY_CERTIFICATE          "reassd.decoder.net_deliver.my_key.certificate"
+#define NR_REG_REASSD_DECODER_NET_DELIVER_MY_KEY_PRIVATE_KEY          "reassd.decoder.net_deliver.my_key.private_key"
+#define NR_REG_REASSD_DECODER_NET_DELIVER_PEER                        "reassd.decoder.net_deliver.peer"
+#define NR_REG_REASSD_DECODER_NET_DELIVER_PEER_PORT_MIN               0
+#define NR_REG_REASSD_DECODER_NET_DELIVER_PEER_PORT_MAX               65535
+
+#define NR_REG_REASSD_DECODER_NET_DELIVER_POLLING_INTERVAL            "reassd.decoder.net_deliver.polling_interval"
+#define NR_REG_REASSD_DECODER_NET_DELIVER_POLLING_INTERVAL_MIN        1
+#define NR_REG_REASSD_DECODER_NET_DELIVER_POLLING_INTERVAL_MAX        1023
+
+#define NR_REG_REASSD_DECODER_NET_DELIVER_STATELESS                   "reassd.decoder.net_deliver.stateless"
+#define NR_REG_REASSD_DECODER_NET_DELIVER_WATCHDOG_TIMER              "reassd.decoder.net_deliver.watchdog_timer"
+#define NR_REG_REASSD_DECODER_NET_DELIVER_WATCHDOG_TIMER_MIN          0
+#define NR_REG_REASSD_DECODER_NET_DELIVER_WATCHDOG_TIMER_MAX          666
+
+/* ASA-only: */
+#define NR_REG_MIGRATE_ENABLED                                        "migrate.enabled"
+
+#define NR_REG_MIGRATE_INACTIVITY_TIMEOUT                             "migrate.inactivity_timeout"
+#define NR_REG_MIGRATE_INACTIVITY_TIMEOUT_MIN                         0
+#define NR_REG_MIGRATE_INACTIVITY_TIMEOUT_MAX                         1023
+
+#define NR_REG_MIGRATE_MIN_LOCAL_SIZE                                 "migrate.min_local_size"
+#define NR_REG_MIGRATE_MIN_OVERLAP_SIZE                               "migrate.min_overlap_size"
+#define NR_REG_MIGRATE_RETRANSMIT_FREQUENCY                           "migrate.retransmit_frequency"
+#define NR_REG_MIGRATE_RETRIES                                        "migrate.retries"
+#define NR_MIGRATE_LOCATION_NUMBER_MIN                                0
+#define NR_MIGRATE_LOCATION_NUMBER_MAX                                255
+
+#define NR_REG_REVELATION_ENABLE                                      "revelation.enabled"
+#define NR_REG_REVELATION_MAX_PER_HOUR                                "revelation.max_per_hour"
+#define NR_REG_REVELATION_MAX_PER_HOUR_PER_PORTAL                     "revelation.max_per_hour_per_portal"
+
+/* Appliance-only: */
+#define NR_REG_SNMP_ENABLED                                           "snmp.enabled"
+#define NR_REG_CLOCK_TIMEZONE                                         "clock.timezone"
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/stats/nrstats.h b/dom/media/webrtc/transport/third_party/nrappkit/src/stats/nrstats.h
new file mode 100644
index 0000000000..0b6c2bc3c3
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/stats/nrstats.h
@@ -0,0 +1,118 @@
+/*
+ *
+ *    nrstats.h
+ *
+ *    $Source: /Users/ekr/tmp/nrappkit-dump/nrappkit/src/stats/nrstats.h,v $
+ *    $Revision: 1.4 $
+ *    $Date: 2007/06/26 22:37:55 $
+ *
+ *    API for keeping and sharing statistics
+ *
+ *
+ *    Copyright (C) 2003-2005, Network Resonance, Inc.
+ *    Copyright (C) 2006, Network Resonance, Inc.
+ *    All Rights Reserved
+ *
+ *    Redistribution and use in source and binary forms, with or without
+ *    modification, are permitted provided that the following conditions
+ *    are met:
+ *
+ *    1. Redistributions of source code must retain the above copyright
+ *       notice, this list of conditions and the following disclaimer.
+ *    2. Redistributions in binary form must reproduce the above copyright
+ *       notice, this list of conditions and the following disclaimer in the
+ *       documentation and/or other materials provided with the distribution.
+ *    3. Neither the name of Network Resonance, Inc. nor the name of any
+ *       contributors to this software may be used to endorse or promote
+ *       products derived from this software without specific prior written
+ *       permission.
+ *
+ *    THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+ *    AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ *    IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ *    ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+ *    LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ *    CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ *    SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+ *    INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+ *    CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ *    ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+ *    POSSIBILITY OF SUCH DAMAGE.
+ *
+ *
+ */
+
+#ifndef __NRSTATS_H__
+#define __NRSTATS_H__
+
+#include <sys/types.h>
+#ifdef WIN32
+#include <time.h>
+#else
+#include <sys/time.h>
+#endif
+#include <r_types.h>
+
+#ifndef CAPTURE_USER
+#define CAPTURE_USER "pcecap"
+#endif
+
+#define NR_MAX_STATS_TYPES         256   /* max number of stats objects */
+#define NR_MAX_STATS_TYPE_NAME     26
+
+typedef struct NR_stats_type_ {
+      char name[NR_MAX_STATS_TYPE_NAME];
+      int (*reset)(void *stats);
+      int (*print)(void *stats, char *stat_namespace, void (*output)(void *handle, const char *fmt, ...), void *handle);
+      int  (*get_lib_name)(char **libname);
+      unsigned int size;
+} NR_stats_type;
+
+typedef struct NR_stats_app_ {
+        time_t               last_counter_reset;
+        time_t               last_restart;
+        UINT8                total_restarts;
+        char                 version[64];
+} NR_stats_app;
+
+extern NR_stats_type *NR_stats_type_app;
+
+/* everything measured in bytes */
+typedef struct NR_stats_memory_ {
+        UINT8                current_size;
+        UINT8                max_size;
+        UINT8                in_use;
+        UINT8                in_use_max;
+} NR_stats_memory;
+
+extern NR_stats_type *NR_stats_type_memory;
+
+/*  all functions below return 0 on success, else they return an
+ *  error code */
+
+/* if errprintf is null, warnings and errors will be sent to /dev/null */
+extern int NR_stats_startup(char *app_name, char *user_name, void (*errprintf)(void *handle, const char *fmt, ...), void *errhandle);
+extern int NR_stats_shutdown(void);
+#define NR_STATS_CREATE  (1<<0)
+extern int NR_stats_get(char *module_name, NR_stats_type *type, int flag, void **stats);
+extern int NR_stats_clear(void *stats);   /* zeroizes */
+extern int NR_stats_reset(void *stats);   /* zeros "speedometers" */
+extern int NR_stats_register(NR_stats_type *type);
+extern int NR_stats_acquire_mutex(void *stats);
+extern int NR_stats_release_mutex(void *stats);
+// TODO: should _get_names take an app_name argument (0==all)????
+extern int NR_stats_get_names(unsigned int *nnames, char ***names);
+extern int NR_stats_get_by_name(char *name, NR_stats_type **type, void **stats);
+extern int NR_stats_get_lib_name(void *stats, char **lib_name);
+extern int NR_stats_rmids(void);
+
+extern char *NR_prefix_to_stats_module(char *prefix);
+
+#define NR_INCREMENT_STAT(stat) do { \
+       stat++; if(stat>stat##_max) stat##_max=stat; \
+     } while (0)
+#define NR_UPDATE_STAT(stat,newval) do { \
+       stat=newval; if(stat>stat##_max) stat##_max=stat; \
+     } while (0)
+
+#endif
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/byteorder.c b/dom/media/webrtc/transport/third_party/nrappkit/src/util/byteorder.c
new file mode 100644
index 0000000000..64689accfa
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/byteorder.c
@@ -0,0 +1,73 @@
+/**
+   byteorder.c
+
+
+   Copyright (C) 2007, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   briank@networkresonance.com  Wed May 16 16:46:00 PDT 2007
+ */
+
+#include "nr_common.h"
+#ifndef WIN32
+#include <arpa/inet.h>
+#endif
+#include "r_types.h"
+#include "byteorder.h"
+
+#define IS_BIG_ENDIAN (htonl(0x1) == 0x1)
+
+#define BYTE(n,i)   (((UCHAR*)&(n))[(i)])
+#define SWAP(n,x,y) tmp=BYTE((n),(x)), \
+                    BYTE((n),(x))=BYTE((n),(y)), \
+                    BYTE((n),(y))=tmp
+
+UINT8
+nr_htonll(UINT8 hostlonglong)
+{
+    UINT8 netlonglong = hostlonglong;
+    UCHAR tmp;
+
+    if (!IS_BIG_ENDIAN) {
+        SWAP(netlonglong, 0, 7);
+        SWAP(netlonglong, 1, 6);
+        SWAP(netlonglong, 2, 5);
+        SWAP(netlonglong, 3, 4);
+    }
+
+    return netlonglong;
+}
+
+UINT8
+nr_ntohll(UINT8 netlonglong)
+{
+    return nr_htonll(netlonglong);
+}
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/byteorder.h b/dom/media/webrtc/transport/third_party/nrappkit/src/util/byteorder.h
new file mode 100644
index 0000000000..8df0589a63
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/byteorder.h
@@ -0,0 +1,47 @@
+/**
+   byteorder.h
+
+
+   Copyright (C) 2007, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   briank@networkresonance.com  Wed May 16 16:46:00 PDT 2007
+ */
+
+
+#ifndef _byteorder_h
+#define _byteorder_h
+
+UINT8 nr_htonll(UINT8 hostlonglong);
+
+UINT8 nr_ntohll(UINT8 netlonglong);
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/hex.c b/dom/media/webrtc/transport/third_party/nrappkit/src/util/hex.c
new file mode 100644
index 0000000000..8212988eda
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/hex.c
@@ -0,0 +1,109 @@
+/**
+   hex.c
+
+
+   Copyright (C) 2001-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   briank@network-resonance.com
+ */
+
+#include <sys/types.h>
+#include <string.h>
+#include <ctype.h>
+#include "r_common.h"
+#include "hex.h"
+#include "r_log.h"
+
+static char bin2hex_map[][3] = { "00", "01", "02", "03", "04", "05", "06", "07", "08", "09", "0a", "0b", "0c", "0d", "0e", "0f", "10", "11", "12", "13", "14", "15", "16", "17", "18", "19", "1a", "1b", "1c", "1d", "1e", "1f", "20", "21", "22", "23", "24", "25", "26", "27", "28", "29", "2a", "2b", "2c", "2d", "2e", "2f", "30", "31", "32", "33", "34", "35", "36", "37", "38", "39", "3a", "3b", "3c", "3d", "3e", "3f", "40", "41", "42", "43", "44", "45", "46", "47", "48", "49", "4a", "4b", "4c", "4d", "4e", "4f", "50", "51", "52", "53", "54", "55", "56", "57", "58", "59", "5a", "5b", "5c", "5d", "5e", "5f", "60", "61", "62", "63", "64", "65", "66", "67", "68", "69", "6a", "6b", "6c", "6d", "6e", "6f", "70", "71", "72", "73", "74", "75", "76", "77", "78", "79", "7a", "7b", "7c", "7d", "7e", "7f", "80", "81", "82", "83", "84", "85", "86", "87", "88", "89", "8a", "8b", "8c", "8d", "8e", "8f", "90", "91", "92", "93", "94", "95", "96", "97", "98", "99", "9a", "9b", "9c", "9d", "9e", "9f", "a0", "a1", "a2", "a3", "a4", "a5", "a6", "a7", "a8", "a9", "aa", "ab", "ac", "ad", "ae", "af", "b0", "b1", "b2", "b3", "b4", "b5", "b6", "b7", "b8", "b9", "ba", "bb", "bc", "bd", "be", "bf", "c0", "c1", "c2", "c3", "c4", "c5", "c6", "c7", "c8", "c9", "ca", "cb", "cc", "cd", "ce", "cf", "d0", "d1", "d2", "d3", "d4", "d5", "d6", "d7", "d8", "d9", "da", "db", "dc", "dd", "de", "df", "e0", "e1", "e2", "e3", "e4", "e5", "e6", "e7", "e8", "e9", "ea", "eb", "ec", "ed", "ee", "ef", "f0", "f1", "f2", "f3", "f4", "f5", "f6", "f7", "f8", "f9", "fa", "fb", "fc", "fd", "fe", "ff" };
+
+static int hex2bin_map[] = { -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, 0 /* '0' */, 1 /* '1' */, 2 /* '2' */, 3 /* '3' */, 4 /* '4' */, 5 /* '5' */, 6 /* '6' */, 7 /* '7' */, 8 /* '8' */, 9 /* '9' */, -1, -1, -1, -1, -1, -1, -1, 10 /* 'A' */, 11 /* 'B' */, 12 /* 'C' */, 13 /* 'D' */, 14 /* 'E' */, 15 /* 'F' */, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, 10 /* 'a' */, 11 /* 'b' */, 12 /* 'c' */, 13 /* 'd' */, 14 /* 'e' */, 15 /* 'f' */, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1 };
+
+int
+nr_nbin2hex(UCHAR *bin, size_t binlen, char hex[], size_t size, size_t *len)
+{
+    int _status;
+    size_t i;
+    size_t hexlen;
+
+    hexlen = 2*binlen;
+    if (size < hexlen)
+        ABORT(R_BAD_ARGS);
+
+    for (i = 0; i < binlen; ++i) {
+        *hex++ = bin2hex_map[bin[i]][0];
+        *hex++ = bin2hex_map[bin[i]][1];
+    }
+
+    if (size >= hexlen+1)
+        *hex = '\0';
+
+    *len = hexlen;
+
+    _status=0;
+  abort:
+    return(_status);
+}
+
+
+int
+nr_nhex2bin(char *hex, size_t hexlen, UCHAR bin[], size_t size, size_t *len)
+{
+    int _status;
+    size_t binlen;
+    int h1;
+    int h2;
+    size_t i;
+
+    if (hexlen % 2)
+        ABORT(R_BAD_ARGS);
+
+    binlen = hexlen/2;
+
+    if (size < binlen)
+        ABORT(R_BAD_ARGS);
+
+    for (i = 0; i < binlen; ++i) {
+        h1 = hex2bin_map[(int)*hex++];
+        h2 = hex2bin_map[(int)*hex++];
+
+        if (h1 == -1 || h2 == -1)
+            ABORT(R_BAD_ARGS);
+
+        bin[i] = (h1 << 4) | h2;
+    }
+
+    *len = binlen;
+
+    _status=0;
+  abort:
+    return(_status);
+}
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/hex.h b/dom/media/webrtc/transport/third_party/nrappkit/src/util/hex.h
new file mode 100644
index 0000000000..8c493b533b
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/hex.h
@@ -0,0 +1,47 @@
+/**
+   hex.h
+
+
+   Copyright (C) 2001-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   briank@network-resonance.com
+ */
+
+
+#ifndef _hex_h
+#define _hex_h
+
+int nr_nbin2hex(UCHAR *bin, size_t binlen, char hex[], size_t size, size_t *len);
+int nr_nhex2bin(char *hex, size_t hexlen, UCHAR bin[], size_t size, size_t *len);
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/assoc.h b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/assoc.h
new file mode 100644
index 0000000000..013e788685
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/assoc.h
@@ -0,0 +1,90 @@
+/**
+   assoc.h
+
+
+   Copyright (C) 2002-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+ */
+
+/**
+   assoc.h
+
+
+   Copyright (C) 1999-2000 RTFM, Inc.
+   All Rights Reserved
+
+   This package is a SSLv3/TLS protocol analyzer written by Eric Rescorla
+   <ekr@rtfm.com> and licensed by RTFM, Inc.
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. All advertising materials mentioning features or use of this software
+      must display the following acknowledgement:
+
+      This product includes software developed by Eric Rescorla for
+      RTFM, Inc.
+
+   4. Neither the name of RTFM, Inc. nor the name of Eric Rescorla may be
+      used to endorse or promote products derived from this
+      software without specific prior written permission.
+
+   THIS SOFTWARE IS PROVIDED BY ERIC RESCORLA AND RTFM, INC. ``AS IS'' AND
+   ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+   FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+   DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+   OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+   LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+   OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY SUCH DAMAGE.
+
+   $Id: assoc.h,v 1.2 2006/08/16 19:39:17 adamcain Exp $
+
+
+   ekr@rtfm.com  Sun Jan 17 17:56:35 1999
+ */
+
+
+#ifndef _assoc_h
+#define _assoc_h
+
+typedef struct assoc_ assoc;
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/debug.c b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/debug.c
new file mode 100644
index 0000000000..fa796217ea
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/debug.c
@@ -0,0 +1,127 @@
+/**
+   debug.c
+
+
+   Copyright (C) 2002-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+ */
+
+/**
+   debug.c
+
+
+   Copyright (C) 1999-2000 RTFM, Inc.
+   All Rights Reserved
+
+   This package is a SSLv3/TLS protocol analyzer written by Eric Rescorla
+   <ekr@rtfm.com> and licensed by RTFM, Inc.
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. All advertising materials mentioning features or use of this software
+      must display the following acknowledgement:
+
+      This product includes software developed by Eric Rescorla for
+      RTFM, Inc.
+
+   4. Neither the name of RTFM, Inc. nor the name of Eric Rescorla may be
+      used to endorse or promote products derived from this
+      software without specific prior written permission.
+
+   THIS SOFTWARE IS PROVIDED BY ERIC RESCORLA AND RTFM, INC. ``AS IS'' AND
+   ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+   FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+   DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+   OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+   LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+   OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY SUCH DAMAGE.
+
+   $Id: debug.c,v 1.3 2007/06/26 22:37:57 adamcain Exp $
+
+
+   ekr@rtfm.com  Wed Jan  6 17:08:58 1999
+ */
+
+#include <stdarg.h>
+#include <stdio.h>
+#include "r_common.h"
+#include "debug.h"
+
+int nr_debug(int class,char *format,...)
+  {
+    va_list ap;
+
+    va_start(ap,format);
+#ifdef WIN32
+    vprintf(format,ap);
+    printf("\n");
+#else
+    vfprintf(stderr,format,ap);
+    fprintf(stderr,"\n");
+#endif
+    return(0);
+  }
+
+int nr_xdump(name,data,len)
+  char *name;
+  UCHAR *data;
+  int len;
+  {
+    int i;
+
+    if(name){
+      printf("%s[%d]=\n",name,len);
+    }
+    for(i=0;i<len;i++){
+
+      if((len>8) && i && !(i%12)){
+        printf("\n");
+      }
+      printf("%.2x ",data[i]&255);
+    }
+    if(i%12)
+      printf("\n");
+    return(0);
+  }
+
+
+
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/debug.h b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/debug.h
new file mode 100644
index 0000000000..34f7b2fb54
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/debug.h
@@ -0,0 +1,94 @@
+/**
+   debug.h
+
+
+   Copyright (C) 2002-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+ */
+
+/**
+   debug.h
+
+
+   Copyright (C) 1999-2000 RTFM, Inc.
+   All Rights Reserved
+
+   This package is a SSLv3/TLS protocol analyzer written by Eric Rescorla
+   <ekr@rtfm.com> and licensed by RTFM, Inc.
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. All advertising materials mentioning features or use of this software
+      must display the following acknowledgement:
+
+      This product includes software developed by Eric Rescorla for
+      RTFM, Inc.
+
+   4. Neither the name of RTFM, Inc. nor the name of Eric Rescorla may be
+      used to endorse or promote products derived from this
+      software without specific prior written permission.
+
+   THIS SOFTWARE IS PROVIDED BY ERIC RESCORLA AND RTFM, INC. ``AS IS'' AND
+   ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+   FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+   DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+   OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+   LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+   OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY SUCH DAMAGE.
+
+   $Id: debug.h,v 1.2 2006/08/16 19:39:17 adamcain Exp $
+
+
+   ekr@rtfm.com  Wed Jan  6 17:13:00 1999
+ */
+
+
+#ifndef _debug_h
+#define _debug_h
+
+// Remove debugging mess from ssldump
+// #define DBG(a)
+
+int nr_debug(int class,char *format,...);
+int nr_xdump(char *name,UCHAR *data, int len);
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_assoc.c b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_assoc.c
new file mode 100644
index 0000000000..25b3827d50
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_assoc.c
@@ -0,0 +1,539 @@
+/**
+   r_assoc.c
+
+
+   Copyright (C) 2002-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+ */
+
+/**
+   r_assoc.c
+
+   This is an associative array implementation, using an open-chained
+   hash bucket technique.
+
+   Note that this implementation permits each data entry to have
+   separate copy constructors and destructors. This currently wastes
+   space, but could be implemented while saving space by using
+   the high order bit of the length value or somesuch.
+
+   The major problem with this code is it's not resizable, though it
+   could be made so.
+
+
+   Copyright (C) 1999-2000 RTFM, Inc.
+   All Rights Reserved
+
+   This package is a SSLv3/TLS protocol analyzer written by Eric Rescorla
+   <ekr@rtfm.com> and licensed by RTFM, Inc.
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. All advertising materials mentioning features or use of this software
+      must display the following acknowledgement:
+
+      This product includes software developed by Eric Rescorla for
+      RTFM, Inc.
+
+   4. Neither the name of RTFM, Inc. nor the name of Eric Rescorla may be
+      used to endorse or promote products derived from this
+      software without specific prior written permission.
+
+   THIS SOFTWARE IS PROVIDED BY ERIC RESCORLA AND RTFM, INC. ``AS IS'' AND
+   ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+   FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+   DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+   OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+   LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+   OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY SUCH DAMAGE.
+
+   $Id: r_assoc.c,v 1.4 2007/06/08 17:41:49 adamcain Exp $
+
+   ekr@rtfm.com  Sun Jan 17 17:57:15 1999
+ */
+
+#include <r_common.h>
+#include <string.h>
+#include "r_assoc.h"
+
+typedef struct r_assoc_el_ {
+     char *key;
+     int key_len;
+     void *data;
+     struct r_assoc_el_ *prev;
+     struct r_assoc_el_ *next;
+     int (*copy)(void **n,void *old);
+     int (*destroy)(void *ptr);
+} r_assoc_el;
+
+struct r_assoc_ {
+     int size;
+     int bits;
+     int (*hash_func)(char *key,int len,int size);
+     r_assoc_el **chains;
+     UINT4 num_elements;
+};
+
+#define DEFAULT_TABLE_BITS 5
+
+static int destroy_assoc_chain(r_assoc_el *chain);
+static int r_assoc_fetch_bucket(r_assoc *assoc,
+  char *key,int len,r_assoc_el **bucketp);
+static int copy_assoc_chain(r_assoc_el **knewp, r_assoc_el *old);
+
+int r_assoc_create(assocp,hash_func,bits)
+  r_assoc **assocp;
+  int (*hash_func)(char *key,int len,int size);
+  int bits;
+  {
+    r_assoc *assoc=0;
+    int _status;
+
+    if(!(assoc=(r_assoc *)RCALLOC(sizeof(r_assoc))))
+      ABORT(R_NO_MEMORY);
+    assoc->size=(1<<bits);
+    assoc->bits=bits;
+    assoc->hash_func=hash_func;
+
+    if(!(assoc->chains=(r_assoc_el **)RCALLOC(sizeof(r_assoc_el *)*
+      assoc->size)))
+      ABORT(R_NO_MEMORY);
+
+    *assocp=assoc;
+
+    _status=0;
+  abort:
+    if(_status){
+      r_assoc_destroy(&assoc);
+    }
+    return(_status);
+  }
+
+int r_assoc_destroy(assocp)
+  r_assoc **assocp;
+  {
+    r_assoc *assoc;
+    int i;
+
+    if(!assocp || !*assocp)
+      return(0);
+
+    assoc=*assocp;
+    for(i=0;i<assoc->size;i++)
+      destroy_assoc_chain(assoc->chains[i]);
+
+    RFREE(assoc->chains);
+    RFREE(*assocp);
+
+    return(0);
+  }
+
+static int destroy_assoc_chain(chain)
+  r_assoc_el *chain;
+  {
+    r_assoc_el *nxt;
+
+    while(chain){
+      nxt=chain->next;
+
+      if(chain->destroy)
+	chain->destroy(chain->data);
+
+      RFREE(chain->key);
+
+      RFREE(chain);
+      chain=nxt;
+    }
+
+    return(0);
+  }
+
+static int copy_assoc_chain(knewp,old)
+  r_assoc_el **knewp;
+  r_assoc_el *old;
+  {
+    r_assoc_el *knew=0,*ptr,*tmp;
+    int r,_status;
+
+    ptr=0; /* Pacify GCC's uninitialized warning.
+              It's not correct */
+    if(!old) {
+      *knewp=0;
+      return(0);
+    }
+    for(;old;old=old->next){
+      if(!(tmp=(r_assoc_el *)RCALLOC(sizeof(r_assoc_el))))
+	ABORT(R_NO_MEMORY);
+
+      if(!knew){
+	knew=tmp;
+	ptr=knew;
+      }
+      else{
+	ptr->next=tmp;
+	tmp->prev=ptr;
+        ptr=tmp;
+      }
+
+      ptr->destroy=old->destroy;
+      ptr->copy=old->copy;
+
+      if(old->copy){
+	if(r=old->copy(&ptr->data,old->data))
+	  ABORT(r);
+      }
+      else
+	ptr->data=old->data;
+
+      if(!(ptr->key=(char *)RMALLOC(old->key_len)))
+	ABORT(R_NO_MEMORY);
+      memcpy(ptr->key,old->key,ptr->key_len=old->key_len);
+    }
+
+    *knewp=knew;
+
+    _status=0;
+  abort:
+    if(_status){
+      destroy_assoc_chain(knew);
+    }
+    return(_status);
+  }
+
+static int r_assoc_fetch_bucket(assoc,key,len,bucketp)
+  r_assoc *assoc;
+  char *key;
+  int len;
+  r_assoc_el **bucketp;
+  {
+    UINT4 hash_value;
+    r_assoc_el *bucket;
+
+    hash_value=assoc->hash_func(key,len,assoc->bits);
+
+    for(bucket=assoc->chains[hash_value];bucket;bucket=bucket->next){
+      if(bucket->key_len == len && !memcmp(bucket->key,key,len)){
+	*bucketp=bucket;
+	return(0);
+      }
+    }
+
+    return(R_NOT_FOUND);
+  }
+
+int r_assoc_fetch(assoc,key,len,datap)
+  r_assoc *assoc;
+  char *key;
+  int len;
+  void **datap;
+  {
+    r_assoc_el *bucket;
+    int r;
+
+    if(r=r_assoc_fetch_bucket(assoc,key,len,&bucket)){
+      if(r!=R_NOT_FOUND)
+	ERETURN(r);
+      return(r);
+    }
+
+    *datap=bucket->data;
+    return(0);
+  }
+
+int r_assoc_insert(assoc,key,len,data,copy,destroy,how)
+  r_assoc *assoc;
+  char *key;
+  int len;
+  void *data;
+  int (*copy)(void **knew,void *old);
+  int (*destroy)(void *ptr);
+  int how;
+  {
+    r_assoc_el *bucket,*new_bucket=0;
+    int r,_status;
+
+    if(r=r_assoc_fetch_bucket(assoc,key,len,&bucket)){
+      /*Note that we compute the hash value twice*/
+      UINT4 hash_value;
+
+      if(r!=R_NOT_FOUND)
+	ABORT(r);
+      hash_value=assoc->hash_func(key,len,assoc->bits);
+
+      if(!(new_bucket=(r_assoc_el *)RCALLOC(sizeof(r_assoc_el))))
+	ABORT(R_NO_MEMORY);
+      if(!(new_bucket->key=(char *)RMALLOC(len)))
+	ABORT(R_NO_MEMORY);
+      memcpy(new_bucket->key,key,len);
+      new_bucket->key_len=len;
+
+      /*Insert at the list head. Is FIFO a good algorithm?*/
+      if(assoc->chains[hash_value])
+        assoc->chains[hash_value]->prev=new_bucket;
+      new_bucket->next=assoc->chains[hash_value];
+      assoc->chains[hash_value]=new_bucket;
+      bucket=new_bucket;
+    }
+    else{
+      if(!(how&R_ASSOC_REPLACE))
+	ABORT(R_ALREADY);
+
+      if(bucket->destroy)
+	bucket->destroy(bucket->data);
+    }
+
+    bucket->data=data;
+    bucket->copy=copy;
+    bucket->destroy=destroy;
+    assoc->num_elements++;
+
+    _status=0;
+  abort:
+    if(_status && new_bucket){
+      RFREE(new_bucket->key);
+      RFREE(new_bucket);
+    }
+    return(_status);
+  }
+
+int r_assoc_delete(assoc,key,len)
+  r_assoc *assoc;
+  char *key;
+  int len;
+  {
+    int r;
+    r_assoc_el *bucket;
+    UINT4 hash_value;
+
+    if(r=r_assoc_fetch_bucket(assoc,key,len,&bucket)){
+      if(r!=R_NOT_FOUND)
+	ERETURN(r);
+      return(r);
+    }
+
+    /* Now remove the element from the hash chain */
+    if(bucket->prev){
+      bucket->prev->next=bucket->next;
+    }
+    else {
+      hash_value=assoc->hash_func(key,len,assoc->bits);
+      assoc->chains[hash_value]=bucket->next;
+    }
+
+    if(bucket->next)
+      bucket->next->prev=bucket->prev;
+
+    /* Remove the data */
+    if(bucket->destroy)
+      bucket->destroy(bucket->data);
+
+    RFREE(bucket->key);
+    RFREE(bucket);
+    assoc->num_elements--;
+
+    return(0);
+  }
+
+int r_assoc_copy(knewp,old)
+  r_assoc **knewp;
+  r_assoc *old;
+  {
+    int r,_status,i;
+    r_assoc *knew;
+
+    if(!(knew=(r_assoc *)RCALLOC(sizeof(r_assoc))))
+      ABORT(R_NO_MEMORY);
+    knew->size=old->size;
+    knew->bits=old->bits;
+    knew->hash_func=old->hash_func;
+
+    if(!(knew->chains=(r_assoc_el **)RCALLOC(sizeof(r_assoc_el)*old->size)))
+      ABORT(R_NO_MEMORY);
+    for(i=0;i<knew->size;i++){
+      if(r=copy_assoc_chain(knew->chains+i,old->chains[i]))
+	ABORT(r);
+    }
+    knew->num_elements=old->num_elements;
+
+    *knewp=knew;
+
+    _status=0;
+  abort:
+    if(_status){
+      r_assoc_destroy(&knew);
+    }
+    return(_status);
+  }
+
+int r_assoc_num_elements(r_assoc *assoc,int *sizep)
+  {
+    *sizep=assoc->num_elements;
+
+    return(0);
+  }
+
+int r_assoc_init_iter(assoc,iter)
+  r_assoc *assoc;
+  r_assoc_iterator *iter;
+  {
+    int i;
+
+    iter->assoc=assoc;
+    iter->prev_chain=-1;
+    iter->prev=0;
+
+    iter->next_chain=assoc->size;
+    iter->next=0;
+
+    for(i=0;i<assoc->size;i++){
+      if(assoc->chains[i]!=0){
+	iter->next_chain=i;
+	iter->next=assoc->chains[i];
+	break;
+      }
+    }
+
+    return(0);
+  }
+
+int r_assoc_iter(iter,key,keyl,val)
+  r_assoc_iterator *iter;
+  void **key;
+  int *keyl;
+  void **val;
+  {
+    int i;
+    r_assoc_el *ret;
+
+    if(!iter->next)
+      return(R_EOD);
+    ret=iter->next;
+
+    *key=ret->key;
+    *keyl=ret->key_len;
+    *val=ret->data;
+
+    /* Now increment */
+    iter->prev_chain=iter->next_chain;
+    iter->prev=iter->next;
+
+    /* More on this chain */
+    if(iter->next->next){
+      iter->next=iter->next->next;
+    }
+    else{
+      iter->next=0;
+
+      /* FInd the next occupied chain*/
+      for(i=iter->next_chain+1;i<iter->assoc->size;i++){
+	if(iter->assoc->chains[i]){
+	  iter->next_chain=i;
+	  iter->next=iter->assoc->chains[i];
+	  break;
+	}
+      }
+    }
+
+    return(0);
+  }
+
+/* Delete the last returned value*/
+int r_assoc_iter_delete(iter)
+  r_assoc_iterator *iter;
+  {
+    /* First unhook it from the list*/
+    if(!iter->prev->prev){
+      /* First element*/
+      iter->assoc->chains[iter->prev_chain]=iter->prev->next;
+    }
+    else{
+      iter->prev->prev->next=iter->prev->next;
+    }
+
+    if(iter->prev->next){
+      iter->prev->next->prev=iter->prev->prev;
+    }
+
+    if (iter->prev->destroy)
+      iter->prev->destroy(iter->prev->data);
+
+    iter->assoc->num_elements--;
+    RFREE(iter->prev->key);
+    RFREE(iter->prev);
+    return(0);
+  }
+
+
+/*This is a hack from AMS. Supposedly, it's pretty good for strings, even
+ though it doesn't take into account all the data*/
+int r_assoc_simple_hash_compute(key,len,bits)
+  char *key;
+  int len;
+  int bits;
+  {
+    UINT4 h=0;
+
+    h=key[0] +(key[len-1] * len);
+
+    h &= (1<<bits) - 1;
+
+    return(h);
+  }
+
+
+int r_crc32(char *data,int len,UINT4 *crcval);
+
+int r_assoc_crc32_hash_compute(data,len,bits)
+  char *data;
+  int len;
+  int bits;
+  {
+    UINT4 res;
+    UINT4 mask;
+
+    /* First compute the CRC value */
+    if(r_crc32(data,len,&res))
+      ERETURN(R_INTERNAL);
+
+    mask=~(0xffffffff<<bits);
+
+    return(res & mask);
+  }
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_assoc.h b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_assoc.h
new file mode 100644
index 0000000000..5311542c9d
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_assoc.h
@@ -0,0 +1,126 @@
+/**
+   r_assoc.h
+
+
+   Copyright (C) 2002-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+ */
+
+/**
+   r_assoc.h
+
+   Associative array code. This code has the advantage that different
+   elements can have different create and destroy operators. Unfortunately,
+   this can waste space.
+
+
+   Copyright (C) 1999-2000 RTFM, Inc.
+   All Rights Reserved
+
+   This package is a SSLv3/TLS protocol analyzer written by Eric Rescorla
+   <ekr@rtfm.com> and licensed by RTFM, Inc.
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. All advertising materials mentioning features or use of this software
+      must display the following acknowledgement:
+
+      This product includes software developed by Eric Rescorla for
+      RTFM, Inc.
+
+   4. Neither the name of RTFM, Inc. nor the name of Eric Rescorla may be
+      used to endorse or promote products derived from this
+      software without specific prior written permission.
+
+   THIS SOFTWARE IS PROVIDED BY ERIC RESCORLA AND RTFM, INC. ``AS IS'' AND
+   ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+   FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+   DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+   OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+   LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+   OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY SUCH DAMAGE.
+
+   $Id: r_assoc.h,v 1.3 2007/06/08 22:16:08 adamcain Exp $
+
+
+   ekr@rtfm.com  Sun Jan 17 17:57:18 1999
+ */
+
+
+#ifndef _r_assoc_h
+#define _r_assoc_h
+
+typedef struct r_assoc_ r_assoc;
+
+int r_assoc_create(r_assoc **assocp,
+                             int (*hash_func)(char *,int,int),
+                             int bits);
+int r_assoc_insert(r_assoc *assoc,char *key,int len,
+  void *value,int (*copy)(void **knew,void *old),
+  int (*destroy)(void *ptr),int how);
+#define R_ASSOC_REPLACE		  0x1
+#define R_ASSOC_NEW		  0x2
+
+int r_assoc_fetch(r_assoc *assoc,char *key, int len, void **value);
+int r_assoc_delete(r_assoc *assoc,char *key, int len);
+
+int r_assoc_copy(r_assoc **knew,r_assoc *old);
+int r_assoc_destroy(r_assoc **assocp);
+int r_assoc_simple_hash_compute(char *key, int len,int bits);
+int r_assoc_crc32_hash_compute(char *key, int len,int bits);
+
+/*We need iterators, but I haven't written them yet*/
+typedef struct r_assoc_iterator_ {
+     r_assoc *assoc;
+     int prev_chain;
+     struct r_assoc_el_ *prev;
+     int next_chain;
+     struct r_assoc_el_ *next;
+} r_assoc_iterator;
+
+int r_assoc_init_iter(r_assoc *assoc,r_assoc_iterator *);
+int r_assoc_iter(r_assoc_iterator *iter,void **key,int *keyl, void **val);
+int r_assoc_iter_delete(r_assoc_iterator *);
+
+int r_assoc_num_elements(r_assoc *assoc,int *sizep);
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_common.h b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_common.h
new file mode 100644
index 0000000000..c11bb3d0fd
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_common.h
@@ -0,0 +1,100 @@
+/**
+   r_common.h
+
+
+   Copyright (C) 2002-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+ */
+
+/**
+   r_common.h
+
+
+   Copyright (C) 1999-2000 RTFM, Inc.
+   All Rights Reserved
+
+   This package is a SSLv3/TLS protocol analyzer written by Eric Rescorla
+   <ekr@rtfm.com> and licensed by RTFM, Inc.
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. All advertising materials mentioning features or use of this software
+      must display the following acknowledgement:
+
+      This product includes software developed by Eric Rescorla for
+      RTFM, Inc.
+
+   4. Neither the name of RTFM, Inc. nor the name of Eric Rescorla may be
+      used to endorse or promote products derived from this
+      software without specific prior written permission.
+
+   THIS SOFTWARE IS PROVIDED BY ERIC RESCORLA AND RTFM, INC. ``AS IS'' AND
+   ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+   FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+   DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+   OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+   LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+   OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY SUCH DAMAGE.
+
+   $Id: r_common.h,v 1.2 2006/08/16 19:39:17 adamcain Exp $
+
+
+   ekr@rtfm.com  Tue Dec 22 10:40:07 1998
+ */
+
+
+#ifndef _r_common_h
+#define _r_common_h
+
+#include "r_defaults.h"
+#include "r_includes.h"
+#include "r_types.h"
+#include "r_macros.h"
+#include "r_errors.h"
+#include "r_data.h"
+
+/* defines for possibly replaced functions */
+#ifndef HAVE_STRDUP
+char *strdup(char *in);
+#endif
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_crc32.c b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_crc32.c
new file mode 100644
index 0000000000..38d3e4da38
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_crc32.c
@@ -0,0 +1,175 @@
+/**
+   r_crc32.c
+
+
+   Copyright (C) 2002-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+ */
+
+/**
+   crc32.c
+
+   Copyright (C) 2003, RTFM, Inc.
+   All Rights Reserved.
+
+   ekr@rtfm.com  Fri Jan 31 13:57:43 2003
+
+   THE FOLLOWING CODE WAS EXTRACTED FROM FreeBSD
+   The comment below was the original header
+
+   The main function was modified to process a buffer
+   rather than a file
+ */
+
+/*
+ * This code implements the AUTODIN II polynomial used by Ethernet,
+ * and can be used to calculate multicast address hash indices.
+ * It assumes that the low order bits will be transmitted first,
+ * and consequently the low byte should be sent first when
+ * the crc computation is finished.  The crc should be complemented
+ * before transmission.
+ * The variable corresponding to the macro argument "crc" should
+ * be an unsigned long and should be preset to all ones for Ethernet
+ * use.  An error-free packet will leave 0xDEBB20E3 in the crc.
+ *			Spencer Garrett <srg@quick.com>
+ */
+
+
+#include <sys/types.h>
+#include <r_types.h>
+
+#ifdef WIN32
+#define u_int32_t  UINT4
+#endif
+
+#define CRC(crc, ch)	 (crc = (crc >> 8) ^ crctab[(crc ^ (ch)) & 0xff])
+
+/* generated using the AUTODIN II polynomial
+ *	x^32 + x^26 + x^23 + x^22 + x^16 +
+ *	x^12 + x^11 + x^10 + x^8 + x^7 + x^5 + x^4 + x^2 + x^1 + 1
+ */
+static const u_int32_t crctab[256] = {
+	0x00000000, 0x77073096, 0xee0e612c, 0x990951ba,
+	0x076dc419, 0x706af48f, 0xe963a535, 0x9e6495a3,
+	0x0edb8832, 0x79dcb8a4, 0xe0d5e91e, 0x97d2d988,
+	0x09b64c2b, 0x7eb17cbd, 0xe7b82d07, 0x90bf1d91,
+	0x1db71064, 0x6ab020f2, 0xf3b97148, 0x84be41de,
+	0x1adad47d, 0x6ddde4eb, 0xf4d4b551, 0x83d385c7,
+	0x136c9856, 0x646ba8c0, 0xfd62f97a, 0x8a65c9ec,
+	0x14015c4f, 0x63066cd9, 0xfa0f3d63, 0x8d080df5,
+	0x3b6e20c8, 0x4c69105e, 0xd56041e4, 0xa2677172,
+	0x3c03e4d1, 0x4b04d447, 0xd20d85fd, 0xa50ab56b,
+	0x35b5a8fa, 0x42b2986c, 0xdbbbc9d6, 0xacbcf940,
+	0x32d86ce3, 0x45df5c75, 0xdcd60dcf, 0xabd13d59,
+	0x26d930ac, 0x51de003a, 0xc8d75180, 0xbfd06116,
+	0x21b4f4b5, 0x56b3c423, 0xcfba9599, 0xb8bda50f,
+	0x2802b89e, 0x5f058808, 0xc60cd9b2, 0xb10be924,
+	0x2f6f7c87, 0x58684c11, 0xc1611dab, 0xb6662d3d,
+	0x76dc4190, 0x01db7106, 0x98d220bc, 0xefd5102a,
+	0x71b18589, 0x06b6b51f, 0x9fbfe4a5, 0xe8b8d433,
+	0x7807c9a2, 0x0f00f934, 0x9609a88e, 0xe10e9818,
+	0x7f6a0dbb, 0x086d3d2d, 0x91646c97, 0xe6635c01,
+	0x6b6b51f4, 0x1c6c6162, 0x856530d8, 0xf262004e,
+	0x6c0695ed, 0x1b01a57b, 0x8208f4c1, 0xf50fc457,
+	0x65b0d9c6, 0x12b7e950, 0x8bbeb8ea, 0xfcb9887c,
+	0x62dd1ddf, 0x15da2d49, 0x8cd37cf3, 0xfbd44c65,
+	0x4db26158, 0x3ab551ce, 0xa3bc0074, 0xd4bb30e2,
+	0x4adfa541, 0x3dd895d7, 0xa4d1c46d, 0xd3d6f4fb,
+	0x4369e96a, 0x346ed9fc, 0xad678846, 0xda60b8d0,
+	0x44042d73, 0x33031de5, 0xaa0a4c5f, 0xdd0d7cc9,
+	0x5005713c, 0x270241aa, 0xbe0b1010, 0xc90c2086,
+	0x5768b525, 0x206f85b3, 0xb966d409, 0xce61e49f,
+	0x5edef90e, 0x29d9c998, 0xb0d09822, 0xc7d7a8b4,
+	0x59b33d17, 0x2eb40d81, 0xb7bd5c3b, 0xc0ba6cad,
+	0xedb88320, 0x9abfb3b6, 0x03b6e20c, 0x74b1d29a,
+	0xead54739, 0x9dd277af, 0x04db2615, 0x73dc1683,
+	0xe3630b12, 0x94643b84, 0x0d6d6a3e, 0x7a6a5aa8,
+	0xe40ecf0b, 0x9309ff9d, 0x0a00ae27, 0x7d079eb1,
+	0xf00f9344, 0x8708a3d2, 0x1e01f268, 0x6906c2fe,
+	0xf762575d, 0x806567cb, 0x196c3671, 0x6e6b06e7,
+	0xfed41b76, 0x89d32be0, 0x10da7a5a, 0x67dd4acc,
+	0xf9b9df6f, 0x8ebeeff9, 0x17b7be43, 0x60b08ed5,
+	0xd6d6a3e8, 0xa1d1937e, 0x38d8c2c4, 0x4fdff252,
+	0xd1bb67f1, 0xa6bc5767, 0x3fb506dd, 0x48b2364b,
+	0xd80d2bda, 0xaf0a1b4c, 0x36034af6, 0x41047a60,
+	0xdf60efc3, 0xa867df55, 0x316e8eef, 0x4669be79,
+	0xcb61b38c, 0xbc66831a, 0x256fd2a0, 0x5268e236,
+	0xcc0c7795, 0xbb0b4703, 0x220216b9, 0x5505262f,
+	0xc5ba3bbe, 0xb2bd0b28, 0x2bb45a92, 0x5cb36a04,
+	0xc2d7ffa7, 0xb5d0cf31, 0x2cd99e8b, 0x5bdeae1d,
+	0x9b64c2b0, 0xec63f226, 0x756aa39c, 0x026d930a,
+	0x9c0906a9, 0xeb0e363f, 0x72076785, 0x05005713,
+	0x95bf4a82, 0xe2b87a14, 0x7bb12bae, 0x0cb61b38,
+	0x92d28e9b, 0xe5d5be0d, 0x7cdcefb7, 0x0bdbdf21,
+	0x86d3d2d4, 0xf1d4e242, 0x68ddb3f8, 0x1fda836e,
+	0x81be16cd, 0xf6b9265b, 0x6fb077e1, 0x18b74777,
+	0x88085ae6, 0xff0f6a70, 0x66063bca, 0x11010b5c,
+	0x8f659eff, 0xf862ae69, 0x616bffd3, 0x166ccf45,
+	0xa00ae278, 0xd70dd2ee, 0x4e048354, 0x3903b3c2,
+	0xa7672661, 0xd06016f7, 0x4969474d, 0x3e6e77db,
+	0xaed16a4a, 0xd9d65adc, 0x40df0b66, 0x37d83bf0,
+	0xa9bcae53, 0xdebb9ec5, 0x47b2cf7f, 0x30b5ffe9,
+	0xbdbdf21c, 0xcabac28a, 0x53b39330, 0x24b4a3a6,
+	0xbad03605, 0xcdd70693, 0x54de5729, 0x23d967bf,
+	0xb3667a2e, 0xc4614ab8, 0x5d681b02, 0x2a6f2b94,
+	0xb40bbe37, 0xc30c8ea1, 0x5a05df1b, 0x2d02ef8d,
+};
+
+#include <stdio.h>
+#include <sys/types.h>
+
+
+int r_crc32 (char *buf,int dlen,u_int32_t *cval);
+
+int
+r_crc32(buf, dlen, cval)
+  char *buf;
+  int dlen;
+  u_int32_t *cval;
+{
+    u_int32_t crc = ~0;
+    char *p ;
+    int i;
+    u_int32_t crc32_total = 0 ;
+
+    p=buf;
+
+    for(i=0;i<dlen;i++){
+      CRC(crc, *p) ;
+      CRC(crc32_total, *p) ;
+      p++;
+    }
+
+    *cval = ~crc ;
+
+    return 0 ;
+}
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_crc32.h b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_crc32.h
new file mode 100644
index 0000000000..c11be4e04d
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_crc32.h
@@ -0,0 +1,14 @@
+/**
+   r_crc32.h
+
+   Copyright (C) 2002-2003, Network Resonance, Inc.
+   All Rights Reserved.
+
+ */
+
+#ifndef _r_crc32_
+#define _r_crc32_
+
+int r_crc32 (char *buf,int dlen, UINT4 *cval);
+
+#endif
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_data.c b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_data.c
new file mode 100644
index 0000000000..dfb7af2d5c
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_data.c
@@ -0,0 +1,248 @@
+/**
+   r_data.c
+
+
+   Copyright (C) 2002-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+ */
+
+/**
+   r_data.c
+
+
+   Copyright (C) 1999-2000 RTFM, Inc.
+   All Rights Reserved
+
+   This package is a SSLv3/TLS protocol analyzer written by Eric Rescorla
+   <ekr@rtfm.com> and licensed by RTFM, Inc.
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. All advertising materials mentioning features or use of this software
+      must display the following acknowledgement:
+
+      This product includes software developed by Eric Rescorla for
+      RTFM, Inc.
+
+   4. Neither the name of RTFM, Inc. nor the name of Eric Rescorla may be
+      used to endorse or promote products derived from this
+      software without specific prior written permission.
+
+   THIS SOFTWARE IS PROVIDED BY ERIC RESCORLA AND RTFM, INC. ``AS IS'' AND
+   ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+   FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+   DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+   OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+   LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+   OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY SUCH DAMAGE.
+
+   $Id: r_data.c,v 1.2 2006/08/16 19:39:17 adamcain Exp $
+
+   ekr@rtfm.com  Tue Aug 17 15:39:50 1999
+ */
+
+#include <string.h>
+#include <r_common.h>
+#include <r_data.h>
+#include <string.h>
+
+int r_data_create(dp,d,l)
+  Data **dp;
+  const UCHAR *d;
+  size_t l;
+  {
+    Data *d_=0;
+    int _status;
+
+    if(!(d_=(Data *)RCALLOC(sizeof(Data))))
+      ABORT(R_NO_MEMORY);
+    if(!(d_->data=(UCHAR *)RMALLOC(l)))
+      ABORT(R_NO_MEMORY);
+
+    if (d) {
+      memcpy(d_->data,d,l);
+    }
+    d_->len=l;
+
+    *dp=d_;
+
+    _status=0;
+  abort:
+    if(_status)
+      r_data_destroy(&d_);
+
+    return(_status);
+  }
+
+
+int r_data_alloc_mem(d,l)
+  Data *d;
+  size_t l;
+  {
+    int _status;
+
+    if(!(d->data=(UCHAR *)RMALLOC(l)))
+      ABORT(R_NO_MEMORY);
+    d->len=l;
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int r_data_alloc(dp,l)
+  Data **dp;
+  size_t l;
+  {
+    Data *d_=0;
+    int _status;
+
+    if(!(d_=(Data *)RCALLOC(sizeof(Data))))
+      ABORT(R_NO_MEMORY);
+    if(!(d_->data=(UCHAR *)RCALLOC(l)))
+      ABORT(R_NO_MEMORY);
+
+    d_->len=l;
+
+    *dp=d_;
+    _status=0;
+  abort:
+    if(_status)
+      r_data_destroy(&d_);
+
+    return(_status);
+  }
+
+int r_data_make(dp,d,l)
+  Data *dp;
+  const UCHAR *d;
+  size_t l;
+  {
+    if(!(dp->data=(UCHAR *)RMALLOC(l)))
+      ERETURN(R_NO_MEMORY);
+
+    memcpy(dp->data,d,l);
+    dp->len=l;
+
+    return(0);
+  }
+
+int r_data_destroy(dp)
+  Data **dp;
+  {
+    if(!dp || !*dp)
+      return(0);
+
+    if((*dp)->data)
+      RFREE((*dp)->data);
+
+    RFREE(*dp);
+    *dp=0;
+
+    return(0);
+  }
+
+int r_data_destroy_v(v)
+  void *v;
+  {
+    Data *d;
+
+    if(!v)
+      return(0);
+
+    d=(Data *)v;
+    r_data_zfree(d);
+
+    RFREE(d);
+
+    return(0);
+  }
+
+int r_data_destroy_vp(v)
+  void **v;
+  {
+    Data *d;
+
+    if(!v || !*v)
+      return(0);
+
+    d=(Data *)*v;
+    r_data_zfree(d);
+
+    *v=0;
+    RFREE(d);
+
+    return(0);
+  }
+
+int r_data_copy(dst,src)
+  Data *dst;
+  Data *src;
+  {
+    if(!(dst->data=(UCHAR *)RMALLOC(src->len)))
+      ERETURN(R_NO_MEMORY);
+    memcpy(dst->data,src->data,dst->len=src->len);
+    return(0);
+  }
+
+int r_data_zfree(d)
+  Data *d;
+  {
+    if(!d)
+      return(0);
+    if(!d->data)
+      return(0);
+    memset(d->data,0,d->len);
+    RFREE(d->data);
+    return(0);
+  }
+
+int r_data_compare(d1,d2)
+  Data *d1;
+  Data *d2;
+  {
+    if(d1->len<d2->len)
+      return(-1);
+    if(d2->len<d1->len)
+      return(-1);
+    return(memcmp(d1->data,d2->data,d1->len));
+  }
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_data.h b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_data.h
new file mode 100644
index 0000000000..3edf7b287a
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_data.h
@@ -0,0 +1,108 @@
+/**
+   r_data.h
+
+
+   Copyright (C) 2002-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+ */
+
+/**
+   r_data.h
+
+
+   Copyright (C) 1999-2000 RTFM, Inc.
+   All Rights Reserved
+
+   This package is a SSLv3/TLS protocol analyzer written by Eric Rescorla
+   <ekr@rtfm.com> and licensed by RTFM, Inc.
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. All advertising materials mentioning features or use of this software
+      must display the following acknowledgement:
+
+      This product includes software developed by Eric Rescorla for
+      RTFM, Inc.
+
+   4. Neither the name of RTFM, Inc. nor the name of Eric Rescorla may be
+      used to endorse or promote products derived from this
+      software without specific prior written permission.
+
+   THIS SOFTWARE IS PROVIDED BY ERIC RESCORLA AND RTFM, INC. ``AS IS'' AND
+   ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+   FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+   DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+   OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+   LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+   OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY SUCH DAMAGE.
+
+   $Id: r_data.h,v 1.2 2006/08/16 19:39:17 adamcain Exp $
+
+
+   ekr@rtfm.com  Wed Feb 10 14:18:19 1999
+ */
+
+
+#ifndef _r_data_h
+#define _r_data_h
+
+typedef struct Data_ {
+     UCHAR *data;
+     size_t len;
+} Data;
+
+int r_data_create(Data **dp,const UCHAR *d,size_t l);
+int r_data_alloc(Data **dp, size_t l);
+int r_data_make(Data *dp, const UCHAR *d,size_t l);
+int r_data_alloc_mem(Data *d,size_t l);
+int r_data_destroy(Data **dp);
+int r_data_destroy_v(void *v);
+int r_data_destroy_vp(void **vp);
+int r_data_copy(Data *dst,Data *src);
+int r_data_zfree(Data *d);
+int r_data_compare(Data *d1,Data *d2);
+
+#define INIT_DATA(a,b,c) (a).data=b; (a).len=c
+#define ATTACH_DATA(a,b) (a).data=b; (a).len=sizeof(b)
+#define ZERO_DATA(a) (a).data=0; (a).len=0
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_defaults.h b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_defaults.h
new file mode 100644
index 0000000000..0ec91a0331
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_defaults.h
@@ -0,0 +1,91 @@
+/**
+   r_defaults.h
+
+
+   Copyright (C) 2002-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+ */
+
+/**
+   r_defaults.h
+
+
+   Copyright (C) 1999-2000 RTFM, Inc.
+   All Rights Reserved
+
+   This package is a SSLv3/TLS protocol analyzer written by Eric Rescorla
+   <ekr@rtfm.com> and licensed by RTFM, Inc.
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. All advertising materials mentioning features or use of this software
+      must display the following acknowledgement:
+
+      This product includes software developed by Eric Rescorla for
+      RTFM, Inc.
+
+   4. Neither the name of RTFM, Inc. nor the name of Eric Rescorla may be
+      used to endorse or promote products derived from this
+      software without specific prior written permission.
+
+   THIS SOFTWARE IS PROVIDED BY ERIC RESCORLA AND RTFM, INC. ``AS IS'' AND
+   ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+   FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+   DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+   OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+   LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+   OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY SUCH DAMAGE.
+
+   $Id: r_defaults.h,v 1.2 2006/08/16 19:39:17 adamcain Exp $
+
+
+   ekr@rtfm.com  Tue Dec 22 10:39:14 1998
+ */
+
+
+#ifndef _r_defaults_h
+#define _r_defaults_h
+
+/*The needs defines don't belong here*/
+#define R_NEEDS_STDLIB_H
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_errors.c b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_errors.c
new file mode 100644
index 0000000000..e770e02438
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_errors.c
@@ -0,0 +1,136 @@
+/**
+   r_errors.c
+
+
+   Copyright (C) 2002-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+ */
+
+/**
+   r_errors.c
+
+
+   Copyright (C) 1999-2000 RTFM, Inc.
+   All Rights Reserved
+
+   This package is a SSLv3/TLS protocol analyzer written by Eric Rescorla
+   <ekr@rtfm.com> and licensed by RTFM, Inc.
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. All advertising materials mentioning features or use of this software
+      must display the following acknowledgement:
+
+      This product includes software developed by Eric Rescorla for
+      RTFM, Inc.
+
+   4. Neither the name of RTFM, Inc. nor the name of Eric Rescorla may be
+      used to endorse or promote products derived from this
+      software without specific prior written permission.
+
+   THIS SOFTWARE IS PROVIDED BY ERIC RESCORLA AND RTFM, INC. ``AS IS'' AND
+   ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+   FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+   DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+   OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+   LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+   OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY SUCH DAMAGE.
+
+   $Id: r_errors.c,v 1.5 2008/11/26 03:22:02 adamcain Exp $
+
+
+   ekr@rtfm.com  Tue Feb 16 16:37:05 1999
+ */
+
+#include <stdio.h>
+#include <stdarg.h>
+#include <string.h>
+#include "r_common.h"
+#include "r_errors.h"
+
+static struct {
+    int    errnum;
+    char  *str;
+} errors[] = NR_ERROR_MAPPING;
+
+int nr_verr_exit(char *fmt,...)
+  {
+    va_list ap;
+
+    va_start(ap,fmt);
+    vfprintf(stderr,fmt,ap);
+
+    if (fmt[0] != '\0' && fmt[strlen(fmt)-1] != '\n')
+        fprintf(stderr,"\n");
+
+    exit(1);
+  }
+
+char *
+nr_strerror(int errnum)
+{
+    static char unknown_error[256];
+    size_t i;
+    char *error = 0;
+
+    for (i = 0; i < sizeof(errors)/sizeof(*errors); ++i) {
+        if (errnum == errors[i].errnum) {
+            error = errors[i].str;
+            break;
+        }
+    }
+
+    if (! error) {
+        snprintf(unknown_error, sizeof(unknown_error), "Unknown error: %d", errnum);
+        error = unknown_error;
+    }
+
+    return error;
+}
+
+int
+nr_strerror_r(int errnum, char *strerrbuf, size_t buflen)
+{
+    char *error = nr_strerror(errnum);
+    snprintf(strerrbuf, buflen, "%s", error);
+    return 0;
+}
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_errors.h b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_errors.h
new file mode 100644
index 0000000000..f52375b98d
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_errors.h
@@ -0,0 +1,127 @@
+/**
+   r_errors.h
+
+
+   Copyright (C) 2002-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+ */
+
+/**
+   r_errors.h
+
+
+   Copyright (C) 1999-2000 RTFM, Inc.
+   All Rights Reserved
+
+   This package is a SSLv3/TLS protocol analyzer written by Eric Rescorla
+   <ekr@rtfm.com> and licensed by RTFM, Inc.
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. All advertising materials mentioning features or use of this software
+      must display the following acknowledgement:
+
+      This product includes software developed by Eric Rescorla for
+      RTFM, Inc.
+
+   4. Neither the name of RTFM, Inc. nor the name of Eric Rescorla may be
+      used to endorse or promote products derived from this
+      software without specific prior written permission.
+
+   THIS SOFTWARE IS PROVIDED BY ERIC RESCORLA AND RTFM, INC. ``AS IS'' AND
+   ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+   FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+   DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+   OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+   LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+   OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY SUCH DAMAGE.
+
+   $Id: r_errors.h,v 1.4 2007/10/12 20:53:24 adamcain Exp $
+
+
+   ekr@rtfm.com  Tue Dec 22 10:59:49 1998
+ */
+
+
+#ifndef _r_errors_h
+#define _r_errors_h
+
+#define R_NO_MEMORY      1 /*out of memory*/
+#define R_NOT_FOUND	 2 /*Item not found*/
+#define R_INTERNAL	 3 /*Unspecified internal error*/
+#define R_ALREADY	 4 /*Action already done*/
+#define R_EOD            5 /*end of data*/
+#define R_BAD_ARGS       6 /*Bad arguments*/
+#define R_BAD_DATA	 7 /*Bad data*/
+#define R_WOULDBLOCK     8 /*Operation would block */
+#define R_QUEUED         9 /*Operation was queued */
+#define R_FAILED        10 /*Operation failed */
+#define R_REJECTED      11 /* We don't care about this */
+#define R_INTERRUPTED   12 /* Operation interrupted */
+#define R_IO_ERROR      13 /* I/O Error */
+#define R_NOT_PERMITTED 14 /* Permission denied */
+#define R_RETRY         15 /* Retry possible */
+
+#define NR_ERROR_MAPPING {\
+    { R_NO_MEMORY,        "Cannot allocate memory" },\
+    { R_NOT_FOUND,        "Item not found" },\
+    { R_INTERNAL,         "Internal failure" },\
+    { R_ALREADY,          "Action already performed" },\
+    { R_EOD,              "End of data" },\
+    { R_BAD_ARGS,         "Invalid argument" },\
+    { R_BAD_DATA,         "Invalid data" },\
+    { R_WOULDBLOCK,       "Operation would block" },\
+    { R_QUEUED,           "Operation queued" },\
+    { R_FAILED,           "Operation failed" },\
+    { R_REJECTED,         "Operation rejected" },\
+    { R_INTERRUPTED,      "Operation interrupted" },\
+    { R_IO_ERROR,         "I/O error" },\
+    { R_NOT_PERMITTED,    "Permission Denied" },\
+    { R_RETRY,            "Retry may be possible" },\
+    }
+
+int nr_verr_exit(char *fmt,...);
+
+char *nr_strerror(int errnum);
+int   nr_strerror_r(int errnum, char *strerrbuf, size_t buflen);
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_includes.h b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_includes.h
new file mode 100644
index 0000000000..4fb7af5643
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_includes.h
@@ -0,0 +1,98 @@
+/**
+   r_includes.h
+
+
+   Copyright (C) 2002-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+ */
+
+/**
+   r_includes.h
+
+
+   Copyright (C) 1999-2000 RTFM, Inc.
+   All Rights Reserved
+
+   This package is a SSLv3/TLS protocol analyzer written by Eric Rescorla
+   <ekr@rtfm.com> and licensed by RTFM, Inc.
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. All advertising materials mentioning features or use of this software
+      must display the following acknowledgement:
+
+      This product includes software developed by Eric Rescorla for
+      RTFM, Inc.
+
+   4. Neither the name of RTFM, Inc. nor the name of Eric Rescorla may be
+      used to endorse or promote products derived from this
+      software without specific prior written permission.
+
+   THIS SOFTWARE IS PROVIDED BY ERIC RESCORLA AND RTFM, INC. ``AS IS'' AND
+   ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+   FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+   DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+   OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+   LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+   OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY SUCH DAMAGE.
+
+   $Id: r_includes.h,v 1.2 2006/08/16 19:39:17 adamcain Exp $
+
+
+   ekr@rtfm.com  Tue Dec 22 11:38:50 1998
+ */
+
+
+#ifndef _r_includes_h
+#define _r_includes_h
+
+#ifdef R_NEEDS_STDLIB_H
+#include <stdlib.h>
+#endif
+
+#ifdef R_NEEDS_MEMORY_H
+#include <memory.h>
+#endif
+
+#include <stdio.h>
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_list.c b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_list.c
new file mode 100644
index 0000000000..4e71d67030
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_list.c
@@ -0,0 +1,273 @@
+/**
+   r_list.c
+
+
+   Copyright (C) 2002-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+ */
+
+/**
+   r_list.c
+
+
+   Copyright (C) 1999-2000 RTFM, Inc.
+   All Rights Reserved
+
+   This package is a SSLv3/TLS protocol analyzer written by Eric Rescorla
+   <ekr@rtfm.com> and licensed by RTFM, Inc.
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. All advertising materials mentioning features or use of this software
+      must display the following acknowledgement:
+
+      This product includes software developed by Eric Rescorla for
+      RTFM, Inc.
+
+   4. Neither the name of RTFM, Inc. nor the name of Eric Rescorla may be
+      used to endorse or promote products derived from this
+      software without specific prior written permission.
+
+   THIS SOFTWARE IS PROVIDED BY ERIC RESCORLA AND RTFM, INC. ``AS IS'' AND
+   ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+   FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+   DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+   OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+   LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+   OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY SUCH DAMAGE.
+
+   $Id: r_list.c,v 1.2 2006/08/16 19:39:17 adamcain Exp $
+
+
+   ekr@rtfm.com  Tue Jan 19 08:36:39 1999
+ */
+
+#include <r_common.h>
+#include "r_list.h"
+
+typedef struct r_list_el_ {
+     void *data;
+     struct r_list_el_ *next;
+     struct r_list_el_ *prev;
+     int (*copy)(void **new,void *old);
+     int (*destroy)(void **ptr);
+} r_list_el;
+
+struct r_list_ {
+     struct r_list_el_ *first;
+     struct r_list_el_ *last;
+};
+
+int r_list_create(listp)
+  r_list **listp;
+  {
+    r_list *list=0;
+    int _status;
+
+    if(!(list=(r_list *)RCALLOC(sizeof(r_list))))
+      ABORT(R_NO_MEMORY);
+
+    list->first=0;
+    list->last=0;
+    *listp=list;
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int r_list_destroy(listp)
+  r_list **listp;
+  {
+    r_list *list;
+    r_list_el *el;
+
+    if(!listp || !*listp)
+      return(0);
+    list=*listp;
+
+    el=list->first;
+
+    while(el){
+      r_list_el *el_t;
+
+      if(el->destroy && el->data)
+        el->destroy(&el->data);
+      el_t=el;
+      el=el->next;
+      RFREE(el_t);
+    }
+
+    RFREE(list);
+    *listp=0;
+
+    return(0);
+  }
+
+int r_list_copy(outp,in)
+  r_list**outp;
+  r_list *in;
+  {
+    r_list *out=0;
+    r_list_el *el,*el2,*last=0;
+    int r, _status;
+
+    if(!in){
+      *outp=0;
+      return(0);
+    }
+
+    if(r=r_list_create(&out))
+      ABORT(r);
+
+    for(el=in->first;el;el=el->next){
+      if(!(el2=(r_list_el *)RCALLOC(sizeof(r_list_el))))
+        ABORT(R_NO_MEMORY);
+
+      if(el->copy && el->data){
+        if(r=el->copy(&el2->data,el->data))
+          ABORT(r);
+      }
+
+      el2->copy=el->copy;
+      el2->destroy=el->destroy;
+
+      if(!(out->first))
+        out->first=el2;
+
+      el2->prev=last;
+      if(last) last->next=el2;
+      last=el2;
+    }
+
+    out->last=last;
+
+    *outp=out;
+
+    _status=0;
+  abort:
+    if(_status)
+      r_list_destroy(&out);
+    return(_status);
+  }
+
+int r_list_insert(list,value,copy,destroy)
+  r_list *list;
+  void *value;
+  int (*copy)(void **out, void *in);
+  int (*destroy)(void **val);
+  {
+    r_list_el *el=0;
+    int _status;
+
+    if(!(el=(r_list_el *)RCALLOC(sizeof(r_list_el))))
+      ABORT(R_NO_MEMORY);
+    el->data=value;
+    el->copy=copy;
+    el->destroy=destroy;
+
+    el->prev=0;
+    el->next=list->first;
+    if(list->first){
+      list->first->prev=el;
+    }
+    list->first=el;
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int r_list_append(list,value,copy,destroy)
+  r_list *list;
+  void *value;
+  int (*copy)(void **out, void *in);
+  int (*destroy)(void **val);
+  {
+    r_list_el *el=0;
+    int _status;
+
+    if(!(el=(r_list_el *)RCALLOC(sizeof(r_list_el))))
+      ABORT(R_NO_MEMORY);
+    el->data=value;
+    el->copy=copy;
+    el->destroy=destroy;
+
+    el->prev=list->last;
+    el->next=0;
+
+    if(list->last) list->last->next=el;
+    else list->first=el;
+
+    list->last=el;
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int r_list_init_iter(list,iter)
+  r_list *list;
+  r_list_iterator *iter;
+  {
+    iter->list=list;
+    iter->ptr=list->first;
+
+    return(0);
+  }
+
+int r_list_iter(iter,val)
+  r_list_iterator *iter;
+  void **val;
+  {
+    if(!iter->ptr)
+      return(R_EOD);
+
+    *val=iter->ptr->data;
+    iter->ptr=iter->ptr->next;
+
+    return(0);
+  }
+
+
+
+
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_list.h b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_list.h
new file mode 100644
index 0000000000..bbefc6c39f
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_list.h
@@ -0,0 +1,106 @@
+/**
+   r_list.h
+
+
+   Copyright (C) 2002-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+ */
+
+/**
+   r_list.h
+
+
+   Copyright (C) 1999-2000 RTFM, Inc.
+   All Rights Reserved
+
+   This package is a SSLv3/TLS protocol analyzer written by Eric Rescorla
+   <ekr@rtfm.com> and licensed by RTFM, Inc.
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. All advertising materials mentioning features or use of this software
+      must display the following acknowledgement:
+
+      This product includes software developed by Eric Rescorla for
+      RTFM, Inc.
+
+   4. Neither the name of RTFM, Inc. nor the name of Eric Rescorla may be
+      used to endorse or promote products derived from this
+      software without specific prior written permission.
+
+   THIS SOFTWARE IS PROVIDED BY ERIC RESCORLA AND RTFM, INC. ``AS IS'' AND
+   ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+   FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+   DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+   OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+   LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+   OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY SUCH DAMAGE.
+
+   $Id: r_list.h,v 1.3 2007/06/08 17:41:49 adamcain Exp $
+
+
+   ekr@rtfm.com  Tue Jan 19 08:36:48 1999
+ */
+
+
+#ifndef _r_list_h
+#define _r_list_h
+
+typedef struct r_list_ r_list;
+
+typedef struct r_list_iterator_ {
+     r_list *list;
+     struct r_list_el_ *ptr;
+} r_list_iterator;
+
+int r_list_create(r_list **listp);
+int r_list_destroy(r_list **listp);
+int r_list_copy(r_list **out,r_list *in);
+int r_list_insert(r_list *list,void *value,
+  int (*copy)(void **knew,void *old),
+  int (*destroy)(void **ptr));
+int r_list_append(r_list *list,void *value,
+  int (*copy)(void **knew,void *old),
+  int (*destroy)(void **ptr));
+int r_list_init_iter(r_list *list,r_list_iterator *iter);
+int r_list_iter(r_list_iterator *iter,void **val);
+
+#endif
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_macros.h b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_macros.h
new file mode 100644
index 0000000000..ddfedd3c36
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_macros.h
@@ -0,0 +1,137 @@
+/**
+   r_macros.h
+
+
+   Copyright (C) 2002-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+ */
+
+/**
+   r_macros.h
+
+
+   Copyright (C) 1999-2000 RTFM, Inc.
+   All Rights Reserved
+
+   This package is a SSLv3/TLS protocol analyzer written by Eric Rescorla
+   <ekr@rtfm.com> and licensed by RTFM, Inc.
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. All advertising materials mentioning features or use of this software
+      must display the following acknowledgement:
+
+      This product includes software developed by Eric Rescorla for
+      RTFM, Inc.
+
+   4. Neither the name of RTFM, Inc. nor the name of Eric Rescorla may be
+      used to endorse or promote products derived from this
+      software without specific prior written permission.
+
+   THIS SOFTWARE IS PROVIDED BY ERIC RESCORLA AND RTFM, INC. ``AS IS'' AND
+   ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+   FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+   DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+   OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+   LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+   OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY SUCH DAMAGE.
+
+   $Id: r_macros.h,v 1.3 2007/06/26 22:37:57 adamcain Exp $
+
+   ekr@rtfm.com  Tue Dec 22 10:37:32 1998
+ */
+
+
+#ifndef _r_macros_h
+#define _r_macros_h
+
+/* Haven't removed all PROTO_LIST defs yet */
+#define PROTO_LIST(a) a
+
+#ifndef WIN32
+#ifndef __GNUC__
+#define __FUNCTION__ "unknown"
+#endif
+#endif
+
+#ifdef R_TRACE_ERRORS
+#ifdef WIN32
+#define REPORT_ERROR_(caller,a) printf("%s: error %d at %s:%d (function %s)\n", \
+	caller,a,__FILE__,__LINE__,__FUNCTION__)
+#else
+#define REPORT_ERROR_(caller,a) fprintf(stderr,"%s: error %d at %s:%d (function %s)\n", \
+	caller,a,__FILE__,__LINE__,__FUNCTION__)
+#endif
+#else
+#define REPORT_ERROR_(caller,a)
+#endif
+
+#ifndef ERETURN
+#define ERETURN(a) do {int _r=a; if(!_r) _r=-1; REPORT_ERROR_("ERETURN",_r); return(_r);} while(0)
+#endif
+
+#ifndef ABORT
+#define ABORT(a) do { int _r=a; if(!_r) _r=-1; REPORT_ERROR_("ABORT",_r); _status=_r; goto abort;} while(0)
+#endif
+
+#ifndef FREE
+#define FREE(a) if(a) free(a)
+#endif
+#ifndef MIN
+#define MIN(a,b) ((a)>(b))?(b):(a)
+#endif
+
+#ifndef MAX
+#define MAX(a,b) ((b)>(a))?(b):(a)
+#endif
+
+#ifdef DEBUG
+#define DBG(a) debug a
+int debug(int cls, char *format,...);
+#else
+#define DBG(a)
+#endif
+
+#define NR_UNIMPLEMENTED do { fprintf(stderr,"%s:%d Function %s unimplemented\n",__FILE__,__LINE__,__FUNCTION__); abort(); } while(0)
+
+#include "r_memory.h"
+
+#endif
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_memory.c b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_memory.c
new file mode 100644
index 0000000000..53846fc019
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_memory.c
@@ -0,0 +1,198 @@
+/**
+   r_memory.c
+
+
+   Copyright (C) 2004, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   ekr@rtfm.com  Thu Apr 22 20:40:45 2004
+ */
+
+#include <string.h>
+#include <stddef.h>
+#include <assert.h>
+#include "r_common.h"
+#include "r_memory.h"
+
+typedef struct r_malloc_chunk_ {
+#ifdef SANITY_CHECKS
+     UINT4 hdr;
+#endif
+     UCHAR type;
+     UINT4 size;
+     UCHAR memory[1];
+} r_malloc_chunk;
+
+#define CHUNK_MEMORY_OFFSET                    offsetof(struct r_malloc_chunk_, memory)
+#define GET_CHUNK_ADDR_FROM_MEM_ADDR(memp) \
+        ((struct r_malloc_chunk *)(((unsigned char*)(memp))-CHUNK_MEMORY_OFFSET))
+#define CHUNK_SIZE(size) (size+sizeof(r_malloc_chunk))
+
+#define HDR_FLAG 0x464c4147
+
+static UINT4 mem_usage;      /* Includes our header */
+static UINT4 mem_stats[256]; /* Does not include our header */
+
+void *r_malloc(type,size)
+  int type;
+  size_t size;
+  {
+    size_t total;
+    r_malloc_chunk *chunk;
+
+    total=size+sizeof(r_malloc_chunk);
+
+    if(!(chunk=malloc(total)))
+      return(0);
+
+#ifdef SANITY_CHECKS
+    chunk->hdr=HDR_FLAG;
+#endif
+    chunk->type=type;
+    chunk->size=size;
+
+    mem_usage+=CHUNK_SIZE(size);
+    mem_stats[type]+=size;
+
+    return(chunk->memory);
+  }
+
+void *r_calloc(type,number,size)
+  int type;
+  size_t number;
+  size_t size;
+  {
+    void *ret;
+    size_t total;
+
+    total=number*size;
+
+    if(!(ret=r_malloc(type,total)))
+      return(0);
+
+    memset(ret,0,size);
+
+    return(ret);
+  }
+
+void r_free(ptr)
+  void *ptr;
+  {
+    r_malloc_chunk *chunk;
+
+    if(!ptr) return;
+
+    chunk=(r_malloc_chunk *)GET_CHUNK_ADDR_FROM_MEM_ADDR(ptr);
+#ifdef SANITY_CHECKS
+    assert(chunk->hdr==HDR_FLAG);
+#endif
+
+    mem_usage-=CHUNK_SIZE(chunk->size);
+    mem_stats[chunk->type]-=chunk->size;
+
+    free(chunk);
+  }
+
+void *r_realloc(ptr,size)
+  void *ptr;
+  size_t size;
+  {
+    r_malloc_chunk *chunk,*nchunk;
+    size_t total;
+
+    if(!ptr) return(r_malloc(255,size));
+
+    chunk=(r_malloc_chunk *)GET_CHUNK_ADDR_FROM_MEM_ADDR(ptr);
+#ifdef SANITY_CHECKS
+    assert(chunk->hdr==HDR_FLAG);
+#endif
+
+    total=size + sizeof(r_malloc_chunk);
+
+    if(!(nchunk=realloc(chunk,total)))
+      return(0);
+
+    mem_usage-=CHUNK_SIZE(nchunk->size);
+    mem_stats[nchunk->type]-=nchunk->size;
+
+    nchunk->size=size;
+    mem_usage+=CHUNK_SIZE(nchunk->size);
+    mem_stats[nchunk->type]+=nchunk->size;
+
+    return(nchunk->memory);
+  }
+
+char *r_strdup(str)
+  const char *str;
+  {
+    int len;
+    char *nstr;
+
+    if(!str)
+      return(0);
+
+    len=strlen(str)+1;
+
+    if(!(nstr=r_malloc(0,len)))
+      return(0);
+
+    memcpy(nstr,str,len);
+
+    return(nstr);
+  }
+
+int r_mem_get_usage(usagep)
+  UINT4 *usagep;
+  {
+    *usagep=mem_usage;
+
+    return(0);
+  }
+
+int r_memory_dump_stats()
+  {
+    int i;
+
+    printf("Total memory usage: %d\n",mem_usage);
+    printf("Memory usage by bucket\n");
+    for(i=0;i<256;i++){
+      if(mem_stats[i]){
+        printf("%d\t%d\n",i,mem_stats[i]);
+      }
+    }
+    return(0);
+  }
+
+void *r_malloc_compat(size)
+  size_t size;
+  {
+    return(r_malloc(255,size));
+  }
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_memory.h b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_memory.h
new file mode 100644
index 0000000000..4357070767
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_memory.h
@@ -0,0 +1,101 @@
+/**
+   r_memory.h
+
+
+   Copyright (C) 2004, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   ekr@rtfm.com  Sat Apr 24 08:30:00 2004
+ */
+
+
+#ifndef _r_memory_h
+#define _r_memory_h
+
+#define R_MALLOC_X 2
+
+#include "r_types.h"
+
+void *r_malloc(int type, size_t size);
+void *r_malloc_compat(size_t size);
+void *r_calloc(int type,size_t number,size_t size);
+void r_free   (void *ptr);
+void *r_realloc(void *ptr,size_t size);
+char *r_strdup(const char *str);
+int r_mem_get_usage(UINT4 *usage);
+int r_memory_dump_stats(void);
+
+#ifdef NO_MALLOC_REPLACE
+
+#ifndef RMALLOC
+#define RMALLOC(a) malloc(a)
+#endif
+
+#ifndef RCALLOC
+#define RCALLOC(a) calloc(1,a)
+#endif
+
+#ifndef RFREE
+#define RFREE(a) if(a) free(a)
+#endif
+
+#ifndef RREALLOC
+#define RREALLOC(a,b) realloc(a,b)
+#endif
+
+#else
+
+
+#ifndef R_MALLOC_TYPE
+#define R_MALLOC_TYPE   0
+#endif
+
+#ifndef RMALLOC
+#define RMALLOC(a) r_malloc(R_MALLOC_TYPE,a)
+#endif
+
+#ifndef RCALLOC
+#define RCALLOC(a) r_calloc(R_MALLOC_TYPE,1,a)
+#endif
+
+#ifndef RFREE
+#define RFREE(a) if(a) r_free(a)
+#endif
+
+#ifndef RREALLOC
+#define RREALLOC(a,b) r_realloc(a,b)
+#endif
+
+#endif
+
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_replace.c b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_replace.c
new file mode 100644
index 0000000000..8916b884cc
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_replace.c
@@ -0,0 +1,107 @@
+/**
+   r_replace.c
+
+
+   Copyright (C) 2002-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+ */
+
+/**
+   r_replace.c
+
+
+   Copyright (C) 1999-2000 RTFM, Inc.
+   All Rights Reserved
+
+   This package is a SSLv3/TLS protocol analyzer written by Eric Rescorla
+   <ekr@rtfm.com> and licensed by RTFM, Inc.
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. All advertising materials mentioning features or use of this software
+      must display the following acknowledgement:
+
+      This product includes software developed by Eric Rescorla for
+      RTFM, Inc.
+
+   4. Neither the name of RTFM, Inc. nor the name of Eric Rescorla may be
+      used to endorse or promote products derived from this
+      software without specific prior written permission.
+
+   THIS SOFTWARE IS PROVIDED BY ERIC RESCORLA AND RTFM, INC. ``AS IS'' AND
+   ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+   FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+   DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+   OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+   LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+   OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY SUCH DAMAGE.
+
+   $Id: r_replace.c,v 1.2 2006/08/16 19:39:17 adamcain Exp $
+
+
+   ekr@rtfm.com  Sun Oct  1 11:18:49 2000
+ */
+
+#include  "r_common.h"
+
+#ifndef HAVE_STRDUP
+
+char *strdup(str)
+  char *str;
+  {
+    int len=strlen(str);
+    char *n;
+
+    if(!(n=(char *)malloc(len+1)))
+      return(0);
+
+    memcpy(n,str,len+1);
+
+    return(n);
+  }
+#endif
+
+
+#ifdef SUPPLY_ATEXIT
+int atexit(void (*func)(void)){
+  ;
+}
+#endif
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_thread.h b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_thread.h
new file mode 100644
index 0000000000..212900bcc4
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_thread.h
@@ -0,0 +1,68 @@
+/**
+   r_thread.h
+
+
+   Copyright (C) 1999, RTFM, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   ekr@rtfm.com  Tue Feb 23 14:58:36 1999
+ */
+
+
+#ifndef _r_thread_h
+#define _r_thread_h
+
+typedef void *r_thread;
+typedef void *r_rwlock;
+typedef void * r_cond;
+
+int r_thread_fork (void (*func)(void *),void *arg,
+  r_thread *tid);
+int r_thread_destroy (r_thread tid);
+int r_thread_yield (void);
+int r_thread_exit (void);
+int r_thread_wait_last (void);
+int r_thread_self (void);
+
+int r_rwlock_create (r_rwlock **lockp);
+int r_rwlock_destroy (r_rwlock **lock);
+int r_rwlock_lock (r_rwlock *lock,int action);
+
+int r_cond_init (r_cond *cond);
+int r_cond_wait (r_cond cond);
+int r_cond_signal (r_cond cond);
+
+#define R_RWLOCK_UNLOCK 0
+#define R_RWLOCK_RLOCK 1
+#define R_RWLOCK_WLOCK 2
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_time.c b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_time.c
new file mode 100644
index 0000000000..f873585c4b
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_time.c
@@ -0,0 +1,235 @@
+/**
+   r_time.c
+
+
+   Copyright (C) 2002-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+ */
+
+/**
+   r_time.c
+
+
+   Copyright (C) 1999-2000 RTFM, Inc.
+   All Rights Reserved
+
+   This package is a SSLv3/TLS protocol analyzer written by Eric Rescorla
+   <ekr@rtfm.com> and licensed by RTFM, Inc.
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. All advertising materials mentioning features or use of this software
+      must display the following acknowledgement:
+
+      This product includes software developed by Eric Rescorla for
+      RTFM, Inc.
+
+   4. Neither the name of RTFM, Inc. nor the name of Eric Rescorla may be
+      used to endorse or promote products derived from this
+      software without specific prior written permission.
+
+   THIS SOFTWARE IS PROVIDED BY ERIC RESCORLA AND RTFM, INC. ``AS IS'' AND
+   ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+   FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+   DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+   OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+   LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+   OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY SUCH DAMAGE.
+
+   $Id: r_time.c,v 1.5 2008/11/26 03:22:02 adamcain Exp $
+
+   ekr@rtfm.com  Thu Mar  4 08:43:46 1999
+ */
+
+#include <r_common.h>
+#include <r_time.h>
+
+/*Note that t1 must be > t0 */
+int r_timeval_diff(t1,t0,diff)
+  struct timeval *t1;
+  struct timeval *t0;
+  struct timeval *diff;
+  {
+    long d;
+
+    if(t0->tv_sec > t1->tv_sec)
+      ERETURN(R_BAD_ARGS);
+    if((t0->tv_sec == t1->tv_sec) && (t0->tv_usec > t1->tv_usec))
+      ERETURN(R_BAD_ARGS);
+
+    /*Easy case*/
+    if(t0->tv_usec <= t1->tv_usec){
+      diff->tv_sec=t1->tv_sec - t0->tv_sec;
+      diff->tv_usec=t1->tv_usec - t0->tv_usec;
+      return(0);
+    }
+
+    /*Hard case*/
+    d=t0->tv_usec - t1->tv_usec;
+    if(t1->tv_sec < (t0->tv_sec + 1))
+      ERETURN(R_BAD_ARGS);
+    diff->tv_sec=t1->tv_sec - (t0->tv_sec + 1);
+    diff->tv_usec=1000000 - d;
+
+    return(0);
+  }
+
+int r_timeval_add(t1,t2,sum)
+  struct timeval *t1;
+  struct timeval *t2;
+  struct timeval *sum;
+  {
+    long tv_sec,tv_usec,d;
+
+    tv_sec=t1->tv_sec + t2->tv_sec;
+
+    d=t1->tv_usec + t2->tv_usec;
+    if(d>1000000){
+      tv_sec++;
+      tv_usec=d-1000000;
+    }
+    else{
+      tv_usec=d;
+    }
+
+    sum->tv_sec=tv_sec;
+    sum->tv_usec=tv_usec;
+
+    return(0);
+  }
+
+int r_timeval_cmp(t1,t2)
+  struct timeval *t1;
+  struct timeval *t2;
+  {
+    if(t1->tv_sec>t2->tv_sec)
+      return(1);
+    if(t1->tv_sec<t2->tv_sec)
+      return(-1);
+    if(t1->tv_usec>t2->tv_usec)
+      return(1);
+    if(t1->tv_usec<t2->tv_usec)
+      return(-1);
+    return(0);
+  }
+
+
+UINT8 r_timeval2int(tv)
+  struct timeval *tv;
+  {
+    UINT8 r=0;
+
+    r=(tv->tv_sec);
+    r*=1000000;
+    r+=tv->tv_usec;
+
+    return r;
+  }
+
+int r_int2timeval(UINT8 t,struct timeval *tv)
+  {
+    tv->tv_sec=t/1000000;
+    tv->tv_usec=t%1000000;
+
+    return(0);
+  }
+
+UINT8 r_gettimeint()
+  {
+    struct timeval tv;
+
+    gettimeofday(&tv,0);
+
+    return r_timeval2int(&tv);
+  }
+
+/* t1-t0 in microseconds */
+int r_timeval_diff_usec(struct timeval *t1, struct timeval *t0, INT8 *diff)
+  {
+    int r,_status;
+    int sign;
+    struct timeval tmp;
+
+    sign = 1;
+    if (r=r_timeval_diff(t1, t0, &tmp)) {
+        if (r == R_BAD_ARGS) {
+            sign = -1;
+            if (r=r_timeval_diff(t0, t1, &tmp))
+                ABORT(r);
+        }
+    }
+
+    /* 1 second = 1000 milliseconds
+     * 1 milliseconds = 1000 microseconds */
+
+    *diff = ((tmp.tv_sec * (1000*1000)) + tmp.tv_usec) * sign;
+
+    _status = 0;
+  abort:
+    return(_status);
+  }
+
+/* t1-t0 in milliseconds */
+int r_timeval_diff_ms(struct timeval *t1, struct timeval *t0, INT8 *diff)
+  {
+    int r,_status;
+    int sign;
+    struct timeval tmp;
+
+    sign = 1;
+    if (r=r_timeval_diff(t1, t0, &tmp)) {
+        if (r == R_BAD_ARGS) {
+            sign = -1;
+            if (r=r_timeval_diff(t0, t1, &tmp))
+                ABORT(r);
+        }
+    }
+
+    /* 1 second = 1000 milliseconds
+     * 1 milliseconds = 1000 microseconds */
+
+    *diff = ((tmp.tv_sec * 1000) + (tmp.tv_usec / 1000)) * sign;
+
+    _status = 0;
+  abort:
+    return(_status);
+  }
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_time.h b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_time.h
new file mode 100644
index 0000000000..1ee1e778fe
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_time.h
@@ -0,0 +1,109 @@
+/**
+   r_time.h
+
+
+   Copyright (C) 2002-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+ */
+
+/**
+   r_time.h
+
+
+   Copyright (C) 1999-2000 RTFM, Inc.
+   All Rights Reserved
+
+   This package is a SSLv3/TLS protocol analyzer written by Eric Rescorla
+   <ekr@rtfm.com> and licensed by RTFM, Inc.
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. All advertising materials mentioning features or use of this software
+      must display the following acknowledgement:
+
+      This product includes software developed by Eric Rescorla for
+      RTFM, Inc.
+
+   4. Neither the name of RTFM, Inc. nor the name of Eric Rescorla may be
+      used to endorse or promote products derived from this
+      software without specific prior written permission.
+
+   THIS SOFTWARE IS PROVIDED BY ERIC RESCORLA AND RTFM, INC. ``AS IS'' AND
+   ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+   FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+   DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+   OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+   LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+   OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY SUCH DAMAGE.
+
+   $Id: r_time.h,v 1.4 2007/06/26 22:37:57 adamcain Exp $
+
+
+   ekr@rtfm.com  Thu Mar  4 08:45:41 1999
+ */
+
+
+#ifndef _r_time_h
+#define _r_time_h
+
+#include <csi_platform.h>
+
+#ifndef WIN32
+#include <sys/time.h>
+#include <time.h>
+#endif
+
+int r_timeval_diff(struct timeval *t1,struct timeval *t0, struct timeval *diff);
+int r_timeval_add(struct timeval *t1,struct timeval *t2, struct timeval *sum);
+int r_timeval_cmp(struct timeval *t1,struct timeval *t2);
+
+UINT8 r_timeval2int(struct timeval *tv);
+int r_int2timeval(UINT8 t,struct timeval *tv);
+UINT8 r_gettimeint(void);
+
+/* t1-t0 in microseconds */
+int r_timeval_diff_usec(struct timeval *t1, struct timeval *t0, INT8 *diff);
+
+/* t1-t0 in milliseconds */
+int r_timeval_diff_ms(struct timeval *t1, struct timeval *t0, INT8 *diff);
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_types.h b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_types.h
new file mode 100644
index 0000000000..d00810a7a2
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/libekr/r_types.h
@@ -0,0 +1,213 @@
+/**
+   r_types.h
+
+
+   Copyright (C) 2002-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+ */
+
+/**
+   r_types.h
+
+
+   Copyright (C) 1999-2000 RTFM, Inc.
+   All Rights Reserved
+
+   This package is a SSLv3/TLS protocol analyzer written by Eric Rescorla
+   <ekr@rtfm.com> and licensed by RTFM, Inc.
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. All advertising materials mentioning features or use of this software
+      must display the following acknowledgement:
+
+      This product includes software developed by Eric Rescorla for
+      RTFM, Inc.
+
+   4. Neither the name of RTFM, Inc. nor the name of Eric Rescorla may be
+      used to endorse or promote products derived from this
+      software without specific prior written permission.
+
+   THIS SOFTWARE IS PROVIDED BY ERIC RESCORLA AND RTFM, INC. ``AS IS'' AND
+   ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+   FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+   DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+   OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+   HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+   LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+   OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY SUCH DAMAGE.
+
+   $Id: r_types.h,v 1.2 2006/08/16 19:39:18 adamcain Exp $
+
+
+   ekr@rtfm.com  Tue Dec 22 10:36:02 1998
+ */
+
+
+#ifndef _r_types_h
+#define _r_types_h
+
+/* Either define R_PLATFORM_INT_TYPES or be on a platform that
+   has stdint.h */
+#ifdef R_PLATFORM_INT_TYPES
+#include R_PLATFORM_INT_TYPES
+#else
+#include <stdint.h>
+#endif
+
+#ifndef R_DEFINED_INT2
+#ifndef SIZEOF_INT
+typedef short INT2;
+#else
+# if (SIZEOF_INT==2)
+typedef int INT2;
+# elif (SIZEOF_SHORT==2)
+typedef short INT2;
+# elif (SIZEOF_LONG==2)
+typedef long INT2;
+# else
+# error no type for INT2
+# endif
+#endif
+#else
+typedef R_DEFINED_INT2 INT2;
+#endif
+
+#ifndef R_DEFINED_UINT2
+#ifndef SIZEOF_UNSIGNED_INT
+typedef unsigned short UINT2;
+#else
+# if (SIZEOF_UNSIGNED_INT==2)
+typedef unsigned int UINT2;
+# elif (SIZEOF_UNSIGNED_SHORT==2)
+typedef unsigned short UINT2;
+# elif (SIZEOF_UNSIGNED_LONG==2)
+typedef unsigned long UINT2;
+# else
+# error no type for UINT2
+# endif
+#endif
+#else
+typedef R_DEFINED_UINT2 UINT2;
+#endif
+
+#ifndef R_DEFINED_INT4
+#ifndef SIZEOF_INT
+typedef int INT4;
+#else
+# if (SIZEOF_INT==4)
+typedef int INT4;
+# elif (SIZEOF_SHORT==4)
+typedef short INT4;
+# elif (SIZEOF_LONG==4)
+typedef long INT4;
+# else
+# error no type for INT4
+# endif
+#endif
+#else
+typedef R_DEFINED_INT4 INT4;
+#endif
+
+#ifndef R_DEFINED_UINT4
+#ifndef SIZEOF_UNSIGNED_INT
+typedef unsigned int UINT4;
+#else
+# if (SIZEOF_UNSIGNED_INT==4)
+typedef unsigned int UINT4;
+# elif (SIZEOF_UNSIGNED_SHORT==4)
+typedef unsigned short UINT4;
+# elif (SIZEOF_UNSIGNED_LONG==4)
+typedef unsigned long UINT4;
+# else
+# error no type for UINT4
+# endif
+#endif
+#else
+typedef R_DEFINED_UINT4 UINT4;
+#endif
+
+#ifndef R_DEFINED_INT8
+#ifndef SIZEOF_INT
+typedef long long INT8;
+#else
+# if (SIZEOF_INT==8)
+typedef int INT8;
+# elif (SIZEOF_SHORT==8)
+typedef short INT8;
+# elif (SIZEOF_LONG==8)
+typedef long INT8;
+# elif (SIZEOF_LONG_LONG==8)
+typedef long long INT8;
+# else
+# error no type for INT8
+# endif
+#endif
+#else
+typedef R_DEFINED_INT8 INT8;
+#endif
+
+#ifndef R_DEFINED_UINT8
+#ifndef SIZEOF_UNSIGNED_INT
+typedef unsigned long long UINT8;
+#else
+# if (SIZEOF_UNSIGNED_INT==8)
+typedef unsigned int UINT8;
+# elif (SIZEOF_UNSIGNED_SHORT==8)
+typedef unsigned short UINT8;
+# elif (SIZEOF_UNSIGNED_LONG==8)
+typedef unsigned long UINT8;
+# elif (SIZEOF_UNSIGNED_LONG_LONG==8)
+typedef unsigned long long UINT8;
+# else
+# error no type for UINT8
+# endif
+#endif
+#else
+typedef R_DEFINED_UINT8 UINT8;
+#endif
+
+#ifndef R_DEFINED_UCHAR
+typedef unsigned char UCHAR;
+#else
+typedef R_DEFINED_UCHAR UCHAR;
+#endif
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/p_buf.c b/dom/media/webrtc/transport/third_party/nrappkit/src/util/p_buf.c
new file mode 100644
index 0000000000..459baecdda
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/p_buf.c
@@ -0,0 +1,215 @@
+/**
+   p_buf.c
+
+
+   Copyright (C) 2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   All Rights Reserved.
+
+   ekr@rtfm.com  Tue Nov 25 16:33:08 2003
+ */
+
+#include <string.h>
+#include <stddef.h>
+#include "nr_common.h"
+#include "p_buf.h"
+
+
+static int nr_p_buf_destroy_chain(nr_p_buf_head *head);
+static int nr_p_buf_destroy(nr_p_buf *buf);
+
+int nr_p_buf_ctx_create(size,ctxp)
+  int size;
+  nr_p_buf_ctx **ctxp;
+  {
+    int _status;
+    nr_p_buf_ctx *ctx=0;
+
+    if(!(ctx=(nr_p_buf_ctx *)RCALLOC(sizeof(nr_p_buf_ctx))))
+      ABORT(R_NO_MEMORY);
+
+    ctx->buf_size=size;
+    STAILQ_INIT(&ctx->free_list);
+
+    *ctxp=ctx;
+    _status=0;
+  abort:
+    if(_status){
+      nr_p_buf_ctx_destroy(&ctx);
+    }
+    return(_status);
+  }
+
+int nr_p_buf_ctx_destroy(ctxp)
+  nr_p_buf_ctx **ctxp;
+  {
+    nr_p_buf_ctx *ctx;
+
+    if(!ctxp || !*ctxp)
+      return(0);
+
+    ctx=*ctxp;
+
+    nr_p_buf_destroy_chain(&ctx->free_list);
+
+    RFREE(ctx);
+    *ctxp=0;
+
+    return(0);
+  }
+
+int nr_p_buf_alloc(ctx,bufp)
+  nr_p_buf_ctx *ctx;
+  nr_p_buf **bufp;
+  {
+    int _status;
+    nr_p_buf *buf=0;
+
+    if(!STAILQ_EMPTY(&ctx->free_list)){
+      buf=STAILQ_FIRST(&ctx->free_list);
+      STAILQ_REMOVE_HEAD(&ctx->free_list,entry);
+      goto ok;
+    }
+    else {
+      if(!(buf=(nr_p_buf *)RCALLOC(sizeof(nr_p_buf))))
+        ABORT(R_NO_MEMORY);
+      if(!(buf->data=(UCHAR *)RMALLOC(ctx->buf_size)))
+        ABORT(R_NO_MEMORY);
+      buf->size=ctx->buf_size;
+    }
+
+  ok:
+     buf->r_offset=0;
+     buf->length=0;
+
+     *bufp=buf;
+    _status=0;
+  abort:
+     if(_status){
+       nr_p_buf_destroy(buf);
+     }
+    return(_status);
+  }
+
+int nr_p_buf_free(ctx,buf)
+  nr_p_buf_ctx *ctx;
+  nr_p_buf *buf;
+  {
+    STAILQ_INSERT_TAIL(&ctx->free_list,buf,entry);
+
+    return(0);
+  }
+
+int nr_p_buf_free_chain(ctx,head)
+  nr_p_buf_ctx *ctx;
+  nr_p_buf_head *head;
+  {
+    nr_p_buf *n1,*n2;
+
+    n1=STAILQ_FIRST(head);
+    while(n1){
+      n2=STAILQ_NEXT(n1,entry);
+
+      nr_p_buf_free(ctx,n1);
+
+      n1=n2;
+    }
+
+    return(0);
+  }
+
+
+int nr_p_buf_write_to_chain(ctx,chain,data,len)
+  nr_p_buf_ctx *ctx;
+  nr_p_buf_head *chain;
+  UCHAR *data;
+  UINT4 len;
+  {
+    int r,_status;
+    nr_p_buf *buf;
+
+    buf=STAILQ_LAST(chain,nr_p_buf_,entry);
+    while(len){
+      int towrite;
+
+      if(!buf){
+        if(r=nr_p_buf_alloc(ctx,&buf))
+          ABORT(r);
+        STAILQ_INSERT_TAIL(chain,buf,entry);
+      }
+
+      towrite=MIN(len,(buf->size-(buf->length+buf->r_offset)));
+
+      memcpy(buf->data+buf->length+buf->r_offset,data,towrite);
+      len-=towrite;
+      data+=towrite;
+      buf->length+=towrite;
+
+      r_log(LOG_COMMON,LOG_DEBUG,"Wrote %d bytes to buffer %p",towrite,buf);
+      buf=0;
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+static int nr_p_buf_destroy_chain(head)
+  nr_p_buf_head *head;
+  {
+    nr_p_buf *n1,*n2;
+
+    n1=STAILQ_FIRST(head);
+    while(n1){
+      n2=STAILQ_NEXT(n1,entry);
+
+      nr_p_buf_destroy(n1);
+
+      n1=n2;
+    }
+
+    return(0);
+  }
+
+static int nr_p_buf_destroy(buf)
+  nr_p_buf *buf;
+  {
+    if(!buf)
+      return(0);
+
+    RFREE(buf->data);
+    RFREE(buf);
+
+    return(0);
+  }
+
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/p_buf.h b/dom/media/webrtc/transport/third_party/nrappkit/src/util/p_buf.h
new file mode 100644
index 0000000000..29881960c6
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/p_buf.h
@@ -0,0 +1,72 @@
+/**
+   p_buf.h
+
+
+   Copyright (C) 2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   ekr@rtfm.com  Tue Nov 25 15:58:37 2003
+ */
+
+
+#ifndef _p_buf_h
+#define _p_buf_h
+
+typedef struct nr_p_buf_ {
+     UCHAR *data;     /*The pointer to the buffer where the data lives */
+     UINT4 size;      /*The size of the buffer */
+     UINT4 r_offset;  /*The offset into the buffer where the data starts
+                        when reading */
+     UINT4 length;    /*The length of the data portion */
+
+     STAILQ_ENTRY(nr_p_buf_) entry;
+} nr_p_buf;
+
+typedef STAILQ_HEAD(nr_p_buf_head_,nr_p_buf_) nr_p_buf_head;
+
+
+typedef struct nr_p_buf_ctx_ {
+     int buf_size;
+
+     nr_p_buf_head free_list;
+} nr_p_buf_ctx;
+
+int nr_p_buf_ctx_create(int size,nr_p_buf_ctx **ctxp);
+int nr_p_buf_ctx_destroy(nr_p_buf_ctx **ctxp);
+int nr_p_buf_alloc(nr_p_buf_ctx *ctx,nr_p_buf **bufp);
+int nr_p_buf_free(nr_p_buf_ctx *ctx,nr_p_buf *buf);
+int nr_p_buf_free_chain(nr_p_buf_ctx *ctx,nr_p_buf_head *chain);
+int nr_p_buf_write_to_chain(nr_p_buf_ctx *ctx,
+                                       nr_p_buf_head *chain,
+                                       UCHAR *data,UINT4 len);
+
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/util.c b/dom/media/webrtc/transport/third_party/nrappkit/src/util/util.c
new file mode 100644
index 0000000000..17d49639fd
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/util.c
@@ -0,0 +1,775 @@
+/**
+   util.c
+
+
+   Copyright (C) 2001-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   ekr@rtfm.com  Wed Dec 26 17:19:36 2001
+ */
+
+#ifndef WIN32
+#include <sys/uio.h>
+#include <pwd.h>
+#include <dirent.h>
+#endif
+#include <string.h>
+#include <errno.h>
+#include <ctype.h>
+#include <sys/stat.h>
+#ifdef OPENSSL
+#include <openssl/evp.h>
+#endif
+#include "nr_common.h"
+#include "r_common.h"
+#include "registry.h"
+#include "util.h"
+#include "r_log.h"
+
+int nr_util_default_log_facility=LOG_COMMON;
+
+int nr_get_filename(base,name,namep)
+  char *base;
+  char *name;
+  char **namep;
+  {
+    int len=strlen(base)+strlen(name)+2;
+    char *ret=0;
+    int _status;
+
+    if(!(ret=(char *)RMALLOC(len)))
+      ABORT(R_NO_MEMORY);
+    if(base[strlen(base)-1]!='/'){
+      sprintf(ret,"%s/%s",base,name);
+    }
+    else{
+      sprintf(ret,"%s%s",base,name);
+    }
+    *namep=ret;
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+#if 0
+int read_RSA_private_key(base,name,keyp)
+  char *base;
+  char *name;
+  RSA **keyp;
+  {
+    char *keyfile=0;
+    BIO *bio=0;
+    FILE *fp=0;
+    RSA *rsa=0;
+    int r,_status;
+
+    /* Load the keyfile */
+    if(r=get_filename(base,name,&keyfile))
+      ABORT(r);
+    if(!(fp=fopen(keyfile,"r")))
+      ABORT(R_NOT_FOUND);
+    if(!(bio=BIO_new(BIO_s_file())))
+      ABORT(R_NO_MEMORY);
+    BIO_set_fp(bio,fp,BIO_NOCLOSE);
+
+    if(!(rsa=PEM_read_bio_RSAPrivateKey(bio,0,0,0)))
+      ABORT(R_NOT_FOUND);
+
+    *keyp=rsa;
+    _status=0;
+  abort:
+    return(_status);
+  }
+#endif
+
+
+void nr_errprintf_log(const char *format,...)
+  {
+    va_list ap;
+
+    va_start(ap,format);
+
+    r_vlog(nr_util_default_log_facility,LOG_ERR,format,ap);
+
+    va_end(ap);
+  }
+
+void nr_errprintf_log2(void *ignore, const char *format,...)
+  {
+    va_list ap;
+
+    va_start(ap,format);
+
+    r_vlog(nr_util_default_log_facility,LOG_ERR,format,ap);
+
+    va_end(ap);
+  }
+
+
+int nr_fwrite_all(FILE *fp,UCHAR *buf,int len)
+  {
+    int r,_status;
+
+    while(len){
+      r=fwrite(buf,1,len,fp);
+      if(r==0)
+        ABORT(R_IO_ERROR);
+
+      len-=r;
+      buf+=r;
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+int nr_read_data(fd,buf,len)
+  int fd;
+  char *buf;
+  int len;
+  {
+    int r,_status;
+
+    while(len){
+      r=NR_SOCKET_READ(fd,buf,len);
+      if(r<=0)
+        ABORT(R_EOD);
+
+      buf+=r;
+      len-=r;
+    }
+
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+
+#ifdef WIN32
+  // TODO
+#else
+int nr_drop_privileges(char *username)
+  {
+    int _status;
+
+    /* Drop privileges */
+    if ((getuid() == 0) || geteuid()==0) {
+      struct passwd *passwd;
+
+      if ((passwd = getpwnam(CAPTURE_USER)) == 0){
+        r_log(LOG_GENERIC,LOG_EMERG,"Couldn't get user %s",CAPTURE_USER);
+        ABORT(R_INTERNAL);
+      }
+
+      if(setuid(passwd->pw_uid)!=0){
+        r_log(LOG_GENERIC,LOG_EMERG,"Couldn't drop privileges");
+        ABORT(R_INTERNAL);
+      }
+    }
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+#endif
+
+int nr_bin2hex(UCHAR *in,int len,UCHAR *out)
+  {
+    while(len){
+      sprintf((char*)out,"%.2x",in[0] & 0xff);
+
+      in+=1;
+      out+=2;
+
+      len--;
+    }
+
+    return(0);
+  }
+
+int nr_hex_ascii_dump(Data *data)
+  {
+    UCHAR *ptr=data->data;
+    int len=data->len;
+
+    while(len){
+      int i;
+      int bytes=MIN(len,16);
+
+      for(i=0;i<bytes;i++)
+        printf("%.2x ",ptr[i]&255);
+      /* Fill */
+      for(i=0;i<(16-bytes);i++)
+        printf("   ");
+      printf("   ");
+
+      for(i=0;i<bytes;i++){
+        if(isprint(ptr[i]))
+          printf("%c",ptr[i]);
+        else
+          printf(".");
+      }
+      printf("\n");
+
+      len-=bytes;
+      ptr+=bytes;
+    }
+    return(0);
+  }
+
+#ifdef OPENSSL
+int nr_sha1_file(char *filename,UCHAR *out)
+  {
+    EVP_MD_CTX md_ctx;
+    FILE *fp=0;
+    int r,_status;
+    UCHAR buf[1024];
+    int out_len;
+
+    EVP_MD_CTX_init(&md_ctx);
+
+    if(!(fp=fopen(filename,"r"))){
+      r_log(LOG_COMMON,LOG_ERR,"Couldn't open file %s",filename);
+      ABORT(R_NOT_FOUND);
+    }
+
+    EVP_DigestInit_ex(&md_ctx,EVP_sha1(),0);
+
+    while(1){
+      r=fread(buf,1,sizeof(buf),fp);
+
+      if(r<0){
+        r_log(LOG_COMMON,LOG_ERR,"Error reading from %s",filename);
+        ABORT(R_INTERNAL);
+      }
+
+      if(!r)
+        break;
+
+      EVP_DigestUpdate(&md_ctx,buf,r);
+    }
+
+    EVP_DigestFinal(&md_ctx,out,(unsigned int*)&out_len);
+    if(out_len!=20)
+      ABORT(R_INTERNAL);
+
+    _status=0;
+  abort:
+    EVP_MD_CTX_cleanup(&md_ctx);
+    if(fp) fclose(fp);
+
+    return(_status);
+  }
+
+#endif
+
+#ifdef WIN32
+  // TODO
+#else
+
+#if 0
+
+#include <fts.h>
+
+int nr_rm_tree(char *path)
+  {
+    FTS *fts=0;
+    FTSENT *p;
+    int failed=0;
+    int _status;
+    char *argv[2];
+
+    argv[0]=path;
+    argv[1]=0;
+
+    if(!(fts=fts_open(argv,0,NULL))){
+      r_log_e(LOG_COMMON,LOG_ERR,"Couldn't open directory %s",path);
+      ABORT(R_FAILED);
+    }
+
+    while(p=fts_read(fts)){
+      switch(p->fts_info){
+        case FTS_D:
+          break;
+        case FTS_DOT:
+          break;
+        case FTS_ERR:
+          r_log_e(LOG_COMMON,LOG_ERR,"Problem reading %s",p->fts_path);
+          break;
+        default:
+          r_log(LOG_COMMON,LOG_DEBUG,"Removing %s",p->fts_path);
+          errno=0;
+          if(remove(p->fts_path)){
+            r_log_e(LOG_COMMON,LOG_ERR,"Problem removing %s",p->fts_path);
+            failed=1;
+          }
+      }
+    }
+
+    if(failed)
+      ABORT(R_FAILED);
+
+    _status=0;
+  abort:
+    if(fts) fts_close(fts);
+    return(_status);
+  }
+#endif
+
+int nr_write_pid_file(char *pid_filename)
+  {
+    FILE *fp;
+    int _status;
+
+    if(!pid_filename)
+      ABORT(R_BAD_ARGS);
+
+    unlink(pid_filename);
+
+    if(!(fp=fopen(pid_filename,"w"))){
+      r_log(LOG_GENERIC,LOG_CRIT,"Couldn't open PID file: %s",strerror(errno));
+      ABORT(R_NOT_FOUND);
+    }
+
+    fprintf(fp,"%d\n",getpid());
+
+    fclose(fp);
+
+    chmod(pid_filename,S_IRUSR | S_IRGRP | S_IROTH);
+
+    _status=0;
+  abort:
+    return(_status);
+  }
+#endif
+
+int nr_reg_uint4_fetch_and_check(NR_registry key, UINT4 min, UINT4 max, int log_fac, int die, UINT4 *val)
+  {
+    int r,_status;
+    UINT4 my_val;
+
+    if(r=NR_reg_get_uint4(key,&my_val)){
+      r_log(log_fac,LOG_ERR,"Couldn't get key '%s', error %d",key,r);
+      ABORT(r);
+    }
+
+    if((min>0) && (my_val<min)){
+      r_log(log_fac,LOG_ERR,"Invalid value for key '%s'=%lu, (min = %lu)",key,(unsigned long)my_val,(unsigned long)min);
+      ABORT(R_BAD_DATA);
+    }
+
+    if(my_val>max){
+      r_log(log_fac,LOG_ERR,"Invalid value for key '%s'=%lu, (max = %lu)",key,(unsigned long)my_val,(unsigned long)max);
+      ABORT(R_BAD_DATA);
+    }
+
+    *val=my_val;
+    _status=0;
+
+  abort:
+    if(die && _status){
+      r_log(log_fac,LOG_CRIT,"Exiting due to invalid configuration (key '%s')",key);
+      exit(1);
+    }
+    return(_status);
+  }
+
+int nr_reg_uint8_fetch_and_check(NR_registry key, UINT8 min, UINT8 max, int log_fac, int die, UINT8 *val)
+  {
+    int r,_status;
+    UINT8 my_val;
+
+    if(r=NR_reg_get_uint8(key,&my_val)){
+      r_log(log_fac,LOG_ERR,"Couldn't get key '%s', error %d",key,r);
+      ABORT(r);
+    }
+
+    if(my_val<min){
+      r_log(log_fac,LOG_ERR,"Invalid value for key '%s'=%llu, (min = %llu)",key,my_val,min);
+      ABORT(R_BAD_DATA);
+    }
+
+    if(my_val>max){
+      r_log(log_fac,LOG_ERR,"Invalid value for key '%s'=%llu, (max = %llu)",key,my_val,max);
+      ABORT(R_BAD_DATA);
+    }
+
+    *val=my_val;
+    _status=0;
+
+  abort:
+    if(die && _status){
+      r_log(log_fac,LOG_CRIT,"Exiting due to invalid configuration (key '%s')",key);
+      exit(1);
+    }
+    return(_status);
+  }
+
+#if defined(LINUX) || defined(WIN32)
+/*-
+ * Copyright (c) 1998 Todd C. Miller <Todd.Miller@courtesan.com>
+ * All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ *    notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ *    notice, this list of conditions and the following disclaimer in the
+ *    documentation and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ *    derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES,
+ * INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY
+ * AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL
+ * THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+
+/*
+ * Appends src to string dst of size siz (unlike strncat, siz is the
+ * full size of dst, not space left).  At most siz-1 characters
+ * will be copied.  Always NUL terminates (unless siz <= strlen(dst)).
+ * Returns strlen(src) + MIN(siz, strlen(initial dst)).
+ * If retval >= siz, truncation occurred.
+ */
+size_t
+strlcat(dst, src, siz)
+        char *dst;
+        const char *src;
+        size_t siz;
+{
+        char *d = dst;
+        const char *s = src;
+        size_t n = siz;
+        size_t dlen;
+
+        /* Find the end of dst and adjust bytes left but don't go past end */
+        while (n-- != 0 && *d != '\0')
+                d++;
+        dlen = d - dst;
+        n = siz - dlen;
+
+        if (n == 0)
+                return(dlen + strlen(s));
+        while (*s != '\0') {
+                if (n != 1) {
+                        *d++ = *s;
+                        n--;
+                }
+                s++;
+        }
+        *d = '\0';
+
+        return(dlen + (s - src));       /* count does not include NUL */
+}
+
+#endif  /* LINUX or WIN32 */
+
+#if defined(USE_OWN_INET_NTOP) || (defined(WIN32) && WINVER < 0x0600)
+#include <errno.h>
+#ifdef WIN32
+#include <Ws2ipdef.h>
+#ifndef EAFNOSUPPORT
+#define EAFNOSUPPORT            WSAEAFNOSUPPORT
+#endif
+#else
+#include <sys/socket.h>
+#endif
+#define INET6
+
+/* inet_ntop implementation from NetBSD */
+
+/*
+ * Copyright (c) 2004 by Internet Systems Consortium, Inc. ("ISC")
+ * Copyright (c) 1996-1999 by Internet Software Consortium.
+ *
+ * Permission to use, copy, modify, and distribute this software for any
+ * purpose with or without fee is hereby granted, provided that the above
+ * copyright notice and this permission notice appear in all copies.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS" AND ISC DISCLAIMS ALL WARRANTIES
+ * WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS.  IN NO EVENT SHALL ISC BE LIABLE FOR
+ * ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
+ * WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
+ * ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT
+ * OF OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
+ */
+
+#if !defined(NS_INADDRSZ)
+# define NS_INADDRSZ  4
+#endif
+#if !defined(NS_IN6ADDRSZ)
+# define NS_IN6ADDRSZ  16
+#endif
+#if !defined(NS_INT16SZ)
+# define NS_INT16SZ  2
+#endif
+
+/*
+ * WARNING: Don't even consider trying to compile this on a system where
+ * sizeof(int) < 4.  sizeof(int) > 4 is fine; all the world's not a VAX.
+ */
+
+static const char *inet_ntop4(const unsigned char *src, char *dst, size_t size);
+#ifdef INET6
+static const char *inet_ntop6(const unsigned char *src, char *dst, size_t size);
+#endif /* INET6 */
+
+/* char *
+ * inet_ntop(af, src, dst, size)
+ *  convert a network format address to presentation format.
+ * return:
+ *  pointer to presentation format address (`dst'), or NULL (see errno).
+ * author:
+ *  Paul Vixie, 1996.
+ */
+const char *
+inet_ntop(int af, const void *src, char *dst, size_t size)
+{
+
+  switch (af) {
+  case AF_INET:
+    return (inet_ntop4(src, dst, size));
+#ifdef INET6
+  case AF_INET6:
+    return (inet_ntop6(src, dst, size));
+#endif /* INET6 */
+  default:
+    errno = EAFNOSUPPORT;
+    return (NULL);
+  }
+  /* NOTREACHED */
+}
+
+/* const char *
+ * inet_ntop4(src, dst, size)
+ *  format an IPv4 address, more or less like inet_ntoa()
+ * return:
+ *  `dst' (as a const)
+ * notes:
+ *  (1) uses no statics
+ *  (2) takes a unsigned char* not an in_addr as input
+ * author:
+ *  Paul Vixie, 1996.
+ */
+static const char *
+inet_ntop4(const unsigned char *src, char *dst, size_t size)
+{
+  char tmp[sizeof "255.255.255.255"];
+  int l;
+
+  l = snprintf(tmp, sizeof(tmp), "%u.%u.%u.%u",
+      src[0], src[1], src[2], src[3]);
+  if (l <= 0 || (size_t) l >= size) {
+    errno = ENOSPC;
+    return (NULL);
+  }
+  strlcpy(dst, tmp, size);
+  return (dst);
+}
+
+#ifdef INET6
+/* const char *
+ * inet_ntop6(src, dst, size)
+ *  convert IPv6 binary address into presentation (printable) format
+ * author:
+ *  Paul Vixie, 1996.
+ */
+static const char *
+inet_ntop6(const unsigned char *src, char *dst, size_t size)
+{
+  /*
+   * Note that int32_t and int16_t need only be "at least" large enough
+   * to contain a value of the specified size.  On some systems, like
+   * Crays, there is no such thing as an integer variable with 16 bits.
+   * Keep this in mind if you think this function should have been coded
+   * to use pointer overlays.  All the world's not a VAX.
+   */
+  char tmp[sizeof "ffff:ffff:ffff:ffff:ffff:ffff:255.255.255.255"];
+  char *tp, *ep;
+  struct { int base, len; } best, cur;
+  unsigned int words[NS_IN6ADDRSZ / NS_INT16SZ];
+  int i;
+  int advance;
+
+  /*
+   * Preprocess:
+   *  Copy the input (bytewise) array into a wordwise array.
+   *  Find the longest run of 0x00's in src[] for :: shorthanding.
+   */
+  memset(words, '\0', sizeof words);
+  for (i = 0; i < NS_IN6ADDRSZ; i++)
+    words[i / 2] |= (src[i] << ((1 - (i % 2)) << 3));
+  best.base = -1;
+  cur.base = -1;
+  best.len = -1;  /* XXX gcc */
+  cur.len = -1;  /* XXX gcc */
+  for (i = 0; i < (NS_IN6ADDRSZ / NS_INT16SZ); i++) {
+    if (words[i] == 0) {
+      if (cur.base == -1)
+        cur.base = i, cur.len = 1;
+      else
+        cur.len++;
+    } else {
+      if (cur.base != -1) {
+        if (best.base == -1 || cur.len > best.len)
+          best = cur;
+        cur.base = -1;
+      }
+    }
+  }
+  if (cur.base != -1) {
+    if (best.base == -1 || cur.len > best.len)
+      best = cur;
+  }
+  if (best.base != -1 && best.len < 2)
+    best.base = -1;
+
+  /*
+   * Format the result.
+   */
+  tp = tmp;
+  ep = tmp + sizeof(tmp);
+  for (i = 0; i < (NS_IN6ADDRSZ / NS_INT16SZ); i++) {
+    /* Are we inside the best run of 0x00's? */
+    if (best.base != -1 && i >= best.base &&
+        i < (best.base + best.len)) {
+      if (i == best.base)
+        *tp++ = ':';
+      continue;
+    }
+    /* Are we following an initial run of 0x00s or any real hex? */
+    if (i != 0) {
+      if (tp + 1 >= ep)
+        return (NULL);
+      *tp++ = ':';
+    }
+    /* Is this address an encapsulated IPv4? */
+    if (i == 6 && best.base == 0 &&
+        (best.len == 6 ||
+        (best.len == 7 && words[7] != 0x0001) ||
+        (best.len == 5 && words[5] == 0xffff))) {
+      if (!inet_ntop4(src+12, tp, (size_t)(ep - tp)))
+        return (NULL);
+      tp += strlen(tp);
+      break;
+    }
+    advance = snprintf(tp, (size_t)(ep - tp), "%x", words[i]);
+    if (advance <= 0 || advance >= ep - tp)
+      return (NULL);
+    tp += advance;
+  }
+  /* Was it a trailing run of 0x00's? */
+  if (best.base != -1 && (best.base + best.len) ==
+      (NS_IN6ADDRSZ / NS_INT16SZ)) {
+    if (tp + 1 >= ep)
+      return (NULL);
+    *tp++ = ':';
+  }
+  if (tp + 1 >= ep)
+    return (NULL);
+  *tp++ = '\0';
+
+  /*
+   * Check for overflow, copy, and we're done.
+   */
+  if ((size_t)(tp - tmp) > size) {
+    errno = ENOSPC;
+    return (NULL);
+  }
+  strlcpy(dst, tmp, size);
+  return (dst);
+}
+#endif /* INET6 */
+
+#ifdef WIN32
+/* Not exactly, will forgive stuff like <addr>:<port> */
+int inet_pton(int af, const char *src, void *dst)
+{
+  struct sockaddr_storage ss;
+  int addrlen = sizeof(ss);
+
+  if (af != AF_INET && af != AF_INET6) {
+    return -1;
+  }
+
+  if (!WSAStringToAddressA(src, af, NULL, (struct sockaddr*)&ss, &addrlen)) {
+    if (af == AF_INET) {
+      struct sockaddr_in *in = (struct sockaddr_in*)&ss;
+      memcpy(dst, &in->sin_addr, sizeof(struct in_addr));
+    } else {
+      struct sockaddr_in6 *in6 = (struct sockaddr_in6*)&ss;
+      memcpy(dst, &in6->sin6_addr, sizeof(struct in6_addr));
+    }
+    return 1;
+  }
+  return 0;
+}
+#endif /* WIN32 */
+
+#endif
+
+#ifdef WIN32
+#include <time.h>
+/* this is only millisecond-accurate, but that should be OK */
+
+int gettimeofday(struct timeval *tv, void *tz)
+  {
+    SYSTEMTIME st;
+    FILETIME ft;
+    ULARGE_INTEGER u;
+
+    GetLocalTime (&st);
+
+    /* strangely, the FILETIME is the number of 100 nanosecond (0.1 us) intervals
+     * since the Epoch */
+    SystemTimeToFileTime(&st, &ft);
+    u.HighPart = ft.dwHighDateTime;
+    u.LowPart = ft.dwLowDateTime;
+
+    tv->tv_sec = (long) (u.QuadPart / 10000000L);
+    tv->tv_usec = (long) (st.wMilliseconds * 1000);;
+
+    return 0;
+  }
+#endif
+
diff --git a/dom/media/webrtc/transport/third_party/nrappkit/src/util/util.h b/dom/media/webrtc/transport/third_party/nrappkit/src/util/util.h
new file mode 100644
index 0000000000..975baa4aa2
--- /dev/null
+++ b/dom/media/webrtc/transport/third_party/nrappkit/src/util/util.h
@@ -0,0 +1,73 @@
+/**
+   util.h
+
+
+   Copyright (C) 2001-2003, Network Resonance, Inc.
+   Copyright (C) 2006, Network Resonance, Inc.
+   All Rights Reserved
+
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   1. Redistributions of source code must retain the above copyright
+      notice, this list of conditions and the following disclaimer.
+   2. Redistributions in binary form must reproduce the above copyright
+      notice, this list of conditions and the following disclaimer in the
+      documentation and/or other materials provided with the distribution.
+   3. Neither the name of Network Resonance, Inc. nor the name of any
+      contributors to this software may be used to endorse or promote
+      products derived from this software without specific prior written
+      permission.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS''
+   AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+   IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+   ARE DISCLAIMED.  IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+   LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+   CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+   SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+   INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+   CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+   ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+   POSSIBILITY OF SUCH DAMAGE.
+
+
+   ekr@rtfm.com  Wed Dec 26 17:20:23 2001
+ */
+
+
+#ifndef _util_h
+#define _util_h
+
+#include "registry.h"
+
+int nr_get_filename(char *base,char *name, char **namep);
+#if 0
+#include <openssl/ssl.h>
+
+int read_RSA_private_key(char *base, char *name,RSA **keyp);
+#endif
+void nr_errprintf_log(const char *fmt,...);
+void nr_errprintf_log2(void *ignore, const char *fmt,...);
+extern int nr_util_default_log_facility;
+
+int nr_read_data(int fd,char *buf,int len);
+int nr_drop_privileges(char *username);
+int nr_hex_ascii_dump(Data *data);
+int nr_fwrite_all(FILE *fp,UCHAR *buf,int len);
+int nr_sha1_file(char *filename,UCHAR *out);
+int nr_bin2hex(UCHAR *in,int len,UCHAR *out);
+int nr_rm_tree(char *path);
+int nr_write_pid_file(char *pid_filename);
+
+int nr_reg_uint4_fetch_and_check(NR_registry key, UINT4 min, UINT4 max, int log_fac, int die, UINT4 *val);
+int nr_reg_uint8_fetch_and_check(NR_registry key, UINT8 min, UINT8 max, int log_fac, int die, UINT8 *val);
+
+#if defined(WIN32) && WINVER < 0x0600
+const char *inet_ntop(int af, const void *src, char *dst, size_t size);
+int inet_pton(int af, const char *src, void *dst);
+#endif
+
+#endif
+
diff --git a/dom/media/webrtc/transport/transportflow.cpp b/dom/media/webrtc/transport/transportflow.cpp
new file mode 100644
index 0000000000..8b263c8cfe
--- /dev/null
+++ b/dom/media/webrtc/transport/transportflow.cpp
@@ -0,0 +1,74 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+#include <deque>
+
+#include "transportflow.h"
+#include "transportlayer.h"
+
+namespace mozilla {
+
+NS_IMPL_ISUPPORTS0(TransportFlow)
+
+// There are some hacks here to allow destruction off of
+// the main thread.
+TransportFlow::~TransportFlow() {
+  // Push the destruction onto the STS thread. Note that there
+  // is still some possibility that someone is accessing this
+  // object simultaneously, but as long as smart pointer discipline
+  // is maintained, it shouldn't be possible to access and
+  // destroy it simultaneously. The conversion to a UniquePtr
+  // ensures automatic destruction of the queue at exit of
+  // DestroyFinal.
+  CheckThread();
+  ClearLayers(layers_.get());
+}
+
+void TransportFlow::DestroyFinal(
+    UniquePtr<std::deque<TransportLayer*>> layers) {
+  ClearLayers(layers.get());
+}
+
+void TransportFlow::ClearLayers(std::deque<TransportLayer*>* layers) {
+  while (!layers->empty()) {
+    delete layers->front();
+    layers->pop_front();
+  }
+}
+
+void TransportFlow::PushLayer(TransportLayer* layer) {
+  CheckThread();
+  layers_->push_front(layer);
+  EnsureSameThread(layer);
+  layer->SetFlowId(id_);
+}
+
+TransportLayer* TransportFlow::GetLayer(const std::string& id) const {
+  CheckThread();
+
+  if (layers_) {
+    for (TransportLayer* layer : *layers_) {
+      if (layer->id() == id) return layer;
+    }
+  }
+
+  return nullptr;
+}
+
+void TransportFlow::EnsureSameThread(TransportLayer* layer) {
+  // Enforce that if any of the layers have a thread binding,
+  // they all have the same binding.
+  if (target_) {
+    const nsCOMPtr<nsIEventTarget>& lthread = layer->GetThread();
+
+    if (lthread && (lthread != target_)) MOZ_CRASH();
+  } else {
+    target_ = layer->GetThread();
+  }
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/transport/transportflow.h b/dom/media/webrtc/transport/transportflow.h
new file mode 100644
index 0000000000..43253a260b
--- /dev/null
+++ b/dom/media/webrtc/transport/transportflow.h
@@ -0,0 +1,105 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+#ifndef transportflow_h__
+#define transportflow_h__
+
+#include <deque>
+#include <string>
+
+#include "nscore.h"
+#include "nsISupportsImpl.h"
+#include "mozilla/UniquePtr.h"
+#include "transportlayer.h"
+#include "m_cpp_utils.h"
+
+// A stack of transport layers acts as a flow.
+// Generally, one reads and writes to the top layer.
+
+// This code has a confusing hybrid threading model which
+// probably needs some eventual refactoring.
+// TODO(ekr@rtfm.com): Bug 844891
+//
+// TransportFlows are not inherently bound to a thread *but*
+// TransportLayers can be. If any layer in a flow is bound
+// to a given thread, then all layers in the flow MUST be
+// bound to that thread and you can only manipulate the
+// flow (push layers, write, etc.) on that thread.
+//
+// The sole official exception to this is that you are
+// allowed to *destroy* a flow off the bound thread provided
+// that there are no listeners on its signals. This exception
+// is designed to allow idioms where you create the flow
+// and then something goes wrong and you destroy it and
+// you don't want to bother with a thread dispatch.
+//
+// Eventually we hope to relax the "no listeners"
+// restriction by thread-locking the signals, but previous
+// attempts have caused deadlocks.
+//
+// Most of these invariants are enforced by hard asserts
+// (i.e., those which fire even in production builds).
+
+namespace mozilla {
+
+class TransportFlow final : public nsISupports {
+ public:
+  TransportFlow()
+      : id_("(anonymous)"), layers_(new std::deque<TransportLayer*>) {}
+  explicit TransportFlow(const std::string id)
+      : id_(id), layers_(new std::deque<TransportLayer*>) {}
+
+  const std::string& id() const { return id_; }
+
+  // Layer management. Note PushLayer() is not thread protected, so
+  // either:
+  // (a) Do it in the thread handling the I/O
+  // (b) Do it before you activate the I/O system
+  //
+  // The flow takes ownership of the layers after a successful
+  // push.
+  void PushLayer(TransportLayer* layer);
+
+  TransportLayer* GetLayer(const std::string& id) const;
+
+  NS_DECL_THREADSAFE_ISUPPORTS
+
+ private:
+  ~TransportFlow();
+
+  DISALLOW_COPY_ASSIGN(TransportFlow);
+
+  // Check if we are on the right thread
+  void CheckThread() const {
+    if (!CheckThreadInt()) MOZ_CRASH();
+  }
+
+  bool CheckThreadInt() const {
+    bool on;
+
+    if (!target_)  // OK if no thread set.
+      return true;
+    if (NS_FAILED(target_->IsOnCurrentThread(&on))) return false;
+
+    return on;
+  }
+
+  void EnsureSameThread(TransportLayer* layer);
+
+  static void DestroyFinal(UniquePtr<std::deque<TransportLayer*>> layers);
+
+  // Overload needed because we use deque internally and queue externally.
+  static void ClearLayers(std::deque<TransportLayer*>* layers);
+
+  std::string id_;
+  UniquePtr<std::deque<TransportLayer*>> layers_;
+  nsCOMPtr<nsIEventTarget> target_;
+};
+
+}  // namespace mozilla
+#endif
diff --git a/dom/media/webrtc/transport/transportlayer.cpp b/dom/media/webrtc/transport/transportlayer.cpp
new file mode 100644
index 0000000000..8816f24754
--- /dev/null
+++ b/dom/media/webrtc/transport/transportlayer.cpp
@@ -0,0 +1,49 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+#include "logging.h"
+#include "transportlayer.h"
+
+// Logging context
+namespace mozilla {
+
+MOZ_MTLOG_MODULE("mtransport")
+
+nsresult TransportLayer::Init() {
+  if (state_ != TS_NONE) return state_ == TS_ERROR ? NS_ERROR_FAILURE : NS_OK;
+
+  nsresult rv = InitInternal();
+
+  if (!NS_SUCCEEDED(rv)) {
+    state_ = TS_ERROR;
+    return rv;
+  }
+  state_ = TS_INIT;
+
+  return NS_OK;
+}
+
+void TransportLayer::Chain(TransportLayer* downward) {
+  downward_ = downward;
+  MOZ_MTLOG(ML_DEBUG, LAYER_INFO << "Inserted: downward='"
+                                 << (downward ? downward->id() : "none")
+                                 << "'");
+
+  WasInserted();
+}
+
+void TransportLayer::SetState(State state, const char* file, unsigned line) {
+  if (state != state_) {
+    MOZ_MTLOG(state == TS_ERROR ? ML_ERROR : ML_DEBUG,
+              file << ":" << line << ": " << LAYER_INFO << "state " << state_
+                   << "->" << state);
+    state_ = state;
+    SignalStateChange(this, state);
+  }
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/transport/transportlayer.h b/dom/media/webrtc/transport/transportlayer.h
new file mode 100644
index 0000000000..47287ab4e0
--- /dev/null
+++ b/dom/media/webrtc/transport/transportlayer.h
@@ -0,0 +1,108 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+#ifndef transportlayer_h__
+#define transportlayer_h__
+
+#include "sigslot.h"
+
+#include "nsCOMPtr.h"
+#include "nsIEventTarget.h"
+
+#include "m_cpp_utils.h"
+#include "mediapacket.h"
+
+namespace mozilla {
+
+class TransportFlow;
+
+typedef int TransportResult;
+
+enum { TE_WOULDBLOCK = -1, TE_ERROR = -2, TE_INTERNAL = -3 };
+
+#define TRANSPORT_LAYER_ID(name)                         \
+  const std::string id() const override { return name; } \
+  static std::string ID() { return name; }
+
+// Abstract base class for network transport layers.
+class TransportLayer : public sigslot::has_slots<> {
+ public:
+  // The state of the transport flow
+  // We can't use "ERROR" because Windows has a macro named "ERROR"
+  enum State { TS_NONE, TS_INIT, TS_CONNECTING, TS_OPEN, TS_CLOSED, TS_ERROR };
+
+  // Is this a stream or datagram flow
+  TransportLayer() : state_(TS_NONE), flow_id_(), downward_(nullptr) {}
+
+  virtual ~TransportLayer() = default;
+
+  // Called to initialize
+  nsresult Init();  // Called by Insert() to set up -- do not override
+  virtual nsresult InitInternal() { return NS_OK; }  // Called by Init
+
+  void SetFlowId(const std::string& flow_id) { flow_id_ = flow_id; }
+
+  virtual void Chain(TransportLayer* downward);
+
+  // Downward interface
+  TransportLayer* downward() { return downward_; }
+
+  // Get the state
+  State state() const { return state_; }
+  // Must be implemented by derived classes
+  virtual TransportResult SendPacket(MediaPacket& packet) = 0;
+
+  // Get the thread.
+  const nsCOMPtr<nsIEventTarget> GetThread() const { return target_; }
+
+  // Event definitions that one can register for
+  // State has changed
+  sigslot::signal2<TransportLayer*, State> SignalStateChange;
+  // Data received on the flow
+  sigslot::signal2<TransportLayer*, MediaPacket&> SignalPacketReceived;
+
+  // Return the layer id for this layer
+  virtual const std::string id() const = 0;
+
+  // The id of the flow
+  const std::string& flow_id() const { return flow_id_; }
+
+ protected:
+  virtual void WasInserted() {}
+  virtual void SetState(State state, const char* file, unsigned line);
+  // Check if we are on the right thread
+  void CheckThread() const { MOZ_ASSERT(CheckThreadInt(), "Wrong thread"); }
+
+  State state_;
+  std::string flow_id_;
+  TransportLayer* downward_;  // The next layer in the stack
+  nsCOMPtr<nsIEventTarget> target_;
+
+ private:
+  DISALLOW_COPY_ASSIGN(TransportLayer);
+
+  bool CheckThreadInt() const {
+    bool on;
+
+    if (!target_)  // OK if no thread set.
+      return true;
+
+    NS_ENSURE_SUCCESS(target_->IsOnCurrentThread(&on), false);
+    NS_ENSURE_TRUE(on, false);
+
+    return true;
+  }
+};
+
+#define LAYER_INFO                 \
+  "Flow[" << flow_id() << "(none)" \
+          << "]; Layer[" << id() << "]: "
+#define TL_SET_STATE(x) SetState((x), __FILE__, __LINE__)
+
+}  // namespace mozilla
+#endif
diff --git a/dom/media/webrtc/transport/transportlayerdtls.cpp b/dom/media/webrtc/transport/transportlayerdtls.cpp
new file mode 100644
index 0000000000..4ab8aaa029
--- /dev/null
+++ b/dom/media/webrtc/transport/transportlayerdtls.cpp
@@ -0,0 +1,1558 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+#include "transportlayerdtls.h"
+
+#include <algorithm>
+#include <queue>
+#include <sstream>
+
+#include "dtlsidentity.h"
+#include "keyhi.h"
+#include "logging.h"
+#include "mozilla/Telemetry.h"
+#include "mozilla/UniquePtr.h"
+#include "mozilla/Unused.h"
+#include "nsCOMPtr.h"
+#include "nsNetCID.h"
+#include "nsServiceManagerUtils.h"
+#include "sslexp.h"
+#include "sslproto.h"
+
+namespace mozilla {
+
+MOZ_MTLOG_MODULE("mtransport")
+
+static PRDescIdentity transport_layer_identity = PR_INVALID_IO_LAYER;
+
+// TODO: Implement a mode for this where
+// the channel is not ready until confirmed externally
+// (e.g., after cert check).
+
+#define UNIMPLEMENTED                                                     \
+  MOZ_MTLOG(ML_ERROR, "Call to unimplemented function " << __FUNCTION__); \
+  MOZ_ASSERT(false);                                                      \
+  PR_SetError(PR_NOT_IMPLEMENTED_ERROR, 0)
+
+#define MAX_ALPN_LENGTH 255
+
+// We need to adapt the NSPR/libssl model to the TransportFlow model.
+// The former wants pull semantics and TransportFlow wants push.
+//
+// - A TransportLayerDtls assumes it is sitting on top of another
+//   TransportLayer, which means that events come in asynchronously.
+// - NSS (libssl) wants to sit on top of a PRFileDesc and poll.
+// - The TransportLayerNSPRAdapter is a PRFileDesc containing a
+//   FIFO.
+// - When TransportLayerDtls.PacketReceived() is called, we insert
+//   the packets in the FIFO and then do a PR_Recv() on the NSS
+//   PRFileDesc, which eventually reads off the FIFO.
+//
+// All of this stuff is assumed to happen solely in a single thread
+// (generally the SocketTransportService thread)
+
+void TransportLayerNSPRAdapter::PacketReceived(MediaPacket& packet) {
+  if (enabled_) {
+    input_.push(new MediaPacket(std::move(packet)));
+  }
+}
+
+int32_t TransportLayerNSPRAdapter::Recv(void* buf, int32_t buflen) {
+  if (input_.empty()) {
+    PR_SetError(PR_WOULD_BLOCK_ERROR, 0);
+    return -1;
+  }
+
+  MediaPacket* front = input_.front();
+  int32_t count = static_cast<int32_t>(front->len());
+
+  if (buflen < count) {
+    MOZ_ASSERT(false, "Not enough buffer space to receive into");
+    PR_SetError(PR_BUFFER_OVERFLOW_ERROR, 0);
+    return -1;
+  }
+
+  memcpy(buf, front->data(), count);
+
+  input_.pop();
+  delete front;
+
+  return count;
+}
+
+int32_t TransportLayerNSPRAdapter::Write(const void* buf, int32_t length) {
+  if (!enabled_) {
+    MOZ_MTLOG(ML_WARNING, "Writing to disabled transport layer");
+    return -1;
+  }
+
+  MediaPacket packet;
+  // Copies. Oh well.
+  packet.Copy(static_cast<const uint8_t*>(buf), static_cast<size_t>(length));
+  packet.SetType(MediaPacket::DTLS);
+
+  TransportResult r = output_->SendPacket(packet);
+  if (r >= 0) {
+    return r;
+  }
+
+  if (r == TE_WOULDBLOCK) {
+    PR_SetError(PR_WOULD_BLOCK_ERROR, 0);
+  } else {
+    PR_SetError(PR_IO_ERROR, 0);
+  }
+
+  return -1;
+}
+
+// Implementation of NSPR methods
+static PRStatus TransportLayerClose(PRFileDesc* f) {
+  f->dtor(f);
+  return PR_SUCCESS;
+}
+
+static int32_t TransportLayerRead(PRFileDesc* f, void* buf, int32_t length) {
+  UNIMPLEMENTED;
+  return -1;
+}
+
+static int32_t TransportLayerWrite(PRFileDesc* f, const void* buf,
+                                   int32_t length) {
+  TransportLayerNSPRAdapter* io =
+      reinterpret_cast<TransportLayerNSPRAdapter*>(f->secret);
+  return io->Write(buf, length);
+}
+
+static int32_t TransportLayerAvailable(PRFileDesc* f) {
+  UNIMPLEMENTED;
+  return -1;
+}
+
+int64_t TransportLayerAvailable64(PRFileDesc* f) {
+  UNIMPLEMENTED;
+  return -1;
+}
+
+static PRStatus TransportLayerSync(PRFileDesc* f) {
+  UNIMPLEMENTED;
+  return PR_FAILURE;
+}
+
+static int32_t TransportLayerSeek(PRFileDesc* f, int32_t offset,
+                                  PRSeekWhence how) {
+  UNIMPLEMENTED;
+  return -1;
+}
+
+static int64_t TransportLayerSeek64(PRFileDesc* f, int64_t offset,
+                                    PRSeekWhence how) {
+  UNIMPLEMENTED;
+  return -1;
+}
+
+static PRStatus TransportLayerFileInfo(PRFileDesc* f, PRFileInfo* info) {
+  UNIMPLEMENTED;
+  return PR_FAILURE;
+}
+
+static PRStatus TransportLayerFileInfo64(PRFileDesc* f, PRFileInfo64* info) {
+  UNIMPLEMENTED;
+  return PR_FAILURE;
+}
+
+static int32_t TransportLayerWritev(PRFileDesc* f, const PRIOVec* iov,
+                                    int32_t iov_size, PRIntervalTime to) {
+  UNIMPLEMENTED;
+  return -1;
+}
+
+static PRStatus TransportLayerConnect(PRFileDesc* f, const PRNetAddr* addr,
+                                      PRIntervalTime to) {
+  UNIMPLEMENTED;
+  return PR_FAILURE;
+}
+
+static PRFileDesc* TransportLayerAccept(PRFileDesc* sd, PRNetAddr* addr,
+                                        PRIntervalTime to) {
+  UNIMPLEMENTED;
+  return nullptr;
+}
+
+static PRStatus TransportLayerBind(PRFileDesc* f, const PRNetAddr* addr) {
+  UNIMPLEMENTED;
+  return PR_FAILURE;
+}
+
+static PRStatus TransportLayerListen(PRFileDesc* f, int32_t depth) {
+  UNIMPLEMENTED;
+  return PR_FAILURE;
+}
+
+static PRStatus TransportLayerShutdown(PRFileDesc* f, int32_t how) {
+  // This is only called from NSS when we are the server and the client refuses
+  // to provide a certificate.  In this case, the handshake is destined for
+  // failure, so we will just let this pass.
+  TransportLayerNSPRAdapter* io =
+      reinterpret_cast<TransportLayerNSPRAdapter*>(f->secret);
+  io->SetEnabled(false);
+  return PR_SUCCESS;
+}
+
+// This function does not support peek, or waiting until `to`
+static int32_t TransportLayerRecv(PRFileDesc* f, void* buf, int32_t buflen,
+                                  int32_t flags, PRIntervalTime to) {
+  MOZ_ASSERT(flags == 0);
+  if (flags != 0) {
+    PR_SetError(PR_NOT_IMPLEMENTED_ERROR, 0);
+    return -1;
+  }
+
+  TransportLayerNSPRAdapter* io =
+      reinterpret_cast<TransportLayerNSPRAdapter*>(f->secret);
+  return io->Recv(buf, buflen);
+}
+
+// Note: this is always nonblocking and assumes a zero timeout.
+static int32_t TransportLayerSend(PRFileDesc* f, const void* buf,
+                                  int32_t amount, int32_t flags,
+                                  PRIntervalTime to) {
+  int32_t written = TransportLayerWrite(f, buf, amount);
+  return written;
+}
+
+static int32_t TransportLayerRecvfrom(PRFileDesc* f, void* buf, int32_t amount,
+                                      int32_t flags, PRNetAddr* addr,
+                                      PRIntervalTime to) {
+  UNIMPLEMENTED;
+  return -1;
+}
+
+static int32_t TransportLayerSendto(PRFileDesc* f, const void* buf,
+                                    int32_t amount, int32_t flags,
+                                    const PRNetAddr* addr, PRIntervalTime to) {
+  UNIMPLEMENTED;
+  return -1;
+}
+
+static int16_t TransportLayerPoll(PRFileDesc* f, int16_t in_flags,
+                                  int16_t* out_flags) {
+  UNIMPLEMENTED;
+  return -1;
+}
+
+static int32_t TransportLayerAcceptRead(PRFileDesc* sd, PRFileDesc** nd,
+                                        PRNetAddr** raddr, void* buf,
+                                        int32_t amount, PRIntervalTime t) {
+  UNIMPLEMENTED;
+  return -1;
+}
+
+static int32_t TransportLayerTransmitFile(PRFileDesc* sd, PRFileDesc* f,
+                                          const void* headers, int32_t hlen,
+                                          PRTransmitFileFlags flags,
+                                          PRIntervalTime t) {
+  UNIMPLEMENTED;
+  return -1;
+}
+
+static PRStatus TransportLayerGetpeername(PRFileDesc* f, PRNetAddr* addr) {
+  // TODO: Modify to return unique names for each channel
+  // somehow, as opposed to always the same static address. The current
+  // implementation messes up the session cache, which is why it's off
+  // elsewhere
+  addr->inet.family = PR_AF_INET;
+  addr->inet.port = 0;
+  addr->inet.ip = 0;
+
+  return PR_SUCCESS;
+}
+
+static PRStatus TransportLayerGetsockname(PRFileDesc* f, PRNetAddr* addr) {
+  UNIMPLEMENTED;
+  return PR_FAILURE;
+}
+
+static PRStatus TransportLayerGetsockoption(PRFileDesc* f,
+                                            PRSocketOptionData* opt) {
+  switch (opt->option) {
+    case PR_SockOpt_Nonblocking:
+      opt->value.non_blocking = PR_TRUE;
+      return PR_SUCCESS;
+    default:
+      UNIMPLEMENTED;
+      break;
+  }
+
+  return PR_FAILURE;
+}
+
+// Imitate setting socket options. These are mostly noops.
+static PRStatus TransportLayerSetsockoption(PRFileDesc* f,
+                                            const PRSocketOptionData* opt) {
+  switch (opt->option) {
+    case PR_SockOpt_Nonblocking:
+      return PR_SUCCESS;
+    case PR_SockOpt_NoDelay:
+      return PR_SUCCESS;
+    default:
+      UNIMPLEMENTED;
+      break;
+  }
+
+  return PR_FAILURE;
+}
+
+static int32_t TransportLayerSendfile(PRFileDesc* out, PRSendFileData* in,
+                                      PRTransmitFileFlags flags,
+                                      PRIntervalTime to) {
+  UNIMPLEMENTED;
+  return -1;
+}
+
+static PRStatus TransportLayerConnectContinue(PRFileDesc* f, int16_t flags) {
+  UNIMPLEMENTED;
+  return PR_FAILURE;
+}
+
+static int32_t TransportLayerReserved(PRFileDesc* f) {
+  UNIMPLEMENTED;
+  return -1;
+}
+
+static const struct PRIOMethods TransportLayerMethods = {
+    PR_DESC_LAYERED,
+    TransportLayerClose,
+    TransportLayerRead,
+    TransportLayerWrite,
+    TransportLayerAvailable,
+    TransportLayerAvailable64,
+    TransportLayerSync,
+    TransportLayerSeek,
+    TransportLayerSeek64,
+    TransportLayerFileInfo,
+    TransportLayerFileInfo64,
+    TransportLayerWritev,
+    TransportLayerConnect,
+    TransportLayerAccept,
+    TransportLayerBind,
+    TransportLayerListen,
+    TransportLayerShutdown,
+    TransportLayerRecv,
+    TransportLayerSend,
+    TransportLayerRecvfrom,
+    TransportLayerSendto,
+    TransportLayerPoll,
+    TransportLayerAcceptRead,
+    TransportLayerTransmitFile,
+    TransportLayerGetsockname,
+    TransportLayerGetpeername,
+    TransportLayerReserved,
+    TransportLayerReserved,
+    TransportLayerGetsockoption,
+    TransportLayerSetsockoption,
+    TransportLayerSendfile,
+    TransportLayerConnectContinue,
+    TransportLayerReserved,
+    TransportLayerReserved,
+    TransportLayerReserved,
+    TransportLayerReserved};
+
+TransportLayerDtls::~TransportLayerDtls() {
+  // Destroy the NSS instance first so it can still send out an alert before
+  // we disable the nspr_io_adapter_.
+  ssl_fd_ = nullptr;
+  nspr_io_adapter_->SetEnabled(false);
+  if (timer_) {
+    timer_->Cancel();
+  }
+}
+
+nsresult TransportLayerDtls::InitInternal() {
+  // Get the transport service as an event target
+  nsresult rv;
+  target_ = do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &rv);
+
+  if (NS_FAILED(rv)) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't get socket transport service");
+    return rv;
+  }
+
+  timer_ = NS_NewTimer();
+  if (!timer_) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't get timer");
+    return rv;
+  }
+
+  return NS_OK;
+}
+
+void TransportLayerDtls::WasInserted() {
+  // Connect to the lower layers
+  if (!Setup()) {
+    TL_SET_STATE(TS_ERROR);
+  }
+}
+
+// Set the permitted and default ALPN identifiers.
+// The default is here to allow for peers that don't want to negotiate ALPN
+// in that case, the default string will be reported from GetNegotiatedAlpn().
+// Setting the default to the empty string causes the transport layer to fail
+// if ALPN is not negotiated.
+// Note: we only support Unicode strings here, which are encoded into UTF-8,
+// even though ALPN ostensibly allows arbitrary octet sequences.
+nsresult TransportLayerDtls::SetAlpn(const std::set<std::string>& alpn_allowed,
+                                     const std::string& alpn_default) {
+  alpn_allowed_ = alpn_allowed;
+  alpn_default_ = alpn_default;
+
+  return NS_OK;
+}
+
+nsresult TransportLayerDtls::SetVerificationAllowAll() {
+  // Defensive programming
+  if (verification_mode_ != VERIFY_UNSET) return NS_ERROR_ALREADY_INITIALIZED;
+
+  verification_mode_ = VERIFY_ALLOW_ALL;
+
+  return NS_OK;
+}
+
+nsresult TransportLayerDtls::SetVerificationDigest(const DtlsDigest& digest) {
+  // Defensive programming
+  if (verification_mode_ != VERIFY_UNSET &&
+      verification_mode_ != VERIFY_DIGEST) {
+    return NS_ERROR_ALREADY_INITIALIZED;
+  }
+
+  digests_.push_back(digest);
+  verification_mode_ = VERIFY_DIGEST;
+  return NS_OK;
+}
+
+void TransportLayerDtls::SetMinMaxVersion(Version min_version,
+                                          Version max_version) {
+  if (min_version < Version::DTLS_1_0 || min_version > Version::DTLS_1_3 ||
+      max_version < Version::DTLS_1_0 || max_version > Version::DTLS_1_3 ||
+      min_version > max_version || max_version < min_version) {
+    return;
+  }
+  minVersion_ = min_version;
+  maxVersion_ = max_version;
+}
+
+// These are the named groups that we will allow.
+static const SSLNamedGroup NamedGroupPreferences[] = {
+    ssl_grp_ec_curve25519, ssl_grp_ec_secp256r1, ssl_grp_ec_secp384r1,
+    ssl_grp_ffdhe_2048, ssl_grp_ffdhe_3072};
+
+// TODO: make sure this is called from STS. Otherwise
+// we have thread safety issues
+bool TransportLayerDtls::Setup() {
+  CheckThread();
+  SECStatus rv;
+
+  if (!downward_) {
+    MOZ_MTLOG(ML_ERROR, "DTLS layer with nothing below. This is useless");
+    return false;
+  }
+  nspr_io_adapter_ = MakeUnique<TransportLayerNSPRAdapter>(downward_);
+
+  if (!identity_) {
+    MOZ_MTLOG(ML_ERROR, "Can't start DTLS without an identity");
+    return false;
+  }
+
+  if (verification_mode_ == VERIFY_UNSET) {
+    MOZ_MTLOG(ML_ERROR,
+              "Can't start DTLS without specifying a verification mode");
+    return false;
+  }
+
+  if (transport_layer_identity == PR_INVALID_IO_LAYER) {
+    transport_layer_identity = PR_GetUniqueIdentity("nssstreamadapter");
+  }
+
+  UniquePRFileDesc pr_fd(
+      PR_CreateIOLayerStub(transport_layer_identity, &TransportLayerMethods));
+  MOZ_ASSERT(pr_fd != nullptr);
+  if (!pr_fd) return false;
+  pr_fd->secret = reinterpret_cast<PRFilePrivate*>(nspr_io_adapter_.get());
+
+  UniquePRFileDesc ssl_fd(DTLS_ImportFD(nullptr, pr_fd.get()));
+  MOZ_ASSERT(ssl_fd != nullptr);  // This should never happen
+  if (!ssl_fd) {
+    return false;
+  }
+
+  Unused << pr_fd.release();  // ownership transfered to ssl_fd;
+
+  if (role_ == CLIENT) {
+    MOZ_MTLOG(ML_INFO, "Setting up DTLS as client");
+    rv = SSL_GetClientAuthDataHook(ssl_fd.get(), GetClientAuthDataHook, this);
+    if (rv != SECSuccess) {
+      MOZ_MTLOG(ML_ERROR, "Couldn't set identity");
+      return false;
+    }
+
+    if (maxVersion_ >= Version::DTLS_1_3) {
+      MOZ_MTLOG(ML_INFO, "Setting DTLS1.3 supported_versions workaround");
+      rv = SSL_SetDtls13VersionWorkaround(ssl_fd.get(), PR_TRUE);
+      if (rv != SECSuccess) {
+        MOZ_MTLOG(ML_ERROR, "Couldn't set DTLS1.3 workaround");
+        return false;
+      }
+    }
+  } else {
+    MOZ_MTLOG(ML_INFO, "Setting up DTLS as server");
+    // Server side
+    rv = SSL_ConfigSecureServer(ssl_fd.get(), identity_->cert().get(),
+                                identity_->privkey().get(),
+                                identity_->auth_type());
+    if (rv != SECSuccess) {
+      MOZ_MTLOG(ML_ERROR, "Couldn't set identity");
+      return false;
+    }
+
+    UniqueCERTCertList zero_certs(CERT_NewCertList());
+    rv = SSL_SetTrustAnchors(ssl_fd.get(), zero_certs.get());
+    if (rv != SECSuccess) {
+      MOZ_MTLOG(ML_ERROR, "Couldn't set trust anchors");
+      return false;
+    }
+
+    // Insist on a certificate from the client
+    rv = SSL_OptionSet(ssl_fd.get(), SSL_REQUEST_CERTIFICATE, PR_TRUE);
+    if (rv != SECSuccess) {
+      MOZ_MTLOG(ML_ERROR, "Couldn't request certificate");
+      return false;
+    }
+
+    rv = SSL_OptionSet(ssl_fd.get(), SSL_REQUIRE_CERTIFICATE, PR_TRUE);
+    if (rv != SECSuccess) {
+      MOZ_MTLOG(ML_ERROR, "Couldn't require certificate");
+      return false;
+    }
+  }
+
+  SSLVersionRange version_range = {static_cast<PRUint16>(minVersion_),
+                                   static_cast<PRUint16>(maxVersion_)};
+
+  rv = SSL_VersionRangeSet(ssl_fd.get(), &version_range);
+  if (rv != SECSuccess) {
+    MOZ_MTLOG(ML_ERROR, "Can't disable SSLv3");
+    return false;
+  }
+
+  rv = SSL_OptionSet(ssl_fd.get(), SSL_ENABLE_SESSION_TICKETS, PR_FALSE);
+  if (rv != SECSuccess) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't disable session tickets");
+    return false;
+  }
+
+  rv = SSL_OptionSet(ssl_fd.get(), SSL_NO_CACHE, PR_TRUE);
+  if (rv != SECSuccess) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't disable session caching");
+    return false;
+  }
+
+  rv = SSL_OptionSet(ssl_fd.get(), SSL_ENABLE_DEFLATE, PR_FALSE);
+  if (rv != SECSuccess) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't disable deflate");
+    return false;
+  }
+
+  rv = SSL_OptionSet(ssl_fd.get(), SSL_ENABLE_RENEGOTIATION,
+                     SSL_RENEGOTIATE_NEVER);
+  if (rv != SECSuccess) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't disable renegotiation");
+    return false;
+  }
+
+  rv = SSL_OptionSet(ssl_fd.get(), SSL_ENABLE_FALSE_START, PR_FALSE);
+  if (rv != SECSuccess) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't disable false start");
+    return false;
+  }
+
+  rv = SSL_OptionSet(ssl_fd.get(), SSL_NO_LOCKS, PR_TRUE);
+  if (rv != SECSuccess) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't disable locks");
+    return false;
+  }
+
+  rv = SSL_OptionSet(ssl_fd.get(), SSL_REUSE_SERVER_ECDHE_KEY, PR_FALSE);
+  if (rv != SECSuccess) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't disable ECDHE key reuse");
+    return false;
+  }
+
+  if (!SetupCipherSuites(ssl_fd)) {
+    return false;
+  }
+
+  rv = SSL_NamedGroupConfig(ssl_fd.get(), NamedGroupPreferences,
+                            mozilla::ArrayLength(NamedGroupPreferences));
+  if (rv != SECSuccess) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't set named groups");
+    return false;
+  }
+
+  // Certificate validation
+  rv = SSL_AuthCertificateHook(ssl_fd.get(), AuthCertificateHook,
+                               reinterpret_cast<void*>(this));
+  if (rv != SECSuccess) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't set certificate validation hook");
+    return false;
+  }
+
+  if (!SetupAlpn(ssl_fd)) {
+    return false;
+  }
+
+  // Now start the handshake
+  rv = SSL_ResetHandshake(ssl_fd.get(), role_ == SERVER ? PR_TRUE : PR_FALSE);
+  if (rv != SECSuccess) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't reset handshake");
+    return false;
+  }
+  ssl_fd_ = std::move(ssl_fd);
+
+  // Finally, get ready to receive data
+  downward_->SignalStateChange.connect(this, &TransportLayerDtls::StateChange);
+  downward_->SignalPacketReceived.connect(this,
+                                          &TransportLayerDtls::PacketReceived);
+
+  if (downward_->state() == TS_OPEN) {
+    TL_SET_STATE(TS_CONNECTING);
+    Handshake();
+  }
+
+  return true;
+}
+
+bool TransportLayerDtls::SetupAlpn(UniquePRFileDesc& ssl_fd) const {
+  if (alpn_allowed_.empty()) {
+    return true;
+  }
+
+  SECStatus rv = SSL_OptionSet(ssl_fd.get(), SSL_ENABLE_NPN, PR_FALSE);
+  if (rv != SECSuccess) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't disable NPN");
+    return false;
+  }
+
+  rv = SSL_OptionSet(ssl_fd.get(), SSL_ENABLE_ALPN, PR_TRUE);
+  if (rv != SECSuccess) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't enable ALPN");
+    return false;
+  }
+
+  unsigned char buf[MAX_ALPN_LENGTH];
+  size_t offset = 0;
+  for (const auto& tag : alpn_allowed_) {
+    if ((offset + 1 + tag.length()) >= sizeof(buf)) {
+      MOZ_MTLOG(ML_ERROR, "ALPN too long");
+      return false;
+    }
+    buf[offset++] = tag.length();
+    memcpy(buf + offset, tag.c_str(), tag.length());
+    offset += tag.length();
+  }
+  rv = SSL_SetNextProtoNego(ssl_fd.get(), buf, offset);
+  if (rv != SECSuccess) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't set ALPN string");
+    return false;
+  }
+  return true;
+}
+
+// Ciphers we need to enable.  These are on by default in standard firefox
+// builds, but can be disabled with prefs and they aren't on in our unit tests
+// since that uses NSS default configuration.
+//
+// Only override prefs to comply with MUST statements in the security-arch doc.
+// Anything outside this list is governed by the usual combination of policy
+// and user preferences.
+static const uint32_t EnabledCiphers[] = {
+    TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256,
+    TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA};
+
+// Disable all NSS suites modes without PFS or with old and rusty ciphersuites.
+// Anything outside this list is governed by the usual combination of policy
+// and user preferences.
+static const uint32_t DisabledCiphers[] = {
+    // Bug 1310061: disable all SHA384 ciphers until fixed
+    TLS_AES_256_GCM_SHA384,
+    TLS_ECDHE_ECDSA_WITH_AES_256_GCM_SHA384,
+    TLS_ECDHE_RSA_WITH_AES_256_GCM_SHA384,
+    TLS_ECDHE_ECDSA_WITH_AES_256_CBC_SHA384,
+    TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA384,
+    TLS_DHE_RSA_WITH_AES_256_GCM_SHA384,
+    TLS_DHE_DSS_WITH_AES_256_GCM_SHA384,
+
+    TLS_DHE_RSA_WITH_AES_128_CBC_SHA,
+    TLS_DHE_RSA_WITH_AES_256_CBC_SHA,
+
+    TLS_ECDHE_ECDSA_WITH_3DES_EDE_CBC_SHA,
+    TLS_ECDHE_RSA_WITH_3DES_EDE_CBC_SHA,
+    TLS_ECDHE_ECDSA_WITH_RC4_128_SHA,
+    TLS_ECDHE_RSA_WITH_RC4_128_SHA,
+
+    TLS_DHE_RSA_WITH_3DES_EDE_CBC_SHA,
+    TLS_DHE_DSS_WITH_3DES_EDE_CBC_SHA,
+    TLS_DHE_DSS_WITH_RC4_128_SHA,
+
+    TLS_ECDH_ECDSA_WITH_AES_128_CBC_SHA,
+    TLS_ECDH_RSA_WITH_AES_128_CBC_SHA,
+    TLS_ECDH_ECDSA_WITH_AES_256_CBC_SHA,
+    TLS_ECDH_RSA_WITH_AES_256_CBC_SHA,
+    TLS_ECDH_ECDSA_WITH_3DES_EDE_CBC_SHA,
+    TLS_ECDH_RSA_WITH_3DES_EDE_CBC_SHA,
+    TLS_ECDH_ECDSA_WITH_RC4_128_SHA,
+    TLS_ECDH_RSA_WITH_RC4_128_SHA,
+
+    TLS_RSA_WITH_AES_128_GCM_SHA256,
+    TLS_RSA_WITH_AES_256_GCM_SHA384,
+    TLS_RSA_WITH_AES_128_CBC_SHA,
+    TLS_RSA_WITH_AES_128_CBC_SHA256,
+    TLS_RSA_WITH_CAMELLIA_128_CBC_SHA,
+    TLS_RSA_WITH_AES_256_CBC_SHA,
+    TLS_RSA_WITH_AES_256_CBC_SHA256,
+    TLS_RSA_WITH_CAMELLIA_256_CBC_SHA,
+    TLS_RSA_WITH_SEED_CBC_SHA,
+    TLS_RSA_WITH_3DES_EDE_CBC_SHA,
+    TLS_RSA_WITH_RC4_128_SHA,
+    TLS_RSA_WITH_RC4_128_MD5,
+
+    TLS_DHE_RSA_WITH_DES_CBC_SHA,
+    TLS_DHE_DSS_WITH_DES_CBC_SHA,
+    TLS_RSA_WITH_DES_CBC_SHA,
+
+    TLS_ECDHE_ECDSA_WITH_NULL_SHA,
+    TLS_ECDHE_RSA_WITH_NULL_SHA,
+    TLS_ECDH_ECDSA_WITH_NULL_SHA,
+    TLS_ECDH_RSA_WITH_NULL_SHA,
+    TLS_RSA_WITH_NULL_SHA,
+    TLS_RSA_WITH_NULL_SHA256,
+    TLS_RSA_WITH_NULL_MD5,
+};
+
+bool TransportLayerDtls::SetupCipherSuites(UniquePRFileDesc& ssl_fd) {
+  SECStatus rv;
+
+  // Set the SRTP ciphers
+  if (!enabled_srtp_ciphers_.empty()) {
+    rv = SSL_InstallExtensionHooks(ssl_fd.get(), ssl_use_srtp_xtn,
+                                   TransportLayerDtls::WriteSrtpXtn, this,
+                                   TransportLayerDtls::HandleSrtpXtn, this);
+    if (rv != SECSuccess) {
+      MOZ_MTLOG(ML_ERROR, LAYER_INFO << "unable to set SRTP extension handler");
+      return false;
+    }
+  }
+
+  for (const auto& cipher : EnabledCiphers) {
+    MOZ_MTLOG(ML_DEBUG, LAYER_INFO << "Enabling: " << cipher);
+    rv = SSL_CipherPrefSet(ssl_fd.get(), cipher, PR_TRUE);
+    if (rv != SECSuccess) {
+      MOZ_MTLOG(ML_ERROR, LAYER_INFO << "Unable to enable suite: " << cipher);
+      return false;
+    }
+  }
+
+  for (const auto& cipher : DisabledCiphers) {
+    MOZ_MTLOG(ML_DEBUG, LAYER_INFO << "Disabling: " << cipher);
+
+    PRBool enabled = false;
+    rv = SSL_CipherPrefGet(ssl_fd.get(), cipher, &enabled);
+    if (rv != SECSuccess) {
+      MOZ_MTLOG(ML_NOTICE, LAYER_INFO << "Unable to check if suite is enabled: "
+                                      << cipher);
+      return false;
+    }
+    if (enabled) {
+      rv = SSL_CipherPrefSet(ssl_fd.get(), cipher, PR_FALSE);
+      if (rv != SECSuccess) {
+        MOZ_MTLOG(ML_NOTICE,
+                  LAYER_INFO << "Unable to disable suite: " << cipher);
+        return false;
+      }
+    }
+  }
+
+  return true;
+}
+
+nsresult TransportLayerDtls::GetCipherSuite(uint16_t* cipherSuite) const {
+  CheckThread();
+  if (!cipherSuite) {
+    MOZ_MTLOG(ML_ERROR, LAYER_INFO << "GetCipherSuite passed a nullptr");
+    return NS_ERROR_NULL_POINTER;
+  }
+  if (state_ != TS_OPEN) {
+    return NS_ERROR_NOT_AVAILABLE;
+  }
+  SSLChannelInfo info;
+  SECStatus rv = SSL_GetChannelInfo(ssl_fd_.get(), &info, sizeof(info));
+  if (rv != SECSuccess) {
+    MOZ_MTLOG(ML_NOTICE, LAYER_INFO << "GetCipherSuite can't get channel info");
+    return NS_ERROR_FAILURE;
+  }
+  *cipherSuite = info.cipherSuite;
+  return NS_OK;
+}
+
+std::vector<uint16_t> TransportLayerDtls::GetDefaultSrtpCiphers() {
+  std::vector<uint16_t> ciphers;
+
+  ciphers.push_back(kDtlsSrtpAeadAes128Gcm);
+  // Since we don't support DTLS 1.3 or SHA384 ciphers (see bug 1312976)
+  // we don't really enough entropy to prefer this over 128 bit
+  ciphers.push_back(kDtlsSrtpAeadAes256Gcm);
+  ciphers.push_back(kDtlsSrtpAes128CmHmacSha1_80);
+#ifndef NIGHTLY_BUILD
+  // To support bug 1491583 lets try to find out if we get bug reports if we no
+  // longer offer this in Nightly builds.
+  ciphers.push_back(kDtlsSrtpAes128CmHmacSha1_32);
+#endif
+
+  return ciphers;
+}
+
+void TransportLayerDtls::StateChange(TransportLayer* layer, State state) {
+  switch (state) {
+    case TS_NONE:
+      MOZ_ASSERT(false);  // Can't happen
+      break;
+
+    case TS_INIT:
+      MOZ_MTLOG(ML_ERROR,
+                LAYER_INFO << "State change of lower layer to INIT forbidden");
+      TL_SET_STATE(TS_ERROR);
+      break;
+
+    case TS_CONNECTING:
+      MOZ_MTLOG(ML_INFO, LAYER_INFO << "Lower layer is connecting.");
+      break;
+
+    case TS_OPEN:
+      if (timer_) {
+        MOZ_MTLOG(ML_INFO,
+                  LAYER_INFO << "Lower layer is now open; starting TLS");
+        timer_->Cancel();
+        timer_->SetTarget(target_);
+        // Async, since the ICE layer might need to send a STUN response, and we
+        // don't want the handshake to start until that is sent.
+        timer_->InitWithNamedFuncCallback(TimerCallback, this, 0,
+                                          nsITimer::TYPE_ONE_SHOT,
+                                          "TransportLayerDtls::TimerCallback");
+        TL_SET_STATE(TS_CONNECTING);
+      } else {
+        // We have already completed DTLS. Can happen if the ICE layer failed
+        // due to a loss of network, and then recovered.
+        TL_SET_STATE(TS_OPEN);
+      }
+      break;
+
+    case TS_CLOSED:
+      MOZ_MTLOG(ML_INFO, LAYER_INFO << "Lower layer is now closed");
+      TL_SET_STATE(TS_CLOSED);
+      break;
+
+    case TS_ERROR:
+      MOZ_MTLOG(ML_ERROR, LAYER_INFO << "Lower layer experienced an error");
+      TL_SET_STATE(TS_ERROR);
+      break;
+  }
+}
+
+void TransportLayerDtls::Handshake() {
+  if (!timer_) {
+    // We are done with DTLS, regardless of the state changes of lower layers
+    return;
+  }
+
+  // Clear the retransmit timer
+  timer_->Cancel();
+
+  MOZ_ASSERT(state_ == TS_CONNECTING);
+
+  SECStatus rv = SSL_ForceHandshake(ssl_fd_.get());
+
+  if (rv == SECSuccess) {
+    MOZ_MTLOG(ML_NOTICE, LAYER_INFO << "****** SSL handshake completed ******");
+    if (!cert_ok_) {
+      MOZ_MTLOG(ML_ERROR, LAYER_INFO << "Certificate check never occurred");
+      TL_SET_STATE(TS_ERROR);
+      return;
+    }
+    if (!CheckAlpn()) {
+      // Despite connecting, the connection doesn't have a valid ALPN label.
+      // Forcibly close the connection so that the peer isn't left hanging
+      // (assuming the close_notify isn't dropped).
+      ssl_fd_ = nullptr;
+      TL_SET_STATE(TS_ERROR);
+      return;
+    }
+
+    TL_SET_STATE(TS_OPEN);
+
+    RecordTlsTelemetry();
+    timer_ = nullptr;
+  } else {
+    int32_t err = PR_GetError();
+    switch (err) {
+      case SSL_ERROR_RX_MALFORMED_HANDSHAKE:
+        MOZ_MTLOG(ML_ERROR, LAYER_INFO << "Malformed DTLS message; ignoring");
+        // If this were TLS (and not DTLS), this would be fatal, but
+        // here we're required to ignore bad messages, so fall through
+        [[fallthrough]];
+      case PR_WOULD_BLOCK_ERROR:
+        MOZ_MTLOG(ML_NOTICE, LAYER_INFO << "Handshake would have blocked");
+        PRIntervalTime timeout;
+        rv = DTLS_GetHandshakeTimeout(ssl_fd_.get(), &timeout);
+        if (rv == SECSuccess) {
+          uint32_t timeout_ms = PR_IntervalToMilliseconds(timeout);
+
+          MOZ_MTLOG(ML_DEBUG,
+                    LAYER_INFO << "Setting DTLS timeout to " << timeout_ms);
+          timer_->SetTarget(target_);
+          timer_->InitWithNamedFuncCallback(
+              TimerCallback, this, timeout_ms, nsITimer::TYPE_ONE_SHOT,
+              "TransportLayerDtls::TimerCallback");
+        }
+        break;
+      default:
+        const char* err_msg = PR_ErrorToName(err);
+        MOZ_MTLOG(ML_ERROR, LAYER_INFO << "DTLS handshake error " << err << " ("
+                                       << err_msg << ")");
+        TL_SET_STATE(TS_ERROR);
+        break;
+    }
+  }
+}
+
+// Checks if ALPN was negotiated correctly and returns false if it wasn't.
+// After this returns successfully, alpn_ will be set to the negotiated
+// protocol.
+bool TransportLayerDtls::CheckAlpn() {
+  if (alpn_allowed_.empty()) {
+    return true;
+  }
+
+  SSLNextProtoState alpnState;
+  char chosenAlpn[MAX_ALPN_LENGTH];
+  unsigned int chosenAlpnLen;
+  SECStatus rv = SSL_GetNextProto(ssl_fd_.get(), &alpnState,
+                                  reinterpret_cast<unsigned char*>(chosenAlpn),
+                                  &chosenAlpnLen, sizeof(chosenAlpn));
+  if (rv != SECSuccess) {
+    MOZ_MTLOG(ML_ERROR, LAYER_INFO << "ALPN error");
+    return false;
+  }
+  switch (alpnState) {
+    case SSL_NEXT_PROTO_SELECTED:
+    case SSL_NEXT_PROTO_NEGOTIATED:
+      break;  // OK
+
+    case SSL_NEXT_PROTO_NO_SUPPORT:
+      MOZ_MTLOG(ML_NOTICE,
+                LAYER_INFO << "ALPN not negotiated, "
+                           << (alpn_default_.empty() ? "failing"
+                                                     : "selecting default"));
+      alpn_ = alpn_default_;
+      return !alpn_.empty();
+
+    case SSL_NEXT_PROTO_NO_OVERLAP:
+      // This only happens if there is a custom NPN/ALPN callback installed and
+      // that callback doesn't properly handle ALPN.
+      MOZ_MTLOG(ML_ERROR, LAYER_INFO << "error in ALPN selection callback");
+      return false;
+
+    case SSL_NEXT_PROTO_EARLY_VALUE:
+      MOZ_CRASH("Unexpected 0-RTT ALPN value");
+      return false;
+  }
+
+  // Warning: NSS won't null terminate the ALPN string for us.
+  std::string chosen(chosenAlpn, chosenAlpnLen);
+  MOZ_MTLOG(ML_NOTICE, LAYER_INFO << "Selected ALPN string: " << chosen);
+  if (alpn_allowed_.find(chosen) == alpn_allowed_.end()) {
+    // Maybe our peer chose a protocol we didn't offer (when we are client), or
+    // something is seriously wrong.
+    std::ostringstream ss;
+    for (auto i = alpn_allowed_.begin(); i != alpn_allowed_.end(); ++i) {
+      ss << (i == alpn_allowed_.begin() ? " '" : ", '") << *i << "'";
+    }
+    MOZ_MTLOG(ML_ERROR, LAYER_INFO << "Bad ALPN string: '" << chosen
+                                   << "'; permitted:" << ss.str());
+    return false;
+  }
+  alpn_ = chosen;
+  return true;
+}
+
+void TransportLayerDtls::PacketReceived(TransportLayer* layer,
+                                        MediaPacket& packet) {
+  CheckThread();
+  MOZ_MTLOG(ML_DEBUG, LAYER_INFO << "PacketReceived(" << packet.len() << ")");
+
+  if (state_ != TS_CONNECTING && state_ != TS_OPEN) {
+    MOZ_MTLOG(ML_DEBUG,
+              LAYER_INFO << "Discarding packet in inappropriate state");
+    return;
+  }
+
+  if (!packet.data()) {
+    // Something ate this, probably the SRTP layer
+    return;
+  }
+
+  if (packet.type() != MediaPacket::DTLS) {
+    return;
+  }
+
+  nspr_io_adapter_->PacketReceived(packet);
+  GetDecryptedPackets();
+}
+
+void TransportLayerDtls::GetDecryptedPackets() {
+  // If we're still connecting, try to handshake
+  if (state_ == TS_CONNECTING) {
+    Handshake();
+  }
+
+  // Now try a recv if we're open, since there might be data left
+  if (state_ == TS_OPEN) {
+    int32_t rv;
+    // One packet might contain several DTLS packets
+    do {
+      // nICEr uses a 9216 bytes buffer to allow support for jumbo frames
+      // Can we peek to get a better idea of the actual size?
+      static const size_t kBufferSize = 9216;
+      auto buffer = MakeUnique<uint8_t[]>(kBufferSize);
+      rv = PR_Recv(ssl_fd_.get(), buffer.get(), kBufferSize, 0,
+                   PR_INTERVAL_NO_WAIT);
+      if (rv > 0) {
+        // We have data
+        MOZ_MTLOG(ML_DEBUG, LAYER_INFO << "Read " << rv << " bytes from NSS");
+        MediaPacket packet;
+        packet.SetType(MediaPacket::SCTP);
+        packet.Take(std::move(buffer), static_cast<size_t>(rv));
+        SignalPacketReceived(this, packet);
+      } else if (rv == 0) {
+        TL_SET_STATE(TS_CLOSED);
+      } else {
+        int32_t err = PR_GetError();
+
+        if (err == PR_WOULD_BLOCK_ERROR) {
+          // This gets ignored
+          MOZ_MTLOG(ML_DEBUG, LAYER_INFO << "Receive would have blocked");
+        } else {
+          MOZ_MTLOG(ML_NOTICE, LAYER_INFO << "NSS Error " << err);
+          TL_SET_STATE(TS_ERROR);
+        }
+      }
+    } while (rv > 0);
+  }
+}
+
+void TransportLayerDtls::SetState(State state, const char* file,
+                                  unsigned line) {
+  if (timer_) {
+    switch (state) {
+      case TS_NONE:
+      case TS_INIT:
+        MOZ_ASSERT(false);
+        break;
+      case TS_CONNECTING:
+        break;
+      case TS_OPEN:
+      case TS_CLOSED:
+      case TS_ERROR:
+        timer_->Cancel();
+        break;
+    }
+  }
+
+  TransportLayer::SetState(state, file, line);
+}
+
+TransportResult TransportLayerDtls::SendPacket(MediaPacket& packet) {
+  CheckThread();
+  if (state_ != TS_OPEN) {
+    MOZ_MTLOG(ML_ERROR,
+              LAYER_INFO << "Can't call SendPacket() in state " << state_);
+    return TE_ERROR;
+  }
+
+  int32_t rv = PR_Send(ssl_fd_.get(), packet.data(), packet.len(), 0,
+                       PR_INTERVAL_NO_WAIT);
+
+  if (rv > 0) {
+    // We have data
+    MOZ_MTLOG(ML_DEBUG, LAYER_INFO << "Wrote " << rv << " bytes to SSL Layer");
+    return rv;
+  }
+
+  if (rv == 0) {
+    TL_SET_STATE(TS_CLOSED);
+    return 0;
+  }
+
+  int32_t err = PR_GetError();
+
+  if (err == PR_WOULD_BLOCK_ERROR) {
+    // This gets ignored
+    MOZ_MTLOG(ML_DEBUG, LAYER_INFO << "Send would have blocked");
+    return TE_WOULDBLOCK;
+  }
+
+  MOZ_MTLOG(ML_NOTICE, LAYER_INFO << "NSS Error " << err);
+  TL_SET_STATE(TS_ERROR);
+  return TE_ERROR;
+}
+
+SECStatus TransportLayerDtls::GetClientAuthDataHook(
+    void* arg, PRFileDesc* fd, CERTDistNames* caNames,
+    CERTCertificate** pRetCert, SECKEYPrivateKey** pRetKey) {
+  MOZ_MTLOG(ML_DEBUG, "Server requested client auth");
+
+  TransportLayerDtls* stream = reinterpret_cast<TransportLayerDtls*>(arg);
+  stream->CheckThread();
+
+  if (!stream->identity_) {
+    MOZ_MTLOG(ML_ERROR, "No identity available");
+    PR_SetError(SSL_ERROR_NO_CERTIFICATE, 0);
+    return SECFailure;
+  }
+
+  *pRetCert = CERT_DupCertificate(stream->identity_->cert().get());
+  if (!*pRetCert) {
+    PR_SetError(PR_OUT_OF_MEMORY_ERROR, 0);
+    return SECFailure;
+  }
+
+  *pRetKey = SECKEY_CopyPrivateKey(stream->identity_->privkey().get());
+  if (!*pRetKey) {
+    CERT_DestroyCertificate(*pRetCert);
+    *pRetCert = nullptr;
+    PR_SetError(PR_OUT_OF_MEMORY_ERROR, 0);
+    return SECFailure;
+  }
+
+  return SECSuccess;
+}
+
+nsresult TransportLayerDtls::SetSrtpCiphers(
+    const std::vector<uint16_t>& ciphers) {
+  enabled_srtp_ciphers_ = std::move(ciphers);
+  return NS_OK;
+}
+
+nsresult TransportLayerDtls::GetSrtpCipher(uint16_t* cipher) const {
+  CheckThread();
+  if (srtp_cipher_ == 0) {
+    return NS_ERROR_NOT_AVAILABLE;
+  }
+  *cipher = srtp_cipher_;
+  return NS_OK;
+}
+
+static uint8_t* WriteUint16(uint8_t* cursor, uint16_t v) {
+  *cursor++ = v >> 8;
+  *cursor++ = v & 0xff;
+  return cursor;
+}
+
+static SSLHandshakeType SrtpXtnServerMessage(PRFileDesc* fd) {
+  SSLPreliminaryChannelInfo preinfo;
+  SECStatus rv = SSL_GetPreliminaryChannelInfo(fd, &preinfo, sizeof(preinfo));
+  if (rv != SECSuccess) {
+    MOZ_ASSERT(false, "Can't get version info");
+    return ssl_hs_client_hello;
+  }
+  return (preinfo.protocolVersion >= SSL_LIBRARY_VERSION_TLS_1_3)
+             ? ssl_hs_encrypted_extensions
+             : ssl_hs_server_hello;
+}
+
+/* static */
+PRBool TransportLayerDtls::WriteSrtpXtn(PRFileDesc* fd,
+                                        SSLHandshakeType message, uint8_t* data,
+                                        unsigned int* len, unsigned int max_len,
+                                        void* arg) {
+  auto self = reinterpret_cast<TransportLayerDtls*>(arg);
+
+  // ClientHello: send all supported versions.
+  if (message == ssl_hs_client_hello) {
+    MOZ_ASSERT(self->role_ == CLIENT);
+    MOZ_ASSERT(self->enabled_srtp_ciphers_.size(), "Haven't enabled SRTP");
+    // We will take 2 octets for each cipher, plus a 2 octet length and 1 octet
+    // for the length of the empty MKI.
+    if (max_len < self->enabled_srtp_ciphers_.size() * 2 + 3) {
+      MOZ_ASSERT(false, "Not enough space to send SRTP extension");
+      return false;
+    }
+    uint8_t* cursor = WriteUint16(data, self->enabled_srtp_ciphers_.size() * 2);
+    for (auto cs : self->enabled_srtp_ciphers_) {
+      cursor = WriteUint16(cursor, cs);
+    }
+    *cursor++ = 0;  // MKI is empty
+    *len = cursor - data;
+    return true;
+  }
+
+  if (message == SrtpXtnServerMessage(fd)) {
+    MOZ_ASSERT(self->role_ == SERVER);
+    if (!self->srtp_cipher_) {
+      // Not negotiated. Definitely bad, but the connection can fail later.
+      return false;
+    }
+    if (max_len < 5) {
+      MOZ_ASSERT(false, "Not enough space to send SRTP extension");
+      return false;
+    }
+
+    uint8_t* cursor = WriteUint16(data, 2);  // Length = 2.
+    cursor = WriteUint16(cursor, self->srtp_cipher_);
+    *cursor++ = 0;  // No MKI
+    *len = cursor - data;
+    return true;
+  }
+
+  return false;
+}
+
+class TlsParser {
+ public:
+  TlsParser(const uint8_t* data, size_t len) : cursor_(data), remaining_(len) {}
+
+  bool error() const { return error_; }
+  size_t remaining() const { return remaining_; }
+
+  template <typename T,
+            class = typename std::enable_if<std::is_unsigned<T>::value>::type>
+  void Read(T* v, size_t sz = sizeof(T)) {
+    MOZ_ASSERT(sz <= sizeof(T),
+               "Type is too small to hold the value requested");
+    if (remaining_ < sz) {
+      error_ = true;
+      return;
+    }
+
+    T result = 0;
+    for (size_t i = 0; i < sz; ++i) {
+      result = (result << 8) | *cursor_++;
+      remaining_--;
+    }
+    *v = result;
+  }
+
+  template <typename T,
+            class = typename std::enable_if<std::is_unsigned<T>::value>::type>
+  void ReadVector(std::vector<T>* v, size_t w) {
+    MOZ_ASSERT(v->empty(), "vector needs to be empty");
+
+    uint32_t len;
+    Read(&len, w);
+    if (error_ || len % sizeof(T) != 0 || len > remaining_) {
+      error_ = true;
+      return;
+    }
+
+    size_t count = len / sizeof(T);
+    v->reserve(count);
+    for (T i = 0; !error_ && i < count; ++i) {
+      T item;
+      Read(&item);
+      if (!error_) {
+        v->push_back(item);
+      }
+    }
+  }
+
+  void Skip(size_t n) {
+    if (remaining_ < n) {
+      error_ = true;
+    } else {
+      cursor_ += n;
+      remaining_ -= n;
+    }
+  }
+
+  size_t SkipVector(size_t w) {
+    uint32_t len = 0;
+    Read(&len, w);
+    Skip(len);
+    return len;
+  }
+
+ private:
+  const uint8_t* cursor_;
+  size_t remaining_;
+  bool error_ = false;
+};
+
+/* static */
+SECStatus TransportLayerDtls::HandleSrtpXtn(
+    PRFileDesc* fd, SSLHandshakeType message, const uint8_t* data,
+    unsigned int len, SSLAlertDescription* alert, void* arg) {
+  static const uint8_t kTlsAlertHandshakeFailure = 40;
+  static const uint8_t kTlsAlertIllegalParameter = 47;
+  static const uint8_t kTlsAlertDecodeError = 50;
+  static const uint8_t kTlsAlertUnsupportedExtension = 110;
+
+  auto self = reinterpret_cast<TransportLayerDtls*>(arg);
+
+  // Parse the extension.
+  TlsParser parser(data, len);
+  std::vector<uint16_t> advertised;
+  parser.ReadVector(&advertised, 2);
+  size_t mki_len = parser.SkipVector(1);
+  if (parser.error() || parser.remaining() > 0) {
+    *alert = kTlsAlertDecodeError;
+    return SECFailure;
+  }
+
+  if (message == ssl_hs_client_hello) {
+    MOZ_ASSERT(self->role_ == SERVER);
+    if (self->enabled_srtp_ciphers_.empty()) {
+      // We don't have SRTP enabled, which is probably bad, but no sense in
+      // having the handshake fail at this point, let the client decide if this
+      // is a problem.
+      return SECSuccess;
+    }
+
+    for (auto supported : self->enabled_srtp_ciphers_) {
+      auto it = std::find(advertised.begin(), advertised.end(), supported);
+      if (it != advertised.end()) {
+        self->srtp_cipher_ = supported;
+        return SECSuccess;
+      }
+    }
+
+    // No common cipher.
+    *alert = kTlsAlertHandshakeFailure;
+    return SECFailure;
+  }
+
+  if (message == SrtpXtnServerMessage(fd)) {
+    MOZ_ASSERT(self->role_ == CLIENT);
+    if (advertised.size() != 1 || mki_len > 0) {
+      *alert = kTlsAlertIllegalParameter;
+      return SECFailure;
+    }
+    self->srtp_cipher_ = advertised[0];
+    return SECSuccess;
+  }
+
+  *alert = kTlsAlertUnsupportedExtension;
+  return SECFailure;
+}
+
+nsresult TransportLayerDtls::ExportKeyingMaterial(const std::string& label,
+                                                  bool use_context,
+                                                  const std::string& context,
+                                                  unsigned char* out,
+                                                  unsigned int outlen) {
+  CheckThread();
+  if (state_ != TS_OPEN) {
+    MOZ_ASSERT(false, "Transport must be open for ExportKeyingMaterial");
+    return NS_ERROR_NOT_AVAILABLE;
+  }
+  SECStatus rv = SSL_ExportKeyingMaterial(
+      ssl_fd_.get(), label.c_str(), label.size(), use_context,
+      reinterpret_cast<const unsigned char*>(context.c_str()), context.size(),
+      out, outlen);
+  if (rv != SECSuccess) {
+    MOZ_MTLOG(ML_ERROR, "Couldn't export SSL keying material");
+    return NS_ERROR_FAILURE;
+  }
+
+  return NS_OK;
+}
+
+SECStatus TransportLayerDtls::AuthCertificateHook(void* arg, PRFileDesc* fd,
+                                                  PRBool checksig,
+                                                  PRBool isServer) {
+  TransportLayerDtls* stream = reinterpret_cast<TransportLayerDtls*>(arg);
+  stream->CheckThread();
+  return stream->AuthCertificateHook(fd, checksig, isServer);
+}
+
+SECStatus TransportLayerDtls::CheckDigest(
+    const DtlsDigest& digest, UniqueCERTCertificate& peer_cert) const {
+  DtlsDigest computed_digest(digest.algorithm_);
+
+  MOZ_MTLOG(ML_DEBUG,
+            LAYER_INFO << "Checking digest, algorithm=" << digest.algorithm_);
+  nsresult res = DtlsIdentity::ComputeFingerprint(peer_cert, &computed_digest);
+  if (NS_FAILED(res)) {
+    MOZ_MTLOG(ML_ERROR, "Could not compute peer fingerprint for digest "
+                            << digest.algorithm_);
+    // Go to end
+    PR_SetError(SSL_ERROR_BAD_CERTIFICATE, 0);
+    return SECFailure;
+  }
+
+  if (computed_digest != digest) {
+    MOZ_MTLOG(ML_ERROR, "Digest does not match");
+    PR_SetError(SSL_ERROR_BAD_CERTIFICATE, 0);
+    return SECFailure;
+  }
+
+  return SECSuccess;
+}
+
+SECStatus TransportLayerDtls::AuthCertificateHook(PRFileDesc* fd,
+                                                  PRBool checksig,
+                                                  PRBool isServer) {
+  CheckThread();
+  UniqueCERTCertificate peer_cert(SSL_PeerCertificate(fd));
+
+  // We are not set up to take this being called multiple
+  // times. Change this if we ever add renegotiation.
+  MOZ_ASSERT(!auth_hook_called_);
+  if (auth_hook_called_) {
+    PR_SetError(PR_UNKNOWN_ERROR, 0);
+    return SECFailure;
+  }
+  auth_hook_called_ = true;
+
+  MOZ_ASSERT(verification_mode_ != VERIFY_UNSET);
+
+  switch (verification_mode_) {
+    case VERIFY_UNSET:
+      // Break out to error exit
+      PR_SetError(PR_UNKNOWN_ERROR, 0);
+      break;
+
+    case VERIFY_ALLOW_ALL:
+      cert_ok_ = true;
+      return SECSuccess;
+
+    case VERIFY_DIGEST: {
+      MOZ_ASSERT(!digests_.empty());
+      // Check all the provided digests
+
+      // Checking functions call PR_SetError()
+      SECStatus rv = SECFailure;
+      for (auto digest : digests_) {
+        rv = CheckDigest(digest, peer_cert);
+
+        // Matches a digest, we are good to go
+        if (rv == SECSuccess) {
+          cert_ok_ = true;
+          return SECSuccess;
+        }
+      }
+    } break;
+    default:
+      MOZ_CRASH();  // Can't happen
+  }
+
+  return SECFailure;
+}
+
+void TransportLayerDtls::TimerCallback(nsITimer* timer, void* arg) {
+  TransportLayerDtls* dtls = reinterpret_cast<TransportLayerDtls*>(arg);
+
+  MOZ_MTLOG(ML_DEBUG, "DTLS timer expired");
+
+  dtls->Handshake();
+}
+
+void TransportLayerDtls::RecordTlsTelemetry() {
+  MOZ_ASSERT(state_ == TS_OPEN);
+  SSLChannelInfo info;
+  SECStatus ss = SSL_GetChannelInfo(ssl_fd_.get(), &info, sizeof(info));
+  if (ss != SECSuccess) {
+    MOZ_MTLOG(ML_NOTICE,
+              LAYER_INFO << "RecordTlsTelemetry failed to get channel info");
+    return;
+  }
+
+  uint16_t telemetry_cipher = 0;
+
+  switch (info.cipherSuite) {
+    /* Old DHE ciphers: candidates for removal, see bug 1227519 */
+    case TLS_DHE_RSA_WITH_AES_128_CBC_SHA:
+      telemetry_cipher = 1;
+      break;
+    case TLS_DHE_RSA_WITH_AES_256_CBC_SHA:
+      telemetry_cipher = 2;
+      break;
+    /* Current ciphers */
+    case TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA:
+      telemetry_cipher = 3;
+      break;
+    case TLS_ECDHE_ECDSA_WITH_AES_256_CBC_SHA:
+      telemetry_cipher = 4;
+      break;
+    case TLS_ECDHE_RSA_WITH_AES_128_CBC_SHA:
+      telemetry_cipher = 5;
+      break;
+    case TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA:
+      telemetry_cipher = 6;
+      break;
+    case TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256:
+      telemetry_cipher = 7;
+      break;
+    case TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256:
+      telemetry_cipher = 8;
+      break;
+    case TLS_ECDHE_RSA_WITH_CHACHA20_POLY1305_SHA256:
+      telemetry_cipher = 9;
+      break;
+    case TLS_ECDHE_ECDSA_WITH_CHACHA20_POLY1305_SHA256:
+      telemetry_cipher = 10;
+      break;
+    /* TLS 1.3 ciphers */
+    case TLS_AES_128_GCM_SHA256:
+      telemetry_cipher = 11;
+      break;
+    case TLS_CHACHA20_POLY1305_SHA256:
+      telemetry_cipher = 12;
+      break;
+    case TLS_AES_256_GCM_SHA384:
+      telemetry_cipher = 13;
+      break;
+  }
+
+  Telemetry::Accumulate(Telemetry::WEBRTC_DTLS_CIPHER, telemetry_cipher);
+
+  uint16_t cipher;
+  nsresult rv = GetSrtpCipher(&cipher);
+
+  if (NS_FAILED(rv)) {
+    MOZ_MTLOG(ML_DEBUG, "No SRTP cipher suite");
+    return;
+  }
+
+  auto cipher_label = mozilla::Telemetry::LABELS_WEBRTC_SRTP_CIPHER::Unknown;
+
+  switch (cipher) {
+    case kDtlsSrtpAes128CmHmacSha1_80:
+      cipher_label = Telemetry::LABELS_WEBRTC_SRTP_CIPHER::Aes128CmHmacSha1_80;
+      break;
+    case kDtlsSrtpAes128CmHmacSha1_32:
+      cipher_label = Telemetry::LABELS_WEBRTC_SRTP_CIPHER::Aes128CmHmacSha1_32;
+      break;
+    case kDtlsSrtpAeadAes128Gcm:
+      cipher_label = Telemetry::LABELS_WEBRTC_SRTP_CIPHER::AeadAes128Gcm;
+      break;
+    case kDtlsSrtpAeadAes256Gcm:
+      cipher_label = Telemetry::LABELS_WEBRTC_SRTP_CIPHER::AeadAes256Gcm;
+      break;
+  }
+
+  Telemetry::AccumulateCategorical(cipher_label);
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/transport/transportlayerdtls.h b/dom/media/webrtc/transport/transportlayerdtls.h
new file mode 100644
index 0000000000..078b266e8b
--- /dev/null
+++ b/dom/media/webrtc/transport/transportlayerdtls.h
@@ -0,0 +1,187 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+#ifndef transportlayerdtls_h__
+#define transportlayerdtls_h__
+
+#include <queue>
+#include <set>
+
+#ifdef XP_MACOSX
+// ensure that Apple Security kit enum goes before "sslproto.h"
+#  include <CoreFoundation/CFAvailability.h>
+#  include <Security/CipherSuite.h>
+#endif
+
+#include "mozilla/RefPtr.h"
+#include "mozilla/UniquePtr.h"
+#include "nsCOMPtr.h"
+#include "nsITimer.h"
+#include "ScopedNSSTypes.h"
+#include "m_cpp_utils.h"
+#include "dtlsidentity.h"
+#include "transportlayer.h"
+#include "ssl.h"
+#include "sslproto.h"
+
+namespace mozilla {
+
+// RFC 5764 (we don't support the NULL cipher)
+static const uint16_t kDtlsSrtpAes128CmHmacSha1_80 = 0x0001;
+static const uint16_t kDtlsSrtpAes128CmHmacSha1_32 = 0x0002;
+// RFC 7714
+static const uint16_t kDtlsSrtpAeadAes128Gcm = 0x0007;
+static const uint16_t kDtlsSrtpAeadAes256Gcm = 0x0008;
+
+struct Packet;
+
+class TransportLayerNSPRAdapter {
+ public:
+  explicit TransportLayerNSPRAdapter(TransportLayer* output)
+      : output_(output), input_(), enabled_(true) {}
+
+  void PacketReceived(MediaPacket& packet);
+  int32_t Recv(void* buf, int32_t buflen);
+  int32_t Write(const void* buf, int32_t length);
+  void SetEnabled(bool enabled) { enabled_ = enabled; }
+
+ private:
+  DISALLOW_COPY_ASSIGN(TransportLayerNSPRAdapter);
+
+  TransportLayer* output_;
+  std::queue<MediaPacket*> input_;
+  bool enabled_;
+};
+
+class TransportLayerDtls final : public TransportLayer {
+ public:
+  TransportLayerDtls() = default;
+
+  virtual ~TransportLayerDtls();
+
+  enum Role { CLIENT, SERVER };
+  enum Verification { VERIFY_UNSET, VERIFY_ALLOW_ALL, VERIFY_DIGEST };
+
+  // DTLS-specific operations
+  void SetRole(Role role) { role_ = role; }
+  Role role() { return role_; }
+
+  enum class Version : uint16_t {
+    DTLS_1_0 = SSL_LIBRARY_VERSION_DTLS_1_0,
+    DTLS_1_2 = SSL_LIBRARY_VERSION_DTLS_1_2,
+    DTLS_1_3 = SSL_LIBRARY_VERSION_DTLS_1_3
+  };
+  void SetMinMaxVersion(Version min_version, Version max_version);
+
+  void SetIdentity(const RefPtr<DtlsIdentity>& identity) {
+    identity_ = identity;
+  }
+  nsresult SetAlpn(const std::set<std::string>& allowedAlpn,
+                   const std::string& alpnDefault);
+  const std::string& GetNegotiatedAlpn() const { return alpn_; }
+
+  nsresult SetVerificationAllowAll();
+
+  nsresult SetVerificationDigest(const DtlsDigest& digest);
+
+  nsresult GetCipherSuite(uint16_t* cipherSuite) const;
+
+  nsresult SetSrtpCiphers(const std::vector<uint16_t>& ciphers);
+  nsresult GetSrtpCipher(uint16_t* cipher) const;
+  static std::vector<uint16_t> GetDefaultSrtpCiphers();
+
+  nsresult ExportKeyingMaterial(const std::string& label, bool use_context,
+                                const std::string& context, unsigned char* out,
+                                unsigned int outlen);
+
+  // Transport layer overrides.
+  nsresult InitInternal() override;
+  void WasInserted() override;
+  TransportResult SendPacket(MediaPacket& packet) override;
+
+  // Signals
+  void StateChange(TransportLayer* layer, State state);
+  void PacketReceived(TransportLayer* layer, MediaPacket& packet);
+
+  // For testing use only.  Returns the fd.
+  PRFileDesc* internal_fd() {
+    CheckThread();
+    return ssl_fd_.get();
+  }
+
+  TRANSPORT_LAYER_ID("dtls")
+
+ protected:
+  void SetState(State state, const char* file, unsigned line) override;
+
+ private:
+  DISALLOW_COPY_ASSIGN(TransportLayerDtls);
+
+  bool Setup();
+  bool SetupCipherSuites(UniquePRFileDesc& ssl_fd);
+  bool SetupAlpn(UniquePRFileDesc& ssl_fd) const;
+  void GetDecryptedPackets();
+  void Handshake();
+
+  bool CheckAlpn();
+
+  static SECStatus GetClientAuthDataHook(void* arg, PRFileDesc* fd,
+                                         CERTDistNames* caNames,
+                                         CERTCertificate** pRetCert,
+                                         SECKEYPrivateKey** pRetKey);
+  static SECStatus AuthCertificateHook(void* arg, PRFileDesc* fd,
+                                       PRBool checksig, PRBool isServer);
+  SECStatus AuthCertificateHook(PRFileDesc* fd, PRBool checksig,
+                                PRBool isServer);
+
+  static void TimerCallback(nsITimer* timer, void* arg);
+
+  SECStatus CheckDigest(const DtlsDigest& digest,
+                        UniqueCERTCertificate& cert) const;
+
+  void RecordHandshakeCompletionTelemetry(TransportLayer::State endState);
+  void RecordTlsTelemetry();
+
+  static PRBool WriteSrtpXtn(PRFileDesc* fd, SSLHandshakeType message,
+                             uint8_t* data, unsigned int* len,
+                             unsigned int max_len, void* arg);
+
+  static SECStatus HandleSrtpXtn(PRFileDesc* fd, SSLHandshakeType message,
+                                 const uint8_t* data, unsigned int len,
+                                 SSLAlertDescription* alert, void* arg);
+
+  RefPtr<DtlsIdentity> identity_;
+  // What ALPN identifiers are permitted.
+  std::set<std::string> alpn_allowed_;
+  // What ALPN identifier is used if ALPN is not supported.
+  // The empty string indicates that ALPN is required.
+  std::string alpn_default_;
+  // What ALPN string was negotiated.
+  std::string alpn_;
+  std::vector<uint16_t> enabled_srtp_ciphers_;
+  uint16_t srtp_cipher_ = 0;
+
+  Role role_ = CLIENT;
+  Verification verification_mode_ = VERIFY_UNSET;
+  std::vector<DtlsDigest> digests_;
+
+  Version minVersion_ = Version::DTLS_1_0;
+  Version maxVersion_ = Version::DTLS_1_2;
+
+  // Must delete nspr_io_adapter after ssl_fd_ b/c ssl_fd_ causes an alert
+  // (ssl_fd_ contains an un-owning pointer to nspr_io_adapter_)
+  UniquePtr<TransportLayerNSPRAdapter> nspr_io_adapter_ = nullptr;
+  UniquePRFileDesc ssl_fd_ = nullptr;
+
+  nsCOMPtr<nsITimer> timer_ = nullptr;
+  bool auth_hook_called_ = false;
+  bool cert_ok_ = false;
+};
+
+}  // namespace mozilla
+#endif
diff --git a/dom/media/webrtc/transport/transportlayerice.cpp b/dom/media/webrtc/transport/transportlayerice.cpp
new file mode 100644
index 0000000000..03d8131ead
--- /dev/null
+++ b/dom/media/webrtc/transport/transportlayerice.cpp
@@ -0,0 +1,168 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+// Some of this code is cut-and-pasted from nICEr. Copyright is:
+
+/*
+Copyright (c) 2007, Adobe Systems, Incorporated
+All rights reserved.
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions are
+met:
+
+* Redistributions of source code must retain the above copyright
+  notice, this list of conditions and the following disclaimer.
+
+* Redistributions in binary form must reproduce the above copyright
+  notice, this list of conditions and the following disclaimer in the
+  documentation and/or other materials provided with the distribution.
+
+* Neither the name of Adobe Systems, Network Resonance nor the names of its
+  contributors may be used to endorse or promote products derived from
+  this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
+OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include <string>
+
+#include "nsComponentManagerUtils.h"
+#include "nsError.h"
+#include "nsNetCID.h"
+
+// nICEr includes
+extern "C" {
+#include "nr_api.h"
+#include "ice_util.h"
+#include "transport_addr.h"
+}
+
+// Local includes
+#include "logging.h"
+#include "nricemediastream.h"
+#include "transportlayerice.h"
+
+namespace mozilla {
+
+#ifdef ERROR
+#  undef ERROR
+#endif
+
+MOZ_MTLOG_MODULE("mtransport")
+
+TransportLayerIce::TransportLayerIce() : stream_(nullptr), component_(0) {
+  // setup happens later
+}
+
+TransportLayerIce::~TransportLayerIce() {
+  // No need to do anything here, since we use smart pointers
+}
+
+void TransportLayerIce::SetParameters(RefPtr<NrIceMediaStream> stream,
+                                      int component) {
+  // Stream could be null in the case of some badly written js that causes
+  // us to be in an ICE restart case, but not have valid streams due to
+  // not calling PeerConnectionImpl::EnsureTransports if
+  // PeerConnectionImpl::SetSignalingState_m thinks the conditions were
+  // not correct.  We also solved a case where an incoming answer was
+  // incorrectly beginning an ICE restart when the offer did not indicate one.
+  if (!stream) {
+    MOZ_ASSERT(false);
+    return;
+  }
+
+  stream_ = stream;
+  component_ = component;
+
+  PostSetup();
+}
+
+void TransportLayerIce::PostSetup() {
+  stream_->SignalReady.connect(this, &TransportLayerIce::IceReady);
+  stream_->SignalFailed.connect(this, &TransportLayerIce::IceFailed);
+  stream_->SignalPacketReceived.connect(this,
+                                        &TransportLayerIce::IcePacketReceived);
+  if (stream_->state() == NrIceMediaStream::ICE_OPEN) {
+    TL_SET_STATE(TS_OPEN);
+  }
+}
+
+TransportResult TransportLayerIce::SendPacket(MediaPacket& packet) {
+  CheckThread();
+  SignalPacketSending(this, packet);
+  nsresult res = stream_->SendPacket(component_, packet.data(), packet.len());
+
+  if (!NS_SUCCEEDED(res)) {
+    return (res == NS_BASE_STREAM_WOULD_BLOCK) ? TE_WOULDBLOCK : TE_ERROR;
+  }
+
+  MOZ_MTLOG(ML_DEBUG,
+            LAYER_INFO << " SendPacket(" << packet.len() << ") succeeded");
+
+  return packet.len();
+}
+
+void TransportLayerIce::IceCandidate(NrIceMediaStream* stream,
+                                     const std::string&) {
+  // NO-OP for now
+}
+
+void TransportLayerIce::IceReady(NrIceMediaStream* stream) {
+  CheckThread();
+  // only handle the current stream (not the old stream during restart)
+  if (stream != stream_) {
+    return;
+  }
+  MOZ_MTLOG(ML_INFO, LAYER_INFO << "ICE Ready(" << stream->name() << ","
+                                << component_ << ")");
+  TL_SET_STATE(TS_OPEN);
+}
+
+void TransportLayerIce::IceFailed(NrIceMediaStream* stream) {
+  CheckThread();
+  // only handle the current stream (not the old stream during restart)
+  if (stream != stream_) {
+    return;
+  }
+  MOZ_MTLOG(ML_INFO, LAYER_INFO << "ICE Failed(" << stream->name() << ","
+                                << component_ << ")");
+  TL_SET_STATE(TS_ERROR);
+}
+
+void TransportLayerIce::IcePacketReceived(NrIceMediaStream* stream,
+                                          int component,
+                                          const unsigned char* data, int len) {
+  CheckThread();
+  // We get packets for both components, so ignore the ones that aren't
+  // for us.
+  if (component_ != component) return;
+
+  MOZ_MTLOG(ML_DEBUG, LAYER_INFO << "PacketReceived(" << stream->name() << ","
+                                 << component << "," << len << ")");
+  // Might be useful to allow MediaPacket to borrow a buffer (ie; not take
+  // ownership, but copy it if the MediaPacket is moved). This could be a
+  // footgun though with MediaPackets that end up on the heap.
+  MediaPacket packet;
+  packet.Copy(data, len);
+  packet.Categorize();
+
+  SignalPacketReceived(this, packet);
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/transport/transportlayerice.h b/dom/media/webrtc/transport/transportlayerice.h
new file mode 100644
index 0000000000..1c55b71ce8
--- /dev/null
+++ b/dom/media/webrtc/transport/transportlayerice.h
@@ -0,0 +1,60 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+// This is a wrapper around the nICEr ICE stack
+#ifndef transportlayerice_h__
+#define transportlayerice_h__
+
+#include "sigslot.h"
+
+#include "mozilla/RefPtr.h"
+
+#include "m_cpp_utils.h"
+
+#include "nricemediastream.h"
+#include "transportlayer.h"
+
+// An ICE transport layer -- corresponds to a single ICE
+namespace mozilla {
+
+class TransportLayerIce : public TransportLayer {
+ public:
+  TransportLayerIce();
+
+  virtual ~TransportLayerIce();
+
+  void SetParameters(RefPtr<NrIceMediaStream> stream, int component);
+
+  void ResetOldStream();    // called after successful ice restart
+  void RestoreOldStream();  // called after unsuccessful ice restart
+
+  // Transport layer overrides.
+  TransportResult SendPacket(MediaPacket& packet) override;
+
+  // Slots for ICE
+  void IceCandidate(NrIceMediaStream* stream, const std::string&);
+  void IceReady(NrIceMediaStream* stream);
+  void IceFailed(NrIceMediaStream* stream);
+  void IcePacketReceived(NrIceMediaStream* stream, int component,
+                         const unsigned char* data, int len);
+
+  // Useful for capturing encrypted packets
+  sigslot::signal2<TransportLayer*, MediaPacket&> SignalPacketSending;
+
+  TRANSPORT_LAYER_ID("ice")
+
+ private:
+  DISALLOW_COPY_ASSIGN(TransportLayerIce);
+  void PostSetup();
+
+  RefPtr<NrIceMediaStream> stream_;
+  int component_;
+};
+
+}  // namespace mozilla
+#endif
diff --git a/dom/media/webrtc/transport/transportlayerlog.cpp b/dom/media/webrtc/transport/transportlayerlog.cpp
new file mode 100644
index 0000000000..d78e722dca
--- /dev/null
+++ b/dom/media/webrtc/transport/transportlayerlog.cpp
@@ -0,0 +1,48 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+#include "logging.h"
+#include "transportlayerlog.h"
+
+namespace mozilla {
+
+MOZ_MTLOG_MODULE("mtransport")
+
+void TransportLayerLogging::WasInserted() {
+  if (downward_) {
+    downward_->SignalStateChange.connect(this,
+                                         &TransportLayerLogging::StateChange);
+    downward_->SignalPacketReceived.connect(
+        this, &TransportLayerLogging::PacketReceived);
+    TL_SET_STATE(downward_->state());
+  }
+}
+
+TransportResult TransportLayerLogging::SendPacket(MediaPacket& packet) {
+  MOZ_MTLOG(ML_DEBUG, LAYER_INFO << "SendPacket(" << packet.len() << ")");
+
+  if (downward_) {
+    return downward_->SendPacket(packet);
+  }
+  return static_cast<TransportResult>(packet.len());
+}
+
+void TransportLayerLogging::StateChange(TransportLayer* layer, State state) {
+  MOZ_MTLOG(ML_DEBUG, LAYER_INFO << "Received StateChange to " << state);
+
+  TL_SET_STATE(state);
+}
+
+void TransportLayerLogging::PacketReceived(TransportLayer* layer,
+                                           MediaPacket& packet) {
+  MOZ_MTLOG(ML_DEBUG, LAYER_INFO << "PacketReceived(" << packet.len() << ")");
+
+  SignalPacketReceived(this, packet);
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/transport/transportlayerlog.h b/dom/media/webrtc/transport/transportlayerlog.h
new file mode 100644
index 0000000000..4d0b3fd930
--- /dev/null
+++ b/dom/media/webrtc/transport/transportlayerlog.h
@@ -0,0 +1,38 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+#ifndef transportlayerlog_h__
+#define transportlayerlog_h__
+
+#include "m_cpp_utils.h"
+#include "transportlayer.h"
+
+namespace mozilla {
+
+class TransportLayerLogging : public TransportLayer {
+ public:
+  TransportLayerLogging() = default;
+
+  // Overrides for TransportLayer
+  TransportResult SendPacket(MediaPacket& packet) override;
+
+  // Signals (forwarded to upper layer)
+  void StateChange(TransportLayer* layer, State state);
+  void PacketReceived(TransportLayer* layer, MediaPacket& packet);
+
+  TRANSPORT_LAYER_ID("log")
+
+ protected:
+  void WasInserted() override;
+
+ private:
+  DISALLOW_COPY_ASSIGN(TransportLayerLogging);
+};
+
+}  // namespace mozilla
+#endif
diff --git a/dom/media/webrtc/transport/transportlayerloopback.cpp b/dom/media/webrtc/transport/transportlayerloopback.cpp
new file mode 100644
index 0000000000..ab2a52c17e
--- /dev/null
+++ b/dom/media/webrtc/transport/transportlayerloopback.cpp
@@ -0,0 +1,119 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+#include "logging.h"
+#include "prlock.h"
+
+#include "nsNetCID.h"
+#include "nsServiceManagerUtils.h"
+#include "nsString.h"
+
+#include "transportlayerloopback.h"
+
+namespace mozilla {
+
+MOZ_MTLOG_MODULE("mtransport")
+
+nsresult TransportLayerLoopback::Init() {
+  nsresult rv;
+  target_ = do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &rv);
+  MOZ_ASSERT(NS_SUCCEEDED(rv));
+  if (!NS_SUCCEEDED(rv)) return rv;
+
+  timer_ = NS_NewTimer(target_);
+  MOZ_ASSERT(timer_);
+  if (!timer_) return NS_ERROR_FAILURE;
+
+  packets_lock_ = PR_NewLock();
+  MOZ_ASSERT(packets_lock_);
+  if (!packets_lock_) return NS_ERROR_FAILURE;
+
+  deliverer_ = new Deliverer(this);
+
+  timer_->InitWithCallback(deliverer_, 100, nsITimer::TYPE_REPEATING_SLACK);
+
+  return NS_OK;
+}
+
+// Connect to the other side
+void TransportLayerLoopback::Connect(TransportLayerLoopback* peer) {
+  peer_ = peer;
+
+  TL_SET_STATE(TS_OPEN);
+}
+
+TransportResult TransportLayerLoopback::SendPacket(MediaPacket& packet) {
+  MOZ_MTLOG(ML_DEBUG, LAYER_INFO << "SendPacket(" << packet.len() << ")");
+
+  if (!peer_) {
+    MOZ_MTLOG(ML_ERROR, "Discarding packet because peer not attached");
+    return TE_ERROR;
+  }
+
+  size_t len = packet.len();
+  nsresult res = peer_->QueuePacket(packet);
+  if (!NS_SUCCEEDED(res)) return TE_ERROR;
+
+  return static_cast<TransportResult>(len);
+}
+
+nsresult TransportLayerLoopback::QueuePacket(MediaPacket& packet) {
+  MOZ_ASSERT(packets_lock_);
+
+  PR_Lock(packets_lock_);
+
+  if (combinePackets_ && !packets_.empty()) {
+    MediaPacket* prevPacket = packets_.front();
+
+    MOZ_MTLOG(ML_DEBUG, LAYER_INFO << " Enqueuing combined packets of length "
+                                   << prevPacket->len() << " and "
+                                   << packet.len());
+    auto combined = MakeUnique<uint8_t[]>(prevPacket->len() + packet.len());
+    memcpy(combined.get(), prevPacket->data(), prevPacket->len());
+    memcpy(combined.get() + prevPacket->len(), packet.data(), packet.len());
+    prevPacket->Take(std::move(combined), prevPacket->len() + packet.len());
+  } else {
+    MOZ_MTLOG(ML_DEBUG,
+              LAYER_INFO << " Enqueuing packet of length " << packet.len());
+    packets_.push(new MediaPacket(std::move(packet)));
+  }
+
+  PRStatus r = PR_Unlock(packets_lock_);
+  MOZ_ASSERT(r == PR_SUCCESS);
+  if (r != PR_SUCCESS) return NS_ERROR_FAILURE;
+
+  return NS_OK;
+}
+
+void TransportLayerLoopback::DeliverPackets() {
+  while (!packets_.empty()) {
+    UniquePtr<MediaPacket> packet(packets_.front());
+    packets_.pop();
+
+    MOZ_MTLOG(ML_DEBUG,
+              LAYER_INFO << " Delivering packet of length " << packet->len());
+    SignalPacketReceived(this, *packet);
+  }
+}
+
+NS_IMPL_ISUPPORTS(TransportLayerLoopback::Deliverer, nsITimerCallback, nsINamed)
+
+NS_IMETHODIMP TransportLayerLoopback::Deliverer::Notify(nsITimer* timer) {
+  if (!layer_) return NS_OK;
+
+  layer_->DeliverPackets();
+
+  return NS_OK;
+}
+
+NS_IMETHODIMP TransportLayerLoopback::Deliverer::GetName(nsACString& aName) {
+  aName.AssignLiteral("TransportLayerLoopback::Deliverer");
+  return NS_OK;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/transport/transportlayerloopback.h b/dom/media/webrtc/transport/transportlayerloopback.h
new file mode 100644
index 0000000000..6801af8189
--- /dev/null
+++ b/dom/media/webrtc/transport/transportlayerloopback.h
@@ -0,0 +1,109 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+#ifndef transportlayerloopback_h__
+#define transportlayerloopback_h__
+
+#include "prlock.h"
+
+#include <queue>
+
+#include "nsCOMPtr.h"
+#include "nsINamed.h"
+#include "nsITimer.h"
+
+#include "m_cpp_utils.h"
+#include "transportlayer.h"
+
+// A simple loopback transport layer that is used for testing.
+namespace mozilla {
+
+class TransportLayerLoopback : public TransportLayer {
+ public:
+  TransportLayerLoopback()
+      : peer_(nullptr),
+        timer_(nullptr),
+        packets_(),
+        packets_lock_(nullptr),
+        deliverer_(nullptr),
+        combinePackets_(false) {}
+
+  ~TransportLayerLoopback() {
+    while (!packets_.empty()) {
+      MediaPacket* packet = packets_.front();
+      packets_.pop();
+      delete packet;
+    }
+    if (packets_lock_) {
+      PR_DestroyLock(packets_lock_);
+    }
+    timer_->Cancel();
+    deliverer_->Detach();
+  }
+
+  // Init
+  nsresult Init();
+
+  // Connect to the other side
+  void Connect(TransportLayerLoopback* peer);
+
+  // Disconnect
+  void Disconnect() {
+    TransportLayerLoopback* peer = peer_;
+
+    peer_ = nullptr;
+    if (peer) {
+      peer->Disconnect();
+    }
+  }
+
+  void CombinePackets(bool combine) { combinePackets_ = combine; }
+
+  // Overrides for TransportLayer
+  TransportResult SendPacket(MediaPacket& packet) override;
+
+  // Deliver queued packets
+  void DeliverPackets();
+
+  TRANSPORT_LAYER_ID("loopback")
+
+ private:
+  DISALLOW_COPY_ASSIGN(TransportLayerLoopback);
+
+  // A timer to deliver packets if some are available
+  // Fires every 100 ms
+  class Deliverer : public nsITimerCallback, public nsINamed {
+   public:
+    explicit Deliverer(TransportLayerLoopback* layer) : layer_(layer) {}
+    void Detach() { layer_ = nullptr; }
+
+    NS_DECL_THREADSAFE_ISUPPORTS
+    NS_DECL_NSITIMERCALLBACK
+    NS_DECL_NSINAMED
+
+   private:
+    virtual ~Deliverer() = default;
+
+    DISALLOW_COPY_ASSIGN(Deliverer);
+
+    TransportLayerLoopback* layer_;
+  };
+
+  // Queue a packet for delivery
+  nsresult QueuePacket(MediaPacket& packet);
+
+  TransportLayerLoopback* peer_;
+  nsCOMPtr<nsITimer> timer_;
+  std::queue<MediaPacket*> packets_;
+  PRLock* packets_lock_;
+  RefPtr<Deliverer> deliverer_;
+  bool combinePackets_;
+};
+
+}  // namespace mozilla
+#endif
diff --git a/dom/media/webrtc/transport/transportlayersrtp.cpp b/dom/media/webrtc/transport/transportlayersrtp.cpp
new file mode 100644
index 0000000000..25830a2dde
--- /dev/null
+++ b/dom/media/webrtc/transport/transportlayersrtp.cpp
@@ -0,0 +1,222 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+#include "transportlayersrtp.h"
+#include "transportlayerdtls.h"
+
+#include "logging.h"
+#include "nsError.h"
+#include "mozilla/Assertions.h"
+
+namespace mozilla {
+
+MOZ_MTLOG_MODULE("mtransport")
+
+static char kDTLSExporterLabel[] = "EXTRACTOR-dtls_srtp";
+
+TransportLayerSrtp::TransportLayerSrtp(TransportLayerDtls& dtls) {
+  // We need to connect to the dtls layer, not the ice layer, because even
+  // though the packets that DTLS decrypts don't flow through us, we do base our
+  // keying information on the keying information established by the DTLS layer.
+  dtls.SignalStateChange.connect(this, &TransportLayerSrtp::StateChange);
+
+  TL_SET_STATE(dtls.state());
+}
+
+void TransportLayerSrtp::WasInserted() {
+  // Connect to the lower layers
+  if (!Setup()) {
+    TL_SET_STATE(TS_ERROR);
+  }
+}
+
+bool TransportLayerSrtp::Setup() {
+  CheckThread();
+  if (!downward_) {
+    MOZ_MTLOG(ML_ERROR, "SRTP layer with nothing below. This is useless");
+    return false;
+  }
+
+  // downward_ is the TransportLayerIce
+  downward_->SignalPacketReceived.connect(this,
+                                          &TransportLayerSrtp::PacketReceived);
+
+  return true;
+}
+
+TransportResult TransportLayerSrtp::SendPacket(MediaPacket& packet) {
+  if (state() != TS_OPEN) {
+    return TE_ERROR;
+  }
+
+  if (packet.len() < 4) {
+    MOZ_ASSERT(false);
+    return TE_ERROR;
+  }
+
+  MOZ_ASSERT(packet.capacity() - packet.len() >= SRTP_MAX_EXPANSION);
+
+  int out_len;
+  nsresult res;
+  switch (packet.type()) {
+    case MediaPacket::RTP:
+      res = mSendSrtp->ProtectRtp(packet.data(), packet.len(),
+                                  packet.capacity(), &out_len);
+      packet.SetType(MediaPacket::SRTP);
+      break;
+    case MediaPacket::RTCP:
+      res = mSendSrtp->ProtectRtcp(packet.data(), packet.len(),
+                                   packet.capacity(), &out_len);
+      packet.SetType(MediaPacket::SRTCP);
+      break;
+    default:
+      MOZ_CRASH("SRTP layer asked to send packet that is neither RTP or RTCP");
+  }
+
+  if (NS_FAILED(res)) {
+    MOZ_MTLOG(ML_ERROR,
+              "Error protecting "
+                  << (packet.type() == MediaPacket::RTP ? "RTP" : "RTCP")
+                  << " len=" << packet.len() << "[" << std::hex
+                  << packet.data()[0] << " " << packet.data()[1] << " "
+                  << packet.data()[2] << " " << packet.data()[3] << "]");
+    return TE_ERROR;
+  }
+
+  size_t unencrypted_len = packet.len();
+  packet.SetLength(out_len);
+
+  TransportResult bytes = downward_->SendPacket(packet);
+  if (bytes == out_len) {
+    // Whole packet was written, but the encrypted length might be different.
+    // Don't confuse the caller.
+    return unencrypted_len;
+  }
+
+  if (bytes == TE_WOULDBLOCK) {
+    return TE_WOULDBLOCK;
+  }
+
+  return TE_ERROR;
+}
+
+void TransportLayerSrtp::StateChange(TransportLayer* layer, State state) {
+  if (state == TS_OPEN && !mSendSrtp) {
+    TransportLayerDtls* dtls = static_cast<TransportLayerDtls*>(layer);
+    MOZ_ASSERT(dtls);  // DTLS is mandatory
+
+    uint16_t cipher_suite;
+    nsresult res = dtls->GetSrtpCipher(&cipher_suite);
+    if (NS_FAILED(res)) {
+      MOZ_MTLOG(ML_DEBUG, "DTLS-SRTP disabled");
+      TL_SET_STATE(TS_ERROR);
+      return;
+    }
+
+    unsigned int key_size = SrtpFlow::KeySize(cipher_suite);
+    unsigned int salt_size = SrtpFlow::SaltSize(cipher_suite);
+    unsigned int master_key_size = key_size + salt_size;
+    MOZ_ASSERT(master_key_size <= SRTP_MAX_KEY_LENGTH);
+
+    // SRTP Key Exporter as per RFC 5764 S 4.2
+    unsigned char srtp_block[SRTP_MAX_KEY_LENGTH * 2];
+    res = dtls->ExportKeyingMaterial(kDTLSExporterLabel, false, "", srtp_block,
+                                     sizeof(srtp_block));
+    if (NS_FAILED(res)) {
+      MOZ_MTLOG(ML_ERROR, "Failed to compute DTLS-SRTP keys. This is an error");
+      TL_SET_STATE(TS_ERROR);
+      return;
+    }
+
+    // Slice and dice as per RFC 5764 S 4.2
+    unsigned char client_write_key[SRTP_MAX_KEY_LENGTH];
+    unsigned char server_write_key[SRTP_MAX_KEY_LENGTH];
+    unsigned int offset = 0;
+    memcpy(client_write_key, srtp_block + offset, key_size);
+    offset += key_size;
+    memcpy(server_write_key, srtp_block + offset, key_size);
+    offset += key_size;
+    memcpy(client_write_key + key_size, srtp_block + offset, salt_size);
+    offset += salt_size;
+    memcpy(server_write_key + key_size, srtp_block + offset, salt_size);
+    MOZ_ASSERT((offset + salt_size) == (2 * master_key_size));
+
+    unsigned char* write_key;
+    unsigned char* read_key;
+
+    if (dtls->role() == TransportLayerDtls::CLIENT) {
+      write_key = client_write_key;
+      read_key = server_write_key;
+    } else {
+      write_key = server_write_key;
+      read_key = client_write_key;
+    }
+
+    MOZ_ASSERT(!mSendSrtp && !mRecvSrtp);
+    mSendSrtp =
+        SrtpFlow::Create(cipher_suite, false, write_key, master_key_size);
+    mRecvSrtp = SrtpFlow::Create(cipher_suite, true, read_key, master_key_size);
+    if (!mSendSrtp || !mRecvSrtp) {
+      MOZ_MTLOG(ML_ERROR, "Couldn't create SRTP flow.");
+      TL_SET_STATE(TS_ERROR);
+      return;
+    }
+
+    MOZ_MTLOG(ML_INFO, "Created SRTP flow!");
+  }
+
+  TL_SET_STATE(state);
+}
+
+void TransportLayerSrtp::PacketReceived(TransportLayer* layer,
+                                        MediaPacket& packet) {
+  if (state() != TS_OPEN) {
+    return;
+  }
+
+  if (!packet.data()) {
+    // Something ate this, probably the DTLS layer
+    return;
+  }
+
+  if (packet.type() != MediaPacket::SRTP &&
+      packet.type() != MediaPacket::SRTCP) {
+    return;
+  }
+
+  // We want to keep the encrypted packet around for packet dumping
+  packet.CopyDataToEncrypted();
+  int outLen;
+  nsresult res;
+
+  if (packet.type() == MediaPacket::SRTP) {
+    packet.SetType(MediaPacket::RTP);
+    res = mRecvSrtp->UnprotectRtp(packet.data(), packet.len(), packet.len(),
+                                  &outLen);
+  } else {
+    packet.SetType(MediaPacket::RTCP);
+    res = mRecvSrtp->UnprotectRtcp(packet.data(), packet.len(), packet.len(),
+                                   &outLen);
+  }
+
+  if (NS_SUCCEEDED(res)) {
+    packet.SetLength(outLen);
+    SignalPacketReceived(this, packet);
+  } else {
+    // TODO: What do we do wrt packet dumping here? Maybe signal an empty
+    // packet? Signal the still-encrypted packet?
+    MOZ_MTLOG(ML_ERROR,
+              "Error unprotecting "
+                  << (packet.type() == MediaPacket::RTP ? "RTP" : "RTCP")
+                  << " len=" << packet.len() << "[" << std::hex
+                  << packet.data()[0] << " " << packet.data()[1] << " "
+                  << packet.data()[2] << " " << packet.data()[3] << "]");
+  }
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/transport/transportlayersrtp.h b/dom/media/webrtc/transport/transportlayersrtp.h
new file mode 100644
index 0000000000..a5ba035930
--- /dev/null
+++ b/dom/media/webrtc/transport/transportlayersrtp.h
@@ -0,0 +1,43 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=2 et sw=2 tw=80: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef transportlayersrtp_h__
+#define transportlayersrtp_h__
+
+#include <string>
+
+#include "transportlayer.h"
+#include "mozilla/RefPtr.h"
+#include "SrtpFlow.h"
+
+namespace mozilla {
+
+class TransportLayerDtls;
+
+class TransportLayerSrtp final : public TransportLayer {
+ public:
+  explicit TransportLayerSrtp(TransportLayerDtls& dtls);
+  virtual ~TransportLayerSrtp() = default;
+
+  // Transport layer overrides.
+  void WasInserted() override;
+  TransportResult SendPacket(MediaPacket& packet) override;
+
+  // Signals
+  void StateChange(TransportLayer* layer, State state);
+  void PacketReceived(TransportLayer* layer, MediaPacket& packet);
+
+  TRANSPORT_LAYER_ID("srtp")
+
+ private:
+  bool Setup();
+  DISALLOW_COPY_ASSIGN(TransportLayerSrtp);
+  RefPtr<SrtpFlow> mSendSrtp;
+  RefPtr<SrtpFlow> mRecvSrtp;
+};
+
+}  // namespace mozilla
+#endif
diff --git a/dom/media/webrtc/transportbridge/MediaPipeline.cpp b/dom/media/webrtc/transportbridge/MediaPipeline.cpp
new file mode 100644
index 0000000000..ec861f69bd
--- /dev/null
+++ b/dom/media/webrtc/transportbridge/MediaPipeline.cpp
@@ -0,0 +1,1655 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+#include "MediaPipeline.h"
+
+#include <inttypes.h>
+#include <math.h>
+#include <sstream>
+
+#include "AudioSegment.h"
+#include "AudioConverter.h"
+#include "DOMMediaStream.h"
+#include "ImageContainer.h"
+#include "ImageTypes.h"
+#include "MediaEngine.h"
+#include "MediaSegment.h"
+#include "MediaTrackGraphImpl.h"
+#include "MediaTrackListener.h"
+#include "MediaStreamTrack.h"
+#include "RtpLogger.h"
+#include "VideoFrameConverter.h"
+#include "VideoSegment.h"
+#include "VideoStreamTrack.h"
+#include "VideoUtils.h"
+#include "mozilla/Logging.h"
+#include "mozilla/NullPrincipal.h"
+#include "mozilla/PeerIdentity.h"
+#include "mozilla/Preferences.h"
+#include "mozilla/SharedThreadPool.h"
+#include "mozilla/Sprintf.h"
+#include "mozilla/StaticPrefs_media.h"
+#include "mozilla/TaskQueue.h"
+#include "mozilla/UniquePtr.h"
+#include "mozilla/UniquePtrExtensions.h"
+#include "mozilla/dom/RTCStatsReportBinding.h"
+#include "mozilla/dom/Document.h"
+#include "mozilla/gfx/Point.h"
+#include "mozilla/gfx/Types.h"
+#include "nsError.h"
+#include "nsThreadUtils.h"
+#include "transport/runnable_utils.h"
+#include "jsapi/MediaTransportHandler.h"
+#include "jsapi/PeerConnectionImpl.h"
+#include "Tracing.h"
+#include "libwebrtcglue/WebrtcImageBuffer.h"
+#include "libwebrtcglue/MediaConduitInterface.h"
+#include "common_video/include/video_frame_buffer.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+
+// Max size given stereo is 480*2*2 = 1920 (10ms of 16-bits stereo audio at
+// 48KHz)
+#define AUDIO_SAMPLE_BUFFER_MAX_BYTES (480 * 2 * 2)
+static_assert((WEBRTC_MAX_SAMPLE_RATE / 100) * sizeof(uint16_t) * 2 <=
+                  AUDIO_SAMPLE_BUFFER_MAX_BYTES,
+              "AUDIO_SAMPLE_BUFFER_MAX_BYTES is not large enough");
+
+using namespace mozilla;
+using namespace mozilla::dom;
+using namespace mozilla::gfx;
+using namespace mozilla::layers;
+
+mozilla::LazyLogModule gMediaPipelineLog("MediaPipeline");
+
+namespace mozilla {
+
+// An async inserter for audio data, to avoid running audio codec encoders
+// on the MTG/input audio thread.  Basically just bounces all the audio
+// data to a single audio processing/input queue.  We could if we wanted to
+// use multiple threads and a TaskQueue.
+class AudioProxyThread {
+ public:
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(AudioProxyThread)
+
+  explicit AudioProxyThread(RefPtr<AudioSessionConduit> aConduit)
+      : mConduit(std::move(aConduit)),
+        mTaskQueue(TaskQueue::Create(
+            GetMediaThreadPool(MediaThreadType::WEBRTC_WORKER), "AudioProxy")),
+        mAudioConverter(nullptr) {
+    MOZ_ASSERT(mConduit);
+    MOZ_COUNT_CTOR(AudioProxyThread);
+  }
+
+  // This function is the identity if aInputRate is supported.
+  // Else, it returns a rate that is supported, that ensure no loss in audio
+  // quality: the sampling rate returned is always greater to the inputed
+  // sampling-rate, if they differ..
+  uint32_t AppropriateSendingRateForInputRate(uint32_t aInputRate) {
+    AudioSessionConduit* conduit =
+        static_cast<AudioSessionConduit*>(mConduit.get());
+    if (conduit->IsSamplingFreqSupported(aInputRate)) {
+      return aInputRate;
+    }
+    if (aInputRate < 16000) {
+      return 16000;
+    }
+    if (aInputRate < 32000) {
+      return 32000;
+    }
+    if (aInputRate < 44100) {
+      return 44100;
+    }
+    return 48000;
+  }
+
+  // From an arbitrary AudioChunk at sampling-rate aRate, process the audio into
+  // something the conduit can work with (or send silence if the track is not
+  // enabled), and send the audio in 10ms chunks to the conduit.
+  void InternalProcessAudioChunk(TrackRate aRate, const AudioChunk& aChunk,
+                                 bool aEnabled) {
+    MOZ_ASSERT(mTaskQueue->IsCurrentThreadIn());
+
+    // Convert to interleaved 16-bits integer audio, with a maximum of two
+    // channels (since the WebRTC.org code below makes the assumption that the
+    // input audio is either mono or stereo), with a sample-rate rate that is
+    // 16, 32, 44.1, or 48kHz.
+    uint32_t outputChannels = aChunk.ChannelCount() == 1 ? 1 : 2;
+    int32_t transmissionRate = AppropriateSendingRateForInputRate(aRate);
+
+    // We take advantage of the fact that the common case (microphone directly
+    // to PeerConnection, that is, a normal call), the samples are already
+    // 16-bits mono, so the representation in interleaved and planar is the
+    // same, and we can just use that.
+    if (aEnabled && outputChannels == 1 &&
+        aChunk.mBufferFormat == AUDIO_FORMAT_S16 && transmissionRate == aRate) {
+      const int16_t* samples = aChunk.ChannelData<int16_t>().Elements()[0];
+      PacketizeAndSend(samples, transmissionRate, outputChannels,
+                       aChunk.mDuration);
+      return;
+    }
+
+    uint32_t sampleCount = aChunk.mDuration * outputChannels;
+    if (mInterleavedAudio.Length() < sampleCount) {
+      mInterleavedAudio.SetLength(sampleCount);
+    }
+
+    if (!aEnabled || aChunk.mBufferFormat == AUDIO_FORMAT_SILENCE) {
+      PodZero(mInterleavedAudio.Elements(), sampleCount);
+    } else if (aChunk.mBufferFormat == AUDIO_FORMAT_FLOAT32) {
+      DownmixAndInterleave(aChunk.ChannelData<float>(), aChunk.mDuration,
+                           aChunk.mVolume, outputChannels,
+                           mInterleavedAudio.Elements());
+    } else if (aChunk.mBufferFormat == AUDIO_FORMAT_S16) {
+      DownmixAndInterleave(aChunk.ChannelData<int16_t>(), aChunk.mDuration,
+                           aChunk.mVolume, outputChannels,
+                           mInterleavedAudio.Elements());
+    }
+    int16_t* inputAudio = mInterleavedAudio.Elements();
+    size_t inputAudioFrameCount = aChunk.mDuration;
+
+    AudioConfig inputConfig(AudioConfig::ChannelLayout(outputChannels), aRate,
+                            AudioConfig::FORMAT_S16);
+    AudioConfig outputConfig(AudioConfig::ChannelLayout(outputChannels),
+                             transmissionRate, AudioConfig::FORMAT_S16);
+    // Resample to an acceptable sample-rate for the sending side
+    if (!mAudioConverter || mAudioConverter->InputConfig() != inputConfig ||
+        mAudioConverter->OutputConfig() != outputConfig) {
+      mAudioConverter = MakeUnique<AudioConverter>(inputConfig, outputConfig);
+    }
+
+    int16_t* processedAudio = nullptr;
+    size_t framesProcessed =
+        mAudioConverter->Process(inputAudio, inputAudioFrameCount);
+
+    if (framesProcessed == 0) {
+      // In place conversion not possible, use a buffer.
+      framesProcessed = mAudioConverter->Process(mOutputAudio, inputAudio,
+                                                 inputAudioFrameCount);
+      processedAudio = mOutputAudio.Data();
+    } else {
+      processedAudio = inputAudio;
+    }
+
+    PacketizeAndSend(processedAudio, transmissionRate, outputChannels,
+                     framesProcessed);
+  }
+
+  // This packetizes aAudioData in 10ms chunks and sends it.
+  // aAudioData is interleaved audio data at a rate and with a channel count
+  // that is appropriate to send with the conduit.
+  void PacketizeAndSend(const int16_t* aAudioData, uint32_t aRate,
+                        uint32_t aChannels, uint32_t aFrameCount) {
+    MOZ_ASSERT(AppropriateSendingRateForInputRate(aRate) == aRate);
+    MOZ_ASSERT(aChannels == 1 || aChannels == 2);
+    MOZ_ASSERT(aAudioData);
+
+    uint32_t audio_10ms = aRate / 100;
+
+    if (!mPacketizer || mPacketizer->mPacketSize != audio_10ms ||
+        mPacketizer->mChannels != aChannels) {
+      // It's the right thing to drop the bit of audio still in the packetizer:
+      // we don't want to send to the conduit audio that has two different
+      // rates while telling it that it has a constante rate.
+      mPacketizer =
+          MakeUnique<AudioPacketizer<int16_t, int16_t>>(audio_10ms, aChannels);
+      mPacket = MakeUnique<int16_t[]>(audio_10ms * aChannels);
+    }
+
+    mPacketizer->Input(aAudioData, aFrameCount);
+
+    while (mPacketizer->PacketsAvailable()) {
+      mPacketizer->Output(mPacket.get());
+      auto frame = std::make_unique<webrtc::AudioFrame>();
+      // UpdateFrame makes a copy of the audio data.
+      frame->UpdateFrame(frame->timestamp_, mPacket.get(),
+                         mPacketizer->mPacketSize, aRate, frame->speech_type_,
+                         frame->vad_activity_, mPacketizer->mChannels);
+      mConduit->SendAudioFrame(std::move(frame));
+    }
+  }
+
+  void QueueAudioChunk(TrackRate aRate, const AudioChunk& aChunk,
+                       bool aEnabled) {
+    RefPtr<AudioProxyThread> self = this;
+    nsresult rv = mTaskQueue->Dispatch(NS_NewRunnableFunction(
+        "AudioProxyThread::QueueAudioChunk", [self, aRate, aChunk, aEnabled]() {
+          self->InternalProcessAudioChunk(aRate, aChunk, aEnabled);
+        }));
+    MOZ_DIAGNOSTIC_ASSERT(NS_SUCCEEDED(rv));
+    Unused << rv;
+  }
+
+ protected:
+  virtual ~AudioProxyThread() { MOZ_COUNT_DTOR(AudioProxyThread); }
+
+  const RefPtr<AudioSessionConduit> mConduit;
+  const RefPtr<TaskQueue> mTaskQueue;
+  // Only accessed on mTaskQueue
+  UniquePtr<AudioPacketizer<int16_t, int16_t>> mPacketizer;
+  // A buffer to hold a single packet of audio.
+  UniquePtr<int16_t[]> mPacket;
+  nsTArray<int16_t> mInterleavedAudio;
+  AlignedShortBuffer mOutputAudio;
+  UniquePtr<AudioConverter> mAudioConverter;
+};
+
+#define INIT_MIRROR(name, val) \
+  name(AbstractThread::MainThread(), val, "MediaPipeline::" #name " (Mirror)")
+
+MediaPipeline::MediaPipeline(const std::string& aPc,
+                             RefPtr<MediaTransportHandler> aTransportHandler,
+                             DirectionType aDirection,
+                             RefPtr<AbstractThread> aCallThread,
+                             RefPtr<nsISerialEventTarget> aStsThread,
+                             RefPtr<MediaSessionConduit> aConduit)
+    : mConduit(std::move(aConduit)),
+      mDirection(aDirection),
+      mCallThread(std::move(aCallThread)),
+      mStsThread(std::move(aStsThread)),
+      INIT_MIRROR(mActive, false),
+      mLevel(0),
+      mTransportHandler(std::move(aTransportHandler)),
+      mRtpPacketsSent(0),
+      mRtcpPacketsSent(0),
+      mRtpPacketsReceived(0),
+      mRtcpPacketsReceived(0),
+      mRtpBytesSent(0),
+      mRtpBytesReceived(0),
+      mPc(aPc),
+      mFilter(),
+      mRtpHeaderExtensionMap(new webrtc::RtpHeaderExtensionMap()),
+      mPacketDumper(PacketDumper::GetPacketDumper(mPc)) {}
+
+#undef INIT_MIRROR
+
+MediaPipeline::~MediaPipeline() {
+  MOZ_LOG(gMediaPipelineLog, LogLevel::Info,
+          ("Destroying MediaPipeline: %s", mDescription.c_str()));
+}
+
+void MediaPipeline::Shutdown() {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  mActive.DisconnectIfConnected();
+  RUN_ON_THREAD(mStsThread,
+                WrapRunnable(RefPtr<MediaPipeline>(this),
+                             &MediaPipeline::DetachTransport_s),
+                NS_DISPATCH_NORMAL);
+}
+
+void MediaPipeline::DetachTransport_s() {
+  ASSERT_ON_THREAD(mStsThread);
+
+  MOZ_LOG(gMediaPipelineLog, LogLevel::Info,
+          ("%s in %s", mDescription.c_str(), __FUNCTION__));
+
+  disconnect_all();
+  mRtpState = TransportLayer::TS_NONE;
+  mRtcpState = TransportLayer::TS_NONE;
+  mTransportId.clear();
+  mConduit->SetTransportActive(false);
+  mRtpSendEventListener.DisconnectIfExists();
+  mSenderRtcpSendEventListener.DisconnectIfExists();
+  mReceiverRtcpSendEventListener.DisconnectIfExists();
+}
+
+void MediaPipeline::UpdateTransport_m(
+    const std::string& aTransportId, UniquePtr<MediaPipelineFilter>&& aFilter) {
+  mStsThread->Dispatch(NS_NewRunnableFunction(
+      __func__, [aTransportId, filter = std::move(aFilter),
+                 self = RefPtr<MediaPipeline>(this)]() mutable {
+        self->UpdateTransport_s(aTransportId, std::move(filter));
+      }));
+}
+
+void MediaPipeline::UpdateTransport_s(
+    const std::string& aTransportId, UniquePtr<MediaPipelineFilter>&& aFilter) {
+  ASSERT_ON_THREAD(mStsThread);
+  if (!mSignalsConnected) {
+    mTransportHandler->SignalStateChange.connect(
+        this, &MediaPipeline::RtpStateChange);
+    mTransportHandler->SignalRtcpStateChange.connect(
+        this, &MediaPipeline::RtcpStateChange);
+    mTransportHandler->SignalEncryptedSending.connect(
+        this, &MediaPipeline::EncryptedPacketSending);
+    mTransportHandler->SignalPacketReceived.connect(
+        this, &MediaPipeline::PacketReceived);
+    mTransportHandler->SignalAlpnNegotiated.connect(
+        this, &MediaPipeline::AlpnNegotiated);
+    mSignalsConnected = true;
+  }
+
+  if (aTransportId != mTransportId) {
+    mTransportId = aTransportId;
+    mRtpState = mTransportHandler->GetState(mTransportId, false);
+    mRtcpState = mTransportHandler->GetState(mTransportId, true);
+    CheckTransportStates();
+  }
+
+  if (mFilter) {
+    for (const auto& extension : mFilter->GetExtmap()) {
+      mRtpHeaderExtensionMap->Deregister(extension.uri);
+    }
+  }
+  if (mFilter && aFilter) {
+    // Use the new filter, but don't forget any remote SSRCs that we've learned
+    // by receiving traffic.
+    mFilter->Update(*aFilter);
+  } else {
+    mFilter = std::move(aFilter);
+  }
+  if (mFilter) {
+    for (const auto& extension : mFilter->GetExtmap()) {
+      mRtpHeaderExtensionMap->RegisterByUri(extension.id, extension.uri);
+    }
+  }
+}
+
+void MediaPipeline::GetContributingSourceStats(
+    const nsString& aInboundRtpStreamId,
+    FallibleTArray<dom::RTCRTPContributingSourceStats>& aArr) const {
+  ASSERT_ON_THREAD(mStsThread);
+  // Get the expiry from now
+  DOMHighResTimeStamp expiry =
+      RtpCSRCStats::GetExpiryFromTime(GetTimestampMaker().GetNow().ToDom());
+  for (auto info : mCsrcStats) {
+    if (!info.second.Expired(expiry)) {
+      RTCRTPContributingSourceStats stats;
+      info.second.GetWebidlInstance(stats, aInboundRtpStreamId);
+      if (!aArr.AppendElement(stats, fallible)) {
+        mozalloc_handle_oom(0);
+      }
+    }
+  }
+}
+
+void MediaPipeline::RtpStateChange(const std::string& aTransportId,
+                                   TransportLayer::State aState) {
+  if (mTransportId != aTransportId) {
+    return;
+  }
+  mRtpState = aState;
+  CheckTransportStates();
+}
+
+void MediaPipeline::RtcpStateChange(const std::string& aTransportId,
+                                    TransportLayer::State aState) {
+  if (mTransportId != aTransportId) {
+    return;
+  }
+  mRtcpState = aState;
+  CheckTransportStates();
+}
+
+void MediaPipeline::CheckTransportStates() {
+  ASSERT_ON_THREAD(mStsThread);
+
+  if (mRtpState == TransportLayer::TS_CLOSED ||
+      mRtpState == TransportLayer::TS_ERROR ||
+      mRtcpState == TransportLayer::TS_CLOSED ||
+      mRtcpState == TransportLayer::TS_ERROR) {
+    MOZ_LOG(gMediaPipelineLog, LogLevel::Warning,
+            ("RTP Transport failed for pipeline %p flow %s", this,
+             mDescription.c_str()));
+
+    NS_WARNING(
+        "MediaPipeline Transport failed. This is not properly cleaned up yet");
+    // TODO(ekr@rtfm.com): SECURITY: Figure out how to clean up if the
+    // connection was good and now it is bad.
+    // TODO(ekr@rtfm.com): Report up so that the PC knows we
+    // have experienced an error.
+    mConduit->SetTransportActive(false);
+    mRtpSendEventListener.DisconnectIfExists();
+    mSenderRtcpSendEventListener.DisconnectIfExists();
+    mReceiverRtcpSendEventListener.DisconnectIfExists();
+    return;
+  }
+
+  if (mRtpState == TransportLayer::TS_OPEN) {
+    MOZ_LOG(gMediaPipelineLog, LogLevel::Info,
+            ("RTP Transport ready for pipeline %p flow %s", this,
+             mDescription.c_str()));
+  }
+
+  if (mRtcpState == TransportLayer::TS_OPEN) {
+    MOZ_LOG(gMediaPipelineLog, LogLevel::Info,
+            ("RTCP Transport ready for pipeline %p flow %s", this,
+             mDescription.c_str()));
+  }
+
+  if (mRtpState == TransportLayer::TS_OPEN && mRtcpState == mRtpState) {
+    if (mDirection == DirectionType::TRANSMIT) {
+      mConduit->ConnectSenderRtcpEvent(mSenderRtcpReceiveEvent);
+      mRtpSendEventListener = mConduit->SenderRtpSendEvent().Connect(
+          mStsThread, this, &MediaPipeline::SendPacket);
+      mSenderRtcpSendEventListener = mConduit->SenderRtcpSendEvent().Connect(
+          mStsThread, this, &MediaPipeline::SendPacket);
+    } else {
+      mConduit->ConnectReceiverRtcpEvent(mReceiverRtcpReceiveEvent);
+      mConduit->ConnectReceiverRtpEvent(mRtpReceiveEvent);
+      mReceiverRtcpSendEventListener =
+          mConduit->ReceiverRtcpSendEvent().Connect(mStsThread, this,
+                                                    &MediaPipeline::SendPacket);
+    }
+    mConduit->SetTransportActive(true);
+    TransportReady_s();
+  }
+}
+
+void MediaPipeline::SendPacket(MediaPacket&& aPacket) {
+  ASSERT_ON_THREAD(mStsThread);
+
+  const bool isRtp = aPacket.type() == MediaPacket::RTP;
+
+  if (isRtp && mRtpState != TransportLayer::TS_OPEN) {
+    return;
+  }
+
+  if (!isRtp && mRtcpState != TransportLayer::TS_OPEN) {
+    return;
+  }
+
+  aPacket.sdp_level() = Some(Level());
+
+  if (RtpLogger::IsPacketLoggingOn()) {
+    RtpLogger::LogPacket(aPacket, false, mDescription);
+  }
+
+  if (isRtp) {
+    mPacketDumper->Dump(Level(), dom::mozPacketDumpType::Rtp, true,
+                        aPacket.data(), aPacket.len());
+    IncrementRtpPacketsSent(aPacket);
+  } else {
+    mPacketDumper->Dump(Level(), dom::mozPacketDumpType::Rtcp, true,
+                        aPacket.data(), aPacket.len());
+    IncrementRtcpPacketsSent();
+  }
+
+  MOZ_LOG(
+      gMediaPipelineLog, LogLevel::Debug,
+      ("%s sending %s packet", mDescription.c_str(), (isRtp ? "RTP" : "RTCP")));
+
+  mTransportHandler->SendPacket(mTransportId, std::move(aPacket));
+}
+
+void MediaPipeline::IncrementRtpPacketsSent(const MediaPacket& aPacket) {
+  ASSERT_ON_THREAD(mStsThread);
+  ++mRtpPacketsSent;
+  mRtpBytesSent += aPacket.len();
+
+  if (!(mRtpPacketsSent % 100)) {
+    MOZ_LOG(gMediaPipelineLog, LogLevel::Info,
+            ("RTP sent packet count for %s Pipeline %p: %u (%" PRId64 " bytes)",
+             mDescription.c_str(), this, mRtpPacketsSent, mRtpBytesSent));
+  }
+}
+
+void MediaPipeline::IncrementRtcpPacketsSent() {
+  ASSERT_ON_THREAD(mStsThread);
+  ++mRtcpPacketsSent;
+  if (!(mRtcpPacketsSent % 100)) {
+    MOZ_LOG(gMediaPipelineLog, LogLevel::Info,
+            ("RTCP sent packet count for %s Pipeline %p: %u",
+             mDescription.c_str(), this, mRtcpPacketsSent));
+  }
+}
+
+void MediaPipeline::IncrementRtpPacketsReceived(int32_t aBytes) {
+  ASSERT_ON_THREAD(mStsThread);
+  ++mRtpPacketsReceived;
+  mRtpBytesReceived += aBytes;
+  if (!(mRtpPacketsReceived % 100)) {
+    MOZ_LOG(
+        gMediaPipelineLog, LogLevel::Info,
+        ("RTP received packet count for %s Pipeline %p: %u (%" PRId64 " bytes)",
+         mDescription.c_str(), this, mRtpPacketsReceived, mRtpBytesReceived));
+  }
+}
+
+void MediaPipeline::IncrementRtcpPacketsReceived() {
+  ASSERT_ON_THREAD(mStsThread);
+  ++mRtcpPacketsReceived;
+  if (!(mRtcpPacketsReceived % 100)) {
+    MOZ_LOG(gMediaPipelineLog, LogLevel::Info,
+            ("RTCP received packet count for %s Pipeline %p: %u",
+             mDescription.c_str(), this, mRtcpPacketsReceived));
+  }
+}
+
+void MediaPipeline::RtpPacketReceived(const MediaPacket& packet) {
+  ASSERT_ON_THREAD(mStsThread);
+
+  if (mDirection == DirectionType::TRANSMIT) {
+    return;
+  }
+
+  if (!packet.len()) {
+    return;
+  }
+
+  webrtc::RTPHeader header;
+  rtc::CopyOnWriteBuffer packet_buffer(packet.data(), packet.len());
+  webrtc::RtpPacketReceived parsedPacket(mRtpHeaderExtensionMap.get());
+  if (!parsedPacket.Parse(packet_buffer)) {
+    return;
+  }
+  parsedPacket.GetHeader(&header);
+
+  if (mFilter && !mFilter->Filter(header)) {
+    return;
+  }
+
+  auto now = GetTimestampMaker().GetNow();
+  parsedPacket.set_arrival_time(now.ToRealtime());
+  if (IsVideo()) {
+    parsedPacket.set_payload_type_frequency(webrtc::kVideoPayloadTypeFrequency);
+  }
+
+  // Remove expired RtpCSRCStats
+  if (!mCsrcStats.empty()) {
+    auto expiry = RtpCSRCStats::GetExpiryFromTime(now.ToDom());
+    for (auto p = mCsrcStats.begin(); p != mCsrcStats.end();) {
+      if (p->second.Expired(expiry)) {
+        p = mCsrcStats.erase(p);
+        continue;
+      }
+      p++;
+    }
+  }
+
+  // Add new RtpCSRCStats
+  if (header.numCSRCs) {
+    for (auto i = 0; i < header.numCSRCs; i++) {
+      auto csrcInfo = mCsrcStats.find(header.arrOfCSRCs[i]);
+      if (csrcInfo == mCsrcStats.end()) {
+        mCsrcStats.insert(
+            std::make_pair(header.arrOfCSRCs[i],
+                           RtpCSRCStats(header.arrOfCSRCs[i], now.ToDom())));
+      } else {
+        csrcInfo->second.SetTimestamp(now.ToDom());
+      }
+    }
+  }
+
+  MOZ_LOG(gMediaPipelineLog, LogLevel::Debug,
+          ("%s received RTP packet.", mDescription.c_str()));
+  IncrementRtpPacketsReceived(packet.len());
+
+  RtpLogger::LogPacket(packet, true, mDescription);
+
+  // Might be nice to pass ownership of the buffer in this case, but it is a
+  // small optimization in a rare case.
+  mPacketDumper->Dump(mLevel, dom::mozPacketDumpType::Srtp, false,
+                      packet.encrypted_data(), packet.encrypted_len());
+
+  mPacketDumper->Dump(mLevel, dom::mozPacketDumpType::Rtp, false, packet.data(),
+                      packet.len());
+
+  mRtpReceiveEvent.Notify(std::move(parsedPacket), header);
+}
+
+void MediaPipeline::RtcpPacketReceived(const MediaPacket& packet) {
+  ASSERT_ON_THREAD(mStsThread);
+
+  if (!packet.len()) {
+    return;
+  }
+
+  // We do not filter RTCP. This is because a compound RTCP packet can contain
+  // any collection of RTCP packets, and webrtc.org already knows how to filter
+  // out what it is interested in, and what it is not. Maybe someday we should
+  // have a TransportLayer that breaks up compound RTCP so we can filter them
+  // individually, but I doubt that will matter much.
+
+  MOZ_LOG(gMediaPipelineLog, LogLevel::Debug,
+          ("%s received RTCP packet.", mDescription.c_str()));
+  IncrementRtcpPacketsReceived();
+
+  RtpLogger::LogPacket(packet, true, mDescription);
+
+  // Might be nice to pass ownership of the buffer in this case, but it is a
+  // small optimization in a rare case.
+  mPacketDumper->Dump(mLevel, dom::mozPacketDumpType::Srtcp, false,
+                      packet.encrypted_data(), packet.encrypted_len());
+
+  mPacketDumper->Dump(mLevel, dom::mozPacketDumpType::Rtcp, false,
+                      packet.data(), packet.len());
+
+  if (StaticPrefs::media_webrtc_net_force_disable_rtcp_reception()) {
+    MOZ_LOG(gMediaPipelineLog, LogLevel::Debug,
+            ("%s RTCP packet forced to be dropped", mDescription.c_str()));
+    return;
+  }
+
+  if (mDirection == DirectionType::TRANSMIT) {
+    mSenderRtcpReceiveEvent.Notify(packet.Clone());
+  } else {
+    mReceiverRtcpReceiveEvent.Notify(packet.Clone());
+  }
+}
+
+void MediaPipeline::PacketReceived(const std::string& aTransportId,
+                                   const MediaPacket& packet) {
+  ASSERT_ON_THREAD(mStsThread);
+
+  if (mTransportId != aTransportId) {
+    return;
+  }
+
+  MOZ_ASSERT(mRtpState == TransportLayer::TS_OPEN);
+  MOZ_ASSERT(mRtcpState == mRtpState);
+
+  switch (packet.type()) {
+    case MediaPacket::RTP:
+      RtpPacketReceived(packet);
+      break;
+    case MediaPacket::RTCP:
+      RtcpPacketReceived(packet);
+      break;
+    default:;
+  }
+}
+
+void MediaPipeline::AlpnNegotiated(const std::string& aAlpn,
+                                   bool aPrivacyRequested) {
+  ASSERT_ON_THREAD(mStsThread);
+
+  if (aPrivacyRequested && Direction() == DirectionType::RECEIVE) {
+    // This will force the receive pipelines to drop data until they have
+    // received a private PrincipalHandle from RTCRtpReceiver (which takes a
+    // detour via main thread).
+    static_cast<MediaPipelineReceive*>(this)->OnPrivacyRequested_s();
+  }
+}
+
+void MediaPipeline::EncryptedPacketSending(const std::string& aTransportId,
+                                           const MediaPacket& aPacket) {
+  ASSERT_ON_THREAD(mStsThread);
+
+  if (mTransportId == aTransportId) {
+    dom::mozPacketDumpType type;
+    if (aPacket.type() == MediaPacket::SRTP) {
+      type = dom::mozPacketDumpType::Srtp;
+    } else if (aPacket.type() == MediaPacket::SRTCP) {
+      type = dom::mozPacketDumpType::Srtcp;
+    } else if (aPacket.type() == MediaPacket::DTLS) {
+      // TODO(bug 1497936): Implement packet dump for DTLS
+      return;
+    } else {
+      MOZ_ASSERT(false);
+      return;
+    }
+    mPacketDumper->Dump(Level(), type, true, aPacket.data(), aPacket.len());
+  }
+}
+
+class MediaPipelineTransmit::PipelineListener
+    : public DirectMediaTrackListener {
+  friend class MediaPipelineTransmit;
+
+ public:
+  explicit PipelineListener(RefPtr<MediaSessionConduit> aConduit)
+      : mConduit(std::move(aConduit)),
+        mActive(false),
+        mEnabled(false),
+        mDirectConnect(false) {}
+
+  ~PipelineListener() {
+    if (mConverter) {
+      mConverter->Shutdown();
+    }
+  }
+
+  void SetActive(bool aActive) {
+    mActive = aActive;
+    if (mConverter) {
+      mConverter->SetActive(aActive);
+    }
+  }
+  void SetEnabled(bool aEnabled) { mEnabled = aEnabled; }
+
+  // These are needed since nested classes don't have access to any particular
+  // instance of the parent
+  void SetAudioProxy(RefPtr<AudioProxyThread> aProxy) {
+    mAudioProcessing = std::move(aProxy);
+  }
+
+  void SetVideoFrameConverter(RefPtr<VideoFrameConverter> aConverter) {
+    mConverter = std::move(aConverter);
+  }
+
+  void OnVideoFrameConverted(webrtc::VideoFrame aVideoFrame) {
+    MOZ_RELEASE_ASSERT(mConduit->type() == MediaSessionConduit::VIDEO);
+    static_cast<VideoSessionConduit*>(mConduit.get())
+        ->SendVideoFrame(std::move(aVideoFrame));
+  }
+
+  // Implement MediaTrackListener
+  void NotifyQueuedChanges(MediaTrackGraph* aGraph, TrackTime aOffset,
+                           const MediaSegment& aQueuedMedia) override;
+  void NotifyEnabledStateChanged(MediaTrackGraph* aGraph,
+                                 bool aEnabled) override;
+
+  // Implement DirectMediaTrackListener
+  void NotifyRealtimeTrackData(MediaTrackGraph* aGraph, TrackTime aOffset,
+                               const MediaSegment& aMedia) override;
+  void NotifyDirectListenerInstalled(InstallationResult aResult) override;
+  void NotifyDirectListenerUninstalled() override;
+
+ private:
+  void NewData(const MediaSegment& aMedia, TrackRate aRate = 0);
+
+  const RefPtr<MediaSessionConduit> mConduit;
+  RefPtr<AudioProxyThread> mAudioProcessing;
+  RefPtr<VideoFrameConverter> mConverter;
+
+  // active is true if there is a transport to send on
+  mozilla::Atomic<bool> mActive;
+  // enabled is true if the media access control permits sending
+  // actual content; when false you get black/silence
+  mozilla::Atomic<bool> mEnabled;
+
+  // Written and read on the MediaTrackGraph thread
+  bool mDirectConnect;
+};
+
+MediaPipelineTransmit::MediaPipelineTransmit(
+    const std::string& aPc, RefPtr<MediaTransportHandler> aTransportHandler,
+    RefPtr<AbstractThread> aCallThread, RefPtr<nsISerialEventTarget> aStsThread,
+    bool aIsVideo, RefPtr<MediaSessionConduit> aConduit)
+    : MediaPipeline(aPc, std::move(aTransportHandler), DirectionType::TRANSMIT,
+                    std::move(aCallThread), std::move(aStsThread),
+                    std::move(aConduit)),
+      mWatchManager(this, AbstractThread::MainThread()),
+      mIsVideo(aIsVideo),
+      mListener(new PipelineListener(mConduit)),
+      mDomTrack(nullptr, "MediaPipelineTransmit::mDomTrack"),
+      mSendTrackOverride(nullptr, "MediaPipelineTransmit::mSendTrackOverride") {
+  if (!IsVideo()) {
+    mAudioProcessing =
+        MakeAndAddRef<AudioProxyThread>(*mConduit->AsAudioSessionConduit());
+    mListener->SetAudioProxy(mAudioProcessing);
+  } else {  // Video
+    mConverter = MakeAndAddRef<VideoFrameConverter>(GetTimestampMaker());
+    mFrameListener = mConverter->VideoFrameConvertedEvent().Connect(
+        mConverter->mTaskQueue,
+        [listener = mListener](webrtc::VideoFrame aFrame) {
+          listener->OnVideoFrameConverted(std::move(aFrame));
+        });
+    mListener->SetVideoFrameConverter(mConverter);
+  }
+
+  mWatchManager.Watch(mActive, &MediaPipelineTransmit::UpdateSendState);
+  mWatchManager.Watch(mDomTrack, &MediaPipelineTransmit::UpdateSendState);
+  mWatchManager.Watch(mSendTrackOverride,
+                      &MediaPipelineTransmit::UpdateSendState);
+
+  mDescription = GenerateDescription();
+}
+
+MediaPipelineTransmit::~MediaPipelineTransmit() {
+  mFrameListener.DisconnectIfExists();
+
+  MOZ_ASSERT(!mTransmitting);
+  MOZ_ASSERT(!mDomTrack.Ref());
+}
+
+void MediaPipelineTransmit::InitControl(
+    MediaPipelineTransmitControlInterface* aControl) {
+  mActive.Connect(aControl->CanonicalTransmitting());
+}
+
+void MediaPipelineTransmit::Shutdown() {
+  MediaPipeline::Shutdown();
+  mWatchManager.Shutdown();
+  if (mDomTrack.Ref()) {
+    mDomTrack.Ref()->RemovePrincipalChangeObserver(this);
+    mDomTrack = nullptr;
+  }
+  mUnsettingSendTrack = false;
+  UpdateSendState();
+  MOZ_ASSERT(!mTransmitting);
+}
+
+void MediaPipeline::SetDescription_s(const std::string& description) {
+  ASSERT_ON_THREAD(mStsThread);
+  mDescription = description;
+}
+
+std::string MediaPipelineTransmit::GenerateDescription() const {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  std::stringstream description;
+  description << mPc << "| ";
+  description << (mIsVideo ? "Transmit video[" : "Transmit audio[");
+
+  if (mDomTrack.Ref()) {
+    nsString nsTrackId;
+    mDomTrack.Ref()->GetId(nsTrackId);
+    description << NS_ConvertUTF16toUTF8(nsTrackId).get();
+  } else if (mSendTrackOverride.Ref()) {
+    description << "override " << mSendTrackOverride.Ref().get();
+  } else {
+    description << "no track";
+  }
+
+  description << "]";
+
+  return description.str();
+}
+
+void MediaPipelineTransmit::UpdateSendState() {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  // This runs because either mActive, mDomTrack or mSendTrackOverride changed,
+  // or because mSendTrack was unset async. Based on these inputs this method
+  // is responsible for hooking up mSendTrack to mListener in order to feed data
+  // to the conduit.
+  //
+  // If we are inactive, or if the send track does not match what we want to
+  // send (mDomTrack or mSendTrackOverride), we must stop feeding data to the
+  // conduit. NB that removing the listener from mSendTrack is async, and we
+  // must wait for it to resolve before adding mListener to another track.
+  // mUnsettingSendTrack gates us until the listener has been removed from
+  // mSendTrack.
+  //
+  // If we are active and the send track does match what we want to send, we
+  // make sure mListener is added to the send track. Either now, or if we're
+  // still waiting for another send track to be removed, during a future call to
+  // this method.
+
+  if (mUnsettingSendTrack) {
+    // We must wait for the send track to be unset before we can set it again,
+    // to avoid races. Once unset this function is triggered again.
+    return;
+  }
+
+  const bool wasTransmitting = mTransmitting;
+
+  const bool haveLiveSendTrack = mSendTrack && !mSendTrack->IsDestroyed();
+  const bool haveLiveDomTrack = mDomTrack.Ref() && !mDomTrack.Ref()->Ended();
+  const bool haveLiveOverrideTrack =
+      mSendTrackOverride.Ref() && !mSendTrackOverride.Ref()->IsDestroyed();
+  const bool mustRemoveSendTrack =
+      haveLiveSendTrack && !mSendTrackOverride.Ref() &&
+      (!haveLiveDomTrack || mDomTrack.Ref()->GetTrack() != mSendPortSource);
+
+  mTransmitting = mActive && (haveLiveDomTrack || haveLiveOverrideTrack) &&
+                  !mustRemoveSendTrack;
+
+  MOZ_LOG(gMediaPipelineLog, LogLevel::Debug,
+          ("MediaPipeline %p UpdateSendState wasTransmitting=%d, active=%d, "
+           "sendTrack=%p (%s), domTrack=%p (%s), "
+           "sendTrackOverride=%p (%s), mustRemove=%d, mTransmitting=%d",
+           this, wasTransmitting, mActive.Ref(), mSendTrack.get(),
+           haveLiveSendTrack ? "live" : "ended", mDomTrack.Ref().get(),
+           haveLiveDomTrack ? "live" : "ended", mSendTrackOverride.Ref().get(),
+           haveLiveOverrideTrack ? "live" : "ended", mustRemoveSendTrack,
+           mTransmitting));
+
+  if (!wasTransmitting && mTransmitting) {
+    MOZ_LOG(gMediaPipelineLog, LogLevel::Debug,
+            ("Attaching pipeline %p to track %p conduit type=%s", this,
+             mDomTrack.Ref().get(), mIsVideo ? "video" : "audio"));
+    if (mDescriptionInvalidated) {
+      // Only update the description when we attach to a track, as detaching is
+      // always a longer async step than updating the description. Updating on
+      // detach would cause the wrong track id to be attributed in logs.
+      RUN_ON_THREAD(mStsThread,
+                    WrapRunnable(RefPtr<MediaPipeline>(this),
+                                 &MediaPipelineTransmit::SetDescription_s,
+                                 GenerateDescription()),
+                    NS_DISPATCH_NORMAL);
+      mDescriptionInvalidated = false;
+    }
+    if (mSendTrackOverride.Ref()) {
+      // Special path that allows unittests to avoid mDomTrack and the graph by
+      // manually calling SetSendTrack.
+      mSendTrack = mSendTrackOverride.Ref();
+    } else {
+      mSendTrack = mDomTrack.Ref()->Graph()->CreateForwardedInputTrack(
+          mDomTrack.Ref()->GetTrack()->mType);
+      mSendPortSource = mDomTrack.Ref()->GetTrack();
+      mSendPort = mSendTrack->AllocateInputPort(mSendPortSource.get());
+    }
+    if (mIsVideo) {
+      mConverter->SetTrackingId(mDomTrack.Ref()->GetSource().mTrackingId);
+    }
+    mSendTrack->QueueSetAutoend(false);
+    if (mIsVideo) {
+      mSendTrack->AddDirectListener(mListener);
+    }
+    mSendTrack->AddListener(mListener);
+  }
+
+  if (wasTransmitting && !mTransmitting) {
+    MOZ_LOG(gMediaPipelineLog, LogLevel::Debug,
+            ("Detaching pipeline %p from track %p conduit type=%s", this,
+             mDomTrack.Ref().get(), mIsVideo ? "video" : "audio"));
+    mUnsettingSendTrack = true;
+    if (mIsVideo) {
+      mSendTrack->RemoveDirectListener(mListener);
+    }
+    mSendTrack->RemoveListener(mListener)->Then(
+        GetMainThreadSerialEventTarget(), __func__,
+        [this, self = RefPtr<MediaPipelineTransmit>(this)] {
+          mUnsettingSendTrack = false;
+          mSendTrack = nullptr;
+          if (!mWatchManager.IsShutdown()) {
+            mWatchManager.ManualNotify(&MediaPipelineTransmit::UpdateSendState);
+          }
+        });
+    if (!mSendTrackOverride.Ref()) {
+      // If an override is set it may be re-used.
+      mSendTrack->Destroy();
+      mSendPort->Destroy();
+      mSendPort = nullptr;
+      mSendPortSource = nullptr;
+    }
+  }
+}
+
+bool MediaPipelineTransmit::Transmitting() const {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  return mActive;
+}
+
+bool MediaPipelineTransmit::IsVideo() const { return mIsVideo; }
+
+void MediaPipelineTransmit::PrincipalChanged(dom::MediaStreamTrack* aTrack) {
+  MOZ_ASSERT(aTrack && aTrack == mDomTrack.Ref());
+
+  PeerConnectionWrapper pcw(mPc);
+  if (pcw.impl()) {
+    Document* doc = pcw.impl()->GetParentObject()->GetExtantDoc();
+    if (doc) {
+      UpdateSinkIdentity(doc->NodePrincipal(), pcw.impl()->GetPeerIdentity());
+    } else {
+      MOZ_LOG(gMediaPipelineLog, LogLevel::Info,
+              ("Can't update sink principal; document gone"));
+    }
+  }
+}
+
+void MediaPipelineTransmit::UpdateSinkIdentity(
+    nsIPrincipal* aPrincipal, const PeerIdentity* aSinkIdentity) {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  if (!mDomTrack.Ref()) {
+    // Nothing to do here
+    return;
+  }
+
+  bool enableTrack = aPrincipal->Subsumes(mDomTrack.Ref()->GetPrincipal());
+  if (!enableTrack) {
+    // first try didn't work, but there's a chance that this is still available
+    // if our track is bound to a peerIdentity, and the peer connection (our
+    // sink) is bound to the same identity, then we can enable the track.
+    const PeerIdentity* trackIdentity = mDomTrack.Ref()->GetPeerIdentity();
+    if (aSinkIdentity && trackIdentity) {
+      enableTrack = (*aSinkIdentity == *trackIdentity);
+    }
+  }
+
+  mListener->SetEnabled(enableTrack);
+}
+
+void MediaPipelineTransmit::TransportReady_s() {
+  ASSERT_ON_THREAD(mStsThread);
+  // Call base ready function.
+  MediaPipeline::TransportReady_s();
+  mListener->SetActive(true);
+}
+
+nsresult MediaPipelineTransmit::SetTrack(
+    const RefPtr<MediaStreamTrack>& aDomTrack) {
+  MOZ_ASSERT(NS_IsMainThread());
+  if (mDomTrack.Ref()) {
+    mDomTrack.Ref()->RemovePrincipalChangeObserver(this);
+  }
+
+  if (aDomTrack) {
+    nsString nsTrackId;
+    aDomTrack->GetId(nsTrackId);
+    MOZ_LOG(gMediaPipelineLog, LogLevel::Debug,
+            ("Reattaching pipeline to track %p track %s conduit type: %s",
+             aDomTrack.get(), NS_ConvertUTF16toUTF8(nsTrackId).get(),
+             mIsVideo ? "video" : "audio"));
+  }
+
+  mDescriptionInvalidated = true;
+  mDomTrack = aDomTrack;
+  if (mDomTrack.Ref()) {
+    mDomTrack.Ref()->AddPrincipalChangeObserver(this);
+    PrincipalChanged(mDomTrack.Ref());
+  }
+
+  return NS_OK;
+}
+
+RefPtr<dom::MediaStreamTrack> MediaPipelineTransmit::GetTrack() const {
+  MOZ_ASSERT(NS_IsMainThread());
+  return mDomTrack;
+}
+
+void MediaPipelineTransmit::SetSendTrackOverride(
+    const RefPtr<ProcessedMediaTrack>& aSendTrack) {
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_RELEASE_ASSERT(!mSendTrack);
+  MOZ_RELEASE_ASSERT(!mSendPort);
+  MOZ_RELEASE_ASSERT(!mSendTrackOverride.Ref());
+  mDescriptionInvalidated = true;
+  mSendTrackOverride = aSendTrack;
+}
+
+// Called if we're attached with AddDirectListener()
+void MediaPipelineTransmit::PipelineListener::NotifyRealtimeTrackData(
+    MediaTrackGraph* aGraph, TrackTime aOffset, const MediaSegment& aMedia) {
+  MOZ_LOG(
+      gMediaPipelineLog, LogLevel::Debug,
+      ("MediaPipeline::NotifyRealtimeTrackData() listener=%p, offset=%" PRId64
+       ", duration=%" PRId64,
+       this, aOffset, aMedia.GetDuration()));
+  TRACE_COMMENT(
+      "MediaPipelineTransmit::PipelineListener::NotifyRealtimeTrackData", "%s",
+      aMedia.GetType() == MediaSegment::VIDEO ? "Video" : "Audio");
+  NewData(aMedia, aGraph->GraphRate());
+}
+
+void MediaPipelineTransmit::PipelineListener::NotifyQueuedChanges(
+    MediaTrackGraph* aGraph, TrackTime aOffset,
+    const MediaSegment& aQueuedMedia) {
+  MOZ_LOG(gMediaPipelineLog, LogLevel::Debug,
+          ("MediaPipeline::NotifyQueuedChanges()"));
+
+  if (aQueuedMedia.GetType() == MediaSegment::VIDEO) {
+    // We always get video from the direct listener.
+    return;
+  }
+
+  TRACE("MediaPipelineTransmit::PipelineListener::NotifyQueuedChanges (Audio)");
+
+  if (mDirectConnect) {
+    // ignore non-direct data if we're also getting direct data
+    return;
+  }
+
+  size_t rate;
+  if (aGraph) {
+    rate = aGraph->GraphRate();
+  } else {
+    // When running tests, graph may be null. In that case use a default.
+    rate = 16000;
+  }
+  NewData(aQueuedMedia, rate);
+}
+
+void MediaPipelineTransmit::PipelineListener::NotifyEnabledStateChanged(
+    MediaTrackGraph* aGraph, bool aEnabled) {
+  if (mConduit->type() != MediaSessionConduit::VIDEO) {
+    return;
+  }
+  MOZ_ASSERT(mConverter);
+  mConverter->SetTrackEnabled(aEnabled);
+}
+
+void MediaPipelineTransmit::PipelineListener::NotifyDirectListenerInstalled(
+    InstallationResult aResult) {
+  MOZ_LOG(gMediaPipelineLog, LogLevel::Info,
+          ("MediaPipeline::NotifyDirectListenerInstalled() listener=%p,"
+           " result=%d",
+           this, static_cast<int32_t>(aResult)));
+
+  mDirectConnect = InstallationResult::SUCCESS == aResult;
+}
+
+void MediaPipelineTransmit::PipelineListener::
+    NotifyDirectListenerUninstalled() {
+  MOZ_LOG(
+      gMediaPipelineLog, LogLevel::Info,
+      ("MediaPipeline::NotifyDirectListenerUninstalled() listener=%p", this));
+
+  if (mConduit->type() == MediaSessionConduit::VIDEO) {
+    // Reset the converter's track-enabled state. If re-added to a new track
+    // later and that track is disabled, we will be signaled explicitly.
+    MOZ_ASSERT(mConverter);
+    mConverter->SetTrackEnabled(true);
+  }
+
+  mDirectConnect = false;
+}
+
+void MediaPipelineTransmit::PipelineListener::NewData(
+    const MediaSegment& aMedia, TrackRate aRate /* = 0 */) {
+  if (mConduit->type() != (aMedia.GetType() == MediaSegment::AUDIO
+                               ? MediaSessionConduit::AUDIO
+                               : MediaSessionConduit::VIDEO)) {
+    MOZ_ASSERT(false,
+               "The media type should always be correct since the "
+               "listener is locked to a specific track");
+    return;
+  }
+
+  // TODO(ekr@rtfm.com): For now assume that we have only one
+  // track type and it's destined for us
+  // See bug 784517
+  if (aMedia.GetType() == MediaSegment::AUDIO) {
+    MOZ_RELEASE_ASSERT(aRate > 0);
+
+    if (!mActive) {
+      MOZ_LOG(gMediaPipelineLog, LogLevel::Debug,
+              ("Discarding audio packets because transport not ready"));
+      return;
+    }
+
+    const AudioSegment* audio = static_cast<const AudioSegment*>(&aMedia);
+    for (AudioSegment::ConstChunkIterator iter(*audio); !iter.IsEnded();
+         iter.Next()) {
+      mAudioProcessing->QueueAudioChunk(aRate, *iter, mEnabled);
+    }
+  } else {
+    const VideoSegment* video = static_cast<const VideoSegment*>(&aMedia);
+
+    for (VideoSegment::ConstChunkIterator iter(*video); !iter.IsEnded();
+         iter.Next()) {
+      mConverter->QueueVideoChunk(*iter, !mEnabled);
+    }
+  }
+}
+
+class GenericReceiveListener : public MediaTrackListener {
+ public:
+  GenericReceiveListener(RefPtr<SourceMediaTrack> aSource,
+                         TrackingId aTrackingId)
+      : mSource(std::move(aSource)),
+        mTrackingId(std::move(aTrackingId)),
+        mIsAudio(mSource->mType == MediaSegment::AUDIO),
+        mEnabled(false) {
+    MOZ_DIAGNOSTIC_ASSERT(NS_IsMainThread());
+    MOZ_DIAGNOSTIC_ASSERT(mSource, "Must be used with a SourceMediaTrack");
+  }
+
+  virtual ~GenericReceiveListener() = default;
+
+  void Init() { mSource->AddListener(this); }
+  void Shutdown() { mSource->RemoveListener(this); }
+
+  void SetEnabled(bool aEnabled) {
+    if (mEnabled == aEnabled) {
+      return;
+    }
+    mEnabled = aEnabled;
+    if (mIsAudio && !mSource->IsDestroyed()) {
+      mSource->SetPullingEnabled(mEnabled);
+    }
+  }
+
+ protected:
+  const RefPtr<SourceMediaTrack> mSource;
+  const TrackingId mTrackingId;
+  const bool mIsAudio;
+  // Main thread only.
+  bool mEnabled;
+};
+
+MediaPipelineReceive::MediaPipelineReceive(
+    const std::string& aPc, RefPtr<MediaTransportHandler> aTransportHandler,
+    RefPtr<AbstractThread> aCallThread, RefPtr<nsISerialEventTarget> aStsThread,
+    RefPtr<MediaSessionConduit> aConduit)
+    : MediaPipeline(aPc, std::move(aTransportHandler), DirectionType::RECEIVE,
+                    std::move(aCallThread), std::move(aStsThread),
+                    std::move(aConduit)),
+      mWatchManager(this, AbstractThread::MainThread()) {
+  mWatchManager.Watch(mActive, &MediaPipelineReceive::UpdateListener);
+}
+
+MediaPipelineReceive::~MediaPipelineReceive() = default;
+
+void MediaPipelineReceive::InitControl(
+    MediaPipelineReceiveControlInterface* aControl) {
+  mActive.Connect(aControl->CanonicalReceiving());
+}
+
+void MediaPipelineReceive::Shutdown() {
+  MOZ_ASSERT(NS_IsMainThread());
+  MediaPipeline::Shutdown();
+  mWatchManager.Shutdown();
+}
+
+class MediaPipelineReceiveAudio::PipelineListener
+    : public GenericReceiveListener {
+ public:
+  PipelineListener(RefPtr<SourceMediaTrack> aSource, TrackingId aTrackingId,
+                   RefPtr<MediaSessionConduit> aConduit,
+                   PrincipalHandle aPrincipalHandle, PrincipalPrivacy aPrivacy)
+      : GenericReceiveListener(std::move(aSource), std::move(aTrackingId)),
+        mConduit(std::move(aConduit)),
+        // AudioSession conduit only supports 16, 32, 44.1 and 48kHz
+        // This is an artificial limitation, it would however require more
+        // changes to support any rates. If the sampling rate is not-supported,
+        // we will use 48kHz instead.
+        mRate(static_cast<AudioSessionConduit*>(mConduit.get())
+                      ->IsSamplingFreqSupported(mSource->Graph()->GraphRate())
+                  ? mSource->Graph()->GraphRate()
+                  : WEBRTC_MAX_SAMPLE_RATE),
+        mTaskQueue(TaskQueue::Create(
+            GetMediaThreadPool(MediaThreadType::WEBRTC_WORKER),
+            "AudioPipelineListener")),
+        mPlayedTicks(0),
+        mAudioFrame(std::make_unique<webrtc::AudioFrame>()),
+        mPrincipalHandle(std::move(aPrincipalHandle)),
+        mPrivacy(aPrivacy),
+        mForceSilence(false) {}
+
+  void Init() {
+    GenericReceiveListener::Init();
+    mSource->SetAppendDataSourceRate(mRate);
+  }
+
+  // Implement MediaTrackListener
+  void NotifyPull(MediaTrackGraph* aGraph, TrackTime aEndOfAppendedData,
+                  TrackTime aDesiredTime) override {
+    NotifyPullImpl(aDesiredTime);
+  }
+
+  void OnPrivacyRequested_s() {
+    if (mPrivacy == PrincipalPrivacy::Private) {
+      return;
+    }
+    mForceSilence = true;
+  }
+
+  void SetPrivatePrincipal(PrincipalHandle aHandle) {
+    MOZ_ASSERT(NS_IsMainThread());
+
+    class Message : public ControlMessage {
+     public:
+      Message(RefPtr<PipelineListener> aListener,
+              PrincipalHandle aPrivatePrincipal)
+          : ControlMessage(nullptr),
+            mListener(std::move(aListener)),
+            mPrivatePrincipal(std::move(aPrivatePrincipal)) {}
+
+      void Run() override {
+        if (mListener->mPrivacy == PrincipalPrivacy::Private) {
+          return;
+        }
+        mListener->mPrincipalHandle = mPrivatePrincipal;
+        mListener->mPrivacy = PrincipalPrivacy::Private;
+        mListener->mForceSilence = false;
+      }
+
+      const RefPtr<PipelineListener> mListener;
+      PrincipalHandle mPrivatePrincipal;
+    };
+
+    if (mSource->IsDestroyed()) {
+      return;
+    }
+
+    mSource->GraphImpl()->AppendMessage(
+        MakeUnique<Message>(this, std::move(aHandle)));
+  }
+
+ private:
+  ~PipelineListener() = default;
+
+  void NotifyPullImpl(TrackTime aDesiredTime) {
+    TRACE_COMMENT("PiplineListener::NotifyPullImpl", "PipelineListener %p",
+                  this);
+    uint32_t samplesPer10ms = mRate / 100;
+
+    // mSource's rate is not necessarily the same as the graph rate, since there
+    // are sample-rate constraints on the inbound audio: only 16, 32, 44.1 and
+    // 48kHz are supported. The audio frames we get here is going to be
+    // resampled when inserted into the graph. aDesiredTime and mPlayedTicks are
+    // in the graph rate.
+
+    while (mPlayedTicks < aDesiredTime) {
+      // This fetches 10ms of data, either mono or stereo
+      MediaConduitErrorCode err =
+          static_cast<AudioSessionConduit*>(mConduit.get())
+              ->GetAudioFrame(mRate, mAudioFrame.get());
+
+      if (err != kMediaConduitNoError) {
+        // Insert silence on conduit/GIPS failure (extremely unlikely)
+        MOZ_LOG(gMediaPipelineLog, LogLevel::Error,
+                ("Audio conduit failed (%d) to return data @ %" PRId64
+                 " (desired %" PRId64 " -> %f)",
+                 err, mPlayedTicks, aDesiredTime,
+                 mSource->TrackTimeToSeconds(aDesiredTime)));
+        constexpr size_t mono = 1;
+        mAudioFrame->UpdateFrame(
+            mAudioFrame->timestamp_, nullptr, samplesPer10ms, mRate,
+            mAudioFrame->speech_type_, mAudioFrame->vad_activity_,
+            std::max(mono, mAudioFrame->num_channels()));
+      }
+
+      MOZ_LOG(
+          gMediaPipelineLog, LogLevel::Debug,
+          ("Audio conduit returned buffer for %zu channels, %zu frames",
+           mAudioFrame->num_channels(), mAudioFrame->samples_per_channel()));
+
+      AudioSegment segment;
+      if (mForceSilence || mAudioFrame->muted()) {
+        segment.AppendNullData(mAudioFrame->samples_per_channel());
+      } else {
+        CheckedInt<size_t> bufferSize(sizeof(uint16_t));
+        bufferSize *= mAudioFrame->samples_per_channel();
+        bufferSize *= mAudioFrame->num_channels();
+        RefPtr<SharedBuffer> samples = SharedBuffer::Create(bufferSize);
+        int16_t* samplesData = static_cast<int16_t*>(samples->Data());
+        AutoTArray<int16_t*, 2> channels;
+        AutoTArray<const int16_t*, 2> outputChannels;
+
+        channels.SetLength(mAudioFrame->num_channels());
+
+        size_t offset = 0;
+        for (size_t i = 0; i < mAudioFrame->num_channels(); i++) {
+          channels[i] = samplesData + offset;
+          offset += mAudioFrame->samples_per_channel();
+        }
+
+        DeinterleaveAndConvertBuffer(
+            mAudioFrame->data(), mAudioFrame->samples_per_channel(),
+            mAudioFrame->num_channels(), channels.Elements());
+
+        outputChannels.AppendElements(channels);
+
+        segment.AppendFrames(samples.forget(), outputChannels,
+                             mAudioFrame->samples_per_channel(),
+                             mPrincipalHandle);
+      }
+
+      // Handle track not actually added yet or removed/finished
+      if (TrackTime appended = mSource->AppendData(&segment)) {
+        mPlayedTicks += appended;
+      } else {
+        MOZ_LOG(gMediaPipelineLog, LogLevel::Error, ("AppendData failed"));
+        // we can't un-read the data, but that's ok since we don't want to
+        // buffer - but don't i-loop!
+        break;
+      }
+    }
+  }
+
+  const RefPtr<MediaSessionConduit> mConduit;
+  // This conduit's sampling rate. This is either 16, 32, 44.1 or 48kHz, and
+  // tries to be the same as the graph rate. If the graph rate is higher than
+  // 48kHz, mRate is capped to 48kHz. If mRate does not match the graph rate,
+  // audio is resampled to the graph rate.
+  const TrackRate mRate;
+  const RefPtr<TaskQueue> mTaskQueue;
+  // Number of frames of data that has been added to the SourceMediaTrack in
+  // the graph's rate. Graph thread only.
+  TrackTicks mPlayedTicks;
+  // Allocation of an audio frame used as a scratch buffer when reading data out
+  // of libwebrtc for forwarding into the graph. Graph thread only.
+  std::unique_ptr<webrtc::AudioFrame> mAudioFrame;
+  // Principal handle used when appending data to the SourceMediaTrack. Graph
+  // thread only.
+  PrincipalHandle mPrincipalHandle;
+  // Privacy of mPrincipalHandle. Graph thread only.
+  PrincipalPrivacy mPrivacy;
+  // Set to true on the sts thread if privacy is requested when ALPN was
+  // negotiated. Set to false again when mPrincipalHandle is private.
+  Atomic<bool> mForceSilence;
+};
+
+MediaPipelineReceiveAudio::MediaPipelineReceiveAudio(
+    const std::string& aPc, RefPtr<MediaTransportHandler> aTransportHandler,
+    RefPtr<AbstractThread> aCallThread, RefPtr<nsISerialEventTarget> aStsThread,
+    RefPtr<AudioSessionConduit> aConduit, RefPtr<SourceMediaTrack> aSource,
+    TrackingId aTrackingId, PrincipalHandle aPrincipalHandle,
+    PrincipalPrivacy aPrivacy)
+    : MediaPipelineReceive(aPc, std::move(aTransportHandler),
+                           std::move(aCallThread), std::move(aStsThread),
+                           std::move(aConduit)),
+      mListener(aSource ? new PipelineListener(
+                              std::move(aSource), std::move(aTrackingId),
+                              mConduit, std::move(aPrincipalHandle), aPrivacy)
+                        : nullptr) {
+  mDescription = mPc + "| Receive audio";
+  if (mListener) {
+    mListener->Init();
+  }
+}
+
+void MediaPipelineReceiveAudio::Shutdown() {
+  MOZ_ASSERT(NS_IsMainThread());
+  MediaPipelineReceive::Shutdown();
+  if (mListener) {
+    mListener->Shutdown();
+  }
+}
+
+void MediaPipelineReceiveAudio::OnPrivacyRequested_s() {
+  ASSERT_ON_THREAD(mStsThread);
+  if (mListener) {
+    mListener->OnPrivacyRequested_s();
+  }
+}
+
+void MediaPipelineReceiveAudio::SetPrivatePrincipal(PrincipalHandle aHandle) {
+  MOZ_ASSERT(NS_IsMainThread());
+  if (mListener) {
+    mListener->SetPrivatePrincipal(std::move(aHandle));
+  }
+}
+
+void MediaPipelineReceiveAudio::UpdateListener() {
+  MOZ_ASSERT(NS_IsMainThread());
+  if (mListener) {
+    mListener->SetEnabled(mActive.Ref());
+  }
+}
+
+class MediaPipelineReceiveVideo::PipelineListener
+    : public GenericReceiveListener {
+ public:
+  PipelineListener(RefPtr<SourceMediaTrack> aSource, TrackingId aTrackingId,
+                   PrincipalHandle aPrincipalHandle, PrincipalPrivacy aPrivacy)
+      : GenericReceiveListener(std::move(aSource), std::move(aTrackingId)),
+        mImageContainer(
+            MakeAndAddRef<ImageContainer>(ImageContainer::ASYNCHRONOUS)),
+        mMutex("MediaPipelineReceiveVideo::PipelineListener::mMutex"),
+        mPrincipalHandle(std::move(aPrincipalHandle)),
+        mPrivacy(aPrivacy) {}
+  void OnPrivacyRequested_s() {
+    MutexAutoLock lock(mMutex);
+    if (mPrivacy == PrincipalPrivacy::Private) {
+      return;
+    }
+    mForceDropFrames = true;
+  }
+
+  void SetPrivatePrincipal(PrincipalHandle aHandle) {
+    MutexAutoLock lock(mMutex);
+    if (mPrivacy == PrincipalPrivacy::Private) {
+      return;
+    }
+    mPrincipalHandle = std::move(aHandle);
+    mPrivacy = PrincipalPrivacy::Private;
+    mForceDropFrames = false;
+  }
+
+  void RenderVideoFrame(const webrtc::VideoFrameBuffer& aBuffer,
+                        uint32_t aTimeStamp, int64_t aRenderTime) {
+    PrincipalHandle principal;
+    {
+      MutexAutoLock lock(mMutex);
+      if (mForceDropFrames) {
+        return;
+      }
+      principal = mPrincipalHandle;
+    }
+    RefPtr<Image> image;
+    if (aBuffer.type() == webrtc::VideoFrameBuffer::Type::kNative) {
+      // We assume that only native handles are used with the
+      // WebrtcMediaDataCodec decoder.
+      const ImageBuffer* imageBuffer =
+          static_cast<const ImageBuffer*>(&aBuffer);
+      image = imageBuffer->GetNativeImage();
+    } else {
+      MOZ_ASSERT(aBuffer.type() == webrtc::VideoFrameBuffer::Type::kI420);
+      rtc::scoped_refptr<const webrtc::I420BufferInterface> i420(
+          aBuffer.GetI420());
+
+      MOZ_ASSERT(i420->DataY());
+      // Create a video frame using |buffer|.
+      PerformanceRecorder<CopyVideoStage> rec(
+          "MediaPipelineReceiveVideo::CopyToImage"_ns, mTrackingId,
+          i420->width(), i420->height());
+
+      RefPtr<PlanarYCbCrImage> yuvImage =
+          mImageContainer->CreatePlanarYCbCrImage();
+
+      PlanarYCbCrData yuvData;
+      yuvData.mYChannel = const_cast<uint8_t*>(i420->DataY());
+      yuvData.mYStride = i420->StrideY();
+      MOZ_ASSERT(i420->StrideU() == i420->StrideV());
+      yuvData.mCbCrStride = i420->StrideU();
+      yuvData.mCbChannel = const_cast<uint8_t*>(i420->DataU());
+      yuvData.mCrChannel = const_cast<uint8_t*>(i420->DataV());
+      yuvData.mPictureRect = IntRect(0, 0, i420->width(), i420->height());
+      yuvData.mStereoMode = StereoMode::MONO;
+      // This isn't the best default.
+      yuvData.mYUVColorSpace = gfx::YUVColorSpace::BT601;
+      yuvData.mChromaSubsampling =
+          gfx::ChromaSubsampling::HALF_WIDTH_AND_HEIGHT;
+
+      if (!yuvImage->CopyData(yuvData)) {
+        MOZ_ASSERT(false);
+        return;
+      }
+      rec.Record();
+
+      image = std::move(yuvImage);
+    }
+
+    VideoSegment segment;
+    auto size = image->GetSize();
+    segment.AppendFrame(image.forget(), size, principal);
+    mSource->AppendData(&segment);
+  }
+
+ private:
+  RefPtr<layers::ImageContainer> mImageContainer;
+  Mutex mMutex;
+  PrincipalHandle mPrincipalHandle MOZ_GUARDED_BY(mMutex);
+  PrincipalPrivacy mPrivacy MOZ_GUARDED_BY(mMutex);
+  // Set to true on the sts thread if privacy is requested when ALPN was
+  // negotiated. Set to false again when mPrincipalHandle is private.
+  bool mForceDropFrames MOZ_GUARDED_BY(mMutex) = false;
+};
+
+class MediaPipelineReceiveVideo::PipelineRenderer
+    : public mozilla::VideoRenderer {
+ public:
+  explicit PipelineRenderer(MediaPipelineReceiveVideo* aPipeline)
+      : mPipeline(aPipeline) {}
+
+  void Detach() { mPipeline = nullptr; }
+
+  // Implement VideoRenderer
+  void FrameSizeChange(unsigned int aWidth, unsigned int aHeight) override {}
+  void RenderVideoFrame(const webrtc::VideoFrameBuffer& aBuffer,
+                        uint32_t aTimeStamp, int64_t aRenderTime) override {
+    mPipeline->mListener->RenderVideoFrame(aBuffer, aTimeStamp, aRenderTime);
+  }
+
+ private:
+  MediaPipelineReceiveVideo* mPipeline;  // Raw pointer to avoid cycles
+};
+
+MediaPipelineReceiveVideo::MediaPipelineReceiveVideo(
+    const std::string& aPc, RefPtr<MediaTransportHandler> aTransportHandler,
+    RefPtr<AbstractThread> aCallThread, RefPtr<nsISerialEventTarget> aStsThread,
+    RefPtr<VideoSessionConduit> aConduit, RefPtr<SourceMediaTrack> aSource,
+    TrackingId aTrackingId, PrincipalHandle aPrincipalHandle,
+    PrincipalPrivacy aPrivacy)
+    : MediaPipelineReceive(aPc, std::move(aTransportHandler),
+                           std::move(aCallThread), std::move(aStsThread),
+                           std::move(aConduit)),
+      mRenderer(new PipelineRenderer(this)),
+      mListener(aSource ? new PipelineListener(
+                              std::move(aSource), std::move(aTrackingId),
+                              std::move(aPrincipalHandle), aPrivacy)
+                        : nullptr) {
+  mDescription = mPc + "| Receive video";
+  if (mListener) {
+    mListener->Init();
+  }
+  static_cast<VideoSessionConduit*>(mConduit.get())->AttachRenderer(mRenderer);
+}
+
+void MediaPipelineReceiveVideo::Shutdown() {
+  MOZ_ASSERT(NS_IsMainThread());
+  MediaPipelineReceive::Shutdown();
+  if (mListener) {
+    mListener->Shutdown();
+  }
+
+  // stop generating video and thus stop invoking the PipelineRenderer
+  // and PipelineListener - the renderer has a raw ptr to the Pipeline to
+  // avoid cycles, and the render callbacks are invoked from a different
+  // thread so simple null-checks would cause TSAN bugs without locks.
+  static_cast<VideoSessionConduit*>(mConduit.get())->DetachRenderer();
+}
+
+void MediaPipelineReceiveVideo::OnPrivacyRequested_s() {
+  ASSERT_ON_THREAD(mStsThread);
+  if (mListener) {
+    mListener->OnPrivacyRequested_s();
+  }
+}
+
+void MediaPipelineReceiveVideo::SetPrivatePrincipal(PrincipalHandle aHandle) {
+  MOZ_ASSERT(NS_IsMainThread());
+  if (mListener) {
+    mListener->SetPrivatePrincipal(std::move(aHandle));
+  }
+}
+
+void MediaPipelineReceiveVideo::UpdateListener() {
+  MOZ_ASSERT(NS_IsMainThread());
+  if (mListener) {
+    mListener->SetEnabled(mActive.Ref());
+  }
+}
+
+const dom::RTCStatsTimestampMaker& MediaPipeline::GetTimestampMaker() const {
+  return mConduit->GetTimestampMaker();
+}
+
+DOMHighResTimeStamp MediaPipeline::RtpCSRCStats::GetExpiryFromTime(
+    const DOMHighResTimeStamp aTime) {
+  // DOMHighResTimeStamp is a unit measured in ms
+  return aTime + EXPIRY_TIME_MILLISECONDS;
+}
+
+MediaPipeline::RtpCSRCStats::RtpCSRCStats(const uint32_t aCsrc,
+                                          const DOMHighResTimeStamp aTime)
+    : mCsrc(aCsrc), mTimestamp(aTime) {}
+
+void MediaPipeline::RtpCSRCStats::GetWebidlInstance(
+    dom::RTCRTPContributingSourceStats& aWebidlObj,
+    const nsString& aInboundRtpStreamId) const {
+  nsString statId = u"csrc_"_ns + aInboundRtpStreamId;
+  statId.AppendLiteral("_");
+  statId.AppendInt(mCsrc);
+  aWebidlObj.mId.Construct(statId);
+  aWebidlObj.mType.Construct(RTCStatsType::Csrc);
+  aWebidlObj.mTimestamp.Construct(mTimestamp);
+  aWebidlObj.mContributorSsrc.Construct(mCsrc);
+  aWebidlObj.mInboundRtpStreamId.Construct(aInboundRtpStreamId);
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/transportbridge/MediaPipeline.h b/dom/media/webrtc/transportbridge/MediaPipeline.h
new file mode 100644
index 0000000000..d58fd12ea3
--- /dev/null
+++ b/dom/media/webrtc/transportbridge/MediaPipeline.h
@@ -0,0 +1,454 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: ekr@rtfm.com
+
+#ifndef mediapipeline_h__
+#define mediapipeline_h__
+
+#include <map>
+
+#include "transport/sigslot.h"
+#include "transport/transportlayer.h"  // For TransportLayer::State
+
+#include "libwebrtcglue/MediaConduitControl.h"
+#include "mozilla/ReentrantMonitor.h"
+#include "mozilla/Atomics.h"
+#include "mozilla/StateMirroring.h"
+#include "transport/mediapacket.h"
+#include "transport/runnable_utils.h"
+#include "AudioPacketizer.h"
+#include "MediaEventSource.h"
+#include "MediaPipelineFilter.h"
+#include "MediaSegment.h"
+#include "PrincipalChangeObserver.h"
+#include "jsapi/PacketDumper.h"
+#include "PerformanceRecorder.h"
+
+// Should come from MediaEngine.h, but that's a pain to include here
+// because of the MOZILLA_EXTERNAL_LINKAGE stuff.
+#define WEBRTC_MAX_SAMPLE_RATE 48000
+
+class nsIPrincipal;
+
+namespace webrtc {
+struct RTPHeader;
+class RtpHeaderExtensionMap;
+class RtpPacketReceived;
+}  // namespace webrtc
+
+namespace mozilla {
+class AudioProxyThread;
+class MediaInputPort;
+class MediaPipelineFilter;
+class MediaTransportHandler;
+class PeerIdentity;
+class ProcessedMediaTrack;
+class SourceMediaTrack;
+class VideoFrameConverter;
+class MediaSessionConduit;
+class AudioSessionConduit;
+class VideoSessionConduit;
+
+namespace dom {
+class MediaStreamTrack;
+struct RTCRTPContributingSourceStats;
+class RTCStatsTimestampMaker;
+}  // namespace dom
+
+struct MediaPipelineReceiveControlInterface {
+  virtual AbstractCanonical<bool>* CanonicalReceiving() = 0;
+};
+
+struct MediaPipelineTransmitControlInterface {
+  virtual AbstractCanonical<bool>* CanonicalTransmitting() = 0;
+};
+
+// A class that represents the pipeline of audio and video
+// The dataflow looks like:
+//
+// TRANSMIT
+// CaptureDevice -> stream -> [us] -> conduit -> [us] -> transport -> network
+//
+// RECEIVE
+// network -> transport -> [us] -> conduit -> [us] -> stream -> Playout
+//
+// The boxes labeled [us] are just bridge logic implemented in this class
+//
+// We have to deal with a number of threads:
+//
+// GSM:
+//   * Assembles the pipeline
+// SocketTransportService
+//   * Receives notification that ICE and DTLS have completed
+//   * Processes incoming network data and passes it to the conduit
+//   * Processes outgoing RTP and RTCP
+// MediaTrackGraph
+//   * Receives outgoing data from the MediaTrackGraph
+//   * Receives pull requests for more data from the
+//     MediaTrackGraph
+// One or another GIPS threads
+//   * Receives RTCP messages to send to the other side
+//   * Processes video frames GIPS wants to render
+//
+// For a transmitting conduit, "output" is RTP and "input" is RTCP.
+// For a receiving conduit, "input" is RTP and "output" is RTCP.
+//
+
+class MediaPipeline : public sigslot::has_slots<> {
+ public:
+  enum class DirectionType { TRANSMIT, RECEIVE };
+  MediaPipeline(const std::string& aPc,
+                RefPtr<MediaTransportHandler> aTransportHandler,
+                DirectionType aDirection, RefPtr<AbstractThread> aCallThread,
+                RefPtr<nsISerialEventTarget> aStsThread,
+                RefPtr<MediaSessionConduit> aConduit);
+
+  void SetLevel(size_t aLevel) { mLevel = aLevel; }
+
+  // Main thread shutdown.
+  virtual void Shutdown();
+
+  void UpdateTransport_m(const std::string& aTransportId,
+                         UniquePtr<MediaPipelineFilter>&& aFilter);
+
+  void UpdateTransport_s(const std::string& aTransportId,
+                         UniquePtr<MediaPipelineFilter>&& aFilter);
+
+  virtual DirectionType Direction() const { return mDirection; }
+  size_t Level() const { return mLevel; }
+  virtual bool IsVideo() const = 0;
+
+  class RtpCSRCStats {
+   public:
+    // Gets an expiration time for CRC info given a reference time,
+    //   this reference time would normally be the time of calling.
+    //   This value can then be used to check if a RtpCSRCStats
+    //   has expired via Expired(...)
+    static DOMHighResTimeStamp GetExpiryFromTime(
+        const DOMHighResTimeStamp aTime);
+
+    RtpCSRCStats(const uint32_t aCsrc, const DOMHighResTimeStamp aTime);
+    ~RtpCSRCStats() = default;
+    // Initialize a webidl representation suitable for adding to a report.
+    //   This assumes that the webidl object is empty.
+    // @param aWebidlObj the webidl binding object to popluate
+    // @param aInboundRtpStreamId the associated RTCInboundRTPStreamStats.id
+    void GetWebidlInstance(dom::RTCRTPContributingSourceStats& aWebidlObj,
+                           const nsString& aInboundRtpStreamId) const;
+    void SetTimestamp(const DOMHighResTimeStamp aTime) { mTimestamp = aTime; }
+    // Check if the RtpCSRCStats has expired, checks against a
+    //   given expiration time.
+    bool Expired(const DOMHighResTimeStamp aExpiry) const {
+      return mTimestamp < aExpiry;
+    }
+
+   private:
+    static const double constexpr EXPIRY_TIME_MILLISECONDS = 10 * 1000;
+    const uint32_t mCsrc;
+    DOMHighResTimeStamp mTimestamp;
+  };
+
+  // Gets the gathered contributing source stats for the last expiration period.
+  // @param aId the stream id to use for populating inboundRtpStreamId field
+  // @param aArr the array to append the stats objects to
+  void GetContributingSourceStats(
+      const nsString& aInboundRtpStreamId,
+      FallibleTArray<dom::RTCRTPContributingSourceStats>& aArr) const;
+
+  int32_t RtpPacketsSent() const { return mRtpPacketsSent; }
+  int64_t RtpBytesSent() const { return mRtpBytesSent; }
+  int32_t RtcpPacketsSent() const { return mRtcpPacketsSent; }
+  int32_t RtpPacketsReceived() const { return mRtpPacketsReceived; }
+  int64_t RtpBytesReceived() const { return mRtpBytesReceived; }
+  int32_t RtcpPacketsReceived() const { return mRtcpPacketsReceived; }
+
+  const dom::RTCStatsTimestampMaker& GetTimestampMaker() const;
+
+  // Thread counting
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(MediaPipeline)
+
+ protected:
+  virtual ~MediaPipeline();
+
+  // The transport is ready
+  virtual void TransportReady_s() {}
+
+  void IncrementRtpPacketsSent(const MediaPacket& aPacket);
+  void IncrementRtcpPacketsSent();
+  void IncrementRtpPacketsReceived(int aBytes);
+  void IncrementRtcpPacketsReceived();
+
+  virtual void SendPacket(MediaPacket&& packet);
+
+  // Process slots on transports
+  void RtpStateChange(const std::string& aTransportId, TransportLayer::State);
+  void RtcpStateChange(const std::string& aTransportId, TransportLayer::State);
+  virtual void CheckTransportStates();
+  void PacketReceived(const std::string& aTransportId,
+                      const MediaPacket& packet);
+  void AlpnNegotiated(const std::string& aAlpn, bool aPrivacyRequested);
+
+  void RtpPacketReceived(const MediaPacket& packet);
+  void RtcpPacketReceived(const MediaPacket& packet);
+
+  void EncryptedPacketSending(const std::string& aTransportId,
+                              const MediaPacket& aPacket);
+
+  void SetDescription_s(const std::string& description);
+
+ public:
+  const RefPtr<MediaSessionConduit> mConduit;
+  const DirectionType mDirection;
+
+  // Pointers to the threads we need. Initialized at creation
+  // and used all over the place.
+  const RefPtr<AbstractThread> mCallThread;
+  const RefPtr<nsISerialEventTarget> mStsThread;
+
+ protected:
+  // True if we should be actively transmitting or receiving data. Main thread
+  // only.
+  Mirror<bool> mActive;
+  Atomic<size_t> mLevel;
+  std::string mTransportId;
+  const RefPtr<MediaTransportHandler> mTransportHandler;
+
+  TransportLayer::State mRtpState = TransportLayer::TS_NONE;
+  TransportLayer::State mRtcpState = TransportLayer::TS_NONE;
+  bool mSignalsConnected = false;
+
+  // Only safe to access from STS thread.
+  int32_t mRtpPacketsSent;
+  int32_t mRtcpPacketsSent;
+  int32_t mRtpPacketsReceived;
+  int32_t mRtcpPacketsReceived;
+  int64_t mRtpBytesSent;
+  int64_t mRtpBytesReceived;
+
+  // Only safe to access from STS thread.
+  std::map<uint32_t, RtpCSRCStats> mCsrcStats;
+
+  // Written in c'tor. Read on STS and main thread.
+  const std::string mPc;
+
+  // String describing this MediaPipeline for logging purposes. Only safe to
+  // access from STS thread.
+  std::string mDescription;
+
+  // Written in c'tor, all following accesses are on the STS thread.
+  UniquePtr<MediaPipelineFilter> mFilter;
+  const UniquePtr<webrtc::RtpHeaderExtensionMap> mRtpHeaderExtensionMap;
+
+  RefPtr<PacketDumper> mPacketDumper;
+
+  MediaEventProducerExc<webrtc::RtpPacketReceived, webrtc::RTPHeader>
+      mRtpReceiveEvent;
+  MediaEventProducerExc<MediaPacket> mSenderRtcpReceiveEvent;
+  MediaEventProducerExc<MediaPacket> mReceiverRtcpReceiveEvent;
+
+  MediaEventListener mRtpSendEventListener;
+  MediaEventListener mSenderRtcpSendEventListener;
+  MediaEventListener mReceiverRtcpSendEventListener;
+
+ private:
+  bool IsRtp(const unsigned char* aData, size_t aLen) const;
+  // Must be called on the STS thread.  Must be called after Shutdown().
+  void DetachTransport_s();
+};
+
+// A specialization of pipeline for reading from an input device
+// and transmitting to the network.
+class MediaPipelineTransmit
+    : public MediaPipeline,
+      public dom::PrincipalChangeObserver<dom::MediaStreamTrack> {
+ public:
+  // Set aRtcpTransport to nullptr to use rtcp-mux
+  MediaPipelineTransmit(const std::string& aPc,
+                        RefPtr<MediaTransportHandler> aTransportHandler,
+                        RefPtr<AbstractThread> aCallThread,
+                        RefPtr<nsISerialEventTarget> aStsThread, bool aIsVideo,
+                        RefPtr<MediaSessionConduit> aConduit);
+
+  void InitControl(MediaPipelineTransmitControlInterface* aControl);
+
+  void Shutdown() override;
+
+  bool Transmitting() const;
+
+  // written and used from MainThread
+  bool IsVideo() const override;
+
+  // When the principal of the domtrack changes, it calls through to here
+  // so that we can determine whether to enable track transmission.
+  // In cases where the peer isn't yet identified, we disable the pipeline (not
+  // the stream, that would potentially affect others), so that it sends
+  // black/silence.  Once the peer is identified, re-enable those streams.
+  virtual void UpdateSinkIdentity(nsIPrincipal* aPrincipal,
+                                  const PeerIdentity* aSinkIdentity);
+
+  // for monitoring changes in track ownership
+  void PrincipalChanged(dom::MediaStreamTrack* aTrack) override;
+
+  // Override MediaPipeline::TransportReady_s.
+  void TransportReady_s() override;
+
+  // Replace a track with a different one.
+  nsresult SetTrack(const RefPtr<dom::MediaStreamTrack>& aDomTrack);
+
+  // Used to correlate stats
+  RefPtr<dom::MediaStreamTrack> GetTrack() const;
+
+  // For test use only. This allows a send track to be set without a
+  // corresponding dom track.
+  void SetSendTrackOverride(const RefPtr<ProcessedMediaTrack>& aSendTrack);
+
+  // Separate classes to allow ref counting
+  class PipelineListener;
+  class VideoFrameFeeder;
+
+ protected:
+  ~MediaPipelineTransmit();
+
+  // Updates mDescription (async) with information about the track we are
+  // transmitting.
+  std::string GenerateDescription() const;
+
+  // Sets up mSendPort and mSendTrack to feed mConduit if we are transmitting
+  // and have a dom track but no send track. Main thread only.
+  void UpdateSendState();
+
+ private:
+  WatchManager<MediaPipelineTransmit> mWatchManager;
+  const bool mIsVideo;
+  const RefPtr<PipelineListener> mListener;
+  RefPtr<AudioProxyThread> mAudioProcessing;
+  RefPtr<VideoFrameConverter> mConverter;
+  MediaEventListener mFrameListener;
+  Watchable<RefPtr<dom::MediaStreamTrack>> mDomTrack;
+  // Input port connecting mDomTrack's MediaTrack to mSendTrack.
+  RefPtr<MediaInputPort> mSendPort;
+  // The source track of the mSendTrack. Main thread only.
+  RefPtr<ProcessedMediaTrack> mSendPortSource;
+  // True if a parameter affecting mDescription has changed. To avoid updating
+  // the description unnecessarily. Main thread only.
+  bool mDescriptionInvalidated = true;
+  // Set true once we trigger the async removal of mSendTrack. Set false once
+  // the async removal is done. Main thread only.
+  bool mUnsettingSendTrack = false;
+  // MediaTrack that we send over the network. This allows changing mDomTrack.
+  // Because changing mSendTrack is async and can be racy (when changing from a
+  // track in one graph to a track in another graph), it is set very strictly.
+  // If mSendTrack is null it can be set by UpdateSendState().
+  // If it is non-null it can only be set to null, and only by the
+  // RemoveListener MozPromise handler, as seen in UpdateSendState.
+  RefPtr<ProcessedMediaTrack> mSendTrack;
+  // When this is set and we are active, this track will be used as mSendTrack.
+  // Allows unittests to insert a send track without requiring a dom track or a
+  // graph. Main thread only.
+  Watchable<RefPtr<ProcessedMediaTrack>> mSendTrackOverride;
+  // True when mSendTrack is set, not destroyed and mActive is true. mListener
+  // is attached to mSendTrack when this is true. Main thread only.
+  bool mTransmitting = false;
+};
+
+// A specialization of pipeline for reading from the network and
+// rendering media.
+class MediaPipelineReceive : public MediaPipeline {
+ public:
+  // Set aRtcpTransport to nullptr to use rtcp-mux
+  MediaPipelineReceive(const std::string& aPc,
+                       RefPtr<MediaTransportHandler> aTransportHandler,
+                       RefPtr<AbstractThread> aCallThread,
+                       RefPtr<nsISerialEventTarget> aStsThread,
+                       RefPtr<MediaSessionConduit> aConduit);
+
+  void InitControl(MediaPipelineReceiveControlInterface* aControl);
+
+  // Called when ALPN is negotiated and is requesting privacy, so receive
+  // pipelines do not enter data into the graph under a content principal.
+  virtual void OnPrivacyRequested_s() = 0;
+
+  // Called after privacy has been requested, with the updated private
+  // principal.
+  virtual void SetPrivatePrincipal(PrincipalHandle aHandle) = 0;
+
+  void Shutdown() override;
+
+ protected:
+  ~MediaPipelineReceive();
+
+  virtual void UpdateListener() = 0;
+
+ private:
+  WatchManager<MediaPipelineReceive> mWatchManager;
+};
+
+// A specialization of pipeline for reading from the network and
+// rendering audio.
+class MediaPipelineReceiveAudio : public MediaPipelineReceive {
+ public:
+  MediaPipelineReceiveAudio(const std::string& aPc,
+                            RefPtr<MediaTransportHandler> aTransportHandler,
+                            RefPtr<AbstractThread> aCallThread,
+                            RefPtr<nsISerialEventTarget> aStsThread,
+                            RefPtr<AudioSessionConduit> aConduit,
+                            RefPtr<SourceMediaTrack> aSource,
+                            TrackingId aTrackingId,
+                            PrincipalHandle aPrincipalHandle,
+                            PrincipalPrivacy aPrivacy);
+
+  void Shutdown() override;
+
+  bool IsVideo() const override { return false; }
+
+  void OnPrivacyRequested_s() override;
+  void SetPrivatePrincipal(PrincipalHandle aHandle) override;
+
+ private:
+  void UpdateListener() override;
+
+  // Separate class to allow ref counting
+  class PipelineListener;
+
+  const RefPtr<PipelineListener> mListener;
+};
+
+// A specialization of pipeline for reading from the network and
+// rendering video.
+class MediaPipelineReceiveVideo : public MediaPipelineReceive {
+ public:
+  MediaPipelineReceiveVideo(const std::string& aPc,
+                            RefPtr<MediaTransportHandler> aTransportHandler,
+                            RefPtr<AbstractThread> aCallThread,
+                            RefPtr<nsISerialEventTarget> aStsThread,
+                            RefPtr<VideoSessionConduit> aConduit,
+                            RefPtr<SourceMediaTrack> aSource,
+                            TrackingId aTrackingId,
+                            PrincipalHandle aPrincipalHandle,
+                            PrincipalPrivacy aPrivacy);
+
+  void Shutdown() override;
+
+  bool IsVideo() const override { return true; }
+
+  void OnPrivacyRequested_s() override;
+  void SetPrivatePrincipal(PrincipalHandle aHandle) override;
+
+ private:
+  void UpdateListener() override;
+
+  class PipelineRenderer;
+  friend class PipelineRenderer;
+
+  // Separate class to allow ref counting
+  class PipelineListener;
+
+  const RefPtr<PipelineRenderer> mRenderer;
+  const RefPtr<PipelineListener> mListener;
+};
+
+}  // namespace mozilla
+#endif
diff --git a/dom/media/webrtc/transportbridge/MediaPipelineFilter.cpp b/dom/media/webrtc/transportbridge/MediaPipelineFilter.cpp
new file mode 100644
index 0000000000..1acb73e9f2
--- /dev/null
+++ b/dom/media/webrtc/transportbridge/MediaPipelineFilter.cpp
@@ -0,0 +1,153 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: softtabstop=2:shiftwidth=2:expandtab
+ * */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: bcampen@mozilla.com
+
+#include "MediaPipelineFilter.h"
+
+#include "api/rtp_headers.h"
+#include "api/rtp_parameters.h"
+#include "mozilla/Logging.h"
+
+// defined in MediaPipeline.cpp
+extern mozilla::LazyLogModule gMediaPipelineLog;
+
+#define DEBUG_LOG(x) MOZ_LOG(gMediaPipelineLog, LogLevel::Debug, x)
+
+namespace mozilla {
+MediaPipelineFilter::MediaPipelineFilter(
+    const std::vector<webrtc::RtpExtension>& aExtMap)
+    : mExtMap(aExtMap) {}
+
+void MediaPipelineFilter::SetRemoteMediaStreamId(
+    const Maybe<std::string>& aMid) {
+  if (aMid != mRemoteMid) {
+    DEBUG_LOG(("MediaPipelineFilter added new remote RTP MID: '%s'.",
+               aMid.valueOr("").c_str()));
+    mRemoteMid = aMid;
+    mRemoteMidBindings.clear();
+  }
+}
+
+bool MediaPipelineFilter::Filter(const webrtc::RTPHeader& header) {
+  DEBUG_LOG(("MediaPipelineFilter inspecting seq# %u SSRC: %u",
+             header.sequenceNumber, header.ssrc));
+
+  auto fromStreamId = [](const std::string& aId) {
+    return Maybe<std::string>(aId.empty() ? Nothing() : Some(aId));
+  };
+
+  //
+  //  MID Based Filtering
+  //
+
+  const auto mid = fromStreamId(header.extension.mid);
+
+  // Check to see if a bound SSRC is moved to a new MID
+  if (mRemoteMidBindings.count(header.ssrc) == 1 && mid && mRemoteMid != mid) {
+    mRemoteMidBindings.erase(header.ssrc);
+  }
+  // Bind an SSRC if a matching MID is found
+  if (mid && mRemoteMid == mid) {
+    DEBUG_LOG(("MediaPipelineFilter learned SSRC: %u for MID: '%s'",
+               header.ssrc, mRemoteMid.value().c_str()));
+    mRemoteMidBindings.insert(header.ssrc);
+  }
+  // Check for matching MID
+  if (!mRemoteMidBindings.empty()) {
+    MOZ_ASSERT(mRemoteMid != Nothing());
+    if (mRemoteMidBindings.count(header.ssrc) == 1) {
+      DEBUG_LOG(
+          ("MediaPipelineFilter SSRC: %u matched for MID: '%s'."
+           " passing packet",
+           header.ssrc, mRemoteMid.value().c_str()));
+      return true;
+    }
+    DEBUG_LOG(
+        ("MediaPipelineFilter SSRC: %u did not match bound SSRC(s) for"
+         " MID: '%s'. ignoring packet",
+         header.ssrc, mRemoteMid.value().c_str()));
+    for (const uint32_t ssrc : mRemoteMidBindings) {
+      DEBUG_LOG(("MID %s is associated with SSRC: %u",
+                 mRemoteMid.value().c_str(), ssrc));
+    }
+    return false;
+  }
+
+  //
+  // RTP-STREAM-ID based filtering (for tests only)
+  //
+
+  //
+  // Remote SSRC based filtering
+  //
+
+  if (remote_ssrc_set_.count(header.ssrc)) {
+    DEBUG_LOG(
+        ("MediaPipelineFilter SSRC: %u matched remote SSRC set."
+         " passing packet",
+         header.ssrc));
+    return true;
+  }
+  DEBUG_LOG(
+      ("MediaPipelineFilter SSRC: %u did not match any of %zu"
+       " remote SSRCS.",
+       header.ssrc, remote_ssrc_set_.size()));
+
+  //
+  // PT, payload type, last ditch effort filtering
+  //
+
+  if (payload_type_set_.count(header.payloadType)) {
+    DEBUG_LOG(
+        ("MediaPipelineFilter payload-type: %u matched %zu"
+         " unique payload type. learning ssrc. passing packet",
+         header.ssrc, remote_ssrc_set_.size()));
+    // Actual match. We need to update the ssrc map so we can route rtcp
+    // sender reports correctly (these use a different payload-type field)
+    AddRemoteSSRC(header.ssrc);
+    return true;
+  }
+  DEBUG_LOG(
+      ("MediaPipelineFilter payload-type: %u did not match any of %zu"
+       " unique payload-types.",
+       header.payloadType, payload_type_set_.size()));
+  DEBUG_LOG(
+      ("MediaPipelineFilter packet failed to match any criteria."
+       " ignoring packet"));
+  return false;
+}
+
+void MediaPipelineFilter::AddRemoteSSRC(uint32_t ssrc) {
+  remote_ssrc_set_.insert(ssrc);
+}
+
+void MediaPipelineFilter::AddUniquePT(uint8_t payload_type) {
+  payload_type_set_.insert(payload_type);
+}
+
+void MediaPipelineFilter::Update(const MediaPipelineFilter& filter_update) {
+  // We will not stomp the remote_ssrc_set_ if the update has no ssrcs,
+  // because we don't want to unlearn any remote ssrcs unless the other end
+  // has explicitly given us a new set.
+  if (!filter_update.remote_ssrc_set_.empty()) {
+    remote_ssrc_set_ = filter_update.remote_ssrc_set_;
+  }
+  // We don't want to overwrite the learned binding unless we have changed MIDs
+  // or the update contains a MID binding.
+  if (!filter_update.mRemoteMidBindings.empty() ||
+      (filter_update.mRemoteMid && filter_update.mRemoteMid != mRemoteMid)) {
+    mRemoteMid = filter_update.mRemoteMid;
+    mRemoteMidBindings = filter_update.mRemoteMidBindings;
+  }
+  payload_type_set_ = filter_update.payload_type_set_;
+
+  // Use extmapping from new filter
+  mExtMap = filter_update.mExtMap;
+}
+
+}  // end namespace mozilla
diff --git a/dom/media/webrtc/transportbridge/MediaPipelineFilter.h b/dom/media/webrtc/transportbridge/MediaPipelineFilter.h
new file mode 100644
index 0000000000..9b40bceda8
--- /dev/null
+++ b/dom/media/webrtc/transportbridge/MediaPipelineFilter.h
@@ -0,0 +1,89 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: softtabstop=2:shiftwidth=2:expandtab
+ * */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: bcampen@mozilla.com
+
+#ifndef mediapipelinefilter_h__
+#define mediapipelinefilter_h__
+
+#include <cstddef>
+#include <stdint.h>
+#include <string>
+
+#include <set>
+#include <vector>
+
+#include "mozilla/Maybe.h"
+
+namespace webrtc {
+struct RTPHeader;
+struct RtpExtension;
+}  // namespace webrtc
+
+namespace mozilla {
+
+// TODO @@NG update documentation after initial review
+
+// A class that handles the work of filtering RTP packets that arrive at a
+// MediaPipeline. This is primarily important for the use of BUNDLE (ie;
+// multiple m-lines share the same RTP stream). There are three ways that this
+// can work;
+//
+// 1) In our SDP, we include a media-level extmap parameter with a unique
+//    integer of our choosing, with the hope that the other side will include
+//    this value in a header in the first few RTP packets it sends us. This
+//    allows us to perform correlation in cases where the other side has not
+//    informed us of the ssrcs it will be sending (either because it did not
+//    include them in its SDP, or their SDP has not arrived yet)
+//    and also gives us the opportunity to learn SSRCs from packets so adorned.
+//
+// 2) If the remote endpoint includes SSRC media-level attributes in its SDP,
+//    we can simply use this information to populate the filter. The only
+//    shortcoming here is when RTP packets arrive before the answer does. See
+//    above.
+//
+// 3) As a fallback, we can try to use payload type IDs to perform correlation,
+//    but only when the type id is unique to this media section.
+//    This too allows us to learn about SSRCs (mostly useful for filtering
+//    sender reports later).
+class MediaPipelineFilter {
+ public:
+  MediaPipelineFilter() = default;
+  explicit MediaPipelineFilter(
+      const std::vector<webrtc::RtpExtension>& aExtMap);
+
+  // Checks whether this packet passes the filter, possibly updating the filter
+  // in the process (if the MID or payload types are used, they can teach
+  // the filter about ssrcs)
+  bool Filter(const webrtc::RTPHeader& header);
+
+  void AddRemoteSSRC(uint32_t ssrc);
+
+  void SetRemoteMediaStreamId(const Maybe<std::string>& aMid);
+
+  // When a payload type id is unique to our media section, add it here.
+  void AddUniquePT(uint8_t payload_type);
+
+  void Update(const MediaPipelineFilter& filter_update);
+
+  std::vector<webrtc::RtpExtension> GetExtmap() const { return mExtMap; }
+
+ private:
+  // The number of filters we manage here is quite small, so I am optimizing
+  // for readability.
+  std::set<uint32_t> remote_ssrc_set_;
+  std::set<uint8_t> payload_type_set_;
+  Maybe<std::string> mRemoteMid;
+  std::set<uint32_t> mRemoteMidBindings;
+  // RID extension can be set by tests and is sticky, the rest of
+  // the mapping is not.
+  std::vector<webrtc::RtpExtension> mExtMap;
+};
+
+}  // end namespace mozilla
+
+#endif  // mediapipelinefilter_h__
diff --git a/dom/media/webrtc/transportbridge/RtpLogger.cpp b/dom/media/webrtc/transportbridge/RtpLogger.cpp
new file mode 100644
index 0000000000..aac1fad197
--- /dev/null
+++ b/dom/media/webrtc/transportbridge/RtpLogger.cpp
@@ -0,0 +1,67 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: nohlmeier@mozilla.com
+
+#include "RtpLogger.h"
+#include "mozilla/Logging.h"
+
+#include <ctime>
+#include <iomanip>
+#include <sstream>
+#ifdef _WIN32
+#  include <time.h>
+#  include <sys/timeb.h>
+#else
+#  include <sys/time.h>
+#endif
+
+// Logging context
+using namespace mozilla;
+
+mozilla::LazyLogModule gRtpLoggerLog("RtpLogger");
+
+namespace mozilla {
+
+bool RtpLogger::IsPacketLoggingOn() {
+  return MOZ_LOG_TEST(gRtpLoggerLog, LogLevel::Debug);
+}
+
+void RtpLogger::LogPacket(const MediaPacket& packet, bool input,
+                          std::string desc) {
+  if (MOZ_LOG_TEST(gRtpLoggerLog, LogLevel::Debug)) {
+    bool isRtp = (packet.type() == MediaPacket::RTP);
+    std::stringstream ss;
+    /* This creates text2pcap compatible format, e.g.:
+     *  RTCP_PACKET O 10:36:26.864934  000000 80 c8 00 06 6d ...
+     */
+    ss << (input ? "I " : "O ");
+    std::time_t t = std::time(nullptr);
+    std::tm tm = *std::localtime(&t);
+    char buf[9];
+    if (0 < strftime(buf, sizeof(buf), "%H:%M:%S", &tm)) {
+      ss << buf;
+    }
+    ss << std::setfill('0');
+#ifdef _WIN32
+    struct timeb tb;
+    ftime(&tb);
+    ss << "." << (tb.millitm) << " ";
+#else
+    struct timeval tv;
+    gettimeofday(&tv, NULL);
+    ss << "." << (tv.tv_usec) << " ";
+#endif
+    ss << " 000000";
+    ss << std::hex << std::setfill('0');
+    for (size_t i = 0; i < packet.len(); ++i) {
+      ss << " " << std::setw(2) << (int)packet.data()[i];
+    }
+    MOZ_LOG(gRtpLoggerLog, LogLevel::Debug,
+            ("%s%s%s", desc.c_str(), (isRtp ? " RTP_PACKET " : " RTCP_PACKET "),
+             ss.str().c_str()));
+  }
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/transportbridge/RtpLogger.h b/dom/media/webrtc/transportbridge/RtpLogger.h
new file mode 100644
index 0000000000..fcfaede6e2
--- /dev/null
+++ b/dom/media/webrtc/transportbridge/RtpLogger.h
@@ -0,0 +1,28 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Original author: nohlmeier@mozilla.com
+
+#ifndef rtplogger_h__
+#define rtplogger_h__
+
+#include "transport/mediapacket.h"
+
+namespace mozilla {
+
+/* This class logs RTP and RTCP packets in hex in a format compatible to
+ * text2pcap.
+ * Example to convert the MOZ log file into a PCAP file:
+ *   egrep '(RTP_PACKET|RTCP_PACKET)' moz.log | \
+ *     text2pcap -D -n -l 1 -i 17 -u 1234,1235 -t '%H:%M:%S.' - rtp.pcap
+ */
+class RtpLogger {
+ public:
+  static bool IsPacketLoggingOn();
+  static void LogPacket(const MediaPacket& packet, bool input,
+                        std::string desc);
+};
+
+}  // namespace mozilla
+#endif
diff --git a/dom/media/webrtc/transportbridge/moz.build b/dom/media/webrtc/transportbridge/moz.build
new file mode 100644
index 0000000000..290dd26a0a
--- /dev/null
+++ b/dom/media/webrtc/transportbridge/moz.build
@@ -0,0 +1,27 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+include("/dom/media/webrtc/third_party_build/webrtc.mozbuild")
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "/dom/media",
+    "/dom/media/webrtc",
+    "/ipc/chromium/src",
+    "/media/libyuv/libyuv/include",
+    "/media/webrtc",
+    "/third_party/libsrtp/src/crypto/include",
+    "/third_party/libsrtp/src/include",
+    "/third_party/libwebrtc",
+    "/third_party/libwebrtc/third_party/abseil-cpp",
+]
+
+UNIFIED_SOURCES += [
+    "MediaPipeline.cpp",
+    "MediaPipelineFilter.cpp",
+    "RtpLogger.cpp",
+]
+
+FINAL_LIBRARY = "xul"