Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

48 files changed, 18103 insertions, 0 deletions

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..847e9fd3de
--- /dev/null
+++ b/dom/media/webrtc/jsapi/MediaTransportHandlerIPC.cpp
@@ -0,0 +1,471 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. 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/Endpoint.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) {}
+
+MediaTransportHandlerIPC::~MediaTransportHandlerIPC() = default;
+
+void MediaTransportHandlerIPC::Initialize() {
+  using EndpointPromise =
+      MozPromise<mozilla::ipc::Endpoint<mozilla::dom::PMediaTransportChild>,
+                 nsCString, true>;
+  mInitPromise =
+      net::SocketProcessBridgeChild::GetSocketProcessBridge()
+          ->Then(
+              GetCurrentSerialEventTarget(), __func__,
+              [](const RefPtr<net::SocketProcessBridgeChild>& aBridge) {
+                mozilla::ipc::Endpoint<mozilla::dom::PMediaTransportParent>
+                    parentEndpoint;
+                mozilla::ipc::Endpoint<mozilla::dom::PMediaTransportChild>
+                    childEndpoint;
+                mozilla::dom::PMediaTransport::CreateEndpoints(&parentEndpoint,
+                                                               &childEndpoint);
+
+                if (!aBridge || !aBridge->SendInitMediaTransport(
+                                    std::move(parentEndpoint))) {
+                  NS_WARNING(
+                      "MediaTransportHandlerIPC async init failed! Webrtc "
+                      "networking "
+                      "will not work!");
+                  return EndpointPromise::CreateAndReject(
+                      nsCString("SendInitMediaTransport failed!"), __func__);
+                }
+
+                return EndpointPromise::CreateAndResolve(
+                    std::move(childEndpoint), __func__);
+              },
+              [](const nsCString& aError) {
+                return EndpointPromise::CreateAndReject(aError, __func__);
+              })
+          ->Then(
+              mCallbackThread, __func__,
+              [this, self = RefPtr<MediaTransportHandlerIPC>(this)](
+                  mozilla::ipc::Endpoint<mozilla::dom::PMediaTransportChild>&&
+                      aEndpoint) {
+                RefPtr<MediaTransportChild> child =
+                    new MediaTransportChild(this);
+                aEndpoint.Bind(child);
+                mChild = 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) {
+    mChild->Shutdown();
+    mCallbackThread->Dispatch(NS_NewRunnableFunction(
+        __func__, [child = std::move(mChild)]() { child->Close(); }));
+  }
+  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)
+    : mMutex("MediaTransportChild"), mUser(aUser) {}
+
+MediaTransportChild::~MediaTransportChild() = default;
+
+mozilla::ipc::IPCResult MediaTransportChild::RecvOnCandidate(
+    const string& transportId, const CandidateInfo& candidateInfo) {
+  MutexAutoLock lock(mMutex);
+  if (mUser) {
+    mUser->OnCandidate(transportId, candidateInfo);
+  }
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportChild::RecvOnAlpnNegotiated(
+    const string& alpn) {
+  MutexAutoLock lock(mMutex);
+  if (mUser) {
+    mUser->OnAlpnNegotiated(alpn);
+  }
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportChild::RecvOnGatheringStateChange(
+    const int& state) {
+  MutexAutoLock lock(mMutex);
+  if (mUser) {
+    mUser->OnGatheringStateChange(
+        static_cast<dom::RTCIceGatheringState>(state));
+  }
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportChild::RecvOnConnectionStateChange(
+    const int& state) {
+  MutexAutoLock lock(mMutex);
+  if (mUser) {
+    mUser->OnConnectionStateChange(
+        static_cast<dom::RTCIceConnectionState>(state));
+  }
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportChild::RecvOnPacketReceived(
+    const string& transportId, const MediaPacket& packet) {
+  MutexAutoLock lock(mMutex);
+  if (mUser) {
+    mUser->OnPacketReceived(transportId, packet);
+  }
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportChild::RecvOnEncryptedSending(
+    const string& transportId, const MediaPacket& packet) {
+  MutexAutoLock lock(mMutex);
+  if (mUser) {
+    mUser->OnEncryptedSending(transportId, packet);
+  }
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportChild::RecvOnStateChange(
+    const string& transportId, const int& state) {
+  MutexAutoLock lock(mMutex);
+  if (mUser) {
+    mUser->OnStateChange(transportId,
+                         static_cast<TransportLayer::State>(state));
+  }
+  return ipc::IPCResult::Ok();
+}
+
+mozilla::ipc::IPCResult MediaTransportChild::RecvOnRtcpStateChange(
+    const string& transportId, const int& state) {
+  MutexAutoLock lock(mMutex);
+  if (mUser) {
+    mUser->OnRtcpStateChange(transportId,
+                             static_cast<TransportLayer::State>(state));
+  }
+  return ipc::IPCResult::Ok();
+}
+
+void MediaTransportChild::Shutdown() {
+  MutexAutoLock lock(mMutex);
+  mUser = nullptr;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/jsapi/MediaTransportHandlerIPC.h b/dom/media/webrtc/jsapi/MediaTransportHandlerIPC.h
new file mode 100644
index 0000000000..53c25c98e8
--- /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;
+  virtual ~MediaTransportHandlerIPC();
+
+  RefPtr<MediaTransportChild> mChild;
+
+  // |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..31d0b9f30f
--- /dev/null
+++ b/dom/media/webrtc/jsapi/MediaTransportParent.cpp
@@ -0,0 +1,238 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. 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();
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webrtc/jsapi/PacketDumper.cpp b/dom/media/webrtc/jsapi/PacketDumper.cpp
new file mode 100644
index 0000000000..9fbb2090d1
--- /dev/null
+++ b/dom/media/webrtc/jsapi/PacketDumper.cpp
@@ -0,0 +1,160 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. 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), 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)) {
+          return NS_OK;
+        }
+
+        PeerConnectionWrapper pcw(mPcHandle);
+        RefPtr<PeerConnectionImpl> pc = pcw.impl();
+        if (!pc) {
+          return NS_OK;
+        }
+
+        if (!aSending && (aType == dom::mozPacketDumpType::Rtcp ||
+                          aType == dom::mozPacketDumpType::Srtcp)) {
+          AutoTArray<size_t, 4> dumpLevels;
+          {
+            MutexAutoLock lock(mPacketDumpFlagsMutex);
+            unsigned flag = 1 << (unsigned)aType;
+            for (size_t i = 0; i < mRecvPacketDumpFlags.size(); ++i) {
+              if (mRecvPacketDumpFlags[i] & flag) {
+                dumpLevels.AppendElement(i);
+              }
+            }
+          }
+          for (size_t level : dumpLevels) {
+            pc->DumpPacket_m(level, aType, aSending, aPacket, aSize);
+          }
+          return NS_OK;
+        }
+
+        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);
+  }
+
+  mPacketDumpRtcpRecvCount += !aSending &&
+                              (aType == dom::mozPacketDumpType::Rtcp ||
+                               aType == dom::mozPacketDumpType::Srtcp) &&
+                              !((*packetDumpFlags)[aLevel] & flag);
+
+  (*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()) {
+    mPacketDumpRtcpRecvCount -= !aSending &&
+                                (aType == dom::mozPacketDumpType::Rtcp ||
+                                 aType == dom::mozPacketDumpType::Srtcp) &&
+                                ((*packetDumpFlags)[aLevel] & flag);
+    (*packetDumpFlags)[aLevel] &= ~flag;
+  }
+
+  return NS_OK;
+}
+
+bool PacketDumper::ShouldDumpPacket(size_t aLevel, dom::mozPacketDumpType aType,
+                                    bool aSending) const {
+  if (!mPacketDumpEnabled) {
+    return false;
+  }
+
+  if (!aSending && (aType == dom::mozPacketDumpType::Rtcp ||
+                    aType == dom::mozPacketDumpType::Srtcp)) {
+    // Received (S)RTCP is dumped per-transport, not per-pipeline.
+    return mPacketDumpRtcpRecvCount > 0;
+  }
+
+  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..6cd83a8a1f
--- /dev/null
+++ b/dom/media/webrtc/jsapi/PacketDumper.h
@@ -0,0 +1,55 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. 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
+      MOZ_GUARDED_BY(mPacketDumpFlagsMutex);
+  std::vector<unsigned> mRecvPacketDumpFlags
+      MOZ_GUARDED_BY(mPacketDumpFlagsMutex);
+  Atomic<bool> mPacketDumpEnabled{false};
+  Atomic<int> mPacketDumpRtcpRecvCount{0};
+  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..d293fa0be6
--- /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/Services.h"
+#include "mozilla/StaticPtr.h"
+#include "mozilla/Telemetry.h"
+#include "mozilla/dom/RTCStatsReportBinding.h"
+#include "nsCRTGlue.h"
+#include "nsIIOService.h"
+#include "nsIObserver.h"
+#include "nsIObserverService.h"
+#include "nsServiceManagerUtils.h"  // do_GetService
+#include "PeerConnectionImpl.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::FieldTrialsView> 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();
+    }
+  }
+
+  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;
+  }
+}
+
+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);
+        // Because this is an integer and we would like some extra precision
+        // the unit is per mille (1/1000) instead of percent (1/100).
+        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::FieldTrialsView> 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();
+    }
+  }
+}
+
+PeerConnectionCtx::PeerConnectionCtx()
+    : mGMPReady(false),
+      mLogHandle(EnsureWebrtcLogging()),
+      mTransportHandler(
+          MediaTransportHandler::Create(GetMainThreadSerialEventTarget())) {}
+
+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..4ef8ea36d4
--- /dev/null
+++ b/dom/media/webrtc/jsapi/PeerConnectionCtx.h
@@ -0,0 +1,192 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. 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 "api/field_trials_view.h"
+#include "call/audio_state.h"
+#include "MediaTransportHandler.h"  // Mostly for IceLogPromise
+#include "mozIGeckoMediaPluginService.h"
+#include "mozilla/StaticPtr.h"
+#include "nsIRunnable.h"
+#include "PeerConnectionImpl.h"
+
+class WebrtcLogSinkHandle;
+
+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();
+
+  // 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;
+
+  const RefPtr<WebrtcLogSinkHandle> mLogHandle;
+
+  // 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..985100153a
--- /dev/null
+++ b/dom/media/webrtc/jsapi/PeerConnectionImpl.cpp
@@ -0,0 +1,4653 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. 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 "transportbridge/RtpLogger.h"
+#include "jsapi/RTCRtpReceiver.h"
+#include "jsapi/RTCRtpSender.h"
+
+#include "mozilla/IntegerPrintfMacros.h"
+#include "mozilla/Sprintf.h"
+#include "mozilla/StaticPrefs_media.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 "nsGlobalWindowInner.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),
+      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) {
+    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);
+  }
+
+  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 = true;
+        }
+      } 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;
+};
+
+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);
+
+  // 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 = GetMainThreadSerialEventTarget();
+  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;
+  DataChannelReliabilityPolicy prPolicy;
+  switch (aType) {
+    case IPeerConnection::kDataChannelReliable:
+      prPolicy = DataChannelReliabilityPolicy::Reliable;
+      break;
+    case IPeerConnection::kDataChannelPartialReliableRexmit:
+      prPolicy = DataChannelReliabilityPolicy::LimitedRetransmissions;
+      break;
+    case IPeerConnection::kDataChannelPartialReliableTimed:
+      prPolicy = DataChannelReliabilityPolicy::LimitedLifetime;
+      break;
+    default:
+      MOZ_ASSERT(false);
+      return NS_ERROR_FAILURE;
+  }
+
+  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), prPolicy,
+      ordered,
+      prPolicy == DataChannelReliabilityPolicy::LimitedRetransmissions
+          ? aMaxNum
+          : (prPolicy == DataChannelReliabilityPolicy::LimitedLifetime
+                 ? 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) {
+          if (jsepTransceiver.IsRemoved()) {
+            return;
+          }
+          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);
+  }
+  // Whenever the state of an RTCDtlsTransport changes or when the [[IsClosed]]
+  // slot turns true, the user agent MUST update the connection state by
+  // queueing a task that runs the following steps:
+  // NOTE: The business about [[IsClosed]] here is probably a bug, because the
+  // rest of the spec makes it very clear that events should never fire when
+  // [[IsClosed]] is true. See https://github.com/w3c/webrtc-pc/issues/2865
+  GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction(
+      __func__, [this, self = RefPtr<PeerConnectionImpl>(this)] {
+        // Let connection be this RTCPeerConnection object.
+        // Let newState be the value of deriving a new state value as described
+        // by the RTCPeerConnectionState enum.
+        // If connection.[[ConnectionState]] is equal to newState, abort these
+        // steps.
+        // Set connection.[[ConnectionState]] to newState.
+        if (UpdateConnectionState()) {
+          // Fire an event named connectionstatechange at connection.
+          JSErrorResult jrv;
+          mPCObserver->OnStateChange(PCObserverStateType::ConnectionState, jrv);
+        }
+      }));
+}
+
+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 either
+  // [[IceConnectionState]] is "failed" or any RTCDtlsTransports are in the
+  // "failed" state.
+  if (mIceConnectionState == RTCIceConnectionState::Failed ||
+      statesFound.count(RTCDtlsTransportState::Failed)) {
+    return RTCPeerConnectionState::Failed;
+  }
+
+  // disconnected 	None of the previous states apply, and
+  // [[IceConnectionState]] is "disconnected".
+  if (mIceConnectionState == RTCIceConnectionState::Disconnected) {
+    return RTCPeerConnectionState::Disconnected;
+  }
+
+  // new 	None of the previous states apply, and either
+  // [[IceConnectionState]] is "new", and all RTCDtlsTransports are in the
+  // "new" or "closed" state...
+  if (mIceConnectionState == RTCIceConnectionState::New &&
+      !statesFound.count(RTCDtlsTransportState::Connecting) &&
+      !statesFound.count(RTCDtlsTransportState::Connected) &&
+      !statesFound.count(RTCDtlsTransportState::Failed)) {
+    return RTCPeerConnectionState::New;
+  }
+
+  // ...or there are no transports.
+  if (statesFound.empty()) {
+    return RTCPeerConnectionState::New;
+  }
+
+  // connected 	None of the previous states apply,
+  // [[IceConnectionState]] is "connected", and all RTCDtlsTransports are in
+  // the "connected" or "closed" state.
+  if (mIceConnectionState == RTCIceConnectionState::Connected &&
+      !statesFound.count(RTCDtlsTransportState::New) &&
+      !statesFound.count(RTCDtlsTransportState::Failed) &&
+      !statesFound.count(RTCDtlsTransportState::Connecting)) {
+    return RTCPeerConnectionState::Connected;
+  }
+
+  // connecting 	None of the previous states apply.
+  return RTCPeerConnectionState::Connecting;
+}
+
+bool 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) {
+      return true;
+    }
+  }
+
+  return false;
+}
+
+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;
+  }
+
+  UniquePtr<void, JS::FreePolicy> packetPtr{packet.release()};
+  JS::Rooted<JSObject*> jsobj(
+      jsapi.cx(),
+      JS::NewArrayBufferWithContents(jsapi.cx(), size, std::move(packetPtr)));
+
+  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);
+
+  const bool redUlpfecEnabled =
+      Preferences::GetBool("media.navigator.video.red_ulpfec_enabled", false);
+
+  // 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 pref (ulpfec and red).
+    if (codec->mName == "webrtc-datachannel" ||
+        (codec->mName == "ulpfec" && !redUlpfecEnabled) ||
+        (codec->mName == "red" && !redUlpfecEnabled)) {
+      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);
+}
+
+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 nsACString& 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());
+  mRtcpReceiveListener.DisconnectIfExists();
+  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;
+  // MOZ_KnownLive due to bug 1506441
+  EventDispatcher::DispatchDOMEvent(
+      MOZ_KnownLive(nsGlobalWindowInner::Cast(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());
+          }
+        }
+
+        auto oldIceCredentials = mJsepSession->GetLocalIceCredentials();
+        auto newIceCredentials =
+            mUncommittedJsepSession->GetLocalIceCredentials();
+
+        bool iceRestartDetected =
+            (!oldIceCredentials.empty() && !newIceCredentials.empty() &&
+             (oldIceCredentials != newIceCredentials));
+
+        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 (iceRestartDetected || 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();
+        }
+
+        bool signalingStateChanged = false;
+        if (newSignalingState != mSignalingState) {
+          mSignalingState = newSignalingState;
+          signalingStateChanged = true;
+        }
+
+        // Spec does not actually tell us to do this, but that is probably a
+        // spec bug.
+        // https://github.com/w3c/webrtc-pc/issues/2817
+        bool connectionStateChanged = UpdateConnectionState();
+
+        // This only gets populated for remote descriptions
+        dom::RTCRtpReceiver::StreamAssociationChanges changes;
+        if (aRemote) {
+          for (const auto& transceiver : mTransceivers) {
+            transceiver->Receiver()->UpdateStreams(&changes);
+          }
+        }
+
+        // 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;
+          }
+        }
+
+        // JS callbacks happen below. DO NOT TOUCH STATE AFTER THIS UNLESS SPEC
+        // EXPLICITLY SAYS TO, OTHERWISE STATES THAT ARE NOT SUPPOSED TO BE
+        // OBSERVABLE TO JS WILL BE!
+
+        JSErrorResult jrv;
+        RefPtr<PeerConnectionObserver> pcObserver(mPCObserver);
+        if (signalingStateChanged) {
+          pcObserver->OnStateChange(PCObserverStateType::SignalingState, jrv);
+        }
+
+        if (connectionStateChanged) {
+          pcObserver->OnStateChange(PCObserverStateType::ConnectionState, jrv);
+        }
+
+        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);
+          }
+        }
+
+        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);
+            }
+          }
+          pcObserver->FireTrackEvent(*trackEvent.mReceiver, streams, jrv);
+        }
+
+        // TODO(Bug 1241291): Legacy event, remove eventually
+        for (const auto& stream : newStreams) {
+          pcObserver->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!");
+  }
+
+  bool connectionStateChanged = UpdateConnectionState();
+  WrappableJSErrorResult rv;
+  RefPtr<PeerConnectionObserver> pcObserver(mPCObserver);
+  pcObserver->OnStateChange(PCObserverStateType::IceConnectionState, rv);
+  if (connectionStateChanged) {
+    pcObserver->OnStateChange(PCObserverStateType::ConnectionState, rv);
+  }
+}
+
+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) {
+    digests.emplace_back(ToString(fingerprint.hashFunc),
+                         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()),
+      mPacketDumper(aPc->GetPacketDumper()) {
+  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);
+  mSource->SignalPacketReceived.connect(
+      this, &PeerConnectionImpl::SignalHandler::OnPacketReceived_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());
+    mRtcpReceiveListener = mSignalHandler->RtcpReceiveEvent().Connect(
+        mCall->mCallThread, [call = mCall](MediaPacket aPacket) {
+          // This might not be initted yet, because the task to do that is tail
+          // dispatched, and STS might beat it to the punch.
+          if (call->Call()) {
+            call->Call()->Receiver()->DeliverRtcpPacket(
+                rtc::CopyOnWriteBuffer(aPacket.data(), aPacket.len()));
+          }
+        });
+  }
+
+  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);
+}
+
+void PeerConnectionImpl::SignalHandler::OnPacketReceived_s(
+    const std::string& aTransportId, const MediaPacket& aPacket) {
+  ASSERT_ON_THREAD(mSTSThread);
+
+  if (!aPacket.len()) {
+    return;
+  }
+
+  if (aPacket.type() != MediaPacket::RTCP) {
+    return;
+  }
+
+  CSFLogVerbose(LOGTAG, "%s received RTCP packet.", mHandle.c_str());
+
+  RtpLogger::LogPacket(aPacket, true, mHandle);
+
+  // Might be nice to pass ownership of the buffer in this case, but it is a
+  // small optimization in a rare case.
+  mPacketDumper->Dump(SIZE_MAX, dom::mozPacketDumpType::Srtcp, false,
+                      aPacket.encrypted_data(), aPacket.encrypted_len());
+
+  mPacketDumper->Dump(SIZE_MAX, dom::mozPacketDumpType::Rtcp, false,
+                      aPacket.data(), aPacket.len());
+
+  if (StaticPrefs::media_webrtc_net_force_disable_rtcp_reception()) {
+    CSFLogVerbose(LOGTAG, "%s RTCP packet forced to be dropped",
+                  mHandle.c_str());
+    return;
+  }
+
+  mRtcpReceiveEvent.Notify(aPacket.Clone());
+}
+
+/**
+ * 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);
+
+  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..085658b206
--- /dev/null
+++ b/dom/media/webrtc/jsapi/PeerConnectionImpl.h
@@ -0,0 +1,980 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. 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 "MediaEventSource.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);
+  dom::RTCPeerConnectionState GetNewConnectionState() const;
+  // Returns whether we need to fire a state change event
+  bool UpdateConnectionState();
+
+  // 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);
+
+  void BreakCycles();
+
+ private:
+  virtual ~PeerConnectionImpl();
+  PeerConnectionImpl(const PeerConnectionImpl& rhs);
+  PeerConnectionImpl& operator=(PeerConnectionImpl);
+
+  RefPtr<dom::RTCStatsPromise> GetDataChannelStats(
+      const RefPtr<DataChannelConnection>& aDataChannelConnection,
+      const DOMHighResTimeStamp aTimestamp);
+  nsresult CalculateFingerprint(const nsACString& 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;
+
+  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);
+    void OnPacketReceived_s(const std::string& aTransportId,
+                            const MediaPacket& aPacket);
+
+    MediaEventSourceExc<MediaPacket>& RtcpReceiveEvent() {
+      return mRtcpReceiveEvent;
+    }
+
+   private:
+    const std::string mHandle;
+    RefPtr<MediaTransportHandler> mSource;
+    RefPtr<nsISerialEventTarget> mSTSThread;
+    RefPtr<PacketDumper> mPacketDumper;
+    MediaEventProducerExc<MediaPacket> mRtcpReceiveEvent;
+  };
+
+  mozilla::UniquePtr<SignalHandler> mSignalHandler;
+  MediaEventListener mRtcpReceiveListener;
+
+  // 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..243f06d2f1
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCDtlsTransport.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 "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::UpdateStateNoEvent(TransportLayer::State aState) {
+  if (mState == RTCDtlsTransportState::Closed) {
+    return;
+  }
+
+  switch (aState) {
+    case TransportLayer::TS_NONE:
+      break;
+    case TransportLayer::TS_INIT:
+      break;
+    case TransportLayer::TS_CONNECTING:
+      mState = RTCDtlsTransportState::Connecting;
+      break;
+    case TransportLayer::TS_OPEN:
+      mState = RTCDtlsTransportState::Connected;
+      break;
+    case TransportLayer::TS_CLOSED:
+      mState = RTCDtlsTransportState::Closed;
+      break;
+    case TransportLayer::TS_ERROR:
+      mState = RTCDtlsTransportState::Failed;
+      break;
+  }
+}
+
+void RTCDtlsTransport::UpdateState(TransportLayer::State aState) {
+  RTCDtlsTransportState oldState = mState;
+  UpdateStateNoEvent(aState);
+  if (oldState == mState) {
+    return;
+  }
+
+  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..3800502154
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCDtlsTransport.h
@@ -0,0 +1,44 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. 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 UpdateStateNoEvent(TransportLayer::State aState);
+  void UpdateState(TransportLayer::State aState);
+
+ private:
+  virtual ~RTCDtlsTransport() = default;
+
+  RTCDtlsTransportState mState;
+};
+
+}  // namespace mozilla::dom
+#endif  // _RTCDtlsTransport_h_
diff --git a/dom/media/webrtc/jsapi/RTCEncodedAudioFrame.cpp b/dom/media/webrtc/jsapi/RTCEncodedAudioFrame.cpp
new file mode 100644
index 0000000000..43f7a1f13b
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCEncodedAudioFrame.cpp
@@ -0,0 +1,96 @@
+/* -*- Mode: C++; tab-width: 2; 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 https://mozilla.org/MPL/2.0/. */
+
+#include "jsapi/RTCEncodedAudioFrame.h"
+
+#include <stdint.h>
+
+#include <memory>
+#include <utility>
+
+#include "api/frame_transformer_interface.h"
+
+#include "jsapi/RTCEncodedFrameBase.h"
+#include "jsapi/RTCRtpScriptTransform.h"
+#include "mozilla/dom/RTCRtpScriptTransformer.h"
+#include "mozilla/dom/RTCEncodedAudioFrameBinding.h"
+#include "nsWrapperCache.h"
+#include "nsISupports.h"
+#include "nsCycleCollectionParticipant.h"
+#include "nsIGlobalObject.h"
+#include "nsContentUtils.h"
+#include "mozilla/HoldDropJSObjects.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/Unused.h"
+#include "mozilla/fallible.h"
+#include "js/RootingAPI.h"
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(RTCEncodedAudioFrame, RTCEncodedFrameBase,
+                                   mOwner)
+NS_IMPL_ADDREF_INHERITED(RTCEncodedAudioFrame, RTCEncodedFrameBase)
+NS_IMPL_RELEASE_INHERITED(RTCEncodedAudioFrame, RTCEncodedFrameBase)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(RTCEncodedAudioFrame)
+  NS_WRAPPERCACHE_INTERFACE_MAP_ENTRY
+NS_INTERFACE_MAP_END_INHERITING(RTCEncodedFrameBase)
+
+RTCEncodedAudioFrame::RTCEncodedAudioFrame(
+    nsIGlobalObject* aGlobal,
+    std::unique_ptr<webrtc::TransformableFrameInterface> aFrame,
+    uint64_t aCounter, RTCRtpScriptTransformer* aOwner)
+    : RTCEncodedFrameBase(aGlobal, std::move(aFrame), aCounter),
+      mOwner(aOwner) {
+  mMetadata.mSynchronizationSource.Construct(mFrame->GetSsrc());
+  mMetadata.mPayloadType.Construct(mFrame->GetPayloadType());
+  // send frames are derived directly from TransformableFrameInterface, not
+  // TransformableAudioFrameInterface! Right now, send frames have no csrcs
+  // or sequence number
+  // TODO(bug 1835076): Fix this
+  if (mFrame->GetDirection() ==
+      webrtc::TransformableFrameInterface::Direction::kReceiver) {
+    const auto& audioFrame(
+        static_cast<webrtc::TransformableAudioFrameInterface&>(*mFrame));
+    mMetadata.mContributingSources.Construct();
+    for (const auto csrc : audioFrame.GetContributingSources()) {
+      Unused << mMetadata.mContributingSources.Value().AppendElement(csrc,
+                                                                     fallible);
+    }
+    if (const auto optionalSeqNum = audioFrame.SequenceNumber()) {
+      mMetadata.mSequenceNumber.Construct(*optionalSeqNum);
+    }
+  }
+
+  // Base class needs this, but can't do it itself because of an assertion in
+  // the cycle-collector.
+  mozilla::HoldJSObjects(this);
+}
+
+RTCEncodedAudioFrame::~RTCEncodedAudioFrame() {
+  // Base class needs this, but can't do it itself because of an assertion in
+  // the cycle-collector.
+  mozilla::DropJSObjects(this);
+}
+
+JSObject* RTCEncodedAudioFrame::WrapObject(JSContext* aCx,
+                                           JS::Handle<JSObject*> aGivenProto) {
+  return RTCEncodedAudioFrame_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+nsIGlobalObject* RTCEncodedAudioFrame::GetParentObject() const {
+  return mGlobal;
+}
+
+void RTCEncodedAudioFrame::GetMetadata(
+    RTCEncodedAudioFrameMetadata& aMetadata) const {
+  aMetadata = mMetadata;
+}
+
+bool RTCEncodedAudioFrame::CheckOwner(RTCRtpScriptTransformer* aOwner) const {
+  return aOwner == mOwner;
+}
+}  // namespace mozilla::dom
diff --git a/dom/media/webrtc/jsapi/RTCEncodedAudioFrame.h b/dom/media/webrtc/jsapi/RTCEncodedAudioFrame.h
new file mode 100644
index 0000000000..339231ccc2
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCEncodedAudioFrame.h
@@ -0,0 +1,52 @@
+/* -*- Mode: C++; tab-width: 2; 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 https://mozilla.org/MPL/2.0/. */
+
+#ifndef MOZILLA_DOM_MEDIA_WEBRTC_JSAPI_RTCENCODEDAUDIOFRAME_H_
+#define MOZILLA_DOM_MEDIA_WEBRTC_JSAPI_RTCENCODEDAUDIOFRAME_H_
+
+#include "mozilla/RefPtr.h"
+#include "nsIGlobalObject.h"
+#include "jsapi/RTCEncodedFrameBase.h"
+#include "mozilla/dom/RTCEncodedAudioFrameBinding.h"
+
+namespace mozilla::dom {
+
+// Wraps a libwebrtc frame, allowing the frame buffer to be modified, and
+// providing read-only access to various metadata. After the libwebrtc frame is
+// extracted (with RTCEncodedFrameBase::TakeFrame), the frame buffer is
+// detached, but the metadata remains accessible.
+class RTCEncodedAudioFrame final : public RTCEncodedFrameBase {
+ public:
+  explicit RTCEncodedAudioFrame(
+      nsIGlobalObject* aGlobal,
+      std::unique_ptr<webrtc::TransformableFrameInterface> aFrame,
+      uint64_t aCounter, RTCRtpScriptTransformer* aOwner);
+
+  // nsISupports
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(RTCEncodedAudioFrame,
+                                           RTCEncodedFrameBase)
+
+  // webidl (timestamp and data accessors live in base class)
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  nsIGlobalObject* GetParentObject() const;
+
+  void GetMetadata(RTCEncodedAudioFrameMetadata& aMetadata) const;
+
+  bool CheckOwner(RTCRtpScriptTransformer* aOwner) const override;
+
+  bool IsVideo() const override { return false; }
+
+ private:
+  virtual ~RTCEncodedAudioFrame();
+  RefPtr<RTCRtpScriptTransformer> mOwner;
+  RTCEncodedAudioFrameMetadata mMetadata;
+};
+
+}  // namespace mozilla::dom
+#endif  // MOZILLA_DOM_MEDIA_WEBRTC_JSAPI_RTCENCODEDAUDIOFRAME_H_
diff --git a/dom/media/webrtc/jsapi/RTCEncodedFrameBase.cpp b/dom/media/webrtc/jsapi/RTCEncodedFrameBase.cpp
new file mode 100644
index 0000000000..79d357d438
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCEncodedFrameBase.cpp
@@ -0,0 +1,73 @@
+/* -*- Mode: C++; tab-width: 2; 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 https://mozilla.org/MPL/2.0/. */
+
+#include "jsapi/RTCEncodedFrameBase.h"
+
+#include "js/GCAPI.h"
+#include "nsIGlobalObject.h"
+#include "mozilla/dom/ScriptSettings.h"
+#include "js/ArrayBuffer.h"
+
+namespace mozilla::dom {
+NS_IMPL_CYCLE_COLLECTION_WITH_JS_MEMBERS(RTCEncodedFrameBase, (mGlobal),
+                                         (mData))
+NS_IMPL_CYCLE_COLLECTING_ADDREF(RTCEncodedFrameBase)
+NS_IMPL_CYCLE_COLLECTING_RELEASE(RTCEncodedFrameBase)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(RTCEncodedFrameBase)
+  NS_INTERFACE_MAP_ENTRY(nsISupports)
+NS_INTERFACE_MAP_END
+
+RTCEncodedFrameBase::RTCEncodedFrameBase(
+    nsIGlobalObject* aGlobal,
+    std::unique_ptr<webrtc::TransformableFrameInterface> aFrame,
+    uint64_t aCounter)
+    : mGlobal(aGlobal),
+      mFrame(std::move(aFrame)),
+      mCounter(aCounter),
+      mTimestamp(mFrame->GetTimestamp()) {
+  AutoJSAPI jsapi;
+  if (NS_WARN_IF(!jsapi.Init(mGlobal))) {
+    return;
+  }
+
+  // Avoid a copy
+  mData = JS::NewArrayBufferWithUserOwnedContents(
+      jsapi.cx(), mFrame->GetData().size(), (void*)(mFrame->GetData().data()));
+}
+
+RTCEncodedFrameBase::~RTCEncodedFrameBase() = default;
+
+unsigned long RTCEncodedFrameBase::Timestamp() const { return mTimestamp; }
+
+void RTCEncodedFrameBase::SetData(const ArrayBuffer& aData) {
+  mData.set(aData.Obj());
+  if (mFrame) {
+    aData.ProcessData([&](const Span<uint8_t>& aData, JS::AutoCheckCannotGC&&) {
+      mFrame->SetData(
+          rtc::ArrayView<const uint8_t>(aData.Elements(), aData.Length()));
+    });
+  }
+}
+
+void RTCEncodedFrameBase::GetData(JSContext* aCx, JS::Rooted<JSObject*>* aObj) {
+  aObj->set(mData);
+}
+
+uint64_t RTCEncodedFrameBase::GetCounter() const { return mCounter; }
+
+std::unique_ptr<webrtc::TransformableFrameInterface>
+RTCEncodedFrameBase::TakeFrame() {
+  AutoJSAPI jsapi;
+  if (!jsapi.Init(mGlobal)) {
+    MOZ_CRASH("Could not init JSAPI!");
+  }
+  JS::Rooted<JSObject*> rootedData(jsapi.cx(), mData);
+  JS::DetachArrayBuffer(jsapi.cx(), rootedData);
+  return std::move(mFrame);
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webrtc/jsapi/RTCEncodedFrameBase.h b/dom/media/webrtc/jsapi/RTCEncodedFrameBase.h
new file mode 100644
index 0000000000..25546257be
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCEncodedFrameBase.h
@@ -0,0 +1,58 @@
+/* -*- Mode: C++; tab-width: 2; 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 https://mozilla.org/MPL/2.0/. */
+
+#ifndef MOZILLA_DOM_MEDIA_WEBRTC_JSAPI_RTCENCODEDFRAMEBASE_H_
+#define MOZILLA_DOM_MEDIA_WEBRTC_JSAPI_RTCENCODEDFRAMEBASE_H_
+
+#include "js/TypeDecls.h"
+#include "mozilla/dom/TypedArray.h"  // ArrayBuffer
+#include "mozilla/Assertions.h"
+#include "api/frame_transformer_interface.h"
+#include <memory>
+
+class nsIGlobalObject;
+
+namespace mozilla::dom {
+
+class RTCRtpScriptTransformer;
+
+class RTCEncodedFrameBase : public nsISupports, public nsWrapperCache {
+ public:
+  explicit RTCEncodedFrameBase(
+      nsIGlobalObject* aGlobal,
+      std::unique_ptr<webrtc::TransformableFrameInterface> aFrame,
+      uint64_t aCounter);
+
+  // nsISupports
+  NS_DECL_CYCLE_COLLECTING_ISUPPORTS
+  NS_DECL_CYCLE_COLLECTION_SCRIPT_HOLDER_CLASS(RTCEncodedFrameBase)
+
+  // Common webidl for RTCEncodedVideoFrame/RTCEncodedAudioFrame
+  unsigned long Timestamp() const;
+
+  void SetData(const ArrayBuffer& aData);
+
+  void GetData(JSContext* aCx, JS::Rooted<JSObject*>* aObj);
+
+  uint64_t GetCounter() const;
+
+  virtual bool CheckOwner(RTCRtpScriptTransformer* aOwner) const = 0;
+
+  std::unique_ptr<webrtc::TransformableFrameInterface> TakeFrame();
+
+  virtual bool IsVideo() const = 0;
+
+ protected:
+  virtual ~RTCEncodedFrameBase();
+  RefPtr<nsIGlobalObject> mGlobal;
+  std::unique_ptr<webrtc::TransformableFrameInterface> mFrame;
+  const uint64_t mCounter = 0;
+  const unsigned long mTimestamp = 0;
+  JS::Heap<JSObject*> mData;
+};
+
+}  // namespace mozilla::dom
+#endif  // MOZILLA_DOM_MEDIA_WEBRTC_JSAPI_RTCENCODEDFRAMEBASE_H_
diff --git a/dom/media/webrtc/jsapi/RTCEncodedVideoFrame.cpp b/dom/media/webrtc/jsapi/RTCEncodedVideoFrame.cpp
new file mode 100644
index 0000000000..8ca7215c98
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCEncodedVideoFrame.cpp
@@ -0,0 +1,116 @@
+/* -*- Mode: C++; tab-width: 2; 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 https://mozilla.org/MPL/2.0/. */
+
+#include "jsapi/RTCEncodedVideoFrame.h"
+
+#include <stdint.h>
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "api/frame_transformer_interface.h"
+
+#include "jsapi/RTCEncodedFrameBase.h"
+#include "mozilla/dom/RTCEncodedVideoFrameBinding.h"
+#include "mozilla/dom/RTCRtpScriptTransformer.h"
+#include "nsWrapperCache.h"
+#include "nsISupports.h"
+#include "nsCycleCollectionParticipant.h"
+#include "nsIGlobalObject.h"
+#include "nsContentUtils.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/Unused.h"
+#include "mozilla/fallible.h"
+#include "mozilla/Maybe.h"
+#include "js/RootingAPI.h"
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(RTCEncodedVideoFrame, RTCEncodedFrameBase,
+                                   mOwner)
+NS_IMPL_ADDREF_INHERITED(RTCEncodedVideoFrame, RTCEncodedFrameBase)
+NS_IMPL_RELEASE_INHERITED(RTCEncodedVideoFrame, RTCEncodedFrameBase)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(RTCEncodedVideoFrame)
+  NS_WRAPPERCACHE_INTERFACE_MAP_ENTRY
+NS_INTERFACE_MAP_END_INHERITING(RTCEncodedFrameBase)
+
+RTCEncodedVideoFrame::RTCEncodedVideoFrame(
+    nsIGlobalObject* aGlobal,
+    std::unique_ptr<webrtc::TransformableFrameInterface> aFrame,
+    uint64_t aCounter, RTCRtpScriptTransformer* aOwner)
+    : RTCEncodedFrameBase(aGlobal, std::move(aFrame), aCounter),
+      mOwner(aOwner) {
+  const auto& videoFrame(
+      static_cast<webrtc::TransformableVideoFrameInterface&>(*mFrame));
+  mType = videoFrame.IsKeyFrame() ? RTCEncodedVideoFrameType::Key
+                                  : RTCEncodedVideoFrameType::Delta;
+  auto metadata = videoFrame.Metadata();
+
+  if (metadata.GetFrameId().has_value()) {
+    mMetadata.mFrameId.Construct(*metadata.GetFrameId());
+  }
+  mMetadata.mDependencies.Construct();
+  for (const auto dep : metadata.GetFrameDependencies()) {
+    Unused << mMetadata.mDependencies.Value().AppendElement(
+        static_cast<unsigned long long>(dep), fallible);
+  }
+  mMetadata.mWidth.Construct(metadata.GetWidth());
+  mMetadata.mHeight.Construct(metadata.GetHeight());
+  if (metadata.GetSpatialIndex() >= 0) {
+    mMetadata.mSpatialIndex.Construct(metadata.GetSpatialIndex());
+  }
+  if (metadata.GetTemporalIndex() >= 0) {
+    mMetadata.mTemporalIndex.Construct(metadata.GetTemporalIndex());
+  }
+  mMetadata.mSynchronizationSource.Construct(videoFrame.GetSsrc());
+  mMetadata.mPayloadType.Construct(videoFrame.GetPayloadType());
+  mMetadata.mContributingSources.Construct();
+  for (const auto csrc : metadata.GetCsrcs()) {
+    Unused << mMetadata.mContributingSources.Value().AppendElement(csrc,
+                                                                   fallible);
+  }
+
+  // The metadata timestamp is different, and not presently present in the
+  // libwebrtc types
+  if (!videoFrame.GetRid().empty()) {
+    mRid = Some(videoFrame.GetRid());
+  }
+
+  // Base class needs this, but can't do it itself because of an assertion in
+  // the cycle-collector.
+  mozilla::HoldJSObjects(this);
+}
+
+RTCEncodedVideoFrame::~RTCEncodedVideoFrame() {
+  // Base class needs this, but can't do it itself because of an assertion in
+  // the cycle-collector.
+  mozilla::DropJSObjects(this);
+}
+
+JSObject* RTCEncodedVideoFrame::WrapObject(JSContext* aCx,
+                                           JS::Handle<JSObject*> aGivenProto) {
+  return RTCEncodedVideoFrame_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+nsIGlobalObject* RTCEncodedVideoFrame::GetParentObject() const {
+  return mGlobal;
+}
+
+RTCEncodedVideoFrameType RTCEncodedVideoFrame::Type() const { return mType; }
+
+void RTCEncodedVideoFrame::GetMetadata(
+    RTCEncodedVideoFrameMetadata& aMetadata) {
+  aMetadata = mMetadata;
+}
+
+bool RTCEncodedVideoFrame::CheckOwner(RTCRtpScriptTransformer* aOwner) const {
+  return aOwner == mOwner;
+}
+
+Maybe<std::string> RTCEncodedVideoFrame::Rid() const { return mRid; }
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webrtc/jsapi/RTCEncodedVideoFrame.h b/dom/media/webrtc/jsapi/RTCEncodedVideoFrame.h
new file mode 100644
index 0000000000..7f1e04db96
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCEncodedVideoFrame.h
@@ -0,0 +1,61 @@
+/* -*- Mode: C++; tab-width: 2; 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 https://mozilla.org/MPL/2.0/. */
+
+#ifndef MOZILLA_DOM_MEDIA_WEBRTC_JSAPI_RTCENCODEDVIDEOFRAME_H_
+#define MOZILLA_DOM_MEDIA_WEBRTC_JSAPI_RTCENCODEDVIDEOFRAME_H_
+
+#include "mozilla/RefPtr.h"
+#include "nsIGlobalObject.h"
+#include "jsapi/RTCEncodedFrameBase.h"
+#include "mozilla/dom/RTCEncodedVideoFrameBinding.h"
+
+namespace mozilla::dom {
+class RTCRtpScriptTransformer;
+
+// Wraps a libwebrtc frame, allowing the frame buffer to be modified, and
+// providing read-only access to various metadata. After the libwebrtc frame is
+// extracted (with RTCEncodedFrameBase::TakeFrame), the frame buffer is
+// detached, but the metadata remains accessible.
+class RTCEncodedVideoFrame final : public RTCEncodedFrameBase {
+ public:
+  explicit RTCEncodedVideoFrame(
+      nsIGlobalObject* aGlobal,
+      std::unique_ptr<webrtc::TransformableFrameInterface> aFrame,
+      uint64_t aCounter, RTCRtpScriptTransformer* aOwner);
+
+  // nsISupports
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(RTCEncodedVideoFrame,
+                                           RTCEncodedFrameBase)
+
+  // webidl (timestamp and data accessors live in base class)
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  nsIGlobalObject* GetParentObject() const;
+
+  RTCEncodedVideoFrameType Type() const;
+
+  void GetMetadata(RTCEncodedVideoFrameMetadata& aMetadata);
+
+  bool CheckOwner(RTCRtpScriptTransformer* aOwner) const override;
+
+  bool IsVideo() const override { return true; }
+
+  // Not in webidl right now. Might change.
+  // https://github.com/w3c/webrtc-encoded-transform/issues/147
+  Maybe<std::string> Rid() const;
+
+ private:
+  virtual ~RTCEncodedVideoFrame();
+  RefPtr<RTCRtpScriptTransformer> mOwner;
+  RTCEncodedVideoFrameType mType;
+  RTCEncodedVideoFrameMetadata mMetadata;
+  Maybe<std::string> mRid;
+};
+
+}  // namespace mozilla::dom
+#endif  // MOZILLA_DOM_MEDIA_WEBRTC_JSAPI_RTCENCODEDVIDEOFRAME_H_
diff --git a/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp b/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp
new file mode 100644
index 0000000000..efe83fb782
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp
@@ -0,0 +1,1046 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. 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 <stdint.h>
+
+#include <vector>
+#include <string>
+#include <set>
+
+#include "call/call.h"
+#include "call/audio_receive_stream.h"
+#include "call/video_receive_stream.h"
+#include "api/rtp_parameters.h"
+#include "api/units/timestamp.h"
+#include "api/units/time_delta.h"
+#include "system_wrappers/include/clock.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+#include "RTCRtpTransceiver.h"
+#include "PeerConnectionImpl.h"
+#include "RTCStatsReport.h"
+#include "mozilla/dom/RTCRtpReceiverBinding.h"
+#include "mozilla/dom/RTCRtpSourcesBinding.h"
+#include "mozilla/dom/RTCStatsReportBinding.h"
+#include "jsep/JsepTransceiver.h"
+#include "libwebrtcglue/MediaConduitControl.h"
+#include "libwebrtcglue/MediaConduitInterface.h"
+#include "transportbridge/MediaPipeline.h"
+#include "sdp/SdpEnum.h"
+#include "sdp/SdpAttribute.h"
+#include "MediaTransportHandler.h"
+#include "RemoteTrackSource.h"
+
+#include "mozilla/dom/RTCRtpCapabilitiesBinding.h"
+#include "mozilla/dom/MediaStreamTrack.h"
+#include "mozilla/dom/Promise.h"
+#include "mozilla/dom/Nullable.h"
+#include "mozilla/dom/Document.h"
+#include "mozilla/dom/AudioStreamTrack.h"
+#include "mozilla/dom/VideoStreamTrack.h"
+#include "mozilla/dom/RTCRtpScriptTransform.h"
+
+#include "nsPIDOMWindow.h"
+#include "PrincipalHandle.h"
+#include "nsIPrincipal.h"
+#include "MediaTrackGraph.h"
+#include "nsStringFwd.h"
+#include "MediaSegment.h"
+#include "nsLiteralString.h"
+#include "nsTArray.h"
+#include "nsDOMNavigationTiming.h"
+#include "MainThreadUtils.h"
+#include "ErrorList.h"
+#include "nsWrapperCache.h"
+#include "nsISupports.h"
+#include "nsCOMPtr.h"
+#include "nsIScriptObjectPrincipal.h"
+#include "nsCycleCollectionParticipant.h"
+#include "nsDebug.h"
+#include "nsThreadUtils.h"
+#include "PerformanceRecorder.h"
+
+#include "mozilla/NullPrincipal.h"
+#include "mozilla/Preferences.h"
+#include "mozilla/StateMirroring.h"
+#include "mozilla/Logging.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/AbstractThread.h"
+#include "mozilla/StateWatching.h"
+#include "mozilla/Maybe.h"
+#include "mozilla/Assertions.h"
+#include "mozilla/AlreadyAddRefed.h"
+#include "mozilla/MozPromise.h"
+#include "mozilla/UniquePtr.h"
+#include "mozilla/fallible.h"
+#include "mozilla/mozalloc_oom.h"
+#include "mozilla/ErrorResult.h"
+#include "js/RootingAPI.h"
+
+namespace mozilla::dom {
+
+LazyLogModule gReceiverLog("RTCRtpReceiver");
+
+NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE_CLASS(RTCRtpReceiver)
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(RTCRtpReceiver)
+  tmp->Unlink();
+  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, mTransform,
+                                    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),
+      INIT_CANONICAL(mFrameTransformerProxy, nullptr) {
+  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.has_value()) {
+                // 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_received);
+              local.mPacketsDiscarded.Construct(audioStats->packets_discarded);
+              local.mBytesReceived.Construct(
+                  audioStats->payload_bytes_received);
+              // 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.has_value()) {
+                local.mLastPacketReceivedTimestamp.Construct(
+                    RTCStatsTimestamp::FromNtp(
+                        aConduit->GetTimestampMaker(),
+                        webrtc::Timestamp::Millis(
+                            audioStats->last_packet_received->ms()) +
+                            webrtc::TimeDelta::Seconds(webrtc::kNtpJan1970))
+                        .ToDom());
+              }
+              local.mHeaderBytesReceived.Construct(
+                  audioStats->header_and_padding_bytes_received);
+              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.has_value()) {
+                // 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<double>());
+              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.has_value()) {
+                local.mLastPacketReceivedTimestamp.Construct(
+                    RTCStatsTimestamp::FromNtp(
+                        aConduit->GetTimestampMaker(),
+                        webrtc::Timestamp::Millis(
+                            videoStats->rtp_stats.last_packet_received->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();
+  if (mTransform) {
+    mTransform->GetProxy().SetReceiver(nullptr);
+  }
+}
+
+void RTCRtpReceiver::BreakCycles() {
+  mWindow = nullptr;
+  mPc = nullptr;
+  mTrack = nullptr;
+  mTrackSource = nullptr;
+}
+
+void RTCRtpReceiver::Unlink() {
+  if (mTransceiver) {
+    mTransceiver->Unlink();
+  }
+}
+
+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() || mTransceiver->Stopping());
+  mReceiving = false;
+  GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction(
+      __func__, [trackSource = mTrackSource] { trackSource->ForceEnded(); }));
+}
+
+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();
+}
+
+void RTCRtpReceiver::SetTransform(RTCRtpScriptTransform* aTransform,
+                                  ErrorResult& aError) {
+  if (aTransform == mTransform.get()) {
+    // Ok... smile and nod
+    // TODO: Depending on spec, this might throw
+    // https://github.com/w3c/webrtc-encoded-transform/issues/189
+    return;
+  }
+
+  if (aTransform && aTransform->IsClaimed()) {
+    aError.ThrowInvalidStateError("transform has already been used elsewhere");
+    return;
+  }
+
+  if (aTransform) {
+    mFrameTransformerProxy = &aTransform->GetProxy();
+  } else {
+    mFrameTransformerProxy = nullptr;
+  }
+
+  if (mTransform) {
+    mTransform->GetProxy().SetReceiver(nullptr);
+  }
+
+  mTransform = const_cast<RTCRtpScriptTransform*>(aTransform);
+
+  if (mTransform) {
+    mTransform->GetProxy().SetReceiver(this);
+    mTransform->SetClaimed();
+  }
+}
+
+void RTCRtpReceiver::RequestKeyFrame() {
+  if (!mTransform || !mPipeline) {
+    return;
+  }
+
+  mPipeline->mConduit->AsVideoSessionConduit().apply([&](const auto& conduit) {
+    conduit->RequestKeyFrame(&mTransform->GetProxy());
+  });
+}
+
+}  // 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..af074ee0f1
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCRtpReceiver.h
@@ -0,0 +1,215 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. 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 RTCRtpScriptTransform;
+
+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);
+
+  RTCRtpScriptTransform* GetTransform() const { return mTransform; }
+
+  void SetTransform(RTCRtpScriptTransform* aTransform, ErrorResult& aError);
+
+  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();
+  void Unlink();
+  // 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);
+
+  // Called by FrameTransformerProxy
+  void RequestKeyFrame();
+
+  void OnRtcpBye();
+  void OnRtcpTimeout();
+
+  void SetTrackMuteFromRemoteSdp();
+  void OnRtpPacket();
+  void UpdateUnmuteBlockingState();
+  void UpdateReceiveTrackMute();
+
+  Canonical<Ssrc>& CanonicalSsrc() { return mSsrc; }
+  Canonical<Ssrc>& CanonicalVideoRtxSsrc() { return mVideoRtxSsrc; }
+  Canonical<RtpExtList>& CanonicalLocalRtpExtensions() {
+    return mLocalRtpExtensions;
+  }
+
+  Canonical<std::vector<AudioCodecConfig>>& CanonicalAudioCodecs() {
+    return mAudioCodecs;
+  }
+
+  Canonical<std::vector<VideoCodecConfig>>& CanonicalVideoCodecs() {
+    return mVideoCodecs;
+  }
+
+  Canonical<Maybe<RtpRtcpConfig>>& CanonicalVideoRtpRtcpConfig() {
+    return mVideoRtpRtcpConfig;
+  }
+
+  Canonical<bool>& CanonicalReceiving() override { return mReceiving; }
+
+  Canonical<RefPtr<FrameTransformerProxy>>& CanonicalFrameTransformerProxy() {
+    return mFrameTransformerProxy;
+  }
+
+ 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;
+  RefPtr<RTCRtpScriptTransform> mTransform;
+  // 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;
+  Canonical<RefPtr<FrameTransformerProxy>> mFrameTransformerProxy;
+};
+
+}  // namespace dom
+}  // namespace mozilla
+#endif  // _RTCRtpReceiver_h_
diff --git a/dom/media/webrtc/jsapi/RTCRtpScriptTransform.cpp b/dom/media/webrtc/jsapi/RTCRtpScriptTransform.cpp
new file mode 100644
index 0000000000..43f34c456f
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCRtpScriptTransform.cpp
@@ -0,0 +1,84 @@
+/* -*- Mode: C++; tab-width: 2; 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 https://mozilla.org/MPL/2.0/. */
+
+#include "RTCRtpScriptTransform.h"
+
+#include "libwebrtcglue/FrameTransformerProxy.h"
+#include "jsapi/RTCTransformEventRunnable.h"
+#include "mozilla/dom/Promise.h"
+#include "mozilla/dom/Worker.h"
+#include "mozilla/dom/RTCRtpScriptTransformBinding.h"
+#include "mozilla/dom/MessagePortBinding.h"
+#include "mozilla/Logging.h"
+#include "mozilla/AlreadyAddRefed.h"
+#include "mozilla/RefPtr.h"
+#include "nsPIDOMWindow.h"
+#include "nsContentUtils.h"
+#include "nsCOMPtr.h"
+#include "nsDebug.h"
+#include "ErrorList.h"
+#include "nsWrapperCache.h"
+#include "nsISupports.h"
+#include "nsCycleCollectionParticipant.h"
+#include "js/RootingAPI.h"
+
+namespace mozilla::dom {
+
+LazyLogModule gScriptTransformLog("RTCRtpScriptTransform");
+
+NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE(RTCRtpScriptTransform, mWindow)
+NS_IMPL_CYCLE_COLLECTING_ADDREF(RTCRtpScriptTransform)
+NS_IMPL_CYCLE_COLLECTING_RELEASE(RTCRtpScriptTransform)
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(RTCRtpScriptTransform)
+  NS_WRAPPERCACHE_INTERFACE_MAP_ENTRY
+  NS_INTERFACE_MAP_ENTRY(nsISupports)
+NS_INTERFACE_MAP_END
+
+already_AddRefed<RTCRtpScriptTransform> RTCRtpScriptTransform::Constructor(
+    const GlobalObject& aGlobal, Worker& aWorker,
+    JS::Handle<JS::Value> aOptions,
+    const Optional<Sequence<JSObject*>>& aTransfer, ErrorResult& aRv) {
+  nsCOMPtr<nsPIDOMWindowInner> ownerWindow =
+      do_QueryInterface(aGlobal.GetAsSupports());
+  if (NS_WARN_IF(!ownerWindow)) {
+    aRv.Throw(NS_ERROR_FAILURE);
+    return nullptr;
+  }
+  auto newTransform = MakeRefPtr<RTCRtpScriptTransform>(ownerWindow);
+  RefPtr<RTCTransformEventRunnable> runnable =
+      new RTCTransformEventRunnable(aWorker, &newTransform->GetProxy());
+
+  if (aTransfer.WasPassed()) {
+    aWorker.PostEventWithOptions(aGlobal.Context(), aOptions, aTransfer.Value(),
+                                 runnable, aRv);
+  } else {
+    StructuredSerializeOptions transferOptions;
+    aWorker.PostEventWithOptions(aGlobal.Context(), aOptions,
+                                 transferOptions.mTransfer, runnable, aRv);
+  }
+
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+
+  return newTransform.forget();
+}
+
+RTCRtpScriptTransform::RTCRtpScriptTransform(nsPIDOMWindowInner* aWindow)
+    : mWindow(aWindow), mProxy(new FrameTransformerProxy) {}
+
+RTCRtpScriptTransform::~RTCRtpScriptTransform() {
+  mProxy->ReleaseScriptTransformer();
+}
+
+JSObject* RTCRtpScriptTransform::WrapObject(JSContext* aCx,
+                                            JS::Handle<JSObject*> aGivenProto) {
+  return RTCRtpScriptTransform_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+}  // namespace mozilla::dom
+
+#undef LOGTAG
diff --git a/dom/media/webrtc/jsapi/RTCRtpScriptTransform.h b/dom/media/webrtc/jsapi/RTCRtpScriptTransform.h
new file mode 100644
index 0000000000..362c59b761
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCRtpScriptTransform.h
@@ -0,0 +1,63 @@
+/* -*- Mode: C++; tab-width: 2; 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 https://mozilla.org/MPL/2.0/. */
+
+#ifndef MOZILLA_DOM_MEDIA_WEBRTC_JSAPI_RTCRTPSCRIPTTRANSFORM_H_
+#define MOZILLA_DOM_MEDIA_WEBRTC_JSAPI_RTCRTPSCRIPTTRANSFORM_H_
+
+#include "nsISupports.h"
+#include "nsWrapperCache.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/Maybe.h"
+#include "js/RootingAPI.h"
+#include "nsTArray.h"
+
+class nsPIDOMWindowInner;
+
+namespace mozilla {
+class FrameTransformerProxy;
+class ErrorResult;
+
+namespace dom {
+class Worker;
+class GlobalObject;
+template <typename T>
+class Sequence;
+template <typename T>
+class Optional;
+
+class RTCRtpScriptTransform : public nsISupports, public nsWrapperCache {
+ public:
+  static already_AddRefed<RTCRtpScriptTransform> Constructor(
+      const GlobalObject& aGlobal, Worker& aWorker,
+      JS::Handle<JS::Value> aOptions,
+      const Optional<Sequence<JSObject*>>& aTransfer, ErrorResult& aRv);
+
+  explicit RTCRtpScriptTransform(nsPIDOMWindowInner* aWindow);
+
+  // nsISupports
+  NS_DECL_CYCLE_COLLECTING_ISUPPORTS
+  NS_DECL_CYCLE_COLLECTION_WRAPPERCACHE_CLASS(RTCRtpScriptTransform)
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  nsPIDOMWindowInner* GetParentObject() const { return mWindow; }
+
+  FrameTransformerProxy& GetProxy() { return *mProxy; }
+
+  bool IsClaimed() const { return mClaimed; }
+  void SetClaimed() { mClaimed = true; }
+
+ private:
+  virtual ~RTCRtpScriptTransform();
+  RefPtr<nsPIDOMWindowInner> mWindow;
+  RefPtr<FrameTransformerProxy> mProxy;
+  bool mClaimed = false;
+};
+
+}  // namespace dom
+}  // namespace mozilla
+#endif  // MOZILLA_DOM_MEDIA_WEBRTC_JSAPI_RTCRTPSCRIPTTRANSFORM_H_
diff --git a/dom/media/webrtc/jsapi/RTCRtpScriptTransformer.cpp b/dom/media/webrtc/jsapi/RTCRtpScriptTransformer.cpp
new file mode 100644
index 0000000000..126020a94f
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCRtpScriptTransformer.cpp
@@ -0,0 +1,449 @@
+/* -*- Mode: C++; tab-width: 2; 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 https://mozilla.org/MPL/2.0/. */
+
+#include "RTCRtpScriptTransformer.h"
+
+#include <stdint.h>
+
+#include <utility>
+#include <memory>
+#include <string>
+
+#include "api/frame_transformer_interface.h"
+
+#include "nsString.h"
+#include "nsCycleCollectionParticipant.h"
+#include "nsISupports.h"
+#include "ErrorList.h"
+#include "nsDebug.h"
+#include "nsCycleCollectionTraversalCallback.h"
+#include "nsTArray.h"
+#include "nsWrapperCache.h"
+#include "nsIGlobalObject.h"
+#include "nsCOMPtr.h"
+#include "nsStringFwd.h"
+#include "nsLiteralString.h"
+#include "nsContentUtils.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/AlreadyAddRefed.h"
+#include "mozilla/ErrorResult.h"
+#include "mozilla/Result.h"
+#include "mozilla/HoldDropJSObjects.h"
+#include "mozilla/Maybe.h"
+#include "mozilla/Assertions.h"
+#include "mozilla/Logging.h"
+#include "mozilla/ErrorResult.h"
+#include "mozilla/Likely.h"
+#include "mozilla/dom/RTCRtpScriptTransformerBinding.h"
+#include "mozilla/dom/BindingUtils.h"
+#include "mozilla/dom/BindingDeclarations.h"
+#include "mozilla/dom/PrototypeList.h"
+#include "mozilla/dom/WorkerRef.h"
+#include "mozilla/dom/RTCEncodedAudioFrame.h"
+#include "mozilla/dom/RTCEncodedVideoFrame.h"
+#include "mozilla/dom/UnderlyingSourceCallbackHelpers.h"
+#include "mozilla/dom/ToJSValue.h"
+#include "mozilla/dom/ReadableStream.h"
+#include "mozilla/dom/WritableStream.h"
+#include "mozilla/dom/UnderlyingSinkCallbackHelpers.h"
+#include "mozilla/dom/WritableStreamDefaultController.h"
+#include "mozilla/dom/ReadableStreamController.h"
+#include "mozilla/dom/Promise.h"
+#include "mozilla/dom/Promise-inl.h"
+#include "js/RootingAPI.h"
+#include "js/Value.h"
+#include "js/CallArgs.h"
+#include "libwebrtcglue/FrameTransformerProxy.h"
+#include "sdp/SdpAttribute.h"  // CheckRidValidity
+
+namespace mozilla::dom {
+
+LazyLogModule gScriptTransformerLog("RTCRtpScriptTransformer");
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(nsISupportsStreamSource,
+                                   UnderlyingSourceAlgorithmsWrapper, mStream,
+                                   mThingQueuedPromise, mQueue)
+NS_IMPL_ADDREF_INHERITED(nsISupportsStreamSource,
+                         UnderlyingSourceAlgorithmsWrapper)
+NS_IMPL_RELEASE_INHERITED(nsISupportsStreamSource,
+                          UnderlyingSourceAlgorithmsWrapper)
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(nsISupportsStreamSource)
+NS_INTERFACE_MAP_END_INHERITING(UnderlyingSourceAlgorithmsWrapper)
+
+nsISupportsStreamSource::nsISupportsStreamSource() = default;
+
+nsISupportsStreamSource::~nsISupportsStreamSource() = default;
+
+void nsISupportsStreamSource::Init(ReadableStream* aStream) {
+  mStream = aStream;
+}
+
+void nsISupportsStreamSource::Enqueue(nsISupports* aThing) {
+  if (!mThingQueuedPromise) {
+    mQueue.AppendElement(aThing);
+    return;
+  }
+
+  // Maybe put a limit here? Or at least some sort of logging if this gets
+  // unreasonably long?
+  AutoJSAPI jsapi;
+  if (NS_WARN_IF(!jsapi.Init(mStream->GetParentObject()))) {
+    return;
+  }
+
+  EnqueueToStream(jsapi.cx(), aThing);
+  mThingQueuedPromise->MaybeResolveWithUndefined();
+  mThingQueuedPromise = nullptr;
+}
+
+already_AddRefed<Promise> nsISupportsStreamSource::PullCallbackImpl(
+    JSContext* aCx, ReadableStreamController& aController, ErrorResult& aRv) {
+  if (!mQueue.IsEmpty()) {
+    EnqueueOneThingFromQueue(aCx);
+    return nullptr;
+  }
+
+  RefPtr<nsISupportsStreamSource> self(this);
+  mThingQueuedPromise = Promise::CreateInfallible(mStream->GetParentObject());
+  return do_AddRef(mThingQueuedPromise);
+}
+
+void nsISupportsStreamSource::EnqueueToStream(JSContext* aCx,
+                                              nsISupports* aThing) {
+  JS::Rooted<JS::Value> jsThing(aCx);
+  if (NS_WARN_IF(MOZ_UNLIKELY(!ToJSValue(aCx, *aThing, &jsThing)))) {
+    // Do we want to add error handling for this?
+    return;
+  }
+  IgnoredErrorResult rv;
+  // EnqueueNative is CAN_RUN_SCRIPT. Need a strong-ref temporary.
+  auto stream = mStream;
+  stream->EnqueueNative(aCx, jsThing, rv);
+}
+
+void nsISupportsStreamSource::EnqueueOneThingFromQueue(JSContext* aCx) {
+  if (!mQueue.IsEmpty()) {
+    RefPtr<nsISupports> thing = mQueue[0];
+    mQueue.RemoveElementAt(0);
+    EnqueueToStream(aCx, thing);
+  }
+}
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(WritableStreamRTCFrameSink,
+                                   UnderlyingSinkAlgorithmsWrapper,
+                                   mTransformer)
+NS_IMPL_ADDREF_INHERITED(WritableStreamRTCFrameSink,
+                         UnderlyingSinkAlgorithmsWrapper)
+NS_IMPL_RELEASE_INHERITED(WritableStreamRTCFrameSink,
+                          UnderlyingSinkAlgorithmsWrapper)
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(WritableStreamRTCFrameSink)
+NS_INTERFACE_MAP_END_INHERITING(UnderlyingSinkAlgorithmsWrapper)
+
+WritableStreamRTCFrameSink::WritableStreamRTCFrameSink(
+    RTCRtpScriptTransformer* aTransformer)
+    : mTransformer(aTransformer) {}
+
+WritableStreamRTCFrameSink::~WritableStreamRTCFrameSink() = default;
+
+already_AddRefed<Promise> WritableStreamRTCFrameSink::WriteCallback(
+    JSContext* aCx, JS::Handle<JS::Value> aChunk,
+    WritableStreamDefaultController& aController, ErrorResult& aError) {
+  // Spec does not say to do this right now. Might be a spec bug, needs
+  // clarification.
+  // https://github.com/w3c/webrtc-encoded-transform/issues/191
+  if (NS_WARN_IF(!aChunk.isObject())) {
+    aError.ThrowTypeError(
+        "Wrong type for RTCRtpScriptTransformer.[[writeable]]: Not an object");
+    return nullptr;
+  }
+
+  // Lame. But, without a webidl base class, this is the only way.
+  RefPtr<RTCEncodedVideoFrame> video;
+  UNWRAP_OBJECT(RTCEncodedVideoFrame, &aChunk.toObject(), video);
+  RefPtr<RTCEncodedAudioFrame> audio;
+  UNWRAP_OBJECT(RTCEncodedAudioFrame, &aChunk.toObject(), audio);
+
+  RefPtr<RTCEncodedFrameBase> frame;
+  if (video) {
+    frame = video;
+  } else if (audio) {
+    frame = audio;
+  }
+
+  if (NS_WARN_IF(!frame)) {
+    aError.ThrowTypeError(
+        "Wrong type for RTCRtpScriptTransformer.[[writeable]]: Not an "
+        "RTCEncodedAudioFrame or RTCEncodedVideoFrame");
+    return nullptr;
+  }
+
+  return mTransformer->OnTransformedFrame(frame, aError);
+}
+
+// There is not presently an implementation of these for nsTHashMap :(
+inline void ImplCycleCollectionUnlink(
+    RTCRtpScriptTransformer::GenerateKeyFramePromises& aMap) {
+  for (auto& tableEntry : aMap) {
+    ImplCycleCollectionUnlink(*tableEntry.GetModifiableData());
+  }
+  aMap.Clear();
+}
+
+inline void ImplCycleCollectionTraverse(
+    nsCycleCollectionTraversalCallback& aCallback,
+    RTCRtpScriptTransformer::GenerateKeyFramePromises& aMap, const char* aName,
+    uint32_t aFlags = 0) {
+  for (auto& tableEntry : aMap) {
+    ImplCycleCollectionTraverse(aCallback, *tableEntry.GetModifiableData(),
+                                aName, aFlags);
+  }
+}
+
+NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE_WITH_JS_MEMBERS(
+    RTCRtpScriptTransformer,
+    (mGlobal, mReadableSource, mReadable, mWritable, mWritableSink,
+     mKeyFrameRequestPromises, mGenerateKeyFramePromises),
+    (mOptions))
+
+NS_IMPL_CYCLE_COLLECTING_ADDREF(RTCRtpScriptTransformer)
+NS_IMPL_CYCLE_COLLECTING_RELEASE(RTCRtpScriptTransformer)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(RTCRtpScriptTransformer)
+  NS_WRAPPERCACHE_INTERFACE_MAP_ENTRY
+  NS_INTERFACE_MAP_ENTRY(nsISupports)
+NS_INTERFACE_MAP_END
+
+RTCRtpScriptTransformer::RTCRtpScriptTransformer(nsIGlobalObject* aGlobal)
+    : mGlobal(aGlobal),
+      mReadableSource(new nsISupportsStreamSource),
+      mWritableSink(new WritableStreamRTCFrameSink(this)),
+      mOptions(JS::UndefinedHandleValue) {
+  mozilla::HoldJSObjects(this);
+}
+
+RTCRtpScriptTransformer::~RTCRtpScriptTransformer() {
+  mozilla::DropJSObjects(this);
+}
+
+nsresult RTCRtpScriptTransformer::Init(JSContext* aCx,
+                                       JS::Handle<JS::Value> aOptions,
+                                       WorkerPrivate* aWorkerPrivate,
+                                       FrameTransformerProxy* aProxy) {
+  ErrorResult rv;
+  RefPtr<nsIGlobalObject> global(mGlobal);
+  auto source = mReadableSource;
+  auto sink = mWritableSink;
+
+  // NOTE: We do not transfer these streams from mainthread, as the spec says,
+  // because there's no JS observable reason to. The spec is likely to change
+  // here, because it is overspecifying implementation details.
+  mReadable = ReadableStream::CreateNative(aCx, global, *source, Some(1.0),
+                                           nullptr, rv);
+  if (rv.Failed()) {
+    return rv.StealNSResult();
+  }
+  mReadableSource->Init(mReadable);
+
+  // WritableStream::CreateNative takes a nsIGlobalObject&, but
+  // ReadableStream::CreateNative takes a nsIGlobalObject*?
+  mWritable =
+      WritableStream::CreateNative(aCx, *global, *sink, Nothing(), nullptr, rv);
+  if (rv.Failed()) {
+    return rv.StealNSResult();
+  }
+
+  mOptions = aOptions;
+  mProxy = aProxy;
+  // This will return null if the worker is already shutting down.
+  // A call to ReleaseScriptTransformer will eventually result in a call to
+  // NotifyReleased.
+  mWorkerRef = StrongWorkerRef::Create(
+      aWorkerPrivate, "RTCRtpScriptTransformer",
+      [this, self = RefPtr(this)]() { mProxy->ReleaseScriptTransformer(); });
+  if (mWorkerRef) {
+    mProxy->SetScriptTransformer(*this);
+  }
+  return NS_OK;
+}
+
+void RTCRtpScriptTransformer::NotifyReleased() {
+  RejectPendingPromises();
+  mWorkerRef = nullptr;
+  mProxy = nullptr;
+}
+
+void RTCRtpScriptTransformer::RejectPendingPromises() {
+  for (const auto& promise : mKeyFrameRequestPromises) {
+    ErrorResult rv;
+    rv.ThrowInvalidStateError(
+        "RTCRtpScriptTransformer is not associated with a receiver");
+    promise->MaybeReject(std::move(rv));
+  }
+  mKeyFrameRequestPromises.Clear();
+
+  // GenerateKeyFrame promises are indexed by rid
+  for (auto& ridAndPromises : mGenerateKeyFramePromises) {
+    for (const auto& promise : ridAndPromises.GetData()) {
+      ErrorResult rv;
+      rv.ThrowInvalidStateError(
+          "RTCRtpScriptTransformer is not associated with a sender");
+      promise->MaybeReject(std::move(rv));
+    }
+  }
+  mGenerateKeyFramePromises.Clear();
+}
+
+void RTCRtpScriptTransformer::TransformFrame(
+    std::unique_ptr<webrtc::TransformableFrameInterface> aFrame) {
+  if (!mVideo.isSome()) {
+    // First frame. mProxy will know whether it's video or not by now.
+    mVideo = mProxy->IsVideo();
+    MOZ_ASSERT(mVideo.isSome());
+  }
+
+  RefPtr<RTCEncodedFrameBase> domFrame;
+  if (*mVideo) {
+    // If this is a send video keyframe, resolve any pending GenerateKeyFrame
+    // promises for its rid.
+    if (aFrame->GetDirection() ==
+        webrtc::TransformableFrameInterface::Direction::kSender) {
+      auto* videoFrame =
+          static_cast<webrtc::TransformableVideoFrameInterface*>(aFrame.get());
+      if (videoFrame->IsKeyFrame()) {
+        ResolveGenerateKeyFramePromises(videoFrame->GetRid(),
+                                        videoFrame->GetTimestamp());
+        if (!videoFrame->GetRid().empty() &&
+            videoFrame->Metadata().GetSimulcastIdx() == 0) {
+          ResolveGenerateKeyFramePromises("", videoFrame->GetTimestamp());
+        }
+      }
+    }
+    domFrame = new RTCEncodedVideoFrame(mGlobal, std::move(aFrame),
+                                        ++mLastEnqueuedFrameCounter, this);
+  } else {
+    domFrame = new RTCEncodedAudioFrame(mGlobal, std::move(aFrame),
+                                        ++mLastEnqueuedFrameCounter, this);
+  }
+  mReadableSource->Enqueue(domFrame);
+}
+
+void RTCRtpScriptTransformer::GetOptions(JSContext* aCx,
+                                         JS::MutableHandle<JS::Value> aVal,
+                                         ErrorResult& aError) {
+  if (!ToJSValue(aCx, mOptions, aVal)) {
+    aError.NoteJSContextException(aCx);
+  }
+}
+
+already_AddRefed<Promise> RTCRtpScriptTransformer::GenerateKeyFrame(
+    const Optional<nsAString>& aRid) {
+  Maybe<std::string> utf8Rid;
+  if (aRid.WasPassed()) {
+    utf8Rid = Some(NS_ConvertUTF16toUTF8(aRid.Value()).get());
+    std::string error;
+    if (!SdpRidAttributeList::CheckRidValidity(*utf8Rid, &error)) {
+      ErrorResult rv;
+      nsCString nsError(error.c_str());
+      rv.ThrowNotAllowedError(nsError);
+      return Promise::CreateRejectedWithErrorResult(GetParentObject(), rv);
+    }
+  }
+
+  nsCString key;
+  if (utf8Rid.isSome()) {
+    key.Assign(utf8Rid->data(), utf8Rid->size());
+  }
+
+  nsTArray<RefPtr<Promise>>& promises =
+      mGenerateKeyFramePromises.LookupOrInsert(key);
+  if (!promises.Length()) {
+    // No pending keyframe generation request for this rid. Make one.
+    if (!mProxy || !mProxy->GenerateKeyFrame(utf8Rid)) {
+      ErrorResult rv;
+      rv.ThrowInvalidStateError(
+          "RTCRtpScriptTransformer is not associated with a video sender");
+      return Promise::CreateRejectedWithErrorResult(GetParentObject(), rv);
+    }
+  }
+  RefPtr<Promise> promise = Promise::CreateInfallible(GetParentObject());
+  promises.AppendElement(promise);
+  return promise.forget();
+}
+
+void RTCRtpScriptTransformer::ResolveGenerateKeyFramePromises(
+    const std::string& aRid, uint64_t aTimestamp) {
+  nsCString key(aRid.data(), aRid.size());
+  nsTArray<RefPtr<Promise>> promises;
+  mGenerateKeyFramePromises.Remove(key, &promises);
+  for (auto& promise : promises) {
+    promise->MaybeResolve(aTimestamp);
+  }
+}
+
+void RTCRtpScriptTransformer::GenerateKeyFrameError(
+    const Maybe<std::string>& aRid, const CopyableErrorResult& aResult) {
+  nsCString key;
+  if (aRid.isSome()) {
+    key.Assign(aRid->data(), aRid->size());
+  }
+  nsTArray<RefPtr<Promise>> promises;
+  mGenerateKeyFramePromises.Remove(key, &promises);
+  for (auto& promise : promises) {
+    CopyableErrorResult rv(aResult);
+    promise->MaybeReject(std::move(rv));
+  }
+}
+
+already_AddRefed<Promise> RTCRtpScriptTransformer::SendKeyFrameRequest() {
+  if (!mKeyFrameRequestPromises.Length()) {
+    if (!mProxy || !mProxy->RequestKeyFrame()) {
+      ErrorResult rv;
+      rv.ThrowInvalidStateError(
+          "RTCRtpScriptTransformer is not associated with a video receiver");
+      return Promise::CreateRejectedWithErrorResult(GetParentObject(), rv);
+    }
+  }
+  RefPtr<Promise> promise = Promise::CreateInfallible(GetParentObject());
+  mKeyFrameRequestPromises.AppendElement(promise);
+  return promise.forget();
+}
+
+void RTCRtpScriptTransformer::KeyFrameRequestDone(bool aSuccess) {
+  auto promises = std::move(mKeyFrameRequestPromises);
+  if (aSuccess) {
+    for (const auto& promise : promises) {
+      promise->MaybeResolveWithUndefined();
+    }
+  } else {
+    for (const auto& promise : promises) {
+      ErrorResult rv;
+      rv.ThrowInvalidStateError(
+          "Depacketizer is not defined, or not processing");
+      promise->MaybeReject(std::move(rv));
+    }
+  }
+}
+
+JSObject* RTCRtpScriptTransformer::WrapObject(
+    JSContext* aCx, JS::Handle<JSObject*> aGivenProto) {
+  return RTCRtpScriptTransformer_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+already_AddRefed<Promise> RTCRtpScriptTransformer::OnTransformedFrame(
+    RTCEncodedFrameBase* aFrame, ErrorResult& aError) {
+  // Spec says to skip frames that are out of order or have wrong owner
+  if (aFrame->GetCounter() > mLastReceivedFrameCounter &&
+      aFrame->CheckOwner(this) && mProxy) {
+    mLastReceivedFrameCounter = aFrame->GetCounter();
+    mProxy->OnTransformedFrame(aFrame->TakeFrame());
+  }
+
+  return Promise::CreateResolvedWithUndefined(GetParentObject(), aError);
+}
+
+}  // namespace mozilla::dom
+
+#undef LOGTAG
diff --git a/dom/media/webrtc/jsapi/RTCRtpScriptTransformer.h b/dom/media/webrtc/jsapi/RTCRtpScriptTransformer.h
new file mode 100644
index 0000000000..6d61ac3cd5
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCRtpScriptTransformer.h
@@ -0,0 +1,197 @@
+/* -*- Mode: C++; tab-width: 2; 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 https://mozilla.org/MPL/2.0/. */
+
+#ifndef MOZILLA_DOM_MEDIA_WEBRTC_JSAPI_RTCRTPSCRIPTTRANSFORMER_H_
+#define MOZILLA_DOM_MEDIA_WEBRTC_JSAPI_RTCRTPSCRIPTTRANSFORMER_H_
+
+#include "nsISupports.h"
+#include "nsWrapperCache.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/dom/ReadableStream.h"
+#include "mozilla/dom/WritableStream.h"
+#include "mozilla/Maybe.h"
+#include "js/RootingAPI.h"
+#include "nsTArray.h"
+#include "nsCOMArray.h"
+#include <memory>
+#include "nsTHashSet.h"
+#include "nsCycleCollectionParticipant.h"
+
+class nsPIDOMWindowInner;
+
+namespace webrtc {
+class TransformableFrameInterface;
+}
+
+namespace mozilla {
+class FrameTransformerProxy;
+
+namespace dom {
+class Worker;
+class WorkerPrivate;
+
+// Dirt simple source for ReadableStream that accepts nsISupports
+// Might be suitable to move someplace else, with some polish.
+class nsISupportsStreamSource final : public UnderlyingSourceAlgorithmsWrapper {
+ public:
+  nsISupportsStreamSource();
+  nsISupportsStreamSource(const nsISupportsStreamSource&) = delete;
+  nsISupportsStreamSource(nsISupportsStreamSource&&) = delete;
+  nsISupportsStreamSource& operator=(const nsISupportsStreamSource&) = delete;
+  nsISupportsStreamSource& operator=(nsISupportsStreamSource&&) = delete;
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(nsISupportsStreamSource,
+                                           UnderlyingSourceAlgorithmsWrapper)
+
+  void Init(ReadableStream* aStream);
+
+  void Enqueue(nsISupports* aThing);
+
+  // From UnderlyingSourceAlgorithmsWrapper
+  already_AddRefed<Promise> PullCallbackImpl(
+      JSContext* aCx, ReadableStreamController& aController,
+      ErrorResult& aRv) override;
+
+  void EnqueueOneThingFromQueue(JSContext* aCx);
+
+ private:
+  virtual ~nsISupportsStreamSource();
+
+  // Calls ReadableStream::EnqueueNative, which is MOZ_CAN_RUN_SCRIPT.
+  MOZ_CAN_RUN_SCRIPT_BOUNDARY void EnqueueToStream(JSContext* aCx,
+                                                   nsISupports* aThing);
+
+  RefPtr<ReadableStream> mStream;
+  RefPtr<Promise> mThingQueuedPromise;
+  // mozilla::Queue is not cycle-collector friendly :(
+  nsCOMArray<nsISupports> mQueue;
+};
+
+class RTCRtpScriptTransformer;
+
+class WritableStreamRTCFrameSink final
+    : public UnderlyingSinkAlgorithmsWrapper {
+ public:
+  explicit WritableStreamRTCFrameSink(RTCRtpScriptTransformer* aTransformer);
+  WritableStreamRTCFrameSink(const WritableStreamRTCFrameSink&) = delete;
+  WritableStreamRTCFrameSink(WritableStreamRTCFrameSink&&) = delete;
+  WritableStreamRTCFrameSink& operator=(const WritableStreamRTCFrameSink&) =
+      delete;
+  WritableStreamRTCFrameSink& operator=(WritableStreamRTCFrameSink&&) = delete;
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(WritableStreamRTCFrameSink,
+                                           UnderlyingSinkAlgorithmsWrapper)
+
+  already_AddRefed<Promise> WriteCallback(
+      JSContext* aCx, JS::Handle<JS::Value> aChunk,
+      WritableStreamDefaultController& aController,
+      ErrorResult& aError) override;
+
+ private:
+  virtual ~WritableStreamRTCFrameSink();
+  RefPtr<RTCRtpScriptTransformer> mTransformer;
+};
+
+class RTCEncodedFrameBase;
+
+// Here's the basic flow. All of this happens on the worker thread.
+// 0. We register with a FrameTransformerProxy.
+// 1. That FrameTransformerProxy dispatches webrtc::TransformableFrameInterface
+//    to us (from either the encoder/depacketizer thread), via our
+//    TransformFrame method.
+// 2. We wrap these frames in RTCEncodedAudio/VideoFrame, and feed them to
+//    mReadableSource, which queues them.
+// 3. mReadableSource.PullCallbackImpl consumes that queue, and feeds the
+//    frames to mReadable.
+// 4. JS worker code consumes from mReadable, does whatever transformation it
+//    wants, then writes the frames to mWritable.
+// 5. mWritableSink.WriteCallback passes those frames to us.
+// 6. We unwrap the webrtc::TransformableFrameInterface from these frames.
+// 7. We pass these unwrapped frames back to the FrameTransformerProxy.
+//    (FrameTransformerProxy handles any dispatching/synchronization necessary)
+// 8. Eventually, that FrameTransformerProxy calls NotifyReleased (possibly at
+//    our prompting).
+class RTCRtpScriptTransformer final : public nsISupports,
+                                      public nsWrapperCache {
+ public:
+  explicit RTCRtpScriptTransformer(nsIGlobalObject* aGlobal);
+
+  MOZ_CAN_RUN_SCRIPT_BOUNDARY nsresult Init(JSContext* aCx,
+                                            JS::Handle<JS::Value> aOptions,
+                                            WorkerPrivate* aWorkerPrivate,
+                                            FrameTransformerProxy* aProxy);
+
+  void NotifyReleased();
+
+  void TransformFrame(
+      std::unique_ptr<webrtc::TransformableFrameInterface> aFrame);
+  already_AddRefed<Promise> OnTransformedFrame(RTCEncodedFrameBase* aFrame,
+                                               ErrorResult& aError);
+
+  // nsISupports
+  NS_DECL_CYCLE_COLLECTING_ISUPPORTS
+  NS_DECL_CYCLE_COLLECTION_SCRIPT_HOLDER_CLASS(RTCRtpScriptTransformer)
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  nsIGlobalObject* GetParentObject() const { return mGlobal; }
+
+  // WebIDL Interface
+  already_AddRefed<mozilla::dom::ReadableStream> Readable() const {
+    return do_AddRef(mReadable);
+  }
+  already_AddRefed<mozilla::dom::WritableStream> Writable() const {
+    return do_AddRef(mWritable);
+  }
+
+  void GetOptions(JSContext* aCx, JS::MutableHandle<JS::Value> aVal,
+                  ErrorResult& aError);
+
+  already_AddRefed<Promise> GenerateKeyFrame(const Optional<nsAString>& aRid);
+  void GenerateKeyFrameError(const Maybe<std::string>& aRid,
+                             const CopyableErrorResult& aResult);
+  already_AddRefed<Promise> SendKeyFrameRequest();
+  void KeyFrameRequestDone(bool aSuccess);
+
+  // Public to ease implementation of cycle collection functions
+  using GenerateKeyFramePromises =
+      nsTHashMap<nsCStringHashKey, nsTArray<RefPtr<Promise>>>;
+
+ private:
+  virtual ~RTCRtpScriptTransformer();
+  void RejectPendingPromises();
+  // empty string means no rid
+  void ResolveGenerateKeyFramePromises(const std::string& aRid,
+                                       uint64_t aTimestamp);
+
+  nsCOMPtr<nsIGlobalObject> mGlobal;
+
+  RefPtr<FrameTransformerProxy> mProxy;
+  RefPtr<nsISupportsStreamSource> mReadableSource;
+  RefPtr<ReadableStream> mReadable;
+  RefPtr<WritableStream> mWritable;
+  RefPtr<WritableStreamRTCFrameSink> mWritableSink;
+
+  JS::Heap<JS::Value> mOptions;
+  uint64_t mLastEnqueuedFrameCounter = 0;
+  uint64_t mLastReceivedFrameCounter = 0;
+  nsTArray<RefPtr<Promise>> mKeyFrameRequestPromises;
+  // Contains the promise returned for each call to GenerateKeyFrame(rid), in
+  // the order in which it was called, keyed by the rid (empty string if not
+  // passed). If there is already a promise in here for a given rid, we do not
+  // ask the FrameTransformerProxy again, and just bulk resolve/reject.
+  GenerateKeyFramePromises mGenerateKeyFramePromises;
+  Maybe<bool> mVideo;
+  RefPtr<StrongWorkerRef> mWorkerRef;
+};
+
+}  // namespace dom
+}  // namespace mozilla
+
+#endif  // MOZILLA_DOM_MEDIA_WEBRTC_JSAPI_RTCRTPSCRIPTTRANSFORMER_H_
diff --git a/dom/media/webrtc/jsapi/RTCRtpSender.cpp b/dom/media/webrtc/jsapi/RTCRtpSender.cpp
new file mode 100644
index 0000000000..4ba0491521
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCRtpSender.cpp
@@ -0,0 +1,1811 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. 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 <stdint.h>
+
+#include <vector>
+#include <string>
+#include <algorithm>
+#include <utility>
+#include <iterator>
+#include <set>
+#include <sstream>
+
+#include "system_wrappers/include/clock.h"
+#include "call/call.h"
+#include "api/rtp_parameters.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "api/video_codecs/video_codec.h"
+#include "api/video/video_codec_constants.h"
+#include "call/audio_send_stream.h"
+#include "call/video_send_stream.h"
+#include "modules/rtp_rtcp/include/report_block_data.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+#include "nsPIDOMWindow.h"
+#include "nsString.h"
+#include "MainThreadUtils.h"
+#include "nsCOMPtr.h"
+#include "nsContentUtils.h"
+#include "nsCycleCollectionParticipant.h"
+#include "nsDebug.h"
+#include "nsISupports.h"
+#include "nsLiteralString.h"
+#include "nsStringFwd.h"
+#include "nsTArray.h"
+#include "nsThreadUtils.h"
+#include "nsWrapperCache.h"
+#include "mozilla/AbstractThread.h"
+#include "mozilla/AlreadyAddRefed.h"
+#include "mozilla/Assertions.h"
+#include "mozilla/Attributes.h"
+#include "mozilla/fallible.h"
+#include "mozilla/Logging.h"
+#include "mozilla/mozalloc_oom.h"
+#include "mozilla/MozPromise.h"
+#include "mozilla/OwningNonNull.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/StateWatching.h"
+#include "mozilla/UniquePtr.h"
+#include "mozilla/Unused.h"
+#include "mozilla/StateMirroring.h"
+#include "mozilla/Maybe.h"
+#include "mozilla/dom/MediaStreamTrack.h"
+#include "mozilla/dom/Promise.h"
+#include "mozilla/dom/RTCRtpScriptTransform.h"
+#include "mozilla/dom/VideoStreamTrack.h"
+#include "mozilla/dom/RTCRtpSenderBinding.h"
+#include "mozilla/dom/MediaStreamTrackBinding.h"
+#include "mozilla/dom/Nullable.h"
+#include "mozilla/dom/RTCRtpParametersBinding.h"
+#include "mozilla/dom/RTCStatsReportBinding.h"
+#include "mozilla/glean/GleanMetrics.h"
+#include "js/RootingAPI.h"
+#include "jsep/JsepTransceiver.h"
+#include "RTCStatsReport.h"
+#include "RTCRtpTransceiver.h"
+#include "PeerConnectionImpl.h"
+#include "libwebrtcglue/CodecConfig.h"
+#include "libwebrtcglue/MediaConduitControl.h"
+#include "libwebrtcglue/MediaConduitInterface.h"
+#include "sdp/SdpAttribute.h"
+#include "sdp/SdpEnum.h"
+
+namespace mozilla::dom {
+
+LazyLogModule gSenderLog("RTCRtpSender");
+
+NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE_CLASS(RTCRtpSender)
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(RTCRtpSender)
+  tmp->Unlink();
+  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, mTransform, 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),
+      INIT_CANONICAL(mFrameTransformerProxy, nullptr) {
+  mPipeline = MediaPipelineTransmit::Create(
+      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.extended_highest_sequence_number() <
+                      *base_seq) {
+                    aRemote.mPacketsReceived.Construct(0);
+                  } else {
+                    aRemote.mPacketsReceived.Construct(
+                        aRtcpData.extended_highest_sequence_number() -
+                        aRtcpData.cumulative_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);
+                }
+              };
+
+          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;
+              };
+
+          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.source_ssrc() == ssrc &&
+                      data.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_rtts().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);
+            }
+
+            // TODO(bug 1804678): Use RTCAudioSourceStats
+            RTCMediaSourceStats mediaSourceStats;
+            constructCommonMediaSourceStats(mediaSourceStats);
+            if (!report->mMediaSourceStats.AppendElement(
+                    std::move(mediaSourceStats), 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 ==
+                webrtc::Timestamp::PlusInfinity()) {
+              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.jitter()) /
+                  webrtc::kVideoPayloadTypeFrequency);
+              remote.mPacketsLost.Construct(rtcpReportData.cumulative_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_rtts().ms() / 1000.0);
+              remote.mFractionLost.Construct(
+                  static_cast<float>(rtcpReportData.fraction_lost_raw()) /
+                  (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.mFramesPerSecond.Construct(streamStats->encode_frame_rate);
+            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);
+            }
+
+            RTCVideoSourceStats videoSourceStats;
+            constructCommonMediaSourceStats(videoSourceStats);
+            // webrtc::VideoSendStream::Stats does not have width/height. We
+            // might be able to get this somewhere else?
+            // videoStats->frames is not documented, but looking at the
+            // implementation it appears to be the number of frames inputted to
+            // the encoder, which ought to work.
+            videoSourceStats.mFrames.Construct(videoStats->frames);
+            videoSourceStats.mFramesPerSecond.Construct(
+                videoStats->input_frame_rate);
+            auto resolution = aConduit->GetLastResolution();
+            resolution.apply(
+                [&](const VideoSessionConduit::Resolution& aResolution) {
+                  videoSourceStats.mWidth.Construct(aResolution.width);
+                  videoSourceStats.mHeight.Construct(aResolution.height);
+                });
+            if (!report->mVideoSourceStats.AppendElement(
+                    std::move(videoSourceStats), 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.[[Stopping]] is true, return a promise rejected with a newly
+  // created InvalidStateError.
+  if (mTransceiver->Stopping()) {
+    p->MaybeRejectWithInvalidStateError(
+        "This sender's transceiver is stopping/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 (!mHaveFailedBecauseStaleTransactionId) {
+      mHaveFailedBecauseStaleTransactionId = true;
+      mozilla::glean::rtcrtpsender_setparameters::fail_stale_transactionid
+          .AddToNumerator(1);
+    }
+    p->MaybeRejectWithInvalidModificationError(error);
+    return p.forget();
+  }
+
+  // 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.mMaxBitrate == a2.mMaxBitrate &&
+         a1.mMaxFramerate == a2.mMaxFramerate && a1.mPriority == a2.mPriority &&
+         a1.mRid == a2.mRid &&
+         a1.mScaleResolutionDownBy == a2.mScaleResolutionDownBy;
+}
+
+// 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.[[Stopping]] is true, return a promise rejected with a
+  // newly created InvalidStateError.
+  if (mTransceiver->Stopped() || mTransceiver->Stopping()) {
+    RefPtr<dom::Promise> error = sender->MakePromise(aError);
+    if (aError.Failed()) {
+      return nullptr;
+    }
+    MOZ_LOG(gSenderLog, LogLevel::Debug,
+            ("%s Cannot call replaceTrack when transceiver is stopping",
+             __FUNCTION__));
+    error->MaybeRejectWithInvalidStateError(
+        "Cannot call replaceTrack when transceiver is stopping");
+    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
+  if (mTransceiver->Stopping()) {
+    return;
+  }
+  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.
+    mSenderTrackSetByAddTrack = 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;
+  if (mTransform) {
+    mTransform->GetProxy().SetSender(nullptr);
+  }
+}
+
+void RTCRtpSender::BreakCycles() {
+  mWindow = nullptr;
+  mPc = nullptr;
+  mSenderTrack = nullptr;
+  mTransceiver = nullptr;
+  mStreams.Clear();
+  mDtmf = nullptr;
+}
+
+void RTCRtpSender::Unlink() {
+  if (mTransceiver) {
+    mTransceiver->Unlink();
+  }
+}
+
+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 (mSenderTrackSetByAddTrack) {
+    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;
+      }
+    }
+  }
+
+  if (!mHaveLoggedUlpfecInfo) {
+    bool ulpfecNegotiated = false;
+    for (const auto& codec : configs) {
+      if (nsCRT::strcasestr(codec.mName.c_str(), "ulpfec")) {
+        ulpfecNegotiated = true;
+      }
+    }
+    mozilla::glean::codec_stats::ulpfec_negotiated
+        .Get(ulpfecNegotiated ? "negotiated"_ns : "not_negotiated"_ns)
+        .Add(1);
+    mHaveLoggedUlpfecInfo = true;
+  }
+
+  // Log codec information we are tracking
+  if (!mHaveLoggedOtherFec &&
+      !GetJsepTransceiver().mSendTrack.GetFecCodecName().empty()) {
+    mozilla::glean::codec_stats::other_fec_signaled
+        .Get(nsDependentCString(
+            GetJsepTransceiver().mSendTrack.GetFecCodecName().c_str()))
+        .Add(1);
+    mHaveLoggedOtherFec = true;
+  }
+  if (!mHaveLoggedVideoPreferredCodec &&
+      !GetJsepTransceiver().mSendTrack.GetVideoPreferredCodec().empty()) {
+    mozilla::glean::codec_stats::video_preferred_codec
+        .Get(nsDependentCString(
+            GetJsepTransceiver().mSendTrack.GetVideoPreferredCodec().c_str()))
+        .Add(1);
+    mHaveLoggedVideoPreferredCodec = true;
+  }
+
+  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 (!mHaveLoggedAudioPreferredCodec &&
+      !GetJsepTransceiver().mSendTrack.GetAudioPreferredCodec().empty()) {
+    mozilla::glean::codec_stats::audio_preferred_codec
+        .Get(nsDependentCString(
+            GetJsepTransceiver().mSendTrack.GetAudioPreferredCodec().c_str()))
+        .Add(1);
+    mHaveLoggedAudioPreferredCodec = true;
+  }
+
+  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) {
+  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->Stopping());
+  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();
+}
+
+void RTCRtpSender::SetTransform(RTCRtpScriptTransform* aTransform,
+                                ErrorResult& aError) {
+  if (aTransform == mTransform.get()) {
+    // Ok... smile and nod
+    // TODO: Depending on spec, this might throw
+    // https://github.com/w3c/webrtc-encoded-transform/issues/189
+    return;
+  }
+
+  if (aTransform && aTransform->IsClaimed()) {
+    aError.ThrowInvalidStateError("transform has already been used elsewhere");
+    return;
+  }
+
+  // Seamless switch for frames
+  if (aTransform) {
+    mFrameTransformerProxy = &aTransform->GetProxy();
+  } else {
+    mFrameTransformerProxy = nullptr;
+  }
+
+  if (mTransform) {
+    mTransform->GetProxy().SetSender(nullptr);
+  }
+
+  mTransform = const_cast<RTCRtpScriptTransform*>(aTransform);
+
+  if (mTransform) {
+    mTransform->GetProxy().SetSender(this);
+    mTransform->SetClaimed();
+  }
+}
+
+bool RTCRtpSender::GenerateKeyFrame(const Maybe<std::string>& aRid) {
+  if (!mTransform || !mPipeline || !mSenderTrack) {
+    return false;
+  }
+
+  mPipeline->mConduit->AsVideoSessionConduit().apply([&](const auto& conduit) {
+    conduit->GenerateKeyFrame(aRid, &mTransform->GetProxy());
+  });
+  return true;
+}
+
+}  // 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..9b26b6897e
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCRtpSender.h
@@ -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/. */
+
+#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 RTCRtpScriptTransform;
+
+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);
+
+  RTCRtpScriptTransform* GetTransform() const { return mTransform; }
+
+  void SetTransform(RTCRtpScriptTransform* aTransform, ErrorResult& aError);
+  bool GenerateKeyFrame(const Maybe<std::string>& aRid);
+
+  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();
+  void Unlink();
+  // 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();
+
+  Canonical<Ssrcs>& CanonicalSsrcs() { return mSsrcs; }
+  Canonical<Ssrcs>& CanonicalVideoRtxSsrcs() { return mVideoRtxSsrcs; }
+  Canonical<RtpExtList>& CanonicalLocalRtpExtensions() {
+    return mLocalRtpExtensions;
+  }
+
+  Canonical<Maybe<AudioCodecConfig>>& CanonicalAudioCodec() {
+    return mAudioCodec;
+  }
+
+  Canonical<Maybe<VideoCodecConfig>>& CanonicalVideoCodec() {
+    return mVideoCodec;
+  }
+  Canonical<Maybe<RtpRtcpConfig>>& CanonicalVideoRtpRtcpConfig() {
+    return mVideoRtpRtcpConfig;
+  }
+  Canonical<webrtc::VideoCodecMode>& CanonicalVideoCodecMode() {
+    return mVideoCodecMode;
+  }
+  Canonical<std::string>& CanonicalCname() { return mCname; }
+  Canonical<bool>& CanonicalTransmitting() override { return mTransmitting; }
+
+  Canonical<RefPtr<FrameTransformerProxy>>& CanonicalFrameTransformerProxy() {
+    return mFrameTransformerProxy;
+  }
+
+  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 mSenderTrackSetByAddTrack = 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;
+  RefPtr<RTCRtpScriptTransform> mTransform;
+  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;
+  // 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;
+
+  // Limits logging of codec information
+  bool mHaveLoggedUlpfecInfo = false;
+  bool mHaveLoggedOtherFec = false;
+  bool mHaveLoggedVideoPreferredCodec = false;
+  bool mHaveLoggedAudioPreferredCodec = 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;
+  Canonical<RefPtr<FrameTransformerProxy>> mFrameTransformerProxy;
+};
+
+}  // 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..b219619f87
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCRtpTransceiver.cpp
@@ -0,0 +1,1167 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. 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 <stdint.h>
+
+#include <algorithm>
+#include <string>
+#include <vector>
+#include <utility>
+#include <set>
+#include <string>
+#include <tuple>
+
+#include "api/video_codecs/video_codec.h"
+
+#include "nsCOMPtr.h"
+#include "nsContentUtils.h"
+#include "nsCycleCollectionParticipant.h"
+#include "nsDebug.h"
+#include "nsISerialEventTarget.h"
+#include "nsISupports.h"
+#include "nsProxyRelease.h"
+#include "nsStringFwd.h"
+#include "nsString.h"
+#include "nsTArray.h"
+#include "nsThreadUtils.h"
+#include "nsWrapperCache.h"
+#include "PrincipalHandle.h"
+#include "ErrorList.h"
+#include "MainThreadUtils.h"
+#include "MediaEventSource.h"
+#include "mozilla/AbstractThread.h"
+#include "mozilla/Assertions.h"
+#include "mozilla/ErrorResult.h"
+#include "mozilla/fallible.h"
+#include "mozilla/Maybe.h"
+#include "mozilla/mozalloc_oom.h"
+#include "mozilla/MozPromise.h"
+#include "mozilla/Preferences.h"
+#include "mozilla/UniquePtr.h"
+#include "mozilla/StateMirroring.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/dom/Nullable.h"
+#include "mozilla/dom/RTCStatsReportBinding.h"
+#include "mozilla/dom/RTCRtpReceiverBinding.h"
+#include "mozilla/dom/RTCRtpSenderBinding.h"
+#include "mozilla/dom/RTCRtpTransceiverBinding.h"
+#include "mozilla/dom/Promise.h"
+#include "utils/PerformanceRecorder.h"
+#include "systemservices/MediaUtils.h"
+#include "MediaTrackGraph.h"
+#include "js/RootingAPI.h"
+#include "libwebrtcglue/AudioConduit.h"
+#include "libwebrtcglue/VideoConduit.h"
+#include "transportbridge/MediaPipeline.h"
+#include "jsep/JsepTrack.h"
+#include "sdp/SdpHelper.h"
+#include "transport/logging.h"
+#include "RemoteTrackSource.h"
+#include "libwebrtcglue/RtpRtcpConfig.h"
+#include "MediaTransportHandler.h"
+#include "RTCDtlsTransport.h"
+#include "RTCRtpReceiver.h"
+#include "RTCRtpSender.h"
+#include "RTCDTMFSender.h"
+#include "PeerConnectionImpl.h"
+#include "RTCStatsIdGenerator.h"
+#include "libwebrtcglue/WebrtcCallWrapper.h"
+#include "libwebrtcglue/FrameTransformerProxy.h"
+#include "jsep/JsepCodecDescription.h"
+#include "jsep/JsepSession.h"
+#include "jsep/JsepTrackEncoding.h"
+#include "libwebrtcglue/CodecConfig.h"
+#include "libwebrtcglue/MediaConduitControl.h"
+#include "libwebrtcglue/MediaConduitInterface.h"
+#include "RTCStatsReport.h"
+#include "sdp/SdpAttribute.h"
+#include "sdp/SdpEnum.h"
+#include "sdp/SdpMediaSection.h"
+#include "transport/transportlayer.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
+  Canonical<bool>& CanonicalReceiving() override {
+    return mReceiver->CanonicalReceiving();
+  }
+  Canonical<bool>& CanonicalTransmitting() override {
+    return mSender->CanonicalTransmitting();
+  }
+  Canonical<Ssrcs>& CanonicalLocalSsrcs() override {
+    return mSender->CanonicalSsrcs();
+  }
+  Canonical<std::string>& CanonicalLocalCname() override {
+    return mSender->CanonicalCname();
+  }
+  Canonical<std::string>& CanonicalMid() override {
+    return mTransceiver->CanonicalMid();
+  }
+  Canonical<Ssrc>& CanonicalRemoteSsrc() override {
+    return mReceiver->CanonicalSsrc();
+  }
+  Canonical<std::string>& CanonicalSyncGroup() override {
+    return mTransceiver->CanonicalSyncGroup();
+  }
+  Canonical<RtpExtList>& CanonicalLocalRecvRtpExtensions() override {
+    return mReceiver->CanonicalLocalRtpExtensions();
+  }
+  Canonical<RtpExtList>& CanonicalLocalSendRtpExtensions() override {
+    return mSender->CanonicalLocalRtpExtensions();
+  }
+
+  // AudioConduitControlInterface
+  Canonical<Maybe<AudioCodecConfig>>& CanonicalAudioSendCodec() override {
+    return mSender->CanonicalAudioCodec();
+  }
+  Canonical<std::vector<AudioCodecConfig>>& CanonicalAudioRecvCodecs()
+      override {
+    return mReceiver->CanonicalAudioCodecs();
+  }
+  MediaEventSource<DtmfEvent>& OnDtmfEvent() override {
+    return mSender->GetDtmf()->OnDtmfEvent();
+  }
+
+  // VideoConduitControlInterface
+  Canonical<Ssrcs>& CanonicalLocalVideoRtxSsrcs() override {
+    return mSender->CanonicalVideoRtxSsrcs();
+  }
+  Canonical<Ssrc>& CanonicalRemoteVideoRtxSsrc() override {
+    return mReceiver->CanonicalVideoRtxSsrc();
+  }
+  Canonical<Maybe<VideoCodecConfig>>& CanonicalVideoSendCodec() override {
+    return mSender->CanonicalVideoCodec();
+  }
+  Canonical<Maybe<RtpRtcpConfig>>& CanonicalVideoSendRtpRtcpConfig() override {
+    return mSender->CanonicalVideoRtpRtcpConfig();
+  }
+  Canonical<std::vector<VideoCodecConfig>>& CanonicalVideoRecvCodecs()
+      override {
+    return mReceiver->CanonicalVideoCodecs();
+  }
+  Canonical<Maybe<RtpRtcpConfig>>& CanonicalVideoRecvRtpRtcpConfig() override {
+    return mReceiver->CanonicalVideoRtpRtcpConfig();
+  }
+  Canonical<webrtc::VideoCodecMode>& CanonicalVideoCodecMode() override {
+    return mSender->CanonicalVideoCodecMode();
+  }
+  Canonical<RefPtr<FrameTransformerProxy>>& CanonicalFrameTransformerProxySend()
+      override {
+    return mSender->CanonicalFrameTransformerProxy();
+  }
+
+  Canonical<RefPtr<FrameTransformerProxy>>& CanonicalFrameTransformerProxyRecv()
+      override {
+    return mReceiver->CanonicalFrameTransformerProxy();
+  }
+};
+}  // namespace
+
+MOZ_MTLOG_MODULE("RTCRtpTransceiver")
+
+NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE_CLASS(RTCRtpTransceiver)
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(RTCRtpTransceiver)
+  tmp->Unlink();
+  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:
+    case dom::RTCRtpTransceiverDirection::Stopped:
+      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);
+  mConduit->AsVideoSessionConduit().apply(
+      [&](auto aConduit) { aConduit->InitControl(&control); });
+  mConduit->AsAudioSessionConduit().apply(
+      [&](auto 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->UpdateStateNoEvent(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;
+}
+
+void RTCRtpTransceiver::Unlink() {
+  if (mHandlingUnlink) {
+    BreakCycles();
+    mHandlingUnlink = false;
+  } else if (mPc) {
+    mPc->Close();
+    mPc->BreakCycles();
+  }
+}
+
+// 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 sRD, so we need to check that
+  if (!mStopped && mJsepTransceiver.IsStopped()) {
+    MOZ_MTLOG(ML_DEBUG, mPc->GetHandle()
+                            << "[" << mMid.Ref() << "]: " << __FUNCTION__
+                            << " JSEP transceiver is stopped");
+    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 = !mStopped && mJsepTransceiver.mTransport.mComponents;
+}
+
+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 (mStopping || 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 transceiver.[[Stopping]] is true, throw an InvalidStateError.
+  if (mStopping) {
+    aRv.ThrowInvalidStateError("Transceiver is stopping/stopped!");
+    return;
+  }
+
+  // If newDirection is equal to transceiver.[[Direction]], abort these steps.
+  if (aDirection == mDirection) {
+    return;
+  }
+
+  // If newDirection is equal to "stopped", throw a TypeError.
+  if (aDirection == RTCRtpTransceiverDirection::Stopped) {
+    aRv.ThrowTypeError("Cannot use \"stopped\" in setDirection!");
+    return;
+  }
+
+  // Set transceiver.[[Direction]] to newDirection.
+  SetDirectionInternal(aDirection);
+
+  // Update the negotiation-needed flag for connection.
+  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() || Stopping()) {
+    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 (SdpHelper::GetPtAsInt(aCodec.mREDRTXPayloadType, &pt)) {
+      (*aConfig)->mREDRTXPayloadType = 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;
+  }
+
+  if (mStopping) {
+    return;
+  }
+
+  StopTransceiving();
+  mPc->UpdateNegotiationNeeded();
+}
+
+void RTCRtpTransceiver::StopTransceiving() {
+  if (mStopping) {
+    MOZ_ASSERT(false);
+    return;
+  }
+  mStopping = true;
+  // This is the "Stop sending and receiving" algorithm from webrtc-pc
+  mSender->Stop();
+  mReceiver->Stop();
+  mDirection = RTCRtpTransceiverDirection::Inactive;
+}
+
+void RTCRtpTransceiver::StopImpl() {
+  // This is the "stop the RTCRtpTransceiver" algorithm from webrtc-pc
+  if (!mStopping) {
+    StopTransceiving();
+  }
+
+  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();
+
+  mHasTransport = false;
+
+  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..afa6142d35
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCRtpTransceiver.h
@@ -0,0 +1,255 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. 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();
+  void Unlink();
+
+  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 {
+    if (mStopping) {
+      return RTCRtpTransceiverDirection::Stopped;
+    }
+    return mDirection;
+  }
+  void SetDirection(RTCRtpTransceiverDirection aDirection, ErrorResult& aRv);
+  Nullable<RTCRtpTransceiverDirection> GetCurrentDirection() {
+    if (mStopped) {
+      return RTCRtpTransceiverDirection::Stopped;
+    }
+    return mCurrentDirection;
+  }
+  void Stop(ErrorResult& aRv);
+  void SetDirectionInternal(RTCRtpTransceiverDirection aDirection);
+  bool HasBeenUsedToSend() const { return mHasBeenUsedToSend; }
+
+  bool CanSendDTMF() const;
+  bool Stopped() const { return mStopped; }
+  bool Stopping() const { return mStopping; }
+  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);
+
+  Canonical<std::string>& CanonicalMid() { return mMid; }
+  Canonical<std::string>& CanonicalSyncGroup() { return mSyncGroup; }
+
+ private:
+  virtual ~RTCRtpTransceiver();
+  void InitAudio();
+  void InitVideo(const TrackingId& aRecvTrackingId);
+  void InitConduitControl();
+  void StopImpl();
+  void StopTransceiving();
+
+  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 mStopping = 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..1224ff99dc
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCStatsIdGenerator.cpp
@@ -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/. */
+/* -*- 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 "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/RTCTransformEventRunnable.cpp b/dom/media/webrtc/jsapi/RTCTransformEventRunnable.cpp
new file mode 100644
index 0000000000..6f41baf80f
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCTransformEventRunnable.cpp
@@ -0,0 +1,80 @@
+/* -*- Mode: C++; tab-width: 2; 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 https://mozilla.org/MPL/2.0/. */
+
+#include "RTCTransformEventRunnable.h"
+
+#include "nsIGlobalObject.h"
+#include "ErrorList.h"
+#include "nsError.h"
+#include "nsDebug.h"
+#include "nsLiteralString.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/AlreadyAddRefed.h"
+// This needs to come before RTCTransformEvent.h, since webidl codegen doesn't
+// include-what-you-use or forward declare.
+#include "mozilla/dom/RTCRtpScriptTransformer.h"
+#include "mozilla/dom/RTCTransformEvent.h"
+#include "mozilla/dom/RTCTransformEventBinding.h"
+#include "mozilla/dom/EventWithOptionsRunnable.h"
+#include "mozilla/dom/Event.h"
+#include "mozilla/dom/EventTarget.h"
+#include "mozilla/dom/RootedDictionary.h"
+#include "js/RootingAPI.h"
+#include "js/Value.h"
+#include "libwebrtcglue/FrameTransformerProxy.h"
+
+namespace mozilla::dom {
+
+RTCTransformEventRunnable::RTCTransformEventRunnable(
+    Worker& aWorker, FrameTransformerProxy* aProxy)
+    : EventWithOptionsRunnable(aWorker, "RTCTransformEventRunnable"),
+      mProxy(aProxy) {}
+
+RTCTransformEventRunnable::~RTCTransformEventRunnable() = default;
+
+already_AddRefed<Event> RTCTransformEventRunnable::BuildEvent(
+    JSContext* aCx, nsIGlobalObject* aGlobal, EventTarget* aTarget,
+    JS::Handle<JS::Value> aTransformerOptions) {
+  // Let transformerOptions be the result of
+  // StructuredDeserialize(serializedOptions, the current Realm).
+
+  // NOTE: We do not do this streams stuff. Spec will likely change here.
+  // The gist here is that we hook [[readable]] and [[writable]] up to the frame
+  // source/sink, which in out case is FrameTransformerProxy.
+  // Let readable be the result of StructuredDeserialize(serializedReadable, the
+  // current Realm). Let writable be the result of
+  // StructuredDeserialize(serializedWritable, the current Realm).
+
+  // Let transformer be a new RTCRtpScriptTransformer.
+
+  // Set transformer.[[options]] to transformerOptions.
+
+  // Set transformer.[[readable]] to readable.
+
+  // Set transformer.[[writable]] to writable.
+  RefPtr<RTCRtpScriptTransformer> transformer =
+      new RTCRtpScriptTransformer(aGlobal);
+  nsresult nrv =
+      transformer->Init(aCx, aTransformerOptions, mWorkerPrivate, mProxy);
+  if (NS_WARN_IF(NS_FAILED(nrv))) {
+    // TODO: Error handling. Currently unspecified.
+    return nullptr;
+  }
+
+  // Fire an event named rtctransform using RTCTransformEvent with transformer
+  // set to transformer on worker’s global scope.
+  RootedDictionary<RTCTransformEventInit> init(aCx);
+  init.mBubbles = false;
+  init.mCancelable = false;
+  init.mTransformer = transformer;
+
+  RefPtr<RTCTransformEvent> event =
+      RTCTransformEvent::Constructor(aTarget, u"rtctransform"_ns, init);
+  event->SetTrusted(true);
+  return event.forget();
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webrtc/jsapi/RTCTransformEventRunnable.h b/dom/media/webrtc/jsapi/RTCTransformEventRunnable.h
new file mode 100644
index 0000000000..763d7c9237
--- /dev/null
+++ b/dom/media/webrtc/jsapi/RTCTransformEventRunnable.h
@@ -0,0 +1,38 @@
+/* -*- Mode: C++; tab-width: 2; 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 https://mozilla.org/MPL/2.0/. */
+
+#ifndef MOZILLA_DOM_MEDIA_WEBRTC_JSAPI_RTCTRANSFORMEVENTRUNNABLE_H_
+#define MOZILLA_DOM_MEDIA_WEBRTC_JSAPI_RTCTRANSFORMEVENTRUNNABLE_H_
+
+#include "mozilla/dom/EventWithOptionsRunnable.h"
+
+namespace mozilla {
+
+class FrameTransformerProxy;
+
+namespace dom {
+
+// Cargo-culted from MesssageEventRunnable.
+// TODO: Maybe could subclass WorkerRunnable instead? Comments on
+// WorkerDebuggeeRunnable indicate that firing an event at JS means we need that
+// class.
+class RTCTransformEventRunnable final : public EventWithOptionsRunnable {
+ public:
+  RTCTransformEventRunnable(Worker& aWorker, FrameTransformerProxy* aProxy);
+
+  already_AddRefed<Event> BuildEvent(
+      JSContext* aCx, nsIGlobalObject* aGlobal, EventTarget* aTarget,
+      JS::Handle<JS::Value> aTransformerOptions) override;
+
+ private:
+  virtual ~RTCTransformEventRunnable();
+  RefPtr<FrameTransformerProxy> mProxy;
+};
+
+}  // namespace dom
+}  // namespace mozilla
+
+#endif  // MOZILLA_DOM_MEDIA_WEBRTC_JSAPI_RTCTRANSFORMEVENTRUNNABLE_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..11247f2cb9
--- /dev/null
+++ b/dom/media/webrtc/jsapi/WebrtcGlobalChild.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 _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;
+
+  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..4e4bf9ab93
--- /dev/null
+++ b/dom/media/webrtc/jsapi/WebrtcGlobalInformation.cpp
@@ -0,0 +1,803 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. 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 "libwebrtcglue/VideoConduit.h"
+#include "mozilla/Assertions.h"
+#include "mozilla/StaticPtr.h"
+#include "mozilla/dom/PWebrtcGlobal.h"
+#include "mozilla/dom/PWebrtcGlobalChild.h"
+#include "WebrtcGlobalChild.h"
+#include "WebrtcGlobalParent.h"
+
+#include <algorithm>
+#include <vector>
+
+#include "mozilla/dom/WebrtcGlobalInformationBinding.h"
+#include "mozilla/dom/RTCStatsReportBinding.h"  // for RTCStatsReportInternal
+#include "mozilla/dom/ContentChild.h"
+
+#include "ErrorList.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/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;
+}
+
+void WebrtcGlobalInformation::GetMediaContext(
+    const GlobalObject& aGlobal, WebrtcGlobalMediaContext& aContext) {
+  aContext.mHasH264Hardware = WebrtcVideoConduit::HasH264Hardware();
+}
+
+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 bool sLastAECDebug = false;
+static Maybe<nsCString> sAecDebugLogDir;
+
+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();
+}
+
+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..0507d5181f
--- /dev/null
+++ b/dom/media/webrtc/jsapi/WebrtcGlobalInformation.h
@@ -0,0 +1,101 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. 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 "mozilla/Attributes.h"
+#include "mozilla/dom/WebrtcGlobalInformationBinding.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 GetMediaContext(const GlobalObject& aGlobal,
+                              WebrtcGlobalMediaContext& aContext);
+
+  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 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..344b11e23d
--- /dev/null
+++ b/dom/media/webrtc/jsapi/WebrtcGlobalParent.h
@@ -0,0 +1,51 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. 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 "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..d17af937ad
--- /dev/null
+++ b/dom/media/webrtc/jsapi/WebrtcGlobalStatsHistory.cpp
@@ -0,0 +1,280 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. 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 "domstubs.h"
+#include "mozilla/LinkedList.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..c596629247
--- /dev/null
+++ b/dom/media/webrtc/jsapi/WebrtcGlobalStatsHistory.h
@@ -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/. */
+
+#pragma once
+
+#include "domstubs.h"
+#include "mozilla/LinkedList.h"
+#include "mozilla/UniquePtr.h"
+#include "mozilla/dom/BindingDeclarations.h"
+#include "mozilla/dom/RTCStatsReportBinding.h"
+#include "nsDOMNavigationTiming.h"
+#include "nsHashKeys.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..78a6241cd6
--- /dev/null
+++ b/dom/media/webrtc/jsapi/moz.build
@@ -0,0 +1,62 @@
+# -*- 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",
+    "RTCEncodedAudioFrame.cpp",
+    "RTCEncodedFrameBase.cpp",
+    "RTCEncodedVideoFrame.cpp",
+    "RTCRtpReceiver.cpp",
+    "RTCRtpScriptTransform.cpp",
+    "RTCRtpScriptTransformer.cpp",
+    "RTCRtpSender.cpp",
+    "RTCRtpTransceiver.cpp",
+    "RTCSctpTransport.cpp",
+    "RTCStatsIdGenerator.cpp",
+    "RTCStatsReport.cpp",
+    "RTCTransformEventRunnable.cpp",
+    "WebrtcGlobalInformation.cpp",
+    "WebrtcGlobalStatsHistory.cpp",
+]
+
+EXPORTS.mozilla.dom += [
+    "RTCDtlsTransport.h",
+    "RTCDTMFSender.h",
+    "RTCEncodedAudioFrame.h",
+    "RTCEncodedFrameBase.h",
+    "RTCEncodedVideoFrame.h",
+    "RTCRtpReceiver.h",
+    "RTCRtpScriptTransform.h",
+    "RTCRtpScriptTransformer.h",
+    "RTCRtpSender.h",
+    "RTCRtpTransceiver.h",
+    "RTCSctpTransport.h",
+    "RTCStatsReport.h",
+]
+
+FINAL_LIBRARY = "xul"