diff options
Diffstat (limited to 'dom/media/webrtc/jsapi/PeerConnectionImpl.cpp')
| -rw-r--r-- | dom/media/webrtc/jsapi/PeerConnectionImpl.cpp | 4640 |
1 files changed, 4640 insertions, 0 deletions
diff --git a/dom/media/webrtc/jsapi/PeerConnectionImpl.cpp b/dom/media/webrtc/jsapi/PeerConnectionImpl.cpp new file mode 100644 index 0000000000..567b682b2a --- /dev/null +++ b/dom/media/webrtc/jsapi/PeerConnectionImpl.cpp @@ -0,0 +1,4640 @@ +/* This Source Code Form is subject to the terms of the Mozilla Public + * License, v. 2.0. If a copy of the MPL was not distributed with this file, + * You can obtain one at http://mozilla.org/MPL/2.0/. */ + +#include <cstdlib> +#include <cerrno> +#include <deque> +#include <set> +#include <sstream> +#include <vector> + +#include "common/browser_logging/CSFLog.h" +#include "base/histogram.h" +#include "common/time_profiling/timecard.h" + +#include "jsapi.h" +#include "nspr.h" +#include "nss.h" +#include "pk11pub.h" + +#include "nsNetCID.h" +#include "nsIIDNService.h" +#include "nsILoadContext.h" +#include "nsEffectiveTLDService.h" +#include "nsServiceManagerUtils.h" +#include "nsThreadUtils.h" +#include "nsIPrefService.h" +#include "nsIPrefBranch.h" +#include "nsProxyRelease.h" +#include "prtime.h" + +#include "libwebrtcglue/AudioConduit.h" +#include "libwebrtcglue/VideoConduit.h" +#include "libwebrtcglue/WebrtcCallWrapper.h" +#include "MediaTrackGraph.h" +#include "transport/runnable_utils.h" +#include "IPeerConnection.h" +#include "PeerConnectionCtx.h" +#include "PeerConnectionImpl.h" +#include "RemoteTrackSource.h" +#include "nsDOMDataChannelDeclarations.h" +#include "transport/dtlsidentity.h" +#include "sdp/SdpAttribute.h" + +#include "jsep/JsepTrack.h" +#include "jsep/JsepSession.h" +#include "jsep/JsepSessionImpl.h" + +#include "transportbridge/MediaPipeline.h" +#include "jsapi/RTCRtpReceiver.h" +#include "jsapi/RTCRtpSender.h" + +#include "mozilla/IntegerPrintfMacros.h" +#include "mozilla/Sprintf.h" + +#ifdef XP_WIN +// We need to undef the MS macro for Document::CreateEvent +# ifdef CreateEvent +# undef CreateEvent +# endif +#endif // XP_WIN + +#include "mozilla/dom/Document.h" +#include "nsGlobalWindow.h" +#include "nsDOMDataChannel.h" +#include "mozilla/dom/Location.h" +#include "mozilla/dom/Promise.h" +#include "mozilla/NullPrincipal.h" +#include "mozilla/TimeStamp.h" +#include "mozilla/Telemetry.h" +#include "mozilla/Preferences.h" +#include "mozilla/PublicSSL.h" +#include "nsXULAppAPI.h" +#include "nsContentUtils.h" +#include "nsDOMJSUtils.h" +#include "nsPrintfCString.h" +#include "nsURLHelper.h" +#include "nsNetUtil.h" +#include "js/ArrayBuffer.h" // JS::NewArrayBufferWithContents +#include "js/GCAnnotations.h" // JS_HAZ_ROOTED +#include "js/RootingAPI.h" // JS::{{,Mutable}Handle,Rooted} +#include "mozilla/PeerIdentity.h" +#include "mozilla/dom/RTCCertificate.h" +#include "mozilla/dom/RTCSctpTransportBinding.h" // RTCSctpTransportState +#include "mozilla/dom/RTCDtlsTransportBinding.h" // RTCDtlsTransportState +#include "mozilla/dom/RTCRtpReceiverBinding.h" +#include "mozilla/dom/RTCRtpSenderBinding.h" +#include "mozilla/dom/RTCStatsReportBinding.h" +#include "mozilla/dom/RTCPeerConnectionBinding.h" +#include "mozilla/dom/PeerConnectionImplBinding.h" +#include "mozilla/dom/RTCDataChannelBinding.h" +#include "mozilla/dom/PluginCrashedEvent.h" +#include "MediaStreamTrack.h" +#include "AudioStreamTrack.h" +#include "VideoStreamTrack.h" +#include "nsIScriptGlobalObject.h" +#include "DOMMediaStream.h" +#include "WebrtcGlobalInformation.h" +#include "mozilla/dom/Event.h" +#include "mozilla/EventDispatcher.h" +#include "mozilla/net/DataChannelProtocol.h" +#include "MediaManager.h" + +#include "transport/nr_socket_proxy_config.h" +#include "RTCSctpTransport.h" +#include "RTCDtlsTransport.h" +#include "jsep/JsepTransport.h" + +#include "nsILoadInfo.h" +#include "nsIPrincipal.h" +#include "mozilla/LoadInfo.h" +#include "nsIProxiedChannel.h" + +#include "mozilla/dom/BrowserChild.h" +#include "mozilla/net/WebrtcProxyConfig.h" + +#ifdef XP_WIN +// We need to undef the MS macro again in case the windows include file +// got imported after we included mozilla/dom/Document.h +# ifdef CreateEvent +# undef CreateEvent +# endif +#endif // XP_WIN + +#include "MediaSegment.h" + +#ifdef USE_FAKE_PCOBSERVER +# include "FakePCObserver.h" +#else +# include "mozilla/dom/PeerConnectionObserverBinding.h" +#endif +#include "mozilla/dom/PeerConnectionObserverEnumsBinding.h" + +#define ICE_PARSING \ + "In RTCConfiguration passed to RTCPeerConnection constructor" + +using namespace mozilla; +using namespace mozilla::dom; + +typedef PCObserverString ObString; + +static const char* pciLogTag = "PeerConnectionImpl"; +#ifdef LOGTAG +# undef LOGTAG +#endif +#define LOGTAG pciLogTag + +static mozilla::LazyLogModule logModuleInfo("signaling"); + +// Getting exceptions back down from PCObserver is generally not harmful. +namespace { +// This is a terrible hack. The problem is that SuppressException is not +// inline, and we link this file without libxul in some cases (e.g. for our test +// setup). So we can't use ErrorResult or IgnoredErrorResult because those call +// SuppressException... And we can't use FastErrorResult because we can't +// include BindingUtils.h, because our linking is completely broken. Use +// BaseErrorResult directly. Please do not let me see _anyone_ doing this +// without really careful review from someone who knows what they are doing. +class JSErrorResult : public binding_danger::TErrorResult< + binding_danger::JustAssertCleanupPolicy> { + public: + ~JSErrorResult() { SuppressException(); } +} JS_HAZ_ROOTED; + +// The WrapRunnable() macros copy passed-in args and passes them to the function +// later on the other thread. ErrorResult cannot be passed like this because it +// disallows copy-semantics. +// +// This WrappableJSErrorResult hack solves this by not actually copying the +// ErrorResult, but creating a new one instead, which works because we don't +// care about the result. +// +// Since this is for JS-calls, these can only be dispatched to the main thread. + +class WrappableJSErrorResult { + public: + WrappableJSErrorResult() : mRv(MakeUnique<JSErrorResult>()), isCopy(false) {} + WrappableJSErrorResult(const WrappableJSErrorResult& other) + : mRv(MakeUnique<JSErrorResult>()), isCopy(true) {} + ~WrappableJSErrorResult() { + if (isCopy) { + MOZ_ASSERT(NS_IsMainThread()); + } + } + operator ErrorResult&() { return *mRv; } + + private: + mozilla::UniquePtr<JSErrorResult> mRv; + bool isCopy; +} JS_HAZ_ROOTED; + +} // namespace + +static nsresult InitNSSInContent() { + NS_ENSURE_TRUE(NS_IsMainThread(), NS_ERROR_NOT_SAME_THREAD); + + if (!XRE_IsContentProcess()) { + MOZ_ASSERT_UNREACHABLE("Must be called in content process"); + return NS_ERROR_FAILURE; + } + + static bool nssStarted = false; + if (nssStarted) { + return NS_OK; + } + + if (NSS_NoDB_Init(nullptr) != SECSuccess) { + CSFLogError(LOGTAG, "NSS_NoDB_Init failed."); + return NS_ERROR_FAILURE; + } + + if (NS_FAILED(mozilla::psm::InitializeCipherSuite())) { + CSFLogError(LOGTAG, "Fail to set up nss cipher suite."); + return NS_ERROR_FAILURE; + } + + mozilla::psm::DisableMD5(); + + nssStarted = true; + + return NS_OK; +} + +namespace mozilla { +class DataChannel; +} + +namespace mozilla { + +void PeerConnectionAutoTimer::RegisterConnection() { mRefCnt++; } + +void PeerConnectionAutoTimer::UnregisterConnection(bool aContainedAV) { + MOZ_ASSERT(mRefCnt); + mRefCnt--; + mUsedAV |= aContainedAV; + if (mRefCnt == 0) { + if (mUsedAV) { + Telemetry::Accumulate( + Telemetry::WEBRTC_AV_CALL_DURATION, + static_cast<uint32_t>((TimeStamp::Now() - mStart).ToSeconds())); + } + Telemetry::Accumulate( + Telemetry::WEBRTC_CALL_DURATION, + static_cast<uint32_t>((TimeStamp::Now() - mStart).ToSeconds())); + } +} + +bool PeerConnectionAutoTimer::IsStopped() { return mRefCnt == 0; } + +NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE_CLASS(PeerConnectionImpl) +NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(PeerConnectionImpl) + tmp->Close(); + tmp->BreakCycles(); + NS_IMPL_CYCLE_COLLECTION_UNLINK(mPCObserver, mWindow, mCertificate, + mSTSThread, mReceiveStreams, mOperations, + mSctpTransport, mKungFuDeathGrip) + NS_IMPL_CYCLE_COLLECTION_UNLINK_PRESERVED_WRAPPER +NS_IMPL_CYCLE_COLLECTION_UNLINK_END +NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN(PeerConnectionImpl) + NS_IMPL_CYCLE_COLLECTION_TRAVERSE( + mPCObserver, mWindow, mCertificate, mSTSThread, mReceiveStreams, + mOperations, mTransceivers, mSctpTransport, mKungFuDeathGrip) +NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END + +NS_IMPL_CYCLE_COLLECTING_ADDREF(PeerConnectionImpl) +NS_IMPL_CYCLE_COLLECTING_RELEASE(PeerConnectionImpl) +NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(PeerConnectionImpl) + NS_WRAPPERCACHE_INTERFACE_MAP_ENTRY + NS_INTERFACE_MAP_ENTRY(nsISupports) +NS_INTERFACE_MAP_END + +already_AddRefed<PeerConnectionImpl> PeerConnectionImpl::Constructor( + const dom::GlobalObject& aGlobal) { + RefPtr<PeerConnectionImpl> pc = new PeerConnectionImpl(&aGlobal); + + CSFLogDebug(LOGTAG, "Created PeerConnection: %p", pc.get()); + + return pc.forget(); +} + +JSObject* PeerConnectionImpl::WrapObject(JSContext* aCx, + JS::Handle<JSObject*> aGivenProto) { + return PeerConnectionImpl_Binding::Wrap(aCx, this, aGivenProto); +} + +nsPIDOMWindowInner* PeerConnectionImpl::GetParentObject() const { + return mWindow; +} + +bool PCUuidGenerator::Generate(std::string* idp) { + nsresult rv; + + if (!mGenerator) { + mGenerator = do_GetService("@mozilla.org/uuid-generator;1", &rv); + if (NS_FAILED(rv)) { + return false; + } + if (!mGenerator) { + return false; + } + } + + nsID id; + rv = mGenerator->GenerateUUIDInPlace(&id); + if (NS_FAILED(rv)) { + return false; + } + char buffer[NSID_LENGTH]; + id.ToProvidedString(buffer); + idp->assign(buffer); + + return true; +} + +bool IsPrivateBrowsing(nsPIDOMWindowInner* aWindow) { + if (!aWindow) { + return false; + } + + Document* doc = aWindow->GetExtantDoc(); + if (!doc) { + return false; + } + + nsILoadContext* loadContext = doc->GetLoadContext(); + return loadContext && loadContext->UsePrivateBrowsing(); +} + +PeerConnectionImpl::PeerConnectionImpl(const GlobalObject* aGlobal) + : mTimeCard(MOZ_LOG_TEST(logModuleInfo, LogLevel::Error) ? create_timecard() + : nullptr), + mJsConfiguration(), + mSignalingState(RTCSignalingState::Stable), + mIceConnectionState(RTCIceConnectionState::New), + mIceGatheringState(RTCIceGatheringState::New), + mConnectionState(RTCPeerConnectionState::New), + mWindow(do_QueryInterface(aGlobal ? aGlobal->GetAsSupports() : nullptr)), + mCertificate(nullptr), + mSTSThread(nullptr), + mForceIceTcp(false), + mTransportHandler(nullptr), + mUuidGen(MakeUnique<PCUuidGenerator>()), + mIceRestartCount(0), + mIceRollbackCount(0), + mHaveConfiguredCodecs(false), + mTrickle(true) // TODO(ekr@rtfm.com): Use pref + , + mPrivateWindow(false), + mActiveOnWindow(false), + mTimestampMaker(dom::RTCStatsTimestampMaker::Create(mWindow)), + mIdGenerator(new RTCStatsIdGenerator()), + listenPort(0), + connectPort(0), + connectStr(nullptr) { + MOZ_ASSERT(NS_IsMainThread()); + MOZ_ASSERT_IF(aGlobal, mWindow); + mKungFuDeathGrip = this; + if (aGlobal) { + if (IsPrivateBrowsing(mWindow)) { + mPrivateWindow = true; + mDisableLongTermStats = true; + } + mWindow->AddPeerConnection(); + mActiveOnWindow = true; + + if (mWindow->GetDocumentURI()) { + mWindow->GetDocumentURI()->GetAsciiHost(mHostname); + nsresult rv; + nsCOMPtr<nsIEffectiveTLDService> eTLDService( + do_GetService(NS_EFFECTIVETLDSERVICE_CONTRACTID, &rv)); + if (eTLDService) { + Unused << eTLDService->GetBaseDomain(mWindow->GetDocumentURI(), 0, + mEffectiveTLDPlus1); + } + + mRtxIsAllowed = !HostnameInPref( + "media.peerconnection.video.use_rtx.blocklist", mHostname); + } + } + + if (!mUuidGen->Generate(&mHandle)) { + MOZ_CRASH(); + } + + CSFLogInfo(LOGTAG, "%s: PeerConnectionImpl constructor for %s", __FUNCTION__, + mHandle.c_str()); + STAMP_TIMECARD(mTimeCard, "Constructor Completed"); + mForceIceTcp = + Preferences::GetBool("media.peerconnection.ice.force_ice_tcp", false); + memset(mMaxReceiving, 0, sizeof(mMaxReceiving)); + memset(mMaxSending, 0, sizeof(mMaxSending)); + mJsConfiguration.mCertificatesProvided = false; + mJsConfiguration.mPeerIdentityProvided = false; +} + +PeerConnectionImpl::~PeerConnectionImpl() { + MOZ_ASSERT(NS_IsMainThread()); + + MOZ_ASSERT(!mTransportHandler, + "PeerConnection should either be closed, or not initted in the " + "first place."); + + if (mTimeCard) { + STAMP_TIMECARD(mTimeCard, "Destructor Invoked"); + STAMP_TIMECARD(mTimeCard, mHandle.c_str()); + print_timecard(mTimeCard); + destroy_timecard(mTimeCard); + mTimeCard = nullptr; + } + + CSFLogInfo(LOGTAG, "%s: PeerConnectionImpl destructor invoked for %s", + __FUNCTION__, mHandle.c_str()); +} + +nsresult PeerConnectionImpl::Initialize(PeerConnectionObserver& aObserver, + nsGlobalWindowInner* aWindow) { + nsresult res; + + MOZ_ASSERT(NS_IsMainThread()); + + mPCObserver = &aObserver; + + // Find the STS thread + + mSTSThread = do_GetService(NS_SOCKETTRANSPORTSERVICE_CONTRACTID, &res); + MOZ_ASSERT(mSTSThread); + + // We do callback handling on STS instead of main to avoid media jank. + // Someday, we may have a dedicated thread for this. + mTransportHandler = MediaTransportHandler::Create(mSTSThread); + if (mPrivateWindow) { + mTransportHandler->EnterPrivateMode(); + } + + // Initialize NSS if we are in content process. For chrome process, NSS should + // already been initialized. + if (XRE_IsParentProcess()) { + // This code interferes with the C++ unit test startup code. + nsCOMPtr<nsISupports> nssDummy = do_GetService("@mozilla.org/psm;1", &res); + NS_ENSURE_SUCCESS(res, res); + } else { + NS_ENSURE_SUCCESS(res = InitNSSInContent(), res); + } + + // Currently no standalone unit tests for DataChannel, + // which is the user of mWindow + MOZ_ASSERT(aWindow); + mWindow = aWindow; + NS_ENSURE_STATE(mWindow); + + PRTime timestamp = PR_Now(); + // Ok if we truncate this, but we want it to be large enough to reliably + // contain the location on the tests we run in CI. + char temp[256]; + + nsAutoCString locationCStr; + + RefPtr<Location> location = mWindow->Location(); + nsAutoString locationAStr; + res = location->ToString(locationAStr); + NS_ENSURE_SUCCESS(res, res); + + CopyUTF16toUTF8(locationAStr, locationCStr); + + SprintfLiteral(temp, "%s %" PRIu64 " (id=%" PRIu64 " url=%s)", + mHandle.c_str(), static_cast<uint64_t>(timestamp), + static_cast<uint64_t>(mWindow ? mWindow->WindowID() : 0), + locationCStr.get() ? locationCStr.get() : "NULL"); + + mName = temp; + + STAMP_TIMECARD(mTimeCard, "Initializing PC Ctx"); + res = PeerConnectionCtx::InitializeGlobal(); + NS_ENSURE_SUCCESS(res, res); + + mTransportHandler->CreateIceCtx("PC:" + GetName()); + + mJsepSession = + MakeUnique<JsepSessionImpl>(mName, MakeUnique<PCUuidGenerator>()); + mJsepSession->SetRtxIsAllowed(mRtxIsAllowed); + + res = mJsepSession->Init(); + if (NS_FAILED(res)) { + CSFLogError(LOGTAG, "%s: Couldn't init JSEP Session, res=%u", __FUNCTION__, + static_cast<unsigned>(res)); + return res; + } + + std::vector<UniquePtr<JsepCodecDescription>> preferredCodecs; + SetupPreferredCodecs(preferredCodecs); + mJsepSession->SetDefaultCodecs(preferredCodecs); + + std::vector<RtpExtensionHeader> preferredHeaders; + SetupPreferredRtpExtensions(preferredHeaders); + + for (const auto& header : preferredHeaders) { + mJsepSession->AddRtpExtension(header.mMediaType, header.extensionname, + header.direction); + } + + if (XRE_IsContentProcess()) { + mStunAddrsRequest = + new net::StunAddrsRequestChild(new StunAddrsHandler(this)); + } + + // Initialize the media object. + mForceProxy = ShouldForceProxy(); + + // We put this here, in case we later want to set this based on a non-standard + // param in RTCConfiguration. + mAllowOldSetParameters = Preferences::GetBool( + "media.peerconnection.allow_old_setParameters", false); + + // setup the stun local addresses IPC async call + InitLocalAddrs(); + + mSignalHandler = MakeUnique<SignalHandler>(this, mTransportHandler.get()); + + PeerConnectionCtx::GetInstance()->AddPeerConnection(mHandle, this); + + return NS_OK; +} + +void PeerConnectionImpl::Initialize(PeerConnectionObserver& aObserver, + nsGlobalWindowInner& aWindow, + ErrorResult& rv) { + MOZ_ASSERT(NS_IsMainThread()); + + nsresult res = Initialize(aObserver, &aWindow); + if (NS_FAILED(res)) { + rv.Throw(res); + return; + } +} + +void PeerConnectionImpl::SetCertificate( + mozilla::dom::RTCCertificate& aCertificate) { + PC_AUTO_ENTER_API_CALL_NO_CHECK(); + MOZ_ASSERT(!mCertificate, "This can only be called once"); + mCertificate = &aCertificate; + + std::vector<uint8_t> fingerprint; + nsresult rv = + CalculateFingerprint(DtlsIdentity::DEFAULT_HASH_ALGORITHM, &fingerprint); + if (NS_FAILED(rv)) { + CSFLogError(LOGTAG, "%s: Couldn't calculate fingerprint, rv=%u", + __FUNCTION__, static_cast<unsigned>(rv)); + mCertificate = nullptr; + return; + } + rv = mJsepSession->AddDtlsFingerprint(DtlsIdentity::DEFAULT_HASH_ALGORITHM, + fingerprint); + if (NS_FAILED(rv)) { + CSFLogError(LOGTAG, "%s: Couldn't set DTLS credentials, rv=%u", + __FUNCTION__, static_cast<unsigned>(rv)); + mCertificate = nullptr; + } + + if (mUncommittedJsepSession) { + Unused << mUncommittedJsepSession->AddDtlsFingerprint( + DtlsIdentity::DEFAULT_HASH_ALGORITHM, fingerprint); + } +} + +const RefPtr<mozilla::dom::RTCCertificate>& PeerConnectionImpl::Certificate() + const { + PC_AUTO_ENTER_API_CALL_NO_CHECK(); + return mCertificate; +} + +RefPtr<DtlsIdentity> PeerConnectionImpl::Identity() const { + PC_AUTO_ENTER_API_CALL_NO_CHECK(); + MOZ_ASSERT(mCertificate); + return mCertificate->CreateDtlsIdentity(); +} + +class CompareCodecPriority { + public: + void SetPreferredCodec(int32_t preferredCodec) { + // This pref really ought to be a string, preferably something like + // "H264" or "VP8" instead of a payload type. + // Bug 1101259. + std::ostringstream os; + os << preferredCodec; + mPreferredCodec = os.str(); + } + + bool operator()(const UniquePtr<JsepCodecDescription>& lhs, + const UniquePtr<JsepCodecDescription>& rhs) const { + if (!mPreferredCodec.empty() && lhs->mDefaultPt == mPreferredCodec && + rhs->mDefaultPt != mPreferredCodec) { + return true; + } + + if (lhs->mStronglyPreferred && !rhs->mStronglyPreferred) { + return true; + } + + return false; + } + + private: + std::string mPreferredCodec; +}; + +class ConfigureCodec { + public: + explicit ConfigureCodec(nsCOMPtr<nsIPrefBranch>& branch) + : mHardwareH264Enabled(false), + mSoftwareH264Enabled(false), + mH264Enabled(false), + mVP9Enabled(true), + mVP9Preferred(false), + mH264Level(13), // minimum suggested for WebRTC spec + mH264MaxBr(0), // Unlimited + mH264MaxMbps(0), // Unlimited + mVP8MaxFs(0), + mVP8MaxFr(0), + mUseTmmbr(false), + mUseRemb(false), + mUseTransportCC(false), + mUseAudioFec(false), + mRedUlpfecEnabled(false), + mDtmfEnabled(false) { + mSoftwareH264Enabled = PeerConnectionCtx::GetInstance()->gmpHasH264(); + + if (WebrtcVideoConduit::HasH264Hardware()) { + Telemetry::Accumulate(Telemetry::WEBRTC_HAS_H264_HARDWARE, true); + branch->GetBoolPref("media.webrtc.hw.h264.enabled", + &mHardwareH264Enabled); + } + + mH264Enabled = mHardwareH264Enabled || mSoftwareH264Enabled; + Telemetry::Accumulate(Telemetry::WEBRTC_SOFTWARE_H264_ENABLED, + mSoftwareH264Enabled); + Telemetry::Accumulate(Telemetry::WEBRTC_HARDWARE_H264_ENABLED, + mHardwareH264Enabled); + Telemetry::Accumulate(Telemetry::WEBRTC_H264_ENABLED, mH264Enabled); + + branch->GetIntPref("media.navigator.video.h264.level", &mH264Level); + mH264Level &= 0xFF; + + branch->GetIntPref("media.navigator.video.h264.max_br", &mH264MaxBr); + + branch->GetIntPref("media.navigator.video.h264.max_mbps", &mH264MaxMbps); + + branch->GetBoolPref("media.peerconnection.video.vp9_enabled", &mVP9Enabled); + + branch->GetBoolPref("media.peerconnection.video.vp9_preferred", + &mVP9Preferred); + + branch->GetIntPref("media.navigator.video.max_fs", &mVP8MaxFs); + if (mVP8MaxFs <= 0) { + mVP8MaxFs = 12288; // We must specify something other than 0 + } + + branch->GetIntPref("media.navigator.video.max_fr", &mVP8MaxFr); + if (mVP8MaxFr <= 0) { + mVP8MaxFr = 60; // We must specify something other than 0 + } + + // TMMBR is enabled from a pref in about:config + branch->GetBoolPref("media.navigator.video.use_tmmbr", &mUseTmmbr); + + // REMB is enabled by default, but can be disabled from about:config + branch->GetBoolPref("media.navigator.video.use_remb", &mUseRemb); + + branch->GetBoolPref("media.navigator.video.use_transport_cc", + &mUseTransportCC); + + branch->GetBoolPref("media.navigator.audio.use_fec", &mUseAudioFec); + + branch->GetBoolPref("media.navigator.video.red_ulpfec_enabled", + &mRedUlpfecEnabled); + + // media.peerconnection.dtmf.enabled controls both sdp generation for + // DTMF support as well as DTMF exposure to DOM + branch->GetBoolPref("media.peerconnection.dtmf.enabled", &mDtmfEnabled); + } + + void operator()(UniquePtr<JsepCodecDescription>& codec) const { + switch (codec->Type()) { + case SdpMediaSection::kAudio: { + JsepAudioCodecDescription& audioCodec = + static_cast<JsepAudioCodecDescription&>(*codec); + if (audioCodec.mName == "opus") { + audioCodec.mFECEnabled = mUseAudioFec; + } else if (audioCodec.mName == "telephone-event") { + audioCodec.mEnabled = mDtmfEnabled; + } + } break; + case SdpMediaSection::kVideo: { + JsepVideoCodecDescription& videoCodec = + static_cast<JsepVideoCodecDescription&>(*codec); + + if (videoCodec.mName == "H264") { + // Override level + videoCodec.mProfileLevelId &= 0xFFFF00; + videoCodec.mProfileLevelId |= mH264Level; + + videoCodec.mConstraints.maxBr = mH264MaxBr; + + videoCodec.mConstraints.maxMbps = mH264MaxMbps; + + // Might disable it, but we set up other params anyway + videoCodec.mEnabled = mH264Enabled; + + if (videoCodec.mPacketizationMode == 0 && !mSoftwareH264Enabled) { + // We're assuming packetization mode 0 is unsupported by + // hardware. + videoCodec.mEnabled = false; + } + + if (mHardwareH264Enabled) { + videoCodec.mStronglyPreferred = true; + } + } else if (videoCodec.mName == "red") { + videoCodec.mEnabled = mRedUlpfecEnabled; + } else if (videoCodec.mName == "ulpfec") { + videoCodec.mEnabled = mRedUlpfecEnabled; + } else if (videoCodec.mName == "VP8" || videoCodec.mName == "VP9") { + if (videoCodec.mName == "VP9") { + if (!mVP9Enabled) { + videoCodec.mEnabled = false; + break; + } + if (mVP9Preferred) { + videoCodec.mStronglyPreferred = true; + } + } + videoCodec.mConstraints.maxFs = mVP8MaxFs; + videoCodec.mConstraints.maxFps = Some(mVP8MaxFr); + } + + if (mUseTmmbr) { + videoCodec.EnableTmmbr(); + } + if (mUseRemb) { + videoCodec.EnableRemb(); + } + if (mUseTransportCC) { + videoCodec.EnableTransportCC(); + } + } break; + case SdpMediaSection::kText: + case SdpMediaSection::kApplication: + case SdpMediaSection::kMessage: { + } // Nothing to configure for these. + } + } + + private: + bool mHardwareH264Enabled; + bool mSoftwareH264Enabled; + bool mH264Enabled; + bool mVP9Enabled; + bool mVP9Preferred; + int32_t mH264Level; + int32_t mH264MaxBr; + int32_t mH264MaxMbps; + int32_t mVP8MaxFs; + int32_t mVP8MaxFr; + bool mUseTmmbr; + bool mUseRemb; + bool mUseTransportCC; + bool mUseAudioFec; + bool mRedUlpfecEnabled; + bool mDtmfEnabled; +}; + +class ConfigureRedCodec { + public: + explicit ConfigureRedCodec(nsCOMPtr<nsIPrefBranch>& branch, + std::vector<uint8_t>* redundantEncodings) + : mRedundantEncodings(redundantEncodings) { + // if we wanted to override or modify which encodings are considered + // for redundant encodings, we'd probably want to handle it here by + // checking prefs modifying the operator() code below + } + + void operator()(UniquePtr<JsepCodecDescription>& codec) const { + if (codec->Type() == SdpMediaSection::kVideo && !codec->mEnabled) { + uint8_t pt = (uint8_t)strtoul(codec->mDefaultPt.c_str(), nullptr, 10); + // don't search for the codec payload type unless we have a valid + // conversion (non-zero) + if (pt != 0) { + std::vector<uint8_t>::iterator it = std::find( + mRedundantEncodings->begin(), mRedundantEncodings->end(), pt); + if (it != mRedundantEncodings->end()) { + mRedundantEncodings->erase(it); + } + } + } + } + + private: + std::vector<uint8_t>* mRedundantEncodings; +}; + +nsresult PeerConnectionImpl::ConfigureJsepSessionCodecs() { + nsresult res; + nsCOMPtr<nsIPrefService> prefs = + do_GetService("@mozilla.org/preferences-service;1", &res); + + if (NS_FAILED(res)) { + CSFLogError(LOGTAG, "%s: Couldn't get prefs service, res=%u", __FUNCTION__, + static_cast<unsigned>(res)); + return res; + } + + nsCOMPtr<nsIPrefBranch> branch = do_QueryInterface(prefs); + if (!branch) { + CSFLogError(LOGTAG, "%s: Couldn't get prefs branch", __FUNCTION__); + return NS_ERROR_FAILURE; + } + + ConfigureCodec configurer(branch); + mJsepSession->ForEachCodec(configurer); + + // if red codec is enabled, configure it for the other enabled codecs + for (auto& codec : mJsepSession->Codecs()) { + if (codec->mName == "red" && codec->mEnabled) { + JsepVideoCodecDescription* redCodec = + static_cast<JsepVideoCodecDescription*>(codec.get()); + ConfigureRedCodec configureRed(branch, &(redCodec->mRedundantEncodings)); + mJsepSession->ForEachCodec(configureRed); + break; + } + } + + // We use this to sort the list of codecs once everything is configured + CompareCodecPriority comparator; + + // Sort by priority + int32_t preferredCodec = 0; + branch->GetIntPref("media.navigator.video.preferred_codec", &preferredCodec); + + if (preferredCodec) { + comparator.SetPreferredCodec(preferredCodec); + } + + mJsepSession->SortCodecs(comparator); + return NS_OK; +} + +// Data channels won't work without a window, so in order for the C++ unit +// tests to work (it doesn't have a window available) we ifdef the following +// two implementations. +// +// Note: 'media.peerconnection.sctp.force_maximum_message_size' changes +// behaviour triggered by these parameters. +NS_IMETHODIMP +PeerConnectionImpl::EnsureDataConnection(uint16_t aLocalPort, + uint16_t aNumstreams, + uint32_t aMaxMessageSize, + bool aMMSSet) { + PC_AUTO_ENTER_API_CALL(false); + + if (mDataConnection) { + CSFLogDebug(LOGTAG, "%s DataConnection already connected", __FUNCTION__); + mDataConnection->SetMaxMessageSize(aMMSSet, aMaxMessageSize); + return NS_OK; + } + + nsCOMPtr<nsISerialEventTarget> target = + mWindow ? mWindow->EventTargetFor(TaskCategory::Other) : nullptr; + Maybe<uint64_t> mms = aMMSSet ? Some(aMaxMessageSize) : Nothing(); + if (auto res = DataChannelConnection::Create(this, target, mTransportHandler, + aLocalPort, aNumstreams, mms)) { + mDataConnection = res.value(); + CSFLogDebug(LOGTAG, "%s DataChannelConnection %p attached to %s", + __FUNCTION__, (void*)mDataConnection.get(), mHandle.c_str()); + return NS_OK; + } + CSFLogError(LOGTAG, "%s DataConnection Create Failed", __FUNCTION__); + return NS_ERROR_FAILURE; +} + +nsresult PeerConnectionImpl::GetDatachannelParameters( + uint32_t* channels, uint16_t* localport, uint16_t* remoteport, + uint32_t* remotemaxmessagesize, bool* mmsset, std::string* transportId, + bool* client) const { + // Clear, just in case we fail. + *channels = 0; + *localport = 0; + *remoteport = 0; + *remotemaxmessagesize = 0; + *mmsset = false; + transportId->clear(); + + Maybe<const JsepTransceiver> datachannelTransceiver = + mJsepSession->FindTransceiver([](const JsepTransceiver& aTransceiver) { + return aTransceiver.GetMediaType() == SdpMediaSection::kApplication; + }); + + if (!datachannelTransceiver || + !datachannelTransceiver->mTransport.mComponents || + !datachannelTransceiver->mSendTrack.GetNegotiatedDetails()) { + return NS_ERROR_FAILURE; + } + + // This will release assert if there is no such index, and that's ok + const JsepTrackEncoding& encoding = + datachannelTransceiver->mSendTrack.GetNegotiatedDetails()->GetEncoding(0); + + if (NS_WARN_IF(encoding.GetCodecs().empty())) { + CSFLogError(LOGTAG, + "%s: Negotiated m=application with no codec. " + "This is likely to be broken.", + __FUNCTION__); + return NS_ERROR_FAILURE; + } + + for (const auto& codec : encoding.GetCodecs()) { + if (codec->Type() != SdpMediaSection::kApplication) { + CSFLogError(LOGTAG, + "%s: Codec type for m=application was %u, this " + "is a bug.", + __FUNCTION__, static_cast<unsigned>(codec->Type())); + MOZ_ASSERT(false, "Codec for m=application was not \"application\""); + return NS_ERROR_FAILURE; + } + + if (codec->mName != "webrtc-datachannel") { + CSFLogWarn(LOGTAG, + "%s: Codec for m=application was not " + "webrtc-datachannel (was instead %s). ", + __FUNCTION__, codec->mName.c_str()); + continue; + } + + if (codec->mChannels) { + *channels = codec->mChannels; + } else { + *channels = WEBRTC_DATACHANNEL_STREAMS_DEFAULT; + } + const JsepApplicationCodecDescription* appCodec = + static_cast<const JsepApplicationCodecDescription*>(codec.get()); + *localport = appCodec->mLocalPort; + *remoteport = appCodec->mRemotePort; + *remotemaxmessagesize = appCodec->mRemoteMaxMessageSize; + *mmsset = appCodec->mRemoteMMSSet; + MOZ_ASSERT(!datachannelTransceiver->mTransport.mTransportId.empty()); + *transportId = datachannelTransceiver->mTransport.mTransportId; + *client = datachannelTransceiver->mTransport.mDtls->GetRole() == + JsepDtlsTransport::kJsepDtlsClient; + return NS_OK; + } + return NS_ERROR_FAILURE; +} + +nsresult PeerConnectionImpl::AddRtpTransceiverToJsepSession( + JsepTransceiver& transceiver) { + nsresult res = ConfigureJsepSessionCodecs(); + if (NS_FAILED(res)) { + CSFLogError(LOGTAG, "Failed to configure codecs"); + return res; + } + + mJsepSession->AddTransceiver(transceiver); + return NS_OK; +} + +static Maybe<SdpMediaSection::MediaType> ToSdpMediaType( + const nsAString& aKind) { + if (aKind.EqualsASCII("audio")) { + return Some(SdpMediaSection::MediaType::kAudio); + } else if (aKind.EqualsASCII("video")) { + return Some(SdpMediaSection::MediaType::kVideo); + } + return Nothing(); +} + +already_AddRefed<RTCRtpTransceiver> PeerConnectionImpl::AddTransceiver( + const dom::RTCRtpTransceiverInit& aInit, const nsAString& aKind, + dom::MediaStreamTrack* aSendTrack, bool aAddTrackMagic, ErrorResult& aRv) { + // Copy, because we might need to modify + RTCRtpTransceiverInit init(aInit); + + Maybe<SdpMediaSection::MediaType> type = ToSdpMediaType(aKind); + if (NS_WARN_IF(!type.isSome())) { + MOZ_ASSERT(false, "Invalid media kind"); + aRv = NS_ERROR_INVALID_ARG; + return nullptr; + } + + JsepTransceiver jsepTransceiver(*type, *mUuidGen); + jsepTransceiver.SetRtxIsAllowed(mRtxIsAllowed); + + // Do this last, since it is not possible to roll back. + nsresult rv = AddRtpTransceiverToJsepSession(jsepTransceiver); + if (NS_FAILED(rv)) { + CSFLogError(LOGTAG, "%s: AddRtpTransceiverToJsepSession failed, res=%u", + __FUNCTION__, static_cast<unsigned>(rv)); + aRv = rv; + return nullptr; + } + + auto& sendEncodings = init.mSendEncodings; + + // CheckAndRectifyEncodings covers these six: + // If any encoding contains a rid member whose value does not conform to the + // grammar requirements specified in Section 10 of [RFC8851], throw a + // TypeError. + + // If some but not all encodings contain a rid member, throw a TypeError. + + // If any encoding contains a rid member whose value is the same as that of a + // rid contained in another encoding in sendEncodings, throw a TypeError. + + // If kind is "audio", remove the scaleResolutionDownBy member from all + // encodings that contain one. + + // If any encoding contains a scaleResolutionDownBy member whose value is + // less than 1.0, throw a RangeError. + + // Verify that the value of each maxFramerate member in sendEncodings that is + // defined is greater than 0.0. If one of the maxFramerate values does not + // meet this requirement, throw a RangeError. + RTCRtpSender::CheckAndRectifyEncodings(sendEncodings, + *type == SdpMediaSection::kVideo, aRv); + if (aRv.Failed()) { + return nullptr; + } + + // If any encoding contains a read-only parameter other than rid, throw an + // InvalidAccessError. + // NOTE: We don't support any additional read-only params right now. Also, + // spec shoehorns this in between checks that setParameters also performs + // (between the rid checks and the scaleResolutionDownBy checks). + + // If any encoding contains a scaleResolutionDownBy member, then for each + // encoding without one, add a scaleResolutionDownBy member with the value + // 1.0. + for (const auto& constEncoding : sendEncodings) { + if (constEncoding.mScaleResolutionDownBy.WasPassed()) { + for (auto& encoding : sendEncodings) { + if (!encoding.mScaleResolutionDownBy.WasPassed()) { + encoding.mScaleResolutionDownBy.Construct(1.0f); + } + } + break; + } + } + + // Let maxN be the maximum number of total simultaneous encodings the user + // agent may support for this kind, at minimum 1.This should be an optimistic + // number since the codec to be used is not known yet. + size_t maxN = + (*type == SdpMediaSection::kVideo) ? webrtc::kMaxSimulcastStreams : 1; + + // If the number of encodings stored in sendEncodings exceeds maxN, then trim + // sendEncodings from the tail until its length is maxN. + // NOTE: Spec has this after all validation steps; even if there are elements + // that we will trim off, we still validate them. + if (sendEncodings.Length() > maxN) { + sendEncodings.TruncateLength(maxN); + } + + // If kind is "video" and none of the encodings contain a + // scaleResolutionDownBy member, then for each encoding, add a + // scaleResolutionDownBy member with the value 2^(length of sendEncodings - + // encoding index - 1). This results in smaller-to-larger resolutions where + // the last encoding has no scaling applied to it, e.g. 4:2:1 if the length + // is 3. + // NOTE: The code above ensures that these are all set, or all unset, so we + // can just check the first one. + if (sendEncodings.Length() && *type == SdpMediaSection::kVideo && + !sendEncodings[0].mScaleResolutionDownBy.WasPassed()) { + double scale = 1.0f; + for (auto it = sendEncodings.rbegin(); it != sendEncodings.rend(); ++it) { + it->mScaleResolutionDownBy.Construct(scale); + scale *= 2; + } + } + + // If the number of encodings now stored in sendEncodings is 1, then remove + // any rid member from the lone entry. + if (sendEncodings.Length() == 1) { + sendEncodings[0].mRid.Reset(); + } + + RefPtr<RTCRtpTransceiver> transceiver = CreateTransceiver( + jsepTransceiver.GetUuid(), + jsepTransceiver.GetMediaType() == SdpMediaSection::kVideo, init, + aSendTrack, aAddTrackMagic, aRv); + + if (aRv.Failed()) { + // Would be nice if we could peek at the rv without stealing it, so we + // could log... + CSFLogError(LOGTAG, "%s: failed", __FUNCTION__); + return nullptr; + } + + mTransceivers.AppendElement(transceiver); + return transceiver.forget(); +} + +bool PeerConnectionImpl::CheckNegotiationNeeded() { + MOZ_ASSERT(mSignalingState == RTCSignalingState::Stable); + SyncToJsep(); + return !mLocalIceCredentialsToReplace.empty() || + mJsepSession->CheckNegotiationNeeded(); +} + +bool PeerConnectionImpl::CreatedSender(const dom::RTCRtpSender& aSender) const { + return aSender.IsMyPc(this); +} + +nsresult PeerConnectionImpl::InitializeDataChannel() { + PC_AUTO_ENTER_API_CALL(false); + CSFLogDebug(LOGTAG, "%s", __FUNCTION__); + + uint32_t channels = 0; + uint16_t localport = 0; + uint16_t remoteport = 0; + uint32_t remotemaxmessagesize = 0; + bool mmsset = false; + std::string transportId; + bool client = false; + nsresult rv = GetDatachannelParameters(&channels, &localport, &remoteport, + &remotemaxmessagesize, &mmsset, + &transportId, &client); + + if (NS_FAILED(rv)) { + CSFLogDebug(LOGTAG, "%s: We did not negotiate datachannel", __FUNCTION__); + return NS_OK; + } + + if (channels > MAX_NUM_STREAMS) { + channels = MAX_NUM_STREAMS; + } + + rv = EnsureDataConnection(localport, channels, remotemaxmessagesize, mmsset); + if (NS_SUCCEEDED(rv)) { + if (mDataConnection->ConnectToTransport(transportId, client, localport, + remoteport)) { + return NS_OK; + } + // If we inited the DataConnection, call Destroy() before releasing it + mDataConnection->Destroy(); + } + mDataConnection = nullptr; + return NS_ERROR_FAILURE; +} + +already_AddRefed<nsDOMDataChannel> PeerConnectionImpl::CreateDataChannel( + const nsAString& aLabel, const nsAString& aProtocol, uint16_t aType, + bool ordered, uint16_t aMaxTime, uint16_t aMaxNum, bool aExternalNegotiated, + uint16_t aStream, ErrorResult& rv) { + RefPtr<nsDOMDataChannel> result; + rv = CreateDataChannel(aLabel, aProtocol, aType, ordered, aMaxTime, aMaxNum, + aExternalNegotiated, aStream, getter_AddRefs(result)); + return result.forget(); +} + +NS_IMETHODIMP +PeerConnectionImpl::CreateDataChannel( + const nsAString& aLabel, const nsAString& aProtocol, uint16_t aType, + bool ordered, uint16_t aMaxTime, uint16_t aMaxNum, bool aExternalNegotiated, + uint16_t aStream, nsDOMDataChannel** aRetval) { + PC_AUTO_ENTER_API_CALL(false); + MOZ_ASSERT(aRetval); + + RefPtr<DataChannel> dataChannel; + DataChannelConnection::Type theType = + static_cast<DataChannelConnection::Type>(aType); + + nsresult rv = EnsureDataConnection( + WEBRTC_DATACHANNEL_PORT_DEFAULT, WEBRTC_DATACHANNEL_STREAMS_DEFAULT, + WEBRTC_DATACHANNEL_MAX_MESSAGE_SIZE_REMOTE_DEFAULT, false); + if (NS_FAILED(rv)) { + return rv; + } + dataChannel = mDataConnection->Open( + NS_ConvertUTF16toUTF8(aLabel), NS_ConvertUTF16toUTF8(aProtocol), theType, + ordered, + aType == DataChannelConnection::PARTIAL_RELIABLE_REXMIT + ? aMaxNum + : (aType == DataChannelConnection::PARTIAL_RELIABLE_TIMED ? aMaxTime + : 0), + nullptr, nullptr, aExternalNegotiated, aStream); + NS_ENSURE_TRUE(dataChannel, NS_ERROR_NOT_AVAILABLE); + + CSFLogDebug(LOGTAG, "%s: making DOMDataChannel", __FUNCTION__); + + Maybe<JsepTransceiver> dcTransceiver = + mJsepSession->FindTransceiver([](const JsepTransceiver& aTransceiver) { + return aTransceiver.GetMediaType() == SdpMediaSection::kApplication; + }); + + if (dcTransceiver) { + dcTransceiver->RestartDatachannelTransceiver(); + mJsepSession->SetTransceiver(*dcTransceiver); + } else { + mJsepSession->AddTransceiver( + JsepTransceiver(SdpMediaSection::MediaType::kApplication, *mUuidGen)); + } + + RefPtr<nsDOMDataChannel> retval; + rv = NS_NewDOMDataChannel(dataChannel.forget(), mWindow, + getter_AddRefs(retval)); + if (NS_FAILED(rv)) { + return rv; + } + retval.forget(aRetval); + return NS_OK; +} + +NS_IMPL_CYCLE_COLLECTION(PeerConnectionImpl::Operation, mPromise, mPc) +NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(PeerConnectionImpl::Operation) + NS_INTERFACE_MAP_ENTRY(nsISupports) +NS_INTERFACE_MAP_END +NS_IMPL_CYCLE_COLLECTING_ADDREF(PeerConnectionImpl::Operation) +NS_IMPL_CYCLE_COLLECTING_RELEASE(PeerConnectionImpl::Operation) + +PeerConnectionImpl::Operation::Operation(PeerConnectionImpl* aPc, + ErrorResult& aError) + : mPromise(aPc->MakePromise(aError)), mPc(aPc) {} + +PeerConnectionImpl::Operation::~Operation() = default; + +void PeerConnectionImpl::Operation::Call(ErrorResult& aError) { + RefPtr<dom::Promise> opPromise = CallImpl(aError); + if (aError.Failed()) { + return; + } + // Upon fulfillment or rejection of the promise returned by the operation, + // run the following steps: + // (NOTE: mPromise is p from https://w3c.github.io/webrtc-pc/#dfn-chain, + // and CallImpl() is what returns the promise for the operation itself) + opPromise->AppendNativeHandler(this); +} + +void PeerConnectionImpl::Operation::ResolvedCallback( + JSContext* aCx, JS::Handle<JS::Value> aValue, ErrorResult& aRv) { + // If connection.[[IsClosed]] is true, abort these steps. + // (the spec wants p to never settle in this event) + if (!mPc->IsClosed()) { + // If the promise returned by operation was fulfilled with a + // value, fulfill p with that value. + mPromise->MaybeResolveWithClone(aCx, aValue); + // Upon fulfillment or rejection of p, execute the following + // steps: + // (Static analysis forces us to use a temporary) + RefPtr<PeerConnectionImpl> pc = mPc; + pc->RunNextOperation(aRv); + } +} + +void PeerConnectionImpl::Operation::RejectedCallback( + JSContext* aCx, JS::Handle<JS::Value> aValue, ErrorResult& aRv) { + // If connection.[[IsClosed]] is true, abort these steps. + // (the spec wants p to never settle in this event) + if (!mPc->IsClosed()) { + // If the promise returned by operation was rejected with a + // value, reject p with that value. + mPromise->MaybeRejectWithClone(aCx, aValue); + // Upon fulfillment or rejection of p, execute the following + // steps: + // (Static analysis forces us to use a temporary) + RefPtr<PeerConnectionImpl> pc = mPc; + pc->RunNextOperation(aRv); + } +} + +NS_IMPL_CYCLE_COLLECTION_INHERITED(PeerConnectionImpl::JSOperation, + PeerConnectionImpl::Operation, mOperation) + +NS_IMPL_ADDREF_INHERITED(PeerConnectionImpl::JSOperation, + PeerConnectionImpl::Operation) +NS_IMPL_RELEASE_INHERITED(PeerConnectionImpl::JSOperation, + PeerConnectionImpl::Operation) + +NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(PeerConnectionImpl::JSOperation) +NS_INTERFACE_MAP_END_INHERITING(PeerConnectionImpl::Operation) + +PeerConnectionImpl::JSOperation::JSOperation(PeerConnectionImpl* aPc, + dom::ChainedOperation& aOp, + ErrorResult& aError) + : Operation(aPc, aError), mOperation(&aOp) {} + +RefPtr<dom::Promise> PeerConnectionImpl::JSOperation::CallImpl( + ErrorResult& aError) { + // Static analysis will not let us call this without a temporary :( + RefPtr<dom::ChainedOperation> op = mOperation; + return op->Call(aError); +} + +already_AddRefed<dom::Promise> PeerConnectionImpl::Chain( + dom::ChainedOperation& aOperation, ErrorResult& aError) { + MOZ_RELEASE_ASSERT(!mChainingOperation); + mChainingOperation = true; + RefPtr<Operation> operation = new JSOperation(this, aOperation, aError); + if (aError.Failed()) { + return nullptr; + } + RefPtr<Promise> promise = Chain(operation, aError); + if (aError.Failed()) { + return nullptr; + } + mChainingOperation = false; + return promise.forget(); +} + +// This is kinda complicated, but it is what the spec requires us to do. The +// core of what makes this complicated is the requirement that |aOperation| be +// run _immediately_ (without any Promise.Then!) if the operations chain is +// empty. +already_AddRefed<dom::Promise> PeerConnectionImpl::Chain( + const RefPtr<Operation>& aOperation, ErrorResult& aError) { + // If connection.[[IsClosed]] is true, return a promise rejected with a newly + // created InvalidStateError. + if (IsClosed()) { + CSFLogDebug(LOGTAG, "%s:%d: Peer connection is closed", __FILE__, __LINE__); + RefPtr<dom::Promise> error = MakePromise(aError); + if (aError.Failed()) { + return nullptr; + } + error->MaybeRejectWithInvalidStateError("Peer connection is closed"); + return error.forget(); + } + + // Append operation to [[Operations]]. + mOperations.AppendElement(aOperation); + + // If the length of [[Operations]] is exactly 1, execute operation. + if (mOperations.Length() == 1) { + aOperation->Call(aError); + if (aError.Failed()) { + return nullptr; + } + } + + // This is the promise p from https://w3c.github.io/webrtc-pc/#dfn-chain + return do_AddRef(aOperation->GetPromise()); +} + +void PeerConnectionImpl::RunNextOperation(ErrorResult& aError) { + // If connection.[[IsClosed]] is true, abort these steps. + if (IsClosed()) { + return; + } + + // Remove the first element of [[Operations]]. + mOperations.RemoveElementAt(0); + + // If [[Operations]] is non-empty, execute the operation represented by the + // first element of [[Operations]], and abort these steps. + if (mOperations.Length()) { + // Cannot call without a temporary :( + RefPtr<Operation> op = mOperations[0]; + op->Call(aError); + return; + } + + // If connection.[[UpdateNegotiationNeededFlagOnEmptyChain]] is false, abort + // these steps. + if (!mUpdateNegotiationNeededFlagOnEmptyChain) { + return; + } + + // Set connection.[[UpdateNegotiationNeededFlagOnEmptyChain]] to false. + mUpdateNegotiationNeededFlagOnEmptyChain = false; + // Update the negotiation-needed flag for connection. + UpdateNegotiationNeeded(); +} + +void PeerConnectionImpl::SyncToJsep() { + for (const auto& transceiver : mTransceivers) { + transceiver->SyncToJsep(*mJsepSession); + } +} + +void PeerConnectionImpl::SyncFromJsep() { + CSFLogDebug(LOGTAG, "%s", __FUNCTION__); + mJsepSession->ForEachTransceiver( + [this, self = RefPtr<PeerConnectionImpl>(this)]( + const JsepTransceiver& jsepTransceiver) { + if (jsepTransceiver.GetMediaType() == + SdpMediaSection::MediaType::kApplication) { + return; + } + + CSFLogDebug(LOGTAG, "%s: Looking for match", __FUNCTION__); + RefPtr<RTCRtpTransceiver> transceiver; + for (auto& temp : mTransceivers) { + if (temp->GetJsepTransceiverId() == jsepTransceiver.GetUuid()) { + CSFLogDebug(LOGTAG, "%s: Found match", __FUNCTION__); + transceiver = temp; + break; + } + } + + if (!transceiver) { + CSFLogDebug(LOGTAG, "%s: No match, making new", __FUNCTION__); + dom::RTCRtpTransceiverInit init; + init.mDirection = RTCRtpTransceiverDirection::Recvonly; + IgnoredErrorResult rv; + transceiver = CreateTransceiver( + jsepTransceiver.GetUuid(), + jsepTransceiver.GetMediaType() == SdpMediaSection::kVideo, init, + nullptr, false, rv); + if (NS_WARN_IF(rv.Failed())) { + MOZ_ASSERT(false); + return; + } + mTransceivers.AppendElement(transceiver); + } + + CSFLogDebug(LOGTAG, "%s: Syncing transceiver", __FUNCTION__); + transceiver->SyncFromJsep(*mJsepSession); + }); +} + +already_AddRefed<dom::Promise> PeerConnectionImpl::MakePromise( + ErrorResult& aError) const { + nsCOMPtr<nsIGlobalObject> global = do_QueryInterface(mWindow); + return dom::Promise::Create(global, aError); +} + +void PeerConnectionImpl::UpdateNegotiationNeeded() { + // If the length of connection.[[Operations]] is not 0, then set + // connection.[[UpdateNegotiationNeededFlagOnEmptyChain]] to true, and abort + // these steps. + if (mOperations.Length() != 0) { + mUpdateNegotiationNeededFlagOnEmptyChain = true; + return; + } + + // Queue a task to run the following steps: + GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction( + __func__, [this, self = RefPtr<PeerConnectionImpl>(this)] { + // If connection.[[IsClosed]] is true, abort these steps. + if (IsClosed()) { + return; + } + // If the length of connection.[[Operations]] is not 0, then set + // connection.[[UpdateNegotiationNeededFlagOnEmptyChain]] to true, and + // abort these steps. + if (mOperations.Length()) { + mUpdateNegotiationNeededFlagOnEmptyChain = true; + return; + } + // If connection's signaling state is not "stable", abort these steps. + if (mSignalingState != RTCSignalingState::Stable) { + return; + } + // If the result of checking if negotiation is needed is false, clear + // the negotiation-needed flag by setting + // connection.[[NegotiationNeeded]] to false, and abort these steps. + if (!CheckNegotiationNeeded()) { + mNegotiationNeeded = false; + return; + } + + // If connection.[[NegotiationNeeded]] is already true, abort these + // steps. + if (mNegotiationNeeded) { + return; + } + + // Set connection.[[NegotiationNeeded]] to true. + mNegotiationNeeded = true; + + // Fire an event named negotiationneeded at connection. + ErrorResult rv; + mPCObserver->FireNegotiationNeededEvent(rv); + })); +} + +void PeerConnectionImpl::NotifyDataChannel( + already_AddRefed<DataChannel> aChannel) { + PC_AUTO_ENTER_API_CALL_NO_CHECK(); + + RefPtr<DataChannel> channel(aChannel); + MOZ_ASSERT(channel); + CSFLogDebug(LOGTAG, "%s: channel: %p", __FUNCTION__, channel.get()); + + RefPtr<nsDOMDataChannel> domchannel; + nsresult rv = NS_NewDOMDataChannel(channel.forget(), mWindow, + getter_AddRefs(domchannel)); + NS_ENSURE_SUCCESS_VOID(rv); + + JSErrorResult jrv; + mPCObserver->NotifyDataChannel(*domchannel, jrv); +} + +void PeerConnectionImpl::NotifyDataChannelOpen(DataChannel*) { + mDataChannelsOpened++; +} + +void PeerConnectionImpl::NotifyDataChannelClosed(DataChannel*) { + mDataChannelsClosed++; +} + +void PeerConnectionImpl::NotifySctpConnected() { + if (!mSctpTransport) { + MOZ_ASSERT(false); + return; + } + + mSctpTransport->UpdateState(RTCSctpTransportState::Connected); +} + +void PeerConnectionImpl::NotifySctpClosed() { + if (!mSctpTransport) { + MOZ_ASSERT(false); + return; + } + + mSctpTransport->UpdateState(RTCSctpTransportState::Closed); +} + +NS_IMETHODIMP +PeerConnectionImpl::CreateOffer(const RTCOfferOptions& aOptions) { + JsepOfferOptions options; + // convert the RTCOfferOptions to JsepOfferOptions + if (aOptions.mOfferToReceiveAudio.WasPassed()) { + options.mOfferToReceiveAudio = + mozilla::Some(size_t(aOptions.mOfferToReceiveAudio.Value())); + } + + if (aOptions.mOfferToReceiveVideo.WasPassed()) { + options.mOfferToReceiveVideo = + mozilla::Some(size_t(aOptions.mOfferToReceiveVideo.Value())); + } + + options.mIceRestart = mozilla::Some(aOptions.mIceRestart || + !mLocalIceCredentialsToReplace.empty()); + + return CreateOffer(options); +} + +static void DeferredCreateOffer(const std::string& aPcHandle, + const JsepOfferOptions& aOptions) { + PeerConnectionWrapper wrapper(aPcHandle); + + if (wrapper.impl()) { + if (!PeerConnectionCtx::GetInstance()->isReady()) { + MOZ_CRASH( + "Why is DeferredCreateOffer being executed when the " + "PeerConnectionCtx isn't ready?"); + } + wrapper.impl()->CreateOffer(aOptions); + } +} + +// Have to use unique_ptr because webidl enums are generated without a +// copy c'tor. +static std::unique_ptr<dom::PCErrorData> buildJSErrorData( + const JsepSession::Result& aResult, const std::string& aMessage) { + std::unique_ptr<dom::PCErrorData> result(new dom::PCErrorData); + result->mName = *aResult.mError; + result->mMessage = NS_ConvertASCIItoUTF16(aMessage.c_str()); + return result; +} + +// Used by unit tests and the IDL CreateOffer. +NS_IMETHODIMP +PeerConnectionImpl::CreateOffer(const JsepOfferOptions& aOptions) { + PC_AUTO_ENTER_API_CALL(true); + + if (!PeerConnectionCtx::GetInstance()->isReady()) { + // Uh oh. We're not ready yet. Enqueue this operation. + PeerConnectionCtx::GetInstance()->queueJSEPOperation( + WrapRunnableNM(DeferredCreateOffer, mHandle, aOptions)); + STAMP_TIMECARD(mTimeCard, "Deferring CreateOffer (not ready)"); + return NS_OK; + } + + CSFLogDebug(LOGTAG, "CreateOffer()"); + + nsresult nrv = ConfigureJsepSessionCodecs(); + if (NS_FAILED(nrv)) { + CSFLogError(LOGTAG, "Failed to configure codecs"); + return nrv; + } + + STAMP_TIMECARD(mTimeCard, "Create Offer"); + + GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction( + __func__, [this, self = RefPtr<PeerConnectionImpl>(this), aOptions] { + std::string offer; + + SyncToJsep(); + UniquePtr<JsepSession> uncommittedJsepSession(mJsepSession->Clone()); + JsepSession::Result result = + uncommittedJsepSession->CreateOffer(aOptions, &offer); + JSErrorResult rv; + if (result.mError.isSome()) { + std::string errorString = uncommittedJsepSession->GetLastError(); + + CSFLogError(LOGTAG, "%s: pc = %s, error = %s", __FUNCTION__, + mHandle.c_str(), errorString.c_str()); + + mPCObserver->OnCreateOfferError( + *buildJSErrorData(result, errorString), rv); + } else { + mJsepSession = std::move(uncommittedJsepSession); + mPCObserver->OnCreateOfferSuccess(ObString(offer.c_str()), rv); + } + })); + + return NS_OK; +} + +NS_IMETHODIMP +PeerConnectionImpl::CreateAnswer() { + PC_AUTO_ENTER_API_CALL(true); + + CSFLogDebug(LOGTAG, "CreateAnswer()"); + + STAMP_TIMECARD(mTimeCard, "Create Answer"); + // TODO(bug 1098015): Once RTCAnswerOptions is standardized, we'll need to + // add it as a param to CreateAnswer, and convert it here. + JsepAnswerOptions options; + + GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction( + __func__, [this, self = RefPtr<PeerConnectionImpl>(this), options] { + std::string answer; + SyncToJsep(); + UniquePtr<JsepSession> uncommittedJsepSession(mJsepSession->Clone()); + JsepSession::Result result = + uncommittedJsepSession->CreateAnswer(options, &answer); + JSErrorResult rv; + if (result.mError.isSome()) { + std::string errorString = uncommittedJsepSession->GetLastError(); + + CSFLogError(LOGTAG, "%s: pc = %s, error = %s", __FUNCTION__, + mHandle.c_str(), errorString.c_str()); + + mPCObserver->OnCreateAnswerError( + *buildJSErrorData(result, errorString), rv); + } else { + mJsepSession = std::move(uncommittedJsepSession); + mPCObserver->OnCreateAnswerSuccess(ObString(answer.c_str()), rv); + } + })); + + return NS_OK; +} + +dom::RTCSdpType ToDomSdpType(JsepSdpType aType) { + switch (aType) { + case kJsepSdpOffer: + return dom::RTCSdpType::Offer; + case kJsepSdpAnswer: + return dom::RTCSdpType::Answer; + case kJsepSdpPranswer: + return dom::RTCSdpType::Pranswer; + case kJsepSdpRollback: + return dom::RTCSdpType::Rollback; + } + + MOZ_CRASH("Nonexistent JsepSdpType"); +} + +JsepSdpType ToJsepSdpType(dom::RTCSdpType aType) { + switch (aType) { + case dom::RTCSdpType::Offer: + return kJsepSdpOffer; + case dom::RTCSdpType::Pranswer: + return kJsepSdpPranswer; + case dom::RTCSdpType::Answer: + return kJsepSdpAnswer; + case dom::RTCSdpType::Rollback: + return kJsepSdpRollback; + case dom::RTCSdpType::EndGuard_:; + } + + MOZ_CRASH("Nonexistent dom::RTCSdpType"); +} + +NS_IMETHODIMP +PeerConnectionImpl::SetLocalDescription(int32_t aAction, const char* aSDP) { + PC_AUTO_ENTER_API_CALL(true); + + if (!aSDP) { + CSFLogError(LOGTAG, "%s - aSDP is NULL", __FUNCTION__); + return NS_ERROR_FAILURE; + } + + STAMP_TIMECARD(mTimeCard, "Set Local Description"); + + if (AnyLocalTrackHasPeerIdentity()) { + mRequestedPrivacy = Some(PrincipalPrivacy::Private); + } + + mozilla::dom::RTCSdpHistoryEntryInternal sdpEntry; + sdpEntry.mIsLocal = true; + sdpEntry.mTimestamp = mTimestampMaker.GetNow().ToDom(); + sdpEntry.mSdp = NS_ConvertASCIItoUTF16(aSDP); + auto appendHistory = [&]() { + if (!mSdpHistory.AppendElement(sdpEntry, fallible)) { + mozalloc_handle_oom(0); + } + }; + + mLocalRequestedSDP = aSDP; + + SyncToJsep(); + + bool wasRestartingIce = mJsepSession->IsIceRestarting(); + JsepSdpType sdpType; + switch (aAction) { + case IPeerConnection::kActionOffer: + sdpType = mozilla::kJsepSdpOffer; + break; + case IPeerConnection::kActionAnswer: + sdpType = mozilla::kJsepSdpAnswer; + break; + case IPeerConnection::kActionPRAnswer: + sdpType = mozilla::kJsepSdpPranswer; + break; + case IPeerConnection::kActionRollback: + sdpType = mozilla::kJsepSdpRollback; + break; + default: + MOZ_ASSERT(false); + appendHistory(); + return NS_ERROR_FAILURE; + } + MOZ_ASSERT(!mUncommittedJsepSession); + mUncommittedJsepSession.reset(mJsepSession->Clone()); + JsepSession::Result result = + mUncommittedJsepSession->SetLocalDescription(sdpType, mLocalRequestedSDP); + JSErrorResult rv; + if (result.mError.isSome()) { + std::string errorString = mUncommittedJsepSession->GetLastError(); + mUncommittedJsepSession = nullptr; + CSFLogError(LOGTAG, "%s: pc = %s, error = %s", __FUNCTION__, + mHandle.c_str(), errorString.c_str()); + mPCObserver->OnSetDescriptionError(*buildJSErrorData(result, errorString), + rv); + sdpEntry.mErrors = GetLastSdpParsingErrors(); + } else { + if (wasRestartingIce) { + RecordIceRestartStatistics(sdpType); + } + + mPCObserver->OnSetDescriptionSuccess(rv); + } + + appendHistory(); + + if (rv.Failed()) { + return rv.StealNSResult(); + } + + return NS_OK; +} + +static void DeferredSetRemote(const std::string& aPcHandle, int32_t aAction, + const std::string& aSdp) { + PeerConnectionWrapper wrapper(aPcHandle); + + if (wrapper.impl()) { + if (!PeerConnectionCtx::GetInstance()->isReady()) { + MOZ_CRASH( + "Why is DeferredSetRemote being executed when the " + "PeerConnectionCtx isn't ready?"); + } + wrapper.impl()->SetRemoteDescription(aAction, aSdp.c_str()); + } +} + +NS_IMETHODIMP +PeerConnectionImpl::SetRemoteDescription(int32_t action, const char* aSDP) { + PC_AUTO_ENTER_API_CALL(true); + + if (!aSDP) { + CSFLogError(LOGTAG, "%s - aSDP is NULL", __FUNCTION__); + return NS_ERROR_FAILURE; + } + + if (action == IPeerConnection::kActionOffer) { + if (!PeerConnectionCtx::GetInstance()->isReady()) { + // Uh oh. We're not ready yet. Enqueue this operation. (This must be a + // remote offer, or else we would not have gotten this far) + PeerConnectionCtx::GetInstance()->queueJSEPOperation(WrapRunnableNM( + DeferredSetRemote, mHandle, action, std::string(aSDP))); + STAMP_TIMECARD(mTimeCard, "Deferring SetRemote (not ready)"); + return NS_OK; + } + + nsresult nrv = ConfigureJsepSessionCodecs(); + if (NS_FAILED(nrv)) { + CSFLogError(LOGTAG, "Failed to configure codecs"); + return nrv; + } + } + + STAMP_TIMECARD(mTimeCard, "Set Remote Description"); + + mozilla::dom::RTCSdpHistoryEntryInternal sdpEntry; + sdpEntry.mIsLocal = false; + sdpEntry.mTimestamp = mTimestampMaker.GetNow().ToDom(); + sdpEntry.mSdp = NS_ConvertASCIItoUTF16(aSDP); + auto appendHistory = [&]() { + if (!mSdpHistory.AppendElement(sdpEntry, fallible)) { + mozalloc_handle_oom(0); + } + }; + + SyncToJsep(); + + mRemoteRequestedSDP = aSDP; + bool wasRestartingIce = mJsepSession->IsIceRestarting(); + JsepSdpType sdpType; + switch (action) { + case IPeerConnection::kActionOffer: + sdpType = mozilla::kJsepSdpOffer; + break; + case IPeerConnection::kActionAnswer: + sdpType = mozilla::kJsepSdpAnswer; + break; + case IPeerConnection::kActionPRAnswer: + sdpType = mozilla::kJsepSdpPranswer; + break; + case IPeerConnection::kActionRollback: + sdpType = mozilla::kJsepSdpRollback; + break; + default: + MOZ_ASSERT(false); + return NS_ERROR_FAILURE; + } + + MOZ_ASSERT(!mUncommittedJsepSession); + mUncommittedJsepSession.reset(mJsepSession->Clone()); + JsepSession::Result result = mUncommittedJsepSession->SetRemoteDescription( + sdpType, mRemoteRequestedSDP); + JSErrorResult jrv; + if (result.mError.isSome()) { + std::string errorString = mUncommittedJsepSession->GetLastError(); + mUncommittedJsepSession = nullptr; + sdpEntry.mErrors = GetLastSdpParsingErrors(); + CSFLogError(LOGTAG, "%s: pc = %s, error = %s", __FUNCTION__, + mHandle.c_str(), errorString.c_str()); + mPCObserver->OnSetDescriptionError(*buildJSErrorData(result, errorString), + jrv); + } else { + if (wasRestartingIce) { + RecordIceRestartStatistics(sdpType); + } + + mPCObserver->OnSetDescriptionSuccess(jrv); + } + + appendHistory(); + + if (jrv.Failed()) { + return jrv.StealNSResult(); + } + + return NS_OK; +} + +already_AddRefed<dom::Promise> PeerConnectionImpl::GetStats( + MediaStreamTrack* aSelector) { + if (!mWindow) { + MOZ_CRASH("Cannot create a promise without a window!"); + } + + nsCOMPtr<nsIGlobalObject> global = do_QueryInterface(mWindow); + ErrorResult rv; + RefPtr<Promise> promise = Promise::Create(global, rv); + if (NS_WARN_IF(rv.Failed())) { + MOZ_CRASH("Failed to create a promise!"); + } + + if (!IsClosed()) { + GetStats(aSelector, false) + ->Then( + GetMainThreadSerialEventTarget(), __func__, + [promise, window = mWindow]( + UniquePtr<dom::RTCStatsReportInternal>&& aReport) { + RefPtr<RTCStatsReport> report(new RTCStatsReport(window)); + report->Incorporate(*aReport); + promise->MaybeResolve(std::move(report)); + }, + [promise, window = mWindow](nsresult aError) { + RefPtr<RTCStatsReport> report(new RTCStatsReport(window)); + promise->MaybeResolve(std::move(report)); + }); + } else { + promise->MaybeReject(NS_ERROR_DOM_INVALID_STATE_ERR); + } + + return promise.forget(); +} + +void PeerConnectionImpl::GetRemoteStreams( + nsTArray<RefPtr<DOMMediaStream>>& aStreamsOut) const { + aStreamsOut = mReceiveStreams.Clone(); +} + +NS_IMETHODIMP +PeerConnectionImpl::AddIceCandidate( + const char* aCandidate, const char* aMid, const char* aUfrag, + const dom::Nullable<unsigned short>& aLevel) { + PC_AUTO_ENTER_API_CALL(true); + + if (mForceIceTcp && + std::string::npos != std::string(aCandidate).find(" UDP ")) { + CSFLogError(LOGTAG, "Blocking remote UDP candidate: %s", aCandidate); + return NS_OK; + } + + STAMP_TIMECARD(mTimeCard, "Add Ice Candidate"); + + CSFLogDebug(LOGTAG, "AddIceCandidate: %s %s", aCandidate, aUfrag); + + std::string transportId; + Maybe<unsigned short> level; + if (!aLevel.IsNull()) { + level = Some(aLevel.Value()); + } + MOZ_DIAGNOSTIC_ASSERT( + !mUncommittedJsepSession, + "AddIceCandidate is chained, which means it should never " + "run while an sRD/sLD is in progress"); + JsepSession::Result result = mJsepSession->AddRemoteIceCandidate( + aCandidate, aMid, level, aUfrag, &transportId); + + if (!result.mError.isSome()) { + // We do not bother the MediaTransportHandler about this before + // offer/answer concludes. Once offer/answer concludes, we will extract + // these candidates from the remote SDP. + if (mSignalingState == RTCSignalingState::Stable && !transportId.empty()) { + AddIceCandidate(aCandidate, transportId, aUfrag); + mRawTrickledCandidates.push_back(aCandidate); + } + // Spec says we queue a task for these updates + GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction( + __func__, [this, self = RefPtr<PeerConnectionImpl>(this)] { + if (IsClosed()) { + return; + } + mPendingRemoteDescription = + mJsepSession->GetRemoteDescription(kJsepDescriptionPending); + mCurrentRemoteDescription = + mJsepSession->GetRemoteDescription(kJsepDescriptionCurrent); + JSErrorResult rv; + mPCObserver->OnAddIceCandidateSuccess(rv); + })); + } else { + std::string errorString = mJsepSession->GetLastError(); + + CSFLogError(LOGTAG, + "Failed to incorporate remote candidate into SDP:" + " res = %u, candidate = %s, level = %i, error = %s", + static_cast<unsigned>(*result.mError), aCandidate, + level.valueOr(-1), errorString.c_str()); + + GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction( + __func__, + [this, self = RefPtr<PeerConnectionImpl>(this), errorString, result] { + if (IsClosed()) { + return; + } + JSErrorResult rv; + mPCObserver->OnAddIceCandidateError( + *buildJSErrorData(result, errorString), rv); + })); + } + + return NS_OK; +} + +NS_IMETHODIMP +PeerConnectionImpl::CloseStreams() { + PC_AUTO_ENTER_API_CALL(false); + + return NS_OK; +} + +NS_IMETHODIMP +PeerConnectionImpl::SetPeerIdentity(const nsAString& aPeerIdentity) { + PC_AUTO_ENTER_API_CALL(true); + MOZ_ASSERT(!aPeerIdentity.IsEmpty()); + + // once set, this can't be changed + if (mPeerIdentity) { + if (!mPeerIdentity->Equals(aPeerIdentity)) { + return NS_ERROR_FAILURE; + } + } else { + mPeerIdentity = new PeerIdentity(aPeerIdentity); + Document* doc = mWindow->GetExtantDoc(); + if (!doc) { + CSFLogInfo(LOGTAG, "Can't update principal on streams; document gone"); + return NS_ERROR_FAILURE; + } + for (const auto& transceiver : mTransceivers) { + transceiver->Sender()->GetPipeline()->UpdateSinkIdentity( + doc->NodePrincipal(), mPeerIdentity); + } + } + return NS_OK; +} + +nsresult PeerConnectionImpl::OnAlpnNegotiated(bool aPrivacyRequested) { + PC_AUTO_ENTER_API_CALL(false); + MOZ_DIAGNOSTIC_ASSERT(!mRequestedPrivacy || + (*mRequestedPrivacy == PrincipalPrivacy::Private) == + aPrivacyRequested); + + mRequestedPrivacy = Some(aPrivacyRequested ? PrincipalPrivacy::Private + : PrincipalPrivacy::NonPrivate); + // This updates the MediaPipelines with a private PrincipalHandle. Note that + // MediaPipelineReceive has its own AlpnNegotiated handler so it can get + // signaled off-main to drop data until it receives the new PrincipalHandle + // from us. + UpdateMediaPipelines(); + return NS_OK; +} + +void PeerConnectionImpl::OnDtlsStateChange(const std::string& aTransportId, + TransportLayer::State aState) { + auto it = mTransportIdToRTCDtlsTransport.find(aTransportId); + if (it != mTransportIdToRTCDtlsTransport.end()) { + it->second->UpdateState(aState); + } + UpdateConnectionState(); +} + +RTCPeerConnectionState PeerConnectionImpl::GetNewConnectionState() const { + // closed The RTCPeerConnection object's [[IsClosed]] slot is true. + if (IsClosed()) { + return RTCPeerConnectionState::Closed; + } + + // Would use a bitset, but that requires lots of static_cast<size_t> + // Oh well. + std::set<RTCDtlsTransportState> statesFound; + for (const auto& [id, dtlsTransport] : mTransportIdToRTCDtlsTransport) { + Unused << id; + statesFound.insert(dtlsTransport->State()); + } + + // failed The previous state doesn't apply and any RTCIceTransports are + // in the "failed" state or any RTCDtlsTransports are in the "failed" state. + // NOTE: "any RTCIceTransports are in the failed state" is equivalent to + // mIceConnectionState == Failed + if (mIceConnectionState == RTCIceConnectionState::Failed || + statesFound.count(RTCDtlsTransportState::Failed)) { + return RTCPeerConnectionState::Failed; + } + + // disconnected None of the previous states apply and any + // RTCIceTransports are in the "disconnected" state. + // NOTE: "any RTCIceTransports are in the disconnected state" is equivalent to + // mIceConnectionState == Disconnected. + if (mIceConnectionState == RTCIceConnectionState::Disconnected) { + return RTCPeerConnectionState::Disconnected; + } + + // new None of the previous states apply and all RTCIceTransports are + // in the "new" or "closed" state, and all RTCDtlsTransports are in the "new" + // or "closed" state, or there are no transports. + // NOTE: "all RTCIceTransports are in the new or closed state" is equivalent + // to mIceConnectionState == New. + if (mIceConnectionState == RTCIceConnectionState::New && + !statesFound.count(RTCDtlsTransportState::Connecting) && + !statesFound.count(RTCDtlsTransportState::Connected) && + !statesFound.count(RTCDtlsTransportState::Failed)) { + return RTCPeerConnectionState::New; + } + + // No transports + if (statesFound.empty()) { + return RTCPeerConnectionState::New; + } + + // connecting None of the previous states apply and any + // RTCIceTransport is in the "new" or "checking" state or any + // RTCDtlsTransport is in the "new" or "connecting" state. + // NOTE: "None of the previous states apply and any RTCIceTransport is in the + // "new" or "checking" state" is equivalent to mIceConnectionState == + // Checking. + if (mIceConnectionState == RTCIceConnectionState::Checking || + statesFound.count(RTCDtlsTransportState::New) || + statesFound.count(RTCDtlsTransportState::Connecting)) { + return RTCPeerConnectionState::Connecting; + } + + // connected None of the previous states apply and all RTCIceTransports are + // in the "connected", "completed" or "closed" state, and all + // RTCDtlsTransports are in the "connected" or "closed" state. + // NOTE: "None of the previous states apply and all RTCIceTransports are in + // the "connected", "completed" or "closed" state" is equivalent to + // mIceConnectionState == Connected. + if (mIceConnectionState == RTCIceConnectionState::Connected && + !statesFound.count(RTCDtlsTransportState::New) && + !statesFound.count(RTCDtlsTransportState::Failed) && + !statesFound.count(RTCDtlsTransportState::Connecting)) { + return RTCPeerConnectionState::Connected; + } + + // !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! + // THERE IS NO CATCH-ALL NONE-OF-THE-ABOVE IN THE SPEC! THIS IS REALLY BAD! !! + // !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! + + // Let's try to figure out how bad, precisely. + // Any one of these will cause us to bail above. + MOZ_ASSERT(mIceConnectionState != RTCIceConnectionState::Failed && + mIceConnectionState != RTCIceConnectionState::Disconnected && + mIceConnectionState != RTCIceConnectionState::Checking); + MOZ_ASSERT(!statesFound.count(RTCDtlsTransportState::New) && + !statesFound.count(RTCDtlsTransportState::Connecting) && + !statesFound.count(RTCDtlsTransportState::Failed)); + + // One of these must be set, or the empty() check would have failed above. + MOZ_ASSERT(statesFound.count(RTCDtlsTransportState::Connected) || + statesFound.count(RTCDtlsTransportState::Closed)); + + // Here are our remaining possibilities: + // ICE connected, !statesFound.count(Connected), statesFound.count(Closed) + // ICE connected, statesFound.count(Connected), !statesFound.count(Closed) + // ICE connected, statesFound.count(Connected), statesFound.count(Closed) + // All three of these would result in returning Connected above. + + // ICE new, !statesFound.count(Connected), statesFound.count(Closed) + // This results in returning New above. Whew. + + // ICE new, statesFound.count(Connected), !statesFound.count(Closed) + // ICE new, statesFound.count(Connected), statesFound.count(Closed) + // These would make it all the way here! Very weird state though, for all + // ICE transports to be new/closed, but having a connected DTLS transport. + // Handle this as a non-transition, just in case. + return mConnectionState; +} + +void PeerConnectionImpl::UpdateConnectionState() { + auto newState = GetNewConnectionState(); + if (newState != mConnectionState) { + CSFLogDebug(LOGTAG, "%s: %d -> %d (%p)", __FUNCTION__, + static_cast<int>(mConnectionState), static_cast<int>(newState), + this); + mConnectionState = newState; + if (mConnectionState != RTCPeerConnectionState::Closed) { + JSErrorResult jrv; + mPCObserver->OnStateChange(PCObserverStateType::ConnectionState, jrv); + } + } +} + +void PeerConnectionImpl::OnMediaError(const std::string& aError) { + CSFLogError(LOGTAG, "Encountered media error! %s", aError.c_str()); + // TODO: Let content know about this somehow. +} + +void PeerConnectionImpl::DumpPacket_m(size_t level, dom::mozPacketDumpType type, + bool sending, + UniquePtr<uint8_t[]>& packet, + size_t size) { + if (IsClosed()) { + return; + } + + // TODO: Is this efficient? Should we try grabbing our JS ctx from somewhere + // else? + AutoJSAPI jsapi; + if (!jsapi.Init(mWindow)) { + return; + } + + JS::Rooted<JSObject*> jsobj( + jsapi.cx(), + JS::NewArrayBufferWithContents(jsapi.cx(), size, packet.release())); + + RootedSpiderMonkeyInterface<ArrayBuffer> arrayBuffer(jsapi.cx()); + if (!arrayBuffer.Init(jsobj)) { + return; + } + + JSErrorResult jrv; + mPCObserver->OnPacket(level, type, sending, arrayBuffer, jrv); +} + +bool PeerConnectionImpl::HostnameInPref(const char* aPref, + const nsCString& aHostName) { + auto HostInDomain = [](const nsCString& aHost, const nsCString& aPattern) { + int32_t patternOffset = 0; + int32_t hostOffset = 0; + + // Act on '*.' wildcard in the left-most position in a domain pattern. + if (StringBeginsWith(aPattern, nsCString("*."))) { + patternOffset = 2; + + // Ignore the lowest level sub-domain for the hostname. + hostOffset = aHost.FindChar('.') + 1; + + if (hostOffset <= 1) { + // Reject a match between a wildcard and a TLD or '.foo' form. + return false; + } + } + + nsDependentCString hostRoot(aHost, hostOffset); + return hostRoot.EqualsIgnoreCase(aPattern.BeginReading() + patternOffset); + }; + + nsCString domainList; + nsresult nr = Preferences::GetCString(aPref, domainList); + + if (NS_FAILED(nr)) { + return false; + } + + domainList.StripWhitespace(); + + if (domainList.IsEmpty() || aHostName.IsEmpty()) { + return false; + } + + // Get UTF8 to ASCII domain name normalization service + nsresult rv; + nsCOMPtr<nsIIDNService> idnService = + do_GetService("@mozilla.org/network/idn-service;1", &rv); + if (NS_WARN_IF(NS_FAILED(rv))) { + return false; + } + + // Test each domain name in the comma separated list + // after converting from UTF8 to ASCII. Each domain + // must match exactly or have a single leading '*.' wildcard. + for (const nsACString& each : domainList.Split(',')) { + nsCString domainPattern; + rv = idnService->ConvertUTF8toACE(each, domainPattern); + if (NS_SUCCEEDED(rv)) { + if (HostInDomain(aHostName, domainPattern)) { + return true; + } + } else { + NS_WARNING("Failed to convert UTF-8 host to ASCII"); + } + } + + return false; +} + +nsresult PeerConnectionImpl::EnablePacketDump(unsigned long level, + dom::mozPacketDumpType type, + bool sending) { + return GetPacketDumper()->EnablePacketDump(level, type, sending); +} + +nsresult PeerConnectionImpl::DisablePacketDump(unsigned long level, + dom::mozPacketDumpType type, + bool sending) { + return GetPacketDumper()->DisablePacketDump(level, type, sending); +} + +void PeerConnectionImpl::StampTimecard(const char* aEvent) { + MOZ_ASSERT(NS_IsMainThread()); + STAMP_TIMECARD(mTimeCard, aEvent); +} + +void PeerConnectionImpl::SendWarningToConsole(const nsCString& aWarning) { + nsAutoString msg = NS_ConvertASCIItoUTF16(aWarning); + nsContentUtils::ReportToConsoleByWindowID(msg, nsIScriptError::warningFlag, + "WebRTC"_ns, mWindow->WindowID()); +} + +void PeerConnectionImpl::GetDefaultVideoCodecs( + std::vector<UniquePtr<JsepCodecDescription>>& aSupportedCodecs, + bool aUseRtx) { + // Supported video codecs. + // Note: order here implies priority for building offers! + aSupportedCodecs.emplace_back( + JsepVideoCodecDescription::CreateDefaultVP8(aUseRtx)); + aSupportedCodecs.emplace_back( + JsepVideoCodecDescription::CreateDefaultVP9(aUseRtx)); + aSupportedCodecs.emplace_back( + JsepVideoCodecDescription::CreateDefaultH264_1(aUseRtx)); + aSupportedCodecs.emplace_back( + JsepVideoCodecDescription::CreateDefaultH264_0(aUseRtx)); + aSupportedCodecs.emplace_back( + JsepVideoCodecDescription::CreateDefaultUlpFec()); + aSupportedCodecs.emplace_back( + JsepApplicationCodecDescription::CreateDefault()); + aSupportedCodecs.emplace_back(JsepVideoCodecDescription::CreateDefaultRed()); +} + +void PeerConnectionImpl::GetDefaultAudioCodecs( + std::vector<UniquePtr<JsepCodecDescription>>& aSupportedCodecs) { + aSupportedCodecs.emplace_back(JsepAudioCodecDescription::CreateDefaultOpus()); + aSupportedCodecs.emplace_back(JsepAudioCodecDescription::CreateDefaultG722()); + aSupportedCodecs.emplace_back(JsepAudioCodecDescription::CreateDefaultPCMU()); + aSupportedCodecs.emplace_back(JsepAudioCodecDescription::CreateDefaultPCMA()); + aSupportedCodecs.emplace_back( + JsepAudioCodecDescription::CreateDefaultTelephoneEvent()); +} + +void PeerConnectionImpl::GetDefaultRtpExtensions( + std::vector<RtpExtensionHeader>& aRtpExtensions) { + RtpExtensionHeader audioLevel = {JsepMediaType::kAudio, + SdpDirectionAttribute::Direction::kSendrecv, + webrtc::RtpExtension::kAudioLevelUri}; + aRtpExtensions.push_back(audioLevel); + + RtpExtensionHeader csrcAudioLevels = { + JsepMediaType::kAudio, SdpDirectionAttribute::Direction::kRecvonly, + webrtc::RtpExtension::kCsrcAudioLevelsUri}; + aRtpExtensions.push_back(csrcAudioLevels); + + RtpExtensionHeader mid = {JsepMediaType::kAudioVideo, + SdpDirectionAttribute::Direction::kSendrecv, + webrtc::RtpExtension::kMidUri}; + aRtpExtensions.push_back(mid); + + RtpExtensionHeader absSendTime = {JsepMediaType::kVideo, + SdpDirectionAttribute::Direction::kSendrecv, + webrtc::RtpExtension::kAbsSendTimeUri}; + aRtpExtensions.push_back(absSendTime); + + RtpExtensionHeader timestampOffset = { + JsepMediaType::kVideo, SdpDirectionAttribute::Direction::kSendrecv, + webrtc::RtpExtension::kTimestampOffsetUri}; + aRtpExtensions.push_back(timestampOffset); + + RtpExtensionHeader playoutDelay = { + JsepMediaType::kVideo, SdpDirectionAttribute::Direction::kRecvonly, + webrtc::RtpExtension::kPlayoutDelayUri}; + aRtpExtensions.push_back(playoutDelay); + + RtpExtensionHeader transportSequenceNumber = { + JsepMediaType::kVideo, SdpDirectionAttribute::Direction::kSendrecv, + webrtc::RtpExtension::kTransportSequenceNumberUri}; + aRtpExtensions.push_back(transportSequenceNumber); +} + +void PeerConnectionImpl::GetCapabilities( + const nsAString& aKind, dom::Nullable<dom::RTCRtpCapabilities>& aResult, + sdp::Direction aDirection) { + std::vector<UniquePtr<JsepCodecDescription>> codecs; + std::vector<RtpExtensionHeader> headers; + auto mediaType = JsepMediaType::kNone; + + if (aKind.EqualsASCII("video")) { + GetDefaultVideoCodecs(codecs, true); + mediaType = JsepMediaType::kVideo; + } else if (aKind.EqualsASCII("audio")) { + GetDefaultAudioCodecs(codecs); + mediaType = JsepMediaType::kAudio; + } else { + return; + } + + GetDefaultRtpExtensions(headers); + + // Use the codecs for kind to fill out the RTCRtpCodecCapability + for (const auto& codec : codecs) { + // To avoid misleading information on codec capabilities skip those + // not signaled for audio/video (webrtc-datachannel) + // and any disabled by default (ulpfec and red). + if (codec->mName == "webrtc-datachannel" || codec->mName == "ulpfec" || + codec->mName == "red") { + continue; + } + + dom::RTCRtpCodecCapability capability; + capability.mMimeType = aKind + NS_ConvertASCIItoUTF16("/" + codec->mName); + capability.mClockRate = codec->mClock; + + if (codec->mChannels) { + capability.mChannels.Construct(codec->mChannels); + } + + UniquePtr<SdpFmtpAttributeList::Parameters> params; + codec->ApplyConfigToFmtp(params); + + if (params != nullptr) { + std::ostringstream paramsString; + params->Serialize(paramsString); + nsTString<char16_t> fmtp; + fmtp.AssignASCII(paramsString.str()); + capability.mSdpFmtpLine.Construct(fmtp); + } + + if (!aResult.SetValue().mCodecs.AppendElement(capability, fallible)) { + mozalloc_handle_oom(0); + } + } + + // We need to manually add rtx for video. + if (mediaType == JsepMediaType::kVideo) { + dom::RTCRtpCodecCapability capability; + capability.mMimeType = aKind + NS_ConvertASCIItoUTF16("/rtx"); + capability.mClockRate = 90000; + if (!aResult.SetValue().mCodecs.AppendElement(capability, fallible)) { + mozalloc_handle_oom(0); + } + } + + // Add headers that match the direction and media type requested. + for (const auto& header : headers) { + if ((header.direction & aDirection) && (header.mMediaType & mediaType)) { + dom::RTCRtpHeaderExtensionCapability rtpHeader; + rtpHeader.mUri.AssignASCII(header.extensionname); + if (!aResult.SetValue().mHeaderExtensions.AppendElement(rtpHeader, + fallible)) { + mozalloc_handle_oom(0); + } + } + } +} + +void PeerConnectionImpl::SetupPreferredCodecs( + std::vector<UniquePtr<JsepCodecDescription>>& aPreferredCodecs) { + bool useRtx = + Preferences::GetBool("media.peerconnection.video.use_rtx", false); + + GetDefaultVideoCodecs(aPreferredCodecs, useRtx); + GetDefaultAudioCodecs(aPreferredCodecs); + + // With red update the redundant encodings list + for (auto& videoCodec : aPreferredCodecs) { + if (videoCodec->mName == "red") { + JsepVideoCodecDescription& red = + static_cast<JsepVideoCodecDescription&>(*videoCodec); + red.UpdateRedundantEncodings(aPreferredCodecs); + } + } +} + +void PeerConnectionImpl::SetupPreferredRtpExtensions( + std::vector<RtpExtensionHeader>& aPreferredheaders) { + GetDefaultRtpExtensions(aPreferredheaders); + + if (!Preferences::GetBool("media.navigator.video.use_transport_cc", false)) { + aPreferredheaders.erase( + std::remove_if( + aPreferredheaders.begin(), aPreferredheaders.end(), + [&](const RtpExtensionHeader& header) { + return header.extensionname == + webrtc::RtpExtension::kTransportSequenceNumberUri; + }), + aPreferredheaders.end()); + } +} + +nsresult PeerConnectionImpl::CalculateFingerprint( + const std::string& algorithm, std::vector<uint8_t>* fingerprint) const { + DtlsDigest digest(algorithm); + + MOZ_ASSERT(fingerprint); + const UniqueCERTCertificate& cert = mCertificate->Certificate(); + nsresult rv = DtlsIdentity::ComputeFingerprint(cert, &digest); + if (NS_FAILED(rv)) { + CSFLogError(LOGTAG, "Unable to calculate certificate fingerprint, rv=%u", + static_cast<unsigned>(rv)); + return rv; + } + *fingerprint = digest.value_; + return NS_OK; +} + +NS_IMETHODIMP +PeerConnectionImpl::GetFingerprint(char** fingerprint) { + MOZ_ASSERT(fingerprint); + MOZ_ASSERT(mCertificate); + std::vector<uint8_t> fp; + nsresult rv = CalculateFingerprint(DtlsIdentity::DEFAULT_HASH_ALGORITHM, &fp); + NS_ENSURE_SUCCESS(rv, rv); + std::ostringstream os; + os << DtlsIdentity::DEFAULT_HASH_ALGORITHM << ' ' + << SdpFingerprintAttributeList::FormatFingerprint(fp); + std::string fpStr = os.str(); + + char* tmp = new char[fpStr.size() + 1]; + std::copy(fpStr.begin(), fpStr.end(), tmp); + tmp[fpStr.size()] = '\0'; + + *fingerprint = tmp; + return NS_OK; +} + +void PeerConnectionImpl::GetCurrentLocalDescription(nsAString& aSDP) const { + aSDP = NS_ConvertASCIItoUTF16(mCurrentLocalDescription.c_str()); +} + +void PeerConnectionImpl::GetPendingLocalDescription(nsAString& aSDP) const { + aSDP = NS_ConvertASCIItoUTF16(mPendingLocalDescription.c_str()); +} + +void PeerConnectionImpl::GetCurrentRemoteDescription(nsAString& aSDP) const { + aSDP = NS_ConvertASCIItoUTF16(mCurrentRemoteDescription.c_str()); +} + +void PeerConnectionImpl::GetPendingRemoteDescription(nsAString& aSDP) const { + aSDP = NS_ConvertASCIItoUTF16(mPendingRemoteDescription.c_str()); +} + +dom::Nullable<bool> PeerConnectionImpl::GetCurrentOfferer() const { + dom::Nullable<bool> result; + if (mCurrentOfferer.isSome()) { + result.SetValue(*mCurrentOfferer); + } + return result; +} + +dom::Nullable<bool> PeerConnectionImpl::GetPendingOfferer() const { + dom::Nullable<bool> result; + if (mPendingOfferer.isSome()) { + result.SetValue(*mPendingOfferer); + } + return result; +} + +NS_IMETHODIMP +PeerConnectionImpl::SignalingState(RTCSignalingState* aState) { + PC_AUTO_ENTER_API_CALL_NO_CHECK(); + MOZ_ASSERT(aState); + + *aState = mSignalingState; + return NS_OK; +} + +NS_IMETHODIMP +PeerConnectionImpl::IceConnectionState(RTCIceConnectionState* aState) { + PC_AUTO_ENTER_API_CALL_NO_CHECK(); + MOZ_ASSERT(aState); + + *aState = mIceConnectionState; + return NS_OK; +} + +NS_IMETHODIMP +PeerConnectionImpl::IceGatheringState(RTCIceGatheringState* aState) { + PC_AUTO_ENTER_API_CALL_NO_CHECK(); + MOZ_ASSERT(aState); + + *aState = mIceGatheringState; + return NS_OK; +} + +NS_IMETHODIMP +PeerConnectionImpl::ConnectionState(RTCPeerConnectionState* aState) { + PC_AUTO_ENTER_API_CALL_NO_CHECK(); + MOZ_ASSERT(aState); + + *aState = mConnectionState; + return NS_OK; +} + +nsresult PeerConnectionImpl::CheckApiState(bool assert_ice_ready) const { + PC_AUTO_ENTER_API_CALL_NO_CHECK(); + MOZ_ASSERT(mTrickle || !assert_ice_ready || + (mIceGatheringState == RTCIceGatheringState::Complete)); + + if (IsClosed()) { + CSFLogError(LOGTAG, "%s: called API while closed", __FUNCTION__); + return NS_ERROR_FAILURE; + } + return NS_OK; +} + +void PeerConnectionImpl::StoreFinalStats( + UniquePtr<RTCStatsReportInternal>&& report) { + using namespace Telemetry; + + report->mClosed = true; + + for (const auto& inboundRtpStats : report->mInboundRtpStreamStats) { + bool isVideo = (inboundRtpStats.mId.Value().Find(u"video") != -1); + if (!isVideo) { + continue; + } + if (inboundRtpStats.mDiscardedPackets.WasPassed() && + report->mCallDurationMs.WasPassed()) { + double mins = report->mCallDurationMs.Value() / (1000 * 60); + if (mins > 0) { + Accumulate( + WEBRTC_VIDEO_DECODER_DISCARDED_PACKETS_PER_CALL_PPM, + uint32_t(double(inboundRtpStats.mDiscardedPackets.Value()) / mins)); + } + } + } + + // Finally, store the stats + mFinalStats = std::move(report); +} + +NS_IMETHODIMP +PeerConnectionImpl::Close() { + CSFLogDebug(LOGTAG, "%s: for %s", __FUNCTION__, mHandle.c_str()); + PC_AUTO_ENTER_API_CALL_NO_CHECK(); + + if (IsClosed()) { + return NS_OK; + } + + STAMP_TIMECARD(mTimeCard, "Close"); + + // When ICE completes, we record some telemetry. We do this at the end of the + // call because we want to make sure we've waited for all trickle ICE + // candidates to come in; this can happen well after we've transitioned to + // connected. As a bonus, this allows us to detect race conditions where a + // stats dispatch happens right as the PC closes. + RecordEndOfCallTelemetry(); + + CSFLogInfo(LOGTAG, + "%s: Closing PeerConnectionImpl %s; " + "ending call", + __FUNCTION__, mHandle.c_str()); + if (mJsepSession) { + mJsepSession->Close(); + } + if (mDataConnection) { + CSFLogInfo(LOGTAG, "%s: Destroying DataChannelConnection %p for %s", + __FUNCTION__, (void*)mDataConnection.get(), mHandle.c_str()); + mDataConnection->Destroy(); + mDataConnection = + nullptr; // it may not go away until the runnables are dead + } + + if (mStunAddrsRequest) { + for (const auto& hostname : mRegisteredMDNSHostnames) { + mStunAddrsRequest->SendUnregisterMDNSHostname( + nsCString(hostname.c_str())); + } + mRegisteredMDNSHostnames.clear(); + mStunAddrsRequest->Cancel(); + mStunAddrsRequest = nullptr; + } + + for (auto& transceiver : mTransceivers) { + transceiver->Close(); + } + + mTransportIdToRTCDtlsTransport.clear(); + + mQueuedIceCtxOperations.clear(); + + mOperations.Clear(); + + // Uncount this connection as active on the inner window upon close. + if (mWindow && mActiveOnWindow) { + mWindow->RemovePeerConnection(); + mActiveOnWindow = false; + } + + mSignalingState = RTCSignalingState::Closed; + mConnectionState = RTCPeerConnectionState::Closed; + + if (!mTransportHandler) { + // We were never initialized, apparently. + return NS_OK; + } + + // Clear any resources held by libwebrtc through our Call instance. + RefPtr<GenericPromise> callDestroyPromise; + if (mCall) { + // Make sure the compiler does not get confused and try to acquire a + // reference to this thread _after_ we null out mCall. + auto callThread = mCall->mCallThread; + callDestroyPromise = + InvokeAsync(callThread, __func__, [call = std::move(mCall)]() { + call->Destroy(); + return GenericPromise::CreateAndResolve( + true, "PCImpl->WebRtcCallWrapper::Destroy"); + }); + } else { + callDestroyPromise = GenericPromise::CreateAndResolve(true, __func__); + } + + mFinalStatsQuery = + GetStats(nullptr, true) + ->Then( + GetMainThreadSerialEventTarget(), __func__, + [this, self = RefPtr<PeerConnectionImpl>(this)]( + UniquePtr<dom::RTCStatsReportInternal>&& aReport) mutable { + StoreFinalStats(std::move(aReport)); + return GenericNonExclusivePromise::CreateAndResolve(true, + __func__); + }, + [](nsresult aError) { + return GenericNonExclusivePromise::CreateAndResolve(true, + __func__); + }); + + // 1. Allow final stats query to complete. + // 2. Tear down call, if necessary. We do this before we shut down the + // transport handler, so RTCP BYE can be sent. + // 3. Unhook from the signal handler (sigslot) for transport stuff. This must + // be done before we tear down the transport handler. + // 4. Tear down the transport handler, and deregister from PeerConnectionCtx. + // When we deregister from PeerConnectionCtx, our final stats (if any) + // will be stored. + MOZ_RELEASE_ASSERT(mSTSThread); + mFinalStatsQuery + ->Then(GetMainThreadSerialEventTarget(), __func__, + [callDestroyPromise]() mutable { return callDestroyPromise; }) + ->Then( + mSTSThread, __func__, + [signalHandler = std::move(mSignalHandler)]() mutable { + CSFLogDebug( + LOGTAG, + "Destroying PeerConnectionImpl::SignalHandler on STS thread"); + return GenericPromise::CreateAndResolve( + true, "PeerConnectionImpl::~SignalHandler"); + }) + ->Then( + GetMainThreadSerialEventTarget(), __func__, + [this, self = RefPtr<PeerConnectionImpl>(this)]() mutable { + CSFLogDebug(LOGTAG, "PCImpl->mTransportHandler::RemoveTransports"); + mTransportHandler->RemoveTransportsExcept(std::set<std::string>()); + if (mPrivateWindow) { + mTransportHandler->ExitPrivateMode(); + } + mTransportHandler = nullptr; + if (PeerConnectionCtx::isActive()) { + // If we're shutting down xpcom, this Instance will be unset + // before calling Close() on all remaining PCs, to avoid + // reentrancy. + PeerConnectionCtx::GetInstance()->RemovePeerConnection(mHandle); + } + }); + + return NS_OK; +} + +void PeerConnectionImpl::BreakCycles() { + for (auto& transceiver : mTransceivers) { + transceiver->BreakCycles(); + } + mTransceivers.Clear(); +} + +bool PeerConnectionImpl::HasPendingSetParameters() const { + for (const auto& transceiver : mTransceivers) { + if (transceiver->Sender()->HasPendingSetParameters()) { + return true; + } + } + return false; +} + +void PeerConnectionImpl::InvalidateLastReturnedParameters() { + for (const auto& transceiver : mTransceivers) { + transceiver->Sender()->InvalidateLastReturnedParameters(); + } +} + +nsresult PeerConnectionImpl::SetConfiguration( + const RTCConfiguration& aConfiguration) { + nsresult rv = mTransportHandler->SetIceConfig( + aConfiguration.mIceServers, aConfiguration.mIceTransportPolicy); + if (NS_WARN_IF(NS_FAILED(rv))) { + return rv; + } + + JsepBundlePolicy bundlePolicy; + switch (aConfiguration.mBundlePolicy) { + case dom::RTCBundlePolicy::Balanced: + bundlePolicy = kBundleBalanced; + break; + case dom::RTCBundlePolicy::Max_compat: + bundlePolicy = kBundleMaxCompat; + break; + case dom::RTCBundlePolicy::Max_bundle: + bundlePolicy = kBundleMaxBundle; + break; + default: + MOZ_CRASH(); + } + + // Ignore errors, since those ought to be handled earlier. + Unused << mJsepSession->SetBundlePolicy(bundlePolicy); + + if (!aConfiguration.mPeerIdentity.IsEmpty()) { + mPeerIdentity = new PeerIdentity(aConfiguration.mPeerIdentity); + mRequestedPrivacy = Some(PrincipalPrivacy::Private); + } + + auto proxyConfig = GetProxyConfig(); + if (proxyConfig) { + // Note that this could check if PrivacyRequested() is set on the PC and + // remove "webrtc" from the ALPN list. But that would only work if the PC + // was constructed with a peerIdentity constraint, not when isolated + // streams are added. If we ever need to signal to the proxy that the + // media is isolated, then we would need to restructure this code. + mTransportHandler->SetProxyConfig(std::move(*proxyConfig)); + } + + // Store the configuration for about:webrtc + StoreConfigurationForAboutWebrtc(aConfiguration); + + return NS_OK; +} + +RTCSctpTransport* PeerConnectionImpl::GetSctp() const { + return mSctpTransport.get(); +} + +void PeerConnectionImpl::RestartIce() { + RestartIceNoRenegotiationNeeded(); + // Update the negotiation-needed flag for connection. + UpdateNegotiationNeeded(); +} + +// webrtc-pc does not specify any situations where this is done, but the JSEP +// spec does, in some situations due to setConfiguration. +void PeerConnectionImpl::RestartIceNoRenegotiationNeeded() { + // Empty connection.[[LocalIceCredentialsToReplace]], and populate it with + // all ICE credentials (ice-ufrag and ice-pwd as defined in section 15.4 of + // [RFC5245]) found in connection.[[CurrentLocalDescription]], as well as all + // ICE credentials found in connection.[[PendingLocalDescription]]. + mLocalIceCredentialsToReplace = mJsepSession->GetLocalIceCredentials(); +} + +bool PeerConnectionImpl::PluginCrash(uint32_t aPluginID, + const nsAString& aPluginName) { + // fire an event to the DOM window if this is "ours" + if (!AnyCodecHasPluginID(aPluginID)) { + return false; + } + + CSFLogError(LOGTAG, "%s: Our plugin %llu crashed", __FUNCTION__, + static_cast<unsigned long long>(aPluginID)); + + RefPtr<Document> doc = mWindow->GetExtantDoc(); + if (!doc) { + NS_WARNING("Couldn't get document for PluginCrashed event!"); + return true; + } + + PluginCrashedEventInit init; + init.mPluginID = aPluginID; + init.mPluginName = aPluginName; + init.mSubmittedCrashReport = false; + init.mGmpPlugin = true; + init.mBubbles = true; + init.mCancelable = true; + + RefPtr<PluginCrashedEvent> event = + PluginCrashedEvent::Constructor(doc, u"PluginCrashed"_ns, init); + + event->SetTrusted(true); + event->WidgetEventPtr()->mFlags.mOnlyChromeDispatch = true; + + nsCOMPtr<nsPIDOMWindowInner> window = mWindow; + EventDispatcher::DispatchDOMEvent(window, nullptr, event, nullptr, nullptr); + + return true; +} + +void PeerConnectionImpl::RecordEndOfCallTelemetry() { + if (!mCallTelemStarted) { + return; + } + MOZ_RELEASE_ASSERT(!mCallTelemEnded, "Don't end telemetry twice"); + MOZ_RELEASE_ASSERT(mJsepSession, + "Call telemetry only starts after jsep session start"); + MOZ_RELEASE_ASSERT(mJsepSession->GetNegotiations() > 0, + "Call telemetry only starts after first connection"); + + // Bitmask used for WEBRTC/LOOP_CALL_TYPE telemetry reporting + static const uint32_t kAudioTypeMask = 1; + static const uint32_t kVideoTypeMask = 2; + static const uint32_t kDataChannelTypeMask = 4; + + // Report end-of-call Telemetry + Telemetry::Accumulate(Telemetry::WEBRTC_RENEGOTIATIONS, + mJsepSession->GetNegotiations() - 1); + Telemetry::Accumulate(Telemetry::WEBRTC_MAX_VIDEO_SEND_TRACK, + mMaxSending[SdpMediaSection::MediaType::kVideo]); + Telemetry::Accumulate(Telemetry::WEBRTC_MAX_VIDEO_RECEIVE_TRACK, + mMaxReceiving[SdpMediaSection::MediaType::kVideo]); + Telemetry::Accumulate(Telemetry::WEBRTC_MAX_AUDIO_SEND_TRACK, + mMaxSending[SdpMediaSection::MediaType::kAudio]); + Telemetry::Accumulate(Telemetry::WEBRTC_MAX_AUDIO_RECEIVE_TRACK, + mMaxReceiving[SdpMediaSection::MediaType::kAudio]); + // DataChannels appear in both Sending and Receiving + Telemetry::Accumulate(Telemetry::WEBRTC_DATACHANNEL_NEGOTIATED, + mMaxSending[SdpMediaSection::MediaType::kApplication]); + // Enumerated/bitmask: 1 = Audio, 2 = Video, 4 = DataChannel + // A/V = 3, A/V/D = 7, etc + uint32_t type = 0; + if (mMaxSending[SdpMediaSection::MediaType::kAudio] || + mMaxReceiving[SdpMediaSection::MediaType::kAudio]) { + type = kAudioTypeMask; + } + if (mMaxSending[SdpMediaSection::MediaType::kVideo] || + mMaxReceiving[SdpMediaSection::MediaType::kVideo]) { + type |= kVideoTypeMask; + } + if (mMaxSending[SdpMediaSection::MediaType::kApplication]) { + type |= kDataChannelTypeMask; + } + Telemetry::Accumulate(Telemetry::WEBRTC_CALL_TYPE, type); + + MOZ_RELEASE_ASSERT(mWindow); + auto found = sCallDurationTimers.find(mWindow->WindowID()); + if (found != sCallDurationTimers.end()) { + found->second.UnregisterConnection((type & kAudioTypeMask) || + (type & kVideoTypeMask)); + if (found->second.IsStopped()) { + sCallDurationTimers.erase(found); + } + } + mCallTelemEnded = true; +} + +DOMMediaStream* PeerConnectionImpl::GetReceiveStream( + const std::string& aId) const { + nsString wanted = NS_ConvertASCIItoUTF16(aId.c_str()); + for (auto& stream : mReceiveStreams) { + nsString id; + stream->GetId(id); + if (id == wanted) { + return stream; + } + } + return nullptr; +} + +DOMMediaStream* PeerConnectionImpl::CreateReceiveStream( + const std::string& aId) { + mReceiveStreams.AppendElement(new DOMMediaStream(mWindow)); + mReceiveStreams.LastElement()->AssignId(NS_ConvertASCIItoUTF16(aId.c_str())); + return mReceiveStreams.LastElement(); +} + +already_AddRefed<dom::Promise> PeerConnectionImpl::OnSetDescriptionSuccess( + dom::RTCSdpType aSdpType, bool aRemote, ErrorResult& aError) { + CSFLogDebug(LOGTAG, __FUNCTION__); + + RefPtr<dom::Promise> p = MakePromise(aError); + if (aError.Failed()) { + return nullptr; + } + + DoSetDescriptionSuccessPostProcessing(aSdpType, aRemote, p); + + return p.forget(); +} + +void PeerConnectionImpl::DoSetDescriptionSuccessPostProcessing( + dom::RTCSdpType aSdpType, bool aRemote, const RefPtr<dom::Promise>& aP) { + // Spec says we queue a task for all the stuff that ends up back in JS + GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction( + __func__, + [this, self = RefPtr<PeerConnectionImpl>(this), aSdpType, aRemote, aP] { + if (IsClosed()) { + // Yes, we do not settle the promise here. Yes, this is what the spec + // wants. + return; + } + + MOZ_ASSERT(mUncommittedJsepSession); + + // sRD/sLD needs to be redone in certain circumstances + bool needsRedo = HasPendingSetParameters(); + if (!needsRedo && aRemote && (aSdpType == dom::RTCSdpType::Offer)) { + for (auto& transceiver : mTransceivers) { + if (!mUncommittedJsepSession->GetTransceiver( + transceiver->GetJsepTransceiverId())) { + needsRedo = true; + break; + } + } + } + + if (needsRedo) { + // Spec says to abort, and re-do the sRD! + // This happens either when there is a SetParameters call in + // flight (that will race against the [[SendEncodings]] + // modification caused by sRD(offer)), or when addTrack has been + // called while sRD(offer) was in progress. + mUncommittedJsepSession.reset(mJsepSession->Clone()); + JsepSession::Result result; + if (aRemote) { + mUncommittedJsepSession->SetRemoteDescription( + ToJsepSdpType(aSdpType), mRemoteRequestedSDP); + } else { + mUncommittedJsepSession->SetLocalDescription( + ToJsepSdpType(aSdpType), mLocalRequestedSDP); + } + if (result.mError.isSome()) { + // wat + nsCString error( + "When redoing sRD/sLD because it raced against " + "addTrack or setParameters, we encountered a failure that " + "did not happen " + "the first time. This should never happen. The error was: "); + error += mUncommittedJsepSession->GetLastError().c_str(); + aP->MaybeRejectWithOperationError(error); + MOZ_ASSERT(false); + } else { + DoSetDescriptionSuccessPostProcessing(aSdpType, aRemote, aP); + } + return; + } + + for (auto& transceiver : mTransceivers) { + if (!mUncommittedJsepSession->GetTransceiver( + transceiver->GetJsepTransceiverId())) { + // sLD, or sRD(answer), just make sure the new transceiver is + // added, no need to re-do anything. + mUncommittedJsepSession->AddTransceiver( + transceiver->GetJsepTransceiver()); + } + } + + mJsepSession = std::move(mUncommittedJsepSession); + + auto newSignalingState = GetSignalingState(); + SyncFromJsep(); + if (aRemote || aSdpType == dom::RTCSdpType::Pranswer || + aSdpType == dom::RTCSdpType::Answer) { + InvalidateLastReturnedParameters(); + } + + // Section 4.4.1.5 Set the RTCSessionDescription: + if (aSdpType == dom::RTCSdpType::Rollback) { + // - step 4.5.10, type is rollback + RollbackRTCDtlsTransports(); + } else if (!(aRemote && aSdpType == dom::RTCSdpType::Offer)) { + // - step 4.5.9 type is not rollback + // - step 4.5.9.1 when remote is false + // - step 4.5.9.2.13 when remote is true, type answer or pranswer + // More simply: not rollback, and not for remote offers. + bool markAsStable = aSdpType == dom::RTCSdpType::Offer && + mSignalingState == RTCSignalingState::Stable; + UpdateRTCDtlsTransports(markAsStable); + } + + // Did we just apply a local description? + if (!aRemote) { + // We'd like to handle this in PeerConnectionImpl::UpdateNetworkState. + // Unfortunately, if the WiFi switch happens quickly, we never see + // that state change. We need to detect the ice restart here and + // reset the PeerConnectionImpl's stun addresses so they are + // regathered when PeerConnectionImpl::GatherIfReady is called. + if (mJsepSession->IsIceRestarting()) { + ResetStunAddrsForIceRestart(); + } + EnsureTransports(*mJsepSession); + } + + if (mJsepSession->GetState() == kJsepStateStable) { + if (aSdpType != dom::RTCSdpType::Rollback) { + // We need this initted for UpdateTransports + InitializeDataChannel(); + } + + // If we're rolling back a local offer, we might need to remove some + // transports, and stomp some MediaPipeline setup, but nothing further + // needs to be done. + UpdateTransports(*mJsepSession, mForceIceTcp); + if (NS_FAILED(UpdateMediaPipelines())) { + CSFLogError(LOGTAG, "Error Updating MediaPipelines"); + NS_ASSERTION( + false, + "Error Updating MediaPipelines in OnSetDescriptionSuccess()"); + aP->MaybeRejectWithOperationError("Error Updating MediaPipelines"); + } + + if (aSdpType != dom::RTCSdpType::Rollback) { + StartIceChecks(*mJsepSession); + } + + // Telemetry: record info on the current state of + // streams/renegotiations/etc Note: this code gets run on rollbacks as + // well! + + // Update the max channels used with each direction for each type + uint16_t receiving[SdpMediaSection::kMediaTypes]; + uint16_t sending[SdpMediaSection::kMediaTypes]; + mJsepSession->CountTracksAndDatachannels(receiving, sending); + for (size_t i = 0; i < SdpMediaSection::kMediaTypes; i++) { + if (mMaxReceiving[i] < receiving[i]) { + mMaxReceiving[i] = receiving[i]; + } + if (mMaxSending[i] < sending[i]) { + mMaxSending[i] = sending[i]; + } + } + } + + mPendingRemoteDescription = + mJsepSession->GetRemoteDescription(kJsepDescriptionPending); + mCurrentRemoteDescription = + mJsepSession->GetRemoteDescription(kJsepDescriptionCurrent); + mPendingLocalDescription = + mJsepSession->GetLocalDescription(kJsepDescriptionPending); + mCurrentLocalDescription = + mJsepSession->GetLocalDescription(kJsepDescriptionCurrent); + mPendingOfferer = mJsepSession->IsPendingOfferer(); + mCurrentOfferer = mJsepSession->IsCurrentOfferer(); + + if (aSdpType == dom::RTCSdpType::Answer) { + std::set<std::pair<std::string, std::string>> iceCredentials = + mJsepSession->GetLocalIceCredentials(); + std::vector<std::pair<std::string, std::string>> + iceCredentialsNotReplaced; + std::set_intersection(mLocalIceCredentialsToReplace.begin(), + mLocalIceCredentialsToReplace.end(), + iceCredentials.begin(), iceCredentials.end(), + std::back_inserter(iceCredentialsNotReplaced)); + + if (iceCredentialsNotReplaced.empty()) { + mLocalIceCredentialsToReplace.clear(); + } + } + + if (newSignalingState == RTCSignalingState::Stable) { + mNegotiationNeeded = false; + UpdateNegotiationNeeded(); + } + + // Spec does not actually tell us to do this, but that is probably a + // spec bug. + UpdateConnectionState(); + + JSErrorResult jrv; + if (newSignalingState != mSignalingState) { + mSignalingState = newSignalingState; + mPCObserver->OnStateChange(PCObserverStateType::SignalingState, jrv); + } + + if (aRemote) { + dom::RTCRtpReceiver::StreamAssociationChanges changes; + for (const auto& transceiver : mTransceivers) { + transceiver->Receiver()->UpdateStreams(&changes); + } + + for (const auto& receiver : changes.mReceiversToMute) { + // This sets the muted state for the recv track and all its clones. + receiver->SetTrackMuteFromRemoteSdp(); + } + + for (const auto& association : changes.mStreamAssociationsRemoved) { + RefPtr<DOMMediaStream> stream = + GetReceiveStream(association.mStreamId); + if (stream && stream->HasTrack(*association.mTrack)) { + stream->RemoveTrackInternal(association.mTrack); + } + } + + // TODO(Bug 1241291): For legacy event, remove eventually + std::vector<RefPtr<DOMMediaStream>> newStreams; + + for (const auto& association : changes.mStreamAssociationsAdded) { + RefPtr<DOMMediaStream> stream = + GetReceiveStream(association.mStreamId); + if (!stream) { + stream = CreateReceiveStream(association.mStreamId); + newStreams.push_back(stream); + } + + if (!stream->HasTrack(*association.mTrack)) { + stream->AddTrackInternal(association.mTrack); + } + } + + // Make sure to wait until after we've calculated track changes before + // doing this. + for (size_t i = 0; i < mTransceivers.Length();) { + auto& transceiver = mTransceivers[i]; + if (transceiver->ShouldRemove()) { + mTransceivers[i]->Close(); + mTransceivers[i]->SetRemovedFromPc(); + mTransceivers.RemoveElementAt(i); + } else { + ++i; + } + } + + for (const auto& trackEvent : changes.mTrackEvents) { + dom::Sequence<OwningNonNull<DOMMediaStream>> streams; + for (const auto& id : trackEvent.mStreamIds) { + RefPtr<DOMMediaStream> stream = GetReceiveStream(id); + if (!stream) { + MOZ_ASSERT(false); + continue; + } + if (!streams.AppendElement(*stream, fallible)) { + // XXX(Bug 1632090) Instead of extending the array 1-by-1 (which + // might involve multiple reallocations) and potentially + // crashing here, SetCapacity could be called outside the loop + // once. + mozalloc_handle_oom(0); + } + } + mPCObserver->FireTrackEvent(*trackEvent.mReceiver, streams, jrv); + } + + // TODO(Bug 1241291): Legacy event, remove eventually + for (const auto& stream : newStreams) { + mPCObserver->FireStreamEvent(*stream, jrv); + } + } + aP->MaybeResolveWithUndefined(); + })); +} + +void PeerConnectionImpl::OnSetDescriptionError() { + mUncommittedJsepSession = nullptr; +} + +RTCSignalingState PeerConnectionImpl::GetSignalingState() const { + switch (mJsepSession->GetState()) { + case kJsepStateStable: + return RTCSignalingState::Stable; + break; + case kJsepStateHaveLocalOffer: + return RTCSignalingState::Have_local_offer; + break; + case kJsepStateHaveRemoteOffer: + return RTCSignalingState::Have_remote_offer; + break; + case kJsepStateHaveLocalPranswer: + return RTCSignalingState::Have_local_pranswer; + break; + case kJsepStateHaveRemotePranswer: + return RTCSignalingState::Have_remote_pranswer; + break; + case kJsepStateClosed: + return RTCSignalingState::Closed; + break; + } + MOZ_CRASH("Invalid JSEP state"); +} + +bool PeerConnectionImpl::IsClosed() const { + return mSignalingState == RTCSignalingState::Closed; +} + +PeerConnectionWrapper::PeerConnectionWrapper(const std::string& handle) + : impl_(nullptr) { + if (PeerConnectionCtx::isActive()) { + impl_ = PeerConnectionCtx::GetInstance()->GetPeerConnection(handle); + } +} + +const RefPtr<MediaTransportHandler> PeerConnectionImpl::GetTransportHandler() + const { + return mTransportHandler; +} + +const std::string& PeerConnectionImpl::GetHandle() { return mHandle; } + +const std::string& PeerConnectionImpl::GetName() { + PC_AUTO_ENTER_API_CALL_NO_CHECK(); + return mName; +} + +void PeerConnectionImpl::CandidateReady(const std::string& candidate, + const std::string& transportId, + const std::string& ufrag) { + STAMP_TIMECARD(mTimeCard, "Ice Candidate gathered"); + PC_AUTO_ENTER_API_CALL_VOID_RETURN(false); + + if (mForceIceTcp && std::string::npos != candidate.find(" UDP ")) { + CSFLogWarn(LOGTAG, "Blocking local UDP candidate: %s", candidate.c_str()); + STAMP_TIMECARD(mTimeCard, "UDP Ice Candidate blocked"); + return; + } + + // One of the very few places we still use level; required by the JSEP API + uint16_t level = 0; + std::string mid; + bool skipped = false; + + if (mUncommittedJsepSession) { + // An sLD or sRD is in progress, and while that is the case, we need to add + // the candidate to both the current JSEP engine, and the uncommitted JSEP + // engine. We ignore errors because the spec says to only take into account + // the current/pending local descriptions when determining whether to + // surface the candidate to content, which does not take into account any + // in-progress sRD/sLD. + Unused << mUncommittedJsepSession->AddLocalIceCandidate( + candidate, transportId, ufrag, &level, &mid, &skipped); + } + + nsresult res = mJsepSession->AddLocalIceCandidate( + candidate, transportId, ufrag, &level, &mid, &skipped); + + if (NS_FAILED(res)) { + std::string errorString = mJsepSession->GetLastError(); + + STAMP_TIMECARD(mTimeCard, "Local Ice Candidate invalid"); + CSFLogError(LOGTAG, + "Failed to incorporate local candidate into SDP:" + " res = %u, candidate = %s, transport-id = %s," + " error = %s", + static_cast<unsigned>(res), candidate.c_str(), + transportId.c_str(), errorString.c_str()); + return; + } + + if (skipped) { + STAMP_TIMECARD(mTimeCard, "Local Ice Candidate skipped"); + CSFLogInfo(LOGTAG, + "Skipped adding local candidate %s (transport-id %s) " + "to SDP, this typically happens because the m-section " + "is bundled, which means it doesn't make sense for it " + "to have its own transport-related attributes.", + candidate.c_str(), transportId.c_str()); + return; + } + + mPendingLocalDescription = + mJsepSession->GetLocalDescription(kJsepDescriptionPending); + mCurrentLocalDescription = + mJsepSession->GetLocalDescription(kJsepDescriptionCurrent); + CSFLogInfo(LOGTAG, "Passing local candidate to content: %s", + candidate.c_str()); + SendLocalIceCandidateToContent(level, mid, candidate, ufrag); +} + +void PeerConnectionImpl::SendLocalIceCandidateToContent( + uint16_t level, const std::string& mid, const std::string& candidate, + const std::string& ufrag) { + STAMP_TIMECARD(mTimeCard, "Send Ice Candidate to content"); + JSErrorResult rv; + mPCObserver->OnIceCandidate(level, ObString(mid.c_str()), + ObString(candidate.c_str()), + ObString(ufrag.c_str()), rv); +} + +void PeerConnectionImpl::IceConnectionStateChange( + dom::RTCIceConnectionState domState) { + PC_AUTO_ENTER_API_CALL_VOID_RETURN(false); + + CSFLogDebug(LOGTAG, "%s: %d -> %d", __FUNCTION__, + static_cast<int>(mIceConnectionState), + static_cast<int>(domState)); + + if (domState == mIceConnectionState) { + // no work to be done since the states are the same. + // this can happen during ICE rollback situations. + return; + } + + mIceConnectionState = domState; + + // Would be nice if we had a means of converting one of these dom enums + // to a string that wasn't almost as much text as this switch statement... + switch (mIceConnectionState) { + case RTCIceConnectionState::New: + STAMP_TIMECARD(mTimeCard, "Ice state: new"); + break; + case RTCIceConnectionState::Checking: + // For telemetry + mIceStartTime = TimeStamp::Now(); + STAMP_TIMECARD(mTimeCard, "Ice state: checking"); + break; + case RTCIceConnectionState::Connected: + STAMP_TIMECARD(mTimeCard, "Ice state: connected"); + StartCallTelem(); + break; + case RTCIceConnectionState::Completed: + STAMP_TIMECARD(mTimeCard, "Ice state: completed"); + break; + case RTCIceConnectionState::Failed: + STAMP_TIMECARD(mTimeCard, "Ice state: failed"); + break; + case RTCIceConnectionState::Disconnected: + STAMP_TIMECARD(mTimeCard, "Ice state: disconnected"); + break; + case RTCIceConnectionState::Closed: + STAMP_TIMECARD(mTimeCard, "Ice state: closed"); + break; + default: + MOZ_ASSERT_UNREACHABLE("Unexpected mIceConnectionState!"); + } + + WrappableJSErrorResult rv; + mPCObserver->OnStateChange(PCObserverStateType::IceConnectionState, rv); + UpdateConnectionState(); +} + +void PeerConnectionImpl::OnCandidateFound(const std::string& aTransportId, + const CandidateInfo& aCandidateInfo) { + if (mStunAddrsRequest && !aCandidateInfo.mMDNSAddress.empty()) { + MOZ_ASSERT(!aCandidateInfo.mActualAddress.empty()); + + if (mCanRegisterMDNSHostnamesDirectly) { + auto itor = mRegisteredMDNSHostnames.find(aCandidateInfo.mMDNSAddress); + + // We'll see the address twice if we're generating both UDP and TCP + // candidates. + if (itor == mRegisteredMDNSHostnames.end()) { + mRegisteredMDNSHostnames.insert(aCandidateInfo.mMDNSAddress); + mStunAddrsRequest->SendRegisterMDNSHostname( + nsCString(aCandidateInfo.mMDNSAddress.c_str()), + nsCString(aCandidateInfo.mActualAddress.c_str())); + } + } else { + mMDNSHostnamesToRegister.emplace(aCandidateInfo.mMDNSAddress, + aCandidateInfo.mActualAddress); + } + } + + if (!aCandidateInfo.mDefaultHostRtp.empty()) { + UpdateDefaultCandidate(aCandidateInfo.mDefaultHostRtp, + aCandidateInfo.mDefaultPortRtp, + aCandidateInfo.mDefaultHostRtcp, + aCandidateInfo.mDefaultPortRtcp, aTransportId); + } + CandidateReady(aCandidateInfo.mCandidate, aTransportId, + aCandidateInfo.mUfrag); +} + +void PeerConnectionImpl::IceGatheringStateChange( + dom::RTCIceGatheringState state) { + PC_AUTO_ENTER_API_CALL_VOID_RETURN(false); + + CSFLogDebug(LOGTAG, "%s %d", __FUNCTION__, static_cast<int>(state)); + if (mIceGatheringState == state) { + return; + } + + mIceGatheringState = state; + + // Would be nice if we had a means of converting one of these dom enums + // to a string that wasn't almost as much text as this switch statement... + switch (mIceGatheringState) { + case RTCIceGatheringState::New: + STAMP_TIMECARD(mTimeCard, "Ice gathering state: new"); + break; + case RTCIceGatheringState::Gathering: + STAMP_TIMECARD(mTimeCard, "Ice gathering state: gathering"); + break; + case RTCIceGatheringState::Complete: + STAMP_TIMECARD(mTimeCard, "Ice gathering state: complete"); + break; + default: + MOZ_ASSERT_UNREACHABLE("Unexpected mIceGatheringState!"); + } + + JSErrorResult rv; + mPCObserver->OnStateChange(PCObserverStateType::IceGatheringState, rv); +} + +void PeerConnectionImpl::UpdateDefaultCandidate( + const std::string& defaultAddr, uint16_t defaultPort, + const std::string& defaultRtcpAddr, uint16_t defaultRtcpPort, + const std::string& transportId) { + CSFLogDebug(LOGTAG, "%s", __FUNCTION__); + mJsepSession->UpdateDefaultCandidate( + defaultAddr, defaultPort, defaultRtcpAddr, defaultRtcpPort, transportId); + if (mUncommittedJsepSession) { + mUncommittedJsepSession->UpdateDefaultCandidate( + defaultAddr, defaultPort, defaultRtcpAddr, defaultRtcpPort, + transportId); + } +} + +static UniquePtr<dom::RTCStatsCollection> GetDataChannelStats_s( + const RefPtr<DataChannelConnection>& aDataConnection, + const DOMHighResTimeStamp aTimestamp) { + UniquePtr<dom::RTCStatsCollection> report(new dom::RTCStatsCollection); + if (aDataConnection) { + aDataConnection->AppendStatsToReport(report, aTimestamp); + } + return report; +} + +RefPtr<dom::RTCStatsPromise> PeerConnectionImpl::GetDataChannelStats( + const RefPtr<DataChannelConnection>& aDataChannelConnection, + const DOMHighResTimeStamp aTimestamp) { + // Gather stats from DataChannels + return InvokeAsync( + GetMainThreadSerialEventTarget(), __func__, + [aDataChannelConnection, aTimestamp]() { + return dom::RTCStatsPromise::CreateAndResolve( + GetDataChannelStats_s(aDataChannelConnection, aTimestamp), + __func__); + }); +} + +void PeerConnectionImpl::CollectConduitTelemetryData() { + MOZ_ASSERT(NS_IsMainThread()); + + nsTArray<RefPtr<VideoSessionConduit>> conduits; + for (const auto& transceiver : mTransceivers) { + if (RefPtr<MediaSessionConduit> conduit = transceiver->GetConduit()) { + conduit->AsVideoSessionConduit().apply( + [&](const auto& aVideo) { conduits.AppendElement(aVideo); }); + } + } + + if (!conduits.IsEmpty() && mCall) { + mCall->mCallThread->Dispatch( + NS_NewRunnableFunction(__func__, [conduits = std::move(conduits)] { + for (const auto& conduit : conduits) { + conduit->CollectTelemetryData(); + } + })); + } +} + +nsTArray<dom::RTCCodecStats> PeerConnectionImpl::GetCodecStats( + DOMHighResTimeStamp aNow) { + MOZ_ASSERT(NS_IsMainThread()); + nsTArray<dom::RTCCodecStats> result; + + struct CodecComparator { + bool operator()(const JsepCodecDescription* aA, + const JsepCodecDescription* aB) const { + return aA->StatsId() < aB->StatsId(); + } + }; + + // transportId -> codec; per direction (whether the codecType + // shall be "encode", "decode" or absent (if a codec exists in both maps for a + // transport)). These do the bookkeeping to ensure codec stats get coalesced + // to transport level. + std::map<std::string, std::set<JsepCodecDescription*, CodecComparator>> + sendCodecMap; + std::map<std::string, std::set<JsepCodecDescription*, CodecComparator>> + recvCodecMap; + + // Find all JsepCodecDescription instances we want to turn into codec stats. + for (const auto& transceiver : mTransceivers) { + // TODO: Grab these from the JSEP transceivers instead + auto sendCodecs = transceiver->GetNegotiatedSendCodecs(); + auto recvCodecs = transceiver->GetNegotiatedRecvCodecs(); + + const std::string transportId = transceiver->GetTransportId(); + // This ensures both codec maps have the same size. + auto& sendMap = sendCodecMap[transportId]; + auto& recvMap = recvCodecMap[transportId]; + + sendCodecs.apply([&](const auto& aCodecs) { + for (const auto& codec : aCodecs) { + sendMap.insert(codec.get()); + } + }); + recvCodecs.apply([&](const auto& aCodecs) { + for (const auto& codec : aCodecs) { + recvMap.insert(codec.get()); + } + }); + } + + auto createCodecStat = [&](const JsepCodecDescription* aCodec, + const nsString& aTransportId, + Maybe<RTCCodecType> aCodecType) { + uint16_t pt; + { + DebugOnly<bool> rv = aCodec->GetPtAsInt(&pt); + MOZ_ASSERT(rv); + } + nsString mimeType; + mimeType.AppendPrintf( + "%s/%s", aCodec->Type() == SdpMediaSection::kVideo ? "video" : "audio", + aCodec->mName.c_str()); + nsString id = aTransportId; + id.Append(u"_"); + id.Append(aCodec->StatsId()); + + dom::RTCCodecStats codec; + codec.mId.Construct(std::move(id)); + codec.mTimestamp.Construct(aNow); + codec.mType.Construct(RTCStatsType::Codec); + codec.mPayloadType = pt; + if (aCodecType) { + codec.mCodecType.Construct(*aCodecType); + } + codec.mTransportId = aTransportId; + codec.mMimeType = std::move(mimeType); + codec.mClockRate.Construct(aCodec->mClock); + if (aCodec->Type() == SdpMediaSection::MediaType::kAudio) { + codec.mChannels.Construct(aCodec->mChannels); + } + if (aCodec->mSdpFmtpLine) { + codec.mSdpFmtpLine.Construct( + NS_ConvertUTF8toUTF16(aCodec->mSdpFmtpLine->c_str())); + } + + result.AppendElement(std::move(codec)); + }; + + // Create codec stats for the gathered codec descriptions, sorted primarily + // by transportId, secondarily by payload type (from StatsId()). + for (const auto& [transportId, sendCodecs] : sendCodecMap) { + const auto& recvCodecs = recvCodecMap[transportId]; + const nsString tid = NS_ConvertASCIItoUTF16(transportId); + AutoTArray<JsepCodecDescription*, 16> bidirectionalCodecs; + AutoTArray<JsepCodecDescription*, 16> unidirectionalCodecs; + std::set_intersection(sendCodecs.cbegin(), sendCodecs.cend(), + recvCodecs.cbegin(), recvCodecs.cend(), + MakeBackInserter(bidirectionalCodecs), + CodecComparator()); + std::set_symmetric_difference(sendCodecs.cbegin(), sendCodecs.cend(), + recvCodecs.cbegin(), recvCodecs.cend(), + MakeBackInserter(unidirectionalCodecs), + CodecComparator()); + for (const auto* codec : bidirectionalCodecs) { + createCodecStat(codec, tid, Nothing()); + } + for (const auto* codec : unidirectionalCodecs) { + createCodecStat( + codec, tid, + Some(codec->mDirection == sdp::kSend ? RTCCodecType::Encode + : RTCCodecType::Decode)); + } + } + + return result; +} + +RefPtr<dom::RTCStatsReportPromise> PeerConnectionImpl::GetStats( + dom::MediaStreamTrack* aSelector, bool aInternalStats) { + MOZ_ASSERT(NS_IsMainThread()); + + if (mFinalStatsQuery) { + // This case should be _extremely_ rare; this will basically only happen + // when WebrtcGlobalInformation tries to get our stats while we are tearing + // down. + return mFinalStatsQuery->Then( + GetMainThreadSerialEventTarget(), __func__, + [this, self = RefPtr<PeerConnectionImpl>(this)]() { + UniquePtr<dom::RTCStatsReportInternal> finalStats = + MakeUnique<dom::RTCStatsReportInternal>(); + // Might not be set if this encountered some error. + if (mFinalStats) { + *finalStats = *mFinalStats; + } + return RTCStatsReportPromise::CreateAndResolve(std::move(finalStats), + __func__); + }); + } + + nsTArray<RefPtr<dom::RTCStatsPromise>> promises; + DOMHighResTimeStamp now = mTimestampMaker.GetNow().ToDom(); + + nsTArray<dom::RTCCodecStats> codecStats = GetCodecStats(now); + std::set<std::string> transportIds; + + if (!aSelector) { + // There might not be any senders/receivers if we're DataChannel only, so we + // don't handle the null selector case in the loop below. + transportIds.insert(""); + } + + nsTArray< + std::tuple<RTCRtpTransceiver*, RefPtr<RTCStatsPromise::AllPromiseType>>> + transceiverStatsPromises; + for (const auto& transceiver : mTransceivers) { + const bool sendSelected = transceiver->Sender()->HasTrack(aSelector); + const bool recvSelected = transceiver->Receiver()->HasTrack(aSelector); + if (!sendSelected && !recvSelected) { + continue; + } + + if (aSelector) { + transportIds.insert(transceiver->GetTransportId()); + } + + nsTArray<RefPtr<RTCStatsPromise>> rtpStreamPromises; + // Get all rtp stream stats for the given selector. Then filter away any + // codec stat not related to the selector, and assign codec ids to the + // stream stats. + // Skips the ICE stats; we do our own queries based on |transportIds| to + // avoid duplicates + if (sendSelected) { + rtpStreamPromises.AppendElements( + transceiver->Sender()->GetStatsInternal(true)); + } + if (recvSelected) { + rtpStreamPromises.AppendElements( + transceiver->Receiver()->GetStatsInternal(true)); + } + transceiverStatsPromises.AppendElement( + std::make_tuple(transceiver.get(), + RTCStatsPromise::All(GetMainThreadSerialEventTarget(), + rtpStreamPromises))); + } + + promises.AppendElement(RTCRtpTransceiver::ApplyCodecStats( + std::move(codecStats), std::move(transceiverStatsPromises))); + + for (const auto& transportId : transportIds) { + promises.AppendElement(mTransportHandler->GetIceStats(transportId, now)); + } + + promises.AppendElement(GetDataChannelStats(mDataConnection, now)); + + auto pcStatsCollection = MakeUnique<dom::RTCStatsCollection>(); + RTCPeerConnectionStats pcStats; + pcStats.mTimestamp.Construct(now); + pcStats.mType.Construct(RTCStatsType::Peer_connection); + pcStats.mId.Construct(NS_ConvertUTF8toUTF16(mHandle.c_str())); + pcStats.mDataChannelsOpened.Construct(mDataChannelsOpened); + pcStats.mDataChannelsClosed.Construct(mDataChannelsClosed); + if (!pcStatsCollection->mPeerConnectionStats.AppendElement(std::move(pcStats), + fallible)) { + mozalloc_handle_oom(0); + } + promises.AppendElement(RTCStatsPromise::CreateAndResolve( + std::move(pcStatsCollection), __func__)); + + // This is what we're going to return; all the stuff in |promises| will be + // accumulated here. + UniquePtr<dom::RTCStatsReportInternal> report( + new dom::RTCStatsReportInternal); + report->mPcid = NS_ConvertASCIItoUTF16(mName.c_str()); + if (mWindow && mWindow->GetBrowsingContext()) { + report->mBrowserId = mWindow->GetBrowsingContext()->BrowserId(); + } + report->mConfiguration.Construct(mJsConfiguration); + // TODO(bug 1589416): We need to do better here. + if (!mIceStartTime.IsNull()) { + report->mCallDurationMs.Construct( + (TimeStamp::Now() - mIceStartTime).ToMilliseconds()); + } + report->mIceRestarts = mIceRestartCount; + report->mIceRollbacks = mIceRollbackCount; + report->mClosed = false; + report->mTimestamp = now; + + if (aInternalStats && mJsepSession) { + for (const auto& candidate : mRawTrickledCandidates) { + if (!report->mRawRemoteCandidates.AppendElement( + NS_ConvertASCIItoUTF16(candidate.c_str()), fallible)) { + // XXX(Bug 1632090) Instead of extending the array 1-by-1 (which might + // involve multiple reallocations) and potentially crashing here, + // SetCapacity could be called outside the loop once. + mozalloc_handle_oom(0); + } + } + + if (mJsepSession) { + // TODO we probably should report Current and Pending SDPs here + // separately. Plus the raw SDP we got from JS (mLocalRequestedSDP). + // And if it's the offer or answer would also be nice. + std::string localDescription = + mJsepSession->GetLocalDescription(kJsepDescriptionPendingOrCurrent); + std::string remoteDescription = + mJsepSession->GetRemoteDescription(kJsepDescriptionPendingOrCurrent); + if (!report->mSdpHistory.AppendElements(mSdpHistory, fallible)) { + mozalloc_handle_oom(0); + } + if (mJsepSession->IsPendingOfferer().isSome()) { + report->mOfferer.Construct(*mJsepSession->IsPendingOfferer()); + } else if (mJsepSession->IsCurrentOfferer().isSome()) { + report->mOfferer.Construct(*mJsepSession->IsCurrentOfferer()); + } else { + // Silly. + report->mOfferer.Construct(false); + } + } + } + + return dom::RTCStatsPromise::All(GetMainThreadSerialEventTarget(), promises) + ->Then( + GetMainThreadSerialEventTarget(), __func__, + [report = std::move(report), idGen = mIdGenerator]( + nsTArray<UniquePtr<dom::RTCStatsCollection>> aStats) mutable { + idGen->RewriteIds(std::move(aStats), report.get()); + return dom::RTCStatsReportPromise::CreateAndResolve( + std::move(report), __func__); + }, + [](nsresult rv) { + return dom::RTCStatsReportPromise::CreateAndReject(rv, __func__); + }); +} + +void PeerConnectionImpl::RecordIceRestartStatistics(JsepSdpType type) { + switch (type) { + case mozilla::kJsepSdpOffer: + case mozilla::kJsepSdpPranswer: + break; + case mozilla::kJsepSdpAnswer: + ++mIceRestartCount; + break; + case mozilla::kJsepSdpRollback: + ++mIceRollbackCount; + break; + } +} + +void PeerConnectionImpl::StoreConfigurationForAboutWebrtc( + const dom::RTCConfiguration& aConfig) { + // This will only be called once, when the PeerConnection is initially + // configured, at least until setConfiguration is implemented + // see https://bugzilla.mozilla.org/show_bug.cgi?id=1253706 + // @TODO bug 1739451 call this from setConfiguration + mJsConfiguration.mIceServers.Clear(); + for (const auto& server : aConfig.mIceServers) { + RTCIceServerInternal internal; + internal.mCredentialProvided = server.mCredential.WasPassed(); + internal.mUserNameProvided = server.mUsername.WasPassed(); + if (server.mUrl.WasPassed()) { + if (!internal.mUrls.AppendElement(server.mUrl.Value(), fallible)) { + mozalloc_handle_oom(0); + } + } + if (server.mUrls.WasPassed()) { + for (const auto& url : server.mUrls.Value().GetAsStringSequence()) { + if (!internal.mUrls.AppendElement(url, fallible)) { + mozalloc_handle_oom(0); + } + } + } + if (!mJsConfiguration.mIceServers.AppendElement(internal, fallible)) { + mozalloc_handle_oom(0); + } + } + mJsConfiguration.mSdpSemantics.Reset(); + if (aConfig.mSdpSemantics.WasPassed()) { + mJsConfiguration.mSdpSemantics.Construct(aConfig.mSdpSemantics.Value()); + } + + mJsConfiguration.mIceTransportPolicy.Reset(); + mJsConfiguration.mIceTransportPolicy.Construct(aConfig.mIceTransportPolicy); + mJsConfiguration.mBundlePolicy.Reset(); + mJsConfiguration.mBundlePolicy.Construct(aConfig.mBundlePolicy); + mJsConfiguration.mPeerIdentityProvided = !aConfig.mPeerIdentity.IsEmpty(); + mJsConfiguration.mCertificatesProvided = !aConfig.mCertificates.Length(); +} + +dom::Sequence<dom::RTCSdpParsingErrorInternal> +PeerConnectionImpl::GetLastSdpParsingErrors() const { + const auto& sdpErrors = mJsepSession->GetLastSdpParsingErrors(); + dom::Sequence<dom::RTCSdpParsingErrorInternal> domErrors; + if (!domErrors.SetCapacity(domErrors.Length(), fallible)) { + mozalloc_handle_oom(0); + } + for (const auto& error : sdpErrors) { + mozilla::dom::RTCSdpParsingErrorInternal internal; + internal.mLineNumber = error.first; + if (!AppendASCIItoUTF16(MakeStringSpan(error.second.c_str()), + internal.mError, fallible)) { + mozalloc_handle_oom(0); + } + if (!domErrors.AppendElement(std::move(internal), fallible)) { + mozalloc_handle_oom(0); + } + } + return domErrors; +} + +// Telemetry for when calls start +void PeerConnectionImpl::StartCallTelem() { + if (mCallTelemStarted) { + return; + } + MOZ_RELEASE_ASSERT(mWindow); + uint64_t windowId = mWindow->WindowID(); + auto found = sCallDurationTimers.find(windowId); + if (found == sCallDurationTimers.end()) { + found = + sCallDurationTimers.emplace(windowId, PeerConnectionAutoTimer()).first; + } + found->second.RegisterConnection(); + mCallTelemStarted = true; + + // Increment session call counter + // If we want to track Loop calls independently here, we need two + // histograms. + // + // NOTE: As of bug 1654248 landing we are no longer counting renegotiations + // as separate calls. Expect numbers to drop compared to + // WEBRTC_CALL_COUNT_2. + Telemetry::Accumulate(Telemetry::WEBRTC_CALL_COUNT_3, 1); +} + +void PeerConnectionImpl::StunAddrsHandler::OnMDNSQueryComplete( + const nsCString& hostname, const Maybe<nsCString>& address) { + MOZ_ASSERT(NS_IsMainThread()); + PeerConnectionWrapper pcw(mPcHandle); + if (!pcw.impl()) { + return; + } + auto itor = pcw.impl()->mQueriedMDNSHostnames.find(hostname.BeginReading()); + if (itor != pcw.impl()->mQueriedMDNSHostnames.end()) { + if (address) { + for (auto& cand : itor->second) { + // Replace obfuscated address with actual address + std::string obfuscatedAddr = cand.mTokenizedCandidate[4]; + cand.mTokenizedCandidate[4] = address->BeginReading(); + std::ostringstream o; + for (size_t i = 0; i < cand.mTokenizedCandidate.size(); ++i) { + o << cand.mTokenizedCandidate[i]; + if (i + 1 != cand.mTokenizedCandidate.size()) { + o << " "; + } + } + std::string mungedCandidate = o.str(); + pcw.impl()->StampTimecard("Done looking up mDNS name"); + pcw.impl()->mTransportHandler->AddIceCandidate( + cand.mTransportId, mungedCandidate, cand.mUfrag, obfuscatedAddr); + } + } else { + pcw.impl()->StampTimecard("Failed looking up mDNS name"); + } + pcw.impl()->mQueriedMDNSHostnames.erase(itor); + } +} + +void PeerConnectionImpl::StunAddrsHandler::OnStunAddrsAvailable( + const mozilla::net::NrIceStunAddrArray& addrs) { + CSFLogInfo(LOGTAG, "%s: receiving (%d) stun addrs", __FUNCTION__, + (int)addrs.Length()); + PeerConnectionWrapper pcw(mPcHandle); + if (!pcw.impl()) { + return; + } + pcw.impl()->mStunAddrs = addrs.Clone(); + pcw.impl()->mLocalAddrsRequestState = STUN_ADDR_REQUEST_COMPLETE; + pcw.impl()->FlushIceCtxOperationQueueIfReady(); + // If parent process returns 0 STUN addresses, change ICE connection + // state to failed. + if (!pcw.impl()->mStunAddrs.Length()) { + pcw.impl()->IceConnectionStateChange(dom::RTCIceConnectionState::Failed); + } +} + +void PeerConnectionImpl::InitLocalAddrs() { + if (mLocalAddrsRequestState == STUN_ADDR_REQUEST_PENDING) { + return; + } + if (mStunAddrsRequest) { + mLocalAddrsRequestState = STUN_ADDR_REQUEST_PENDING; + mStunAddrsRequest->SendGetStunAddrs(); + } else { + mLocalAddrsRequestState = STUN_ADDR_REQUEST_COMPLETE; + } +} + +bool PeerConnectionImpl::ShouldForceProxy() const { + if (Preferences::GetBool("media.peerconnection.ice.proxy_only", false)) { + return true; + } + + bool isPBM = false; + // This complicated null check is being extra conservative to avoid + // introducing crashes. It may not be needed. + if (mWindow && mWindow->GetExtantDoc() && + mWindow->GetExtantDoc()->GetPrincipal() && + mWindow->GetExtantDoc() + ->GetPrincipal() + ->OriginAttributesRef() + .mPrivateBrowsingId > 0) { + isPBM = true; + } + + if (isPBM && Preferences::GetBool( + "media.peerconnection.ice.proxy_only_if_pbmode", false)) { + return true; + } + + if (!Preferences::GetBool( + "media.peerconnection.ice.proxy_only_if_behind_proxy", false)) { + return false; + } + + // Ok, we're supposed to be proxy_only, but only if a proxy is configured. + // Let's just see if the document was loaded via a proxy. + + nsCOMPtr<nsIHttpChannelInternal> httpChannelInternal = GetChannel(); + if (!httpChannelInternal) { + return false; + } + + bool proxyUsed = false; + Unused << httpChannelInternal->GetIsProxyUsed(&proxyUsed); + return proxyUsed; +} + +void PeerConnectionImpl::EnsureTransports(const JsepSession& aSession) { + mJsepSession->ForEachTransceiver([this, + self = RefPtr<PeerConnectionImpl>(this)]( + const JsepTransceiver& transceiver) { + if (transceiver.HasOwnTransport()) { + mTransportHandler->EnsureProvisionalTransport( + transceiver.mTransport.mTransportId, + transceiver.mTransport.mLocalUfrag, transceiver.mTransport.mLocalPwd, + transceiver.mTransport.mComponents); + } + }); + + GatherIfReady(); +} + +void PeerConnectionImpl::UpdateRTCDtlsTransports(bool aMarkAsStable) { + mJsepSession->ForEachTransceiver( + [this, self = RefPtr<PeerConnectionImpl>(this)]( + const JsepTransceiver& jsepTransceiver) { + std::string transportId = jsepTransceiver.mTransport.mTransportId; + if (transportId.empty()) { + return; + } + if (!mTransportIdToRTCDtlsTransport.count(transportId)) { + mTransportIdToRTCDtlsTransport.emplace( + transportId, new RTCDtlsTransport(GetParentObject())); + } + }); + + for (auto& transceiver : mTransceivers) { + std::string transportId = transceiver->GetTransportId(); + if (transportId.empty()) { + continue; + } + if (mTransportIdToRTCDtlsTransport.count(transportId)) { + transceiver->SetDtlsTransport(mTransportIdToRTCDtlsTransport[transportId], + aMarkAsStable); + } + } + + // Spec says we only update the RTCSctpTransport when negotiation completes +} + +void PeerConnectionImpl::RollbackRTCDtlsTransports() { + for (auto& transceiver : mTransceivers) { + transceiver->RollbackToStableDtlsTransport(); + } +} + +void PeerConnectionImpl::RemoveRTCDtlsTransportsExcept( + const std::set<std::string>& aTransportIds) { + for (auto iter = mTransportIdToRTCDtlsTransport.begin(); + iter != mTransportIdToRTCDtlsTransport.end();) { + if (!aTransportIds.count(iter->first)) { + iter = mTransportIdToRTCDtlsTransport.erase(iter); + } else { + ++iter; + } + } +} + +nsresult PeerConnectionImpl::UpdateTransports(const JsepSession& aSession, + const bool forceIceTcp) { + std::set<std::string> finalTransports; + Maybe<std::string> sctpTransport; + mJsepSession->ForEachTransceiver( + [&, this, self = RefPtr<PeerConnectionImpl>(this)]( + const JsepTransceiver& transceiver) { + if (transceiver.GetMediaType() == SdpMediaSection::kApplication && + transceiver.HasTransport()) { + sctpTransport = Some(transceiver.mTransport.mTransportId); + } + + if (transceiver.HasOwnTransport()) { + finalTransports.insert(transceiver.mTransport.mTransportId); + UpdateTransport(transceiver, forceIceTcp); + } + }); + + // clean up the unused RTCDtlsTransports + RemoveRTCDtlsTransportsExcept(finalTransports); + + mTransportHandler->RemoveTransportsExcept(finalTransports); + + for (const auto& transceiverImpl : mTransceivers) { + transceiverImpl->UpdateTransport(); + } + + if (sctpTransport.isSome()) { + auto it = mTransportIdToRTCDtlsTransport.find(*sctpTransport); + if (it == mTransportIdToRTCDtlsTransport.end()) { + // What? + MOZ_ASSERT(false); + return NS_ERROR_FAILURE; + } + if (!mDataConnection) { + // What? + MOZ_ASSERT(false); + return NS_ERROR_FAILURE; + } + RefPtr<RTCDtlsTransport> dtlsTransport = it->second; + // Why on earth does the spec use a floating point for this? + double maxMessageSize = + static_cast<double>(mDataConnection->GetMaxMessageSize()); + Nullable<uint16_t> maxChannels; + + if (!mSctpTransport) { + mSctpTransport = new RTCSctpTransport(GetParentObject(), *dtlsTransport, + maxMessageSize, maxChannels); + } else { + mSctpTransport->SetTransport(*dtlsTransport); + mSctpTransport->SetMaxMessageSize(maxMessageSize); + mSctpTransport->SetMaxChannels(maxChannels); + } + } else { + mSctpTransport = nullptr; + } + + return NS_OK; +} + +void PeerConnectionImpl::UpdateTransport(const JsepTransceiver& aTransceiver, + bool aForceIceTcp) { + std::string ufrag; + std::string pwd; + std::vector<std::string> candidates; + size_t components = 0; + + const JsepTransport& transport = aTransceiver.mTransport; + unsigned level = aTransceiver.GetLevel(); + + CSFLogDebug(LOGTAG, "ACTIVATING TRANSPORT! - PC %s: level=%u components=%u", + mHandle.c_str(), (unsigned)level, + (unsigned)transport.mComponents); + + ufrag = transport.mIce->GetUfrag(); + pwd = transport.mIce->GetPassword(); + candidates = transport.mIce->GetCandidates(); + components = transport.mComponents; + if (aForceIceTcp) { + candidates.erase( + std::remove_if(candidates.begin(), candidates.end(), + [](const std::string& s) { + return s.find(" UDP ") != std::string::npos || + s.find(" udp ") != std::string::npos; + }), + candidates.end()); + } + + nsTArray<uint8_t> keyDer; + nsTArray<uint8_t> certDer; + nsresult rv = Identity()->Serialize(&keyDer, &certDer); + if (NS_FAILED(rv)) { + CSFLogError(LOGTAG, "%s: Failed to serialize DTLS identity: %d", + __FUNCTION__, (int)rv); + return; + } + + DtlsDigestList digests; + for (const auto& fingerprint : + transport.mDtls->GetFingerprints().mFingerprints) { + std::ostringstream ss; + ss << fingerprint.hashFunc; + digests.emplace_back(ss.str(), fingerprint.fingerprint); + } + + mTransportHandler->ActivateTransport( + transport.mTransportId, transport.mLocalUfrag, transport.mLocalPwd, + components, ufrag, pwd, keyDer, certDer, Identity()->auth_type(), + transport.mDtls->GetRole() == JsepDtlsTransport::kJsepDtlsClient, digests, + PrivacyRequested()); + + for (auto& candidate : candidates) { + AddIceCandidate("candidate:" + candidate, transport.mTransportId, ufrag); + } +} + +nsresult PeerConnectionImpl::UpdateMediaPipelines() { + for (RefPtr<RTCRtpTransceiver>& transceiver : mTransceivers) { + transceiver->ResetSync(); + } + + for (RefPtr<RTCRtpTransceiver>& transceiver : mTransceivers) { + if (!transceiver->IsVideo()) { + nsresult rv = transceiver->SyncWithMatchingVideoConduits(mTransceivers); + if (NS_FAILED(rv)) { + return rv; + } + } + + transceiver->UpdatePrincipalPrivacy(PrivacyRequested() + ? PrincipalPrivacy::Private + : PrincipalPrivacy::NonPrivate); + + nsresult rv = transceiver->UpdateConduit(); + if (NS_FAILED(rv)) { + return rv; + } + } + + return NS_OK; +} + +void PeerConnectionImpl::StartIceChecks(const JsepSession& aSession) { + MOZ_ASSERT(NS_IsMainThread()); + + if (!mCanRegisterMDNSHostnamesDirectly) { + for (auto& pair : mMDNSHostnamesToRegister) { + mRegisteredMDNSHostnames.insert(pair.first); + mStunAddrsRequest->SendRegisterMDNSHostname( + nsCString(pair.first.c_str()), nsCString(pair.second.c_str())); + } + mMDNSHostnamesToRegister.clear(); + mCanRegisterMDNSHostnamesDirectly = true; + } + + std::vector<std::string> attributes; + if (aSession.RemoteIsIceLite()) { + attributes.push_back("ice-lite"); + } + + if (!aSession.GetIceOptions().empty()) { + attributes.push_back("ice-options:"); + for (const auto& option : aSession.GetIceOptions()) { + attributes.back() += option + ' '; + } + } + + nsCOMPtr<nsIRunnable> runnable( + WrapRunnable(mTransportHandler, &MediaTransportHandler::StartIceChecks, + aSession.IsIceControlling(), attributes)); + + PerformOrEnqueueIceCtxOperation(runnable); +} + +bool PeerConnectionImpl::GetPrefDefaultAddressOnly() const { + MOZ_ASSERT(NS_IsMainThread()); + + uint64_t winId = mWindow->WindowID(); + + bool default_address_only = Preferences::GetBool( + "media.peerconnection.ice.default_address_only", false); + default_address_only |= + !MediaManager::Get()->IsActivelyCapturingOrHasAPermission(winId); + return default_address_only; +} + +bool PeerConnectionImpl::GetPrefObfuscateHostAddresses() const { + MOZ_ASSERT(NS_IsMainThread()); + + uint64_t winId = mWindow->WindowID(); + + bool obfuscate_host_addresses = Preferences::GetBool( + "media.peerconnection.ice.obfuscate_host_addresses", false); + obfuscate_host_addresses &= + !MediaManager::Get()->IsActivelyCapturingOrHasAPermission(winId); + obfuscate_host_addresses &= !PeerConnectionImpl::HostnameInPref( + "media.peerconnection.ice.obfuscate_host_addresses.blocklist", mHostname); + obfuscate_host_addresses &= XRE_IsContentProcess(); + + return obfuscate_host_addresses; +} + +PeerConnectionImpl::SignalHandler::SignalHandler(PeerConnectionImpl* aPc, + MediaTransportHandler* aSource) + : mHandle(aPc->GetHandle()), + mSource(aSource), + mSTSThread(aPc->GetSTSThread()) { + ConnectSignals(); +} + +PeerConnectionImpl::SignalHandler::~SignalHandler() { + ASSERT_ON_THREAD(mSTSThread); +} + +void PeerConnectionImpl::SignalHandler::ConnectSignals() { + mSource->SignalGatheringStateChange.connect( + this, &PeerConnectionImpl::SignalHandler::IceGatheringStateChange_s); + mSource->SignalConnectionStateChange.connect( + this, &PeerConnectionImpl::SignalHandler::IceConnectionStateChange_s); + mSource->SignalCandidate.connect( + this, &PeerConnectionImpl::SignalHandler::OnCandidateFound_s); + mSource->SignalAlpnNegotiated.connect( + this, &PeerConnectionImpl::SignalHandler::AlpnNegotiated_s); + mSource->SignalStateChange.connect( + this, &PeerConnectionImpl::SignalHandler::ConnectionStateChange_s); + mSource->SignalRtcpStateChange.connect( + this, &PeerConnectionImpl::SignalHandler::ConnectionStateChange_s); +} + +void PeerConnectionImpl::AddIceCandidate(const std::string& aCandidate, + const std::string& aTransportId, + const std::string& aUfrag) { + MOZ_ASSERT(NS_IsMainThread()); + MOZ_ASSERT(!aTransportId.empty()); + + bool obfuscate_host_addresses = Preferences::GetBool( + "media.peerconnection.ice.obfuscate_host_addresses", false); + + if (obfuscate_host_addresses && !RelayOnly()) { + std::vector<std::string> tokens; + TokenizeCandidate(aCandidate, tokens); + + if (tokens.size() > 4) { + std::string addr = tokens[4]; + + // Check for address ending with .local + size_t nPeriods = std::count(addr.begin(), addr.end(), '.'); + size_t dotLocalLength = 6; // length of ".local" + + if (nPeriods == 1 && + addr.rfind(".local") + dotLocalLength == addr.length()) { + if (mStunAddrsRequest) { + PendingIceCandidate cand; + cand.mTokenizedCandidate = std::move(tokens); + cand.mTransportId = aTransportId; + cand.mUfrag = aUfrag; + mQueriedMDNSHostnames[addr].push_back(cand); + + GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction( + "PeerConnectionImpl::SendQueryMDNSHostname", + [self = RefPtr<PeerConnectionImpl>(this), addr]() mutable { + if (self->mStunAddrsRequest) { + self->StampTimecard("Look up mDNS name"); + self->mStunAddrsRequest->SendQueryMDNSHostname( + nsCString(nsAutoCString(addr.c_str()))); + } + NS_ReleaseOnMainThread( + "PeerConnectionImpl::SendQueryMDNSHostname", self.forget()); + })); + } + // TODO: Bug 1535690, we don't want to tell the ICE context that remote + // trickle is done if we are waiting to resolve a mDNS candidate. + return; + } + } + } + + mTransportHandler->AddIceCandidate(aTransportId, aCandidate, aUfrag, ""); +} + +void PeerConnectionImpl::UpdateNetworkState(bool online) { + if (mTransportHandler) { + mTransportHandler->UpdateNetworkState(online); + } +} + +void PeerConnectionImpl::FlushIceCtxOperationQueueIfReady() { + MOZ_ASSERT(NS_IsMainThread()); + + if (IsIceCtxReady()) { + for (auto& queuedIceCtxOperation : mQueuedIceCtxOperations) { + queuedIceCtxOperation->Run(); + } + mQueuedIceCtxOperations.clear(); + } +} + +void PeerConnectionImpl::PerformOrEnqueueIceCtxOperation( + nsIRunnable* runnable) { + MOZ_ASSERT(NS_IsMainThread()); + + if (IsIceCtxReady()) { + runnable->Run(); + } else { + mQueuedIceCtxOperations.push_back(runnable); + } +} + +void PeerConnectionImpl::GatherIfReady() { + MOZ_ASSERT(NS_IsMainThread()); + + // Init local addrs here so that if we re-gather after an ICE restart + // resulting from changing WiFi networks, we get new local addrs. + // Otherwise, we would reuse the addrs from the original WiFi network + // and the ICE restart will fail. + if (!mStunAddrs.Length()) { + InitLocalAddrs(); + } + + // If we had previously queued gathering or ICE start, unqueue them + mQueuedIceCtxOperations.clear(); + nsCOMPtr<nsIRunnable> runnable(WrapRunnable( + RefPtr<PeerConnectionImpl>(this), &PeerConnectionImpl::EnsureIceGathering, + GetPrefDefaultAddressOnly(), GetPrefObfuscateHostAddresses())); + + PerformOrEnqueueIceCtxOperation(runnable); +} + +already_AddRefed<nsIHttpChannelInternal> PeerConnectionImpl::GetChannel() + const { + Document* doc = mWindow->GetExtantDoc(); + if (NS_WARN_IF(!doc)) { + NS_WARNING("Unable to get document from window"); + return nullptr; + } + + if (!doc->GetDocumentURI()->SchemeIs("file")) { + nsIChannel* channel = doc->GetChannel(); + if (!channel) { + NS_WARNING("Unable to get channel from document"); + return nullptr; + } + + nsCOMPtr<nsIHttpChannelInternal> httpChannelInternal = + do_QueryInterface(channel); + if (NS_WARN_IF(!httpChannelInternal)) { + CSFLogInfo(LOGTAG, "%s: Document does not have an HTTP channel", + __FUNCTION__); + return nullptr; + } + return httpChannelInternal.forget(); + } + return nullptr; +} + +nsresult PeerConnectionImpl::SetTargetForDefaultLocalAddressLookup() { + nsCOMPtr<nsIHttpChannelInternal> httpChannelInternal = GetChannel(); + if (!httpChannelInternal) { + return NS_OK; + } + + nsCString remoteIp; + nsresult rv = httpChannelInternal->GetRemoteAddress(remoteIp); + if (NS_FAILED(rv) || remoteIp.IsEmpty()) { + CSFLogError(LOGTAG, "%s: Failed to get remote IP address: %d", __FUNCTION__, + (int)rv); + return rv; + } + + int32_t remotePort; + rv = httpChannelInternal->GetRemotePort(&remotePort); + if (NS_FAILED(rv)) { + CSFLogError(LOGTAG, "%s: Failed to get remote port number: %d", + __FUNCTION__, (int)rv); + return rv; + } + + mTransportHandler->SetTargetForDefaultLocalAddressLookup(remoteIp.get(), + remotePort); + + return NS_OK; +} + +void PeerConnectionImpl::EnsureIceGathering(bool aDefaultRouteOnly, + bool aObfuscateHostAddresses) { + if (!mTargetForDefaultLocalAddressLookupIsSet) { + nsresult rv = SetTargetForDefaultLocalAddressLookup(); + if (NS_FAILED(rv)) { + NS_WARNING("Unable to set target for default local address lookup"); + } + mTargetForDefaultLocalAddressLookupIsSet = true; + } + + // Make sure we don't call StartIceGathering if we're in e10s mode + // and we received no STUN addresses from the parent process. In the + // absence of previously provided STUN addresses, StartIceGathering will + // attempt to gather them (as in non-e10s mode), and this will cause a + // sandboxing exception in e10s mode. + if (!mStunAddrs.Length() && XRE_IsContentProcess()) { + CSFLogInfo(LOGTAG, "%s: No STUN addresses returned from parent process", + __FUNCTION__); + return; + } + + mTransportHandler->StartIceGathering(aDefaultRouteOnly, + aObfuscateHostAddresses, mStunAddrs); +} + +already_AddRefed<dom::RTCRtpTransceiver> PeerConnectionImpl::CreateTransceiver( + const std::string& aId, bool aIsVideo, const RTCRtpTransceiverInit& aInit, + dom::MediaStreamTrack* aSendTrack, bool aAddTrackMagic, ErrorResult& aRv) { + PeerConnectionCtx* ctx = PeerConnectionCtx::GetInstance(); + if (!mCall) { + mCall = WebrtcCallWrapper::Create( + GetTimestampMaker(), + media::ShutdownBlockingTicket::Create( + u"WebrtcCallWrapper shutdown blocker"_ns, + NS_LITERAL_STRING_FROM_CSTRING(__FILE__), __LINE__), + ctx->GetSharedWebrtcState()); + } + + if (aAddTrackMagic) { + mJsepSession->ApplyToTransceiver(aId, [](JsepTransceiver& aTransceiver) { + aTransceiver.SetAddTrackMagic(); + }); + } + + RefPtr<RTCRtpTransceiver> transceiver = new RTCRtpTransceiver( + mWindow, PrivacyRequested(), this, mTransportHandler, mJsepSession.get(), + aId, aIsVideo, mSTSThread.get(), aSendTrack, mCall.get(), mIdGenerator); + + transceiver->Init(aInit, aRv); + if (aRv.Failed()) { + return nullptr; + } + + if (aSendTrack) { + // implement checking for peerIdentity (where failure == black/silence) + Document* doc = mWindow->GetExtantDoc(); + if (doc) { + transceiver->Sender()->GetPipeline()->UpdateSinkIdentity( + doc->NodePrincipal(), GetPeerIdentity()); + } else { + MOZ_CRASH(); + aRv = NS_ERROR_FAILURE; + return nullptr; // Don't remove this till we know it's safe. + } + } + + return transceiver.forget(); +} + +std::string PeerConnectionImpl::GetTransportIdMatchingSendTrack( + const dom::MediaStreamTrack& aTrack) const { + for (const RefPtr<RTCRtpTransceiver>& transceiver : mTransceivers) { + if (transceiver->Sender()->HasTrack(&aTrack)) { + return transceiver->GetTransportId(); + } + } + return std::string(); +} + +void PeerConnectionImpl::SignalHandler::IceGatheringStateChange_s( + dom::RTCIceGatheringState aState) { + ASSERT_ON_THREAD(mSTSThread); + + GetMainThreadSerialEventTarget()->Dispatch( + NS_NewRunnableFunction(__func__, + [handle = mHandle, aState] { + PeerConnectionWrapper wrapper(handle); + if (wrapper.impl()) { + wrapper.impl()->IceGatheringStateChange( + aState); + } + }), + NS_DISPATCH_NORMAL); +} + +void PeerConnectionImpl::SignalHandler::IceConnectionStateChange_s( + dom::RTCIceConnectionState aState) { + ASSERT_ON_THREAD(mSTSThread); + + GetMainThreadSerialEventTarget()->Dispatch( + NS_NewRunnableFunction(__func__, + [handle = mHandle, aState] { + PeerConnectionWrapper wrapper(handle); + if (wrapper.impl()) { + wrapper.impl()->IceConnectionStateChange( + aState); + } + }), + NS_DISPATCH_NORMAL); +} + +void PeerConnectionImpl::SignalHandler::OnCandidateFound_s( + const std::string& aTransportId, const CandidateInfo& aCandidateInfo) { + ASSERT_ON_THREAD(mSTSThread); + CSFLogDebug(LOGTAG, "%s: %s", __FUNCTION__, aTransportId.c_str()); + + MOZ_ASSERT(!aCandidateInfo.mUfrag.empty()); + + GetMainThreadSerialEventTarget()->Dispatch( + NS_NewRunnableFunction(__func__, + [handle = mHandle, aTransportId, aCandidateInfo] { + PeerConnectionWrapper wrapper(handle); + if (wrapper.impl()) { + wrapper.impl()->OnCandidateFound( + aTransportId, aCandidateInfo); + } + }), + NS_DISPATCH_NORMAL); +} + +void PeerConnectionImpl::SignalHandler::AlpnNegotiated_s( + const std::string& aAlpn, bool aPrivacyRequested) { + MOZ_DIAGNOSTIC_ASSERT((aAlpn == "c-webrtc") == aPrivacyRequested); + GetMainThreadSerialEventTarget()->Dispatch( + NS_NewRunnableFunction(__func__, + [handle = mHandle, aPrivacyRequested] { + PeerConnectionWrapper wrapper(handle); + if (wrapper.impl()) { + wrapper.impl()->OnAlpnNegotiated( + aPrivacyRequested); + } + }), + NS_DISPATCH_NORMAL); +} + +void PeerConnectionImpl::SignalHandler::ConnectionStateChange_s( + const std::string& aTransportId, TransportLayer::State aState) { + GetMainThreadSerialEventTarget()->Dispatch( + NS_NewRunnableFunction(__func__, + [handle = mHandle, aTransportId, aState] { + PeerConnectionWrapper wrapper(handle); + if (wrapper.impl()) { + wrapper.impl()->OnDtlsStateChange(aTransportId, + aState); + } + }), + NS_DISPATCH_NORMAL); +} + +/** + * Tells you if any local track is isolated to a specific peer identity. + * Obviously, we want all the tracks to be isolated equally so that they can + * all be sent or not. We check once when we are setting a local description + * and that determines if we flip the "privacy requested" bit on. Once the bit + * is on, all media originating from this peer connection is isolated. + * + * @returns true if any track has a peerIdentity set on it + */ +bool PeerConnectionImpl::AnyLocalTrackHasPeerIdentity() const { + MOZ_ASSERT(NS_IsMainThread()); + + for (const RefPtr<RTCRtpTransceiver>& transceiver : mTransceivers) { + if (transceiver->Sender()->GetTrack() && + transceiver->Sender()->GetTrack()->GetPeerIdentity()) { + return true; + } + } + return false; +} + +bool PeerConnectionImpl::AnyCodecHasPluginID(uint64_t aPluginID) { + for (RefPtr<RTCRtpTransceiver>& transceiver : mTransceivers) { + if (transceiver->ConduitHasPluginID(aPluginID)) { + return true; + } + } + return false; +} + +std::unique_ptr<NrSocketProxyConfig> PeerConnectionImpl::GetProxyConfig() + const { + MOZ_ASSERT(NS_IsMainThread()); + + if (!mForceProxy && + Preferences::GetBool("media.peerconnection.disable_http_proxy", false)) { + return nullptr; + } + + nsCString alpn = "webrtc,c-webrtc"_ns; + auto* browserChild = BrowserChild::GetFrom(mWindow); + if (!browserChild) { + // Android doesn't have browser child apparently... + return nullptr; + } + + Document* doc = mWindow->GetExtantDoc(); + if (NS_WARN_IF(!doc)) { + NS_WARNING("Unable to get document from window"); + return nullptr; + } + + TabId id = browserChild->GetTabId(); + nsCOMPtr<nsILoadInfo> loadInfo = + new net::LoadInfo(doc->NodePrincipal(), doc->NodePrincipal(), doc, 0, + nsIContentPolicy::TYPE_INVALID); + + Maybe<net::LoadInfoArgs> loadInfoArgs; + MOZ_ALWAYS_SUCCEEDS( + mozilla::ipc::LoadInfoToLoadInfoArgs(loadInfo, &loadInfoArgs)); + return std::unique_ptr<NrSocketProxyConfig>(new NrSocketProxyConfig( + net::WebrtcProxyConfig(id, alpn, *loadInfoArgs, mForceProxy))); +} + +std::map<uint64_t, PeerConnectionAutoTimer> + PeerConnectionImpl::sCallDurationTimers; +} // namespace mozilla |