b1c1de17d4
This changes all internal code to use the media_description() helper for ContentInfo along with the as_audio, as_video, and as_data casting methods introduced in a previous CL. Reduces the total number of pointer static_casts in pc/ from 351 to 122. Bug: webrtc:8620 Change-Id: I996f49b55f1501c758a9e5223e30539a9f8d4eac Reviewed-on: https://webrtc-review.googlesource.com/35921 Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21419}
5053 lines
186 KiB
C++
5053 lines
186 KiB
C++
/*
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* Copyright 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "pc/peerconnection.h"
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#include <algorithm>
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#include <set>
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#include <utility>
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#include <vector>
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#include "api/jsepicecandidate.h"
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#include "api/jsepsessiondescription.h"
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#include "api/mediaconstraintsinterface.h"
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#include "api/mediastreamproxy.h"
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#include "api/mediastreamtrackproxy.h"
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#include "call/call.h"
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#include "logging/rtc_event_log/output/rtc_event_log_output_file.h"
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#include "logging/rtc_event_log/rtc_event_log.h"
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#include "media/sctp/sctptransport.h"
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#include "pc/audiotrack.h"
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#include "pc/channel.h"
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#include "pc/channelmanager.h"
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#include "pc/dtmfsender.h"
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#include "pc/mediastream.h"
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#include "pc/mediastreamobserver.h"
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#include "pc/remoteaudiosource.h"
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#include "pc/rtpmediautils.h"
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#include "pc/rtpreceiver.h"
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#include "pc/rtpsender.h"
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#include "pc/sctputils.h"
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#include "pc/sdputils.h"
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#include "pc/streamcollection.h"
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#include "pc/videocapturertracksource.h"
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#include "pc/videotrack.h"
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#include "rtc_base/bind.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/numerics/safe_conversions.h"
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#include "rtc_base/ptr_util.h"
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#include "rtc_base/stringencode.h"
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#include "rtc_base/stringutils.h"
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#include "rtc_base/trace_event.h"
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#include "system_wrappers/include/clock.h"
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#include "system_wrappers/include/field_trial.h"
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using cricket::ContentInfo;
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using cricket::ContentInfos;
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using cricket::MediaContentDescription;
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using cricket::SessionDescription;
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using cricket::MediaProtocolType;
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using cricket::TransportInfo;
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using cricket::LOCAL_PORT_TYPE;
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using cricket::STUN_PORT_TYPE;
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using cricket::RELAY_PORT_TYPE;
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using cricket::PRFLX_PORT_TYPE;
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namespace webrtc {
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// Error messages
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const char kBundleWithoutRtcpMux[] =
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"rtcp-mux must be enabled when BUNDLE "
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"is enabled.";
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const char kInvalidCandidates[] = "Description contains invalid candidates.";
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const char kInvalidSdp[] = "Invalid session description.";
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const char kMlineMismatchInAnswer[] =
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"The order of m-lines in answer doesn't match order in offer. Rejecting "
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"answer.";
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const char kMlineMismatchInSubsequentOffer[] =
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"The order of m-lines in subsequent offer doesn't match order from "
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"previous offer/answer.";
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const char kSdpWithoutDtlsFingerprint[] =
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"Called with SDP without DTLS fingerprint.";
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const char kSdpWithoutSdesCrypto[] = "Called with SDP without SDES crypto.";
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const char kSdpWithoutIceUfragPwd[] =
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"Called with SDP without ice-ufrag and ice-pwd.";
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const char kSessionError[] = "Session error code: ";
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const char kSessionErrorDesc[] = "Session error description: ";
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const char kDtlsSrtpSetupFailureRtp[] =
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"Couldn't set up DTLS-SRTP on RTP channel.";
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const char kDtlsSrtpSetupFailureRtcp[] =
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"Couldn't set up DTLS-SRTP on RTCP channel.";
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const char kEnableBundleFailed[] = "Failed to enable BUNDLE.";
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namespace {
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static const char kDefaultStreamLabel[] = "default";
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static const char kDefaultAudioSenderId[] = "defaulta0";
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static const char kDefaultVideoSenderId[] = "defaultv0";
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// The length of RTCP CNAMEs.
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static const int kRtcpCnameLength = 16;
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enum {
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MSG_SET_SESSIONDESCRIPTION_SUCCESS = 0,
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MSG_SET_SESSIONDESCRIPTION_FAILED,
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MSG_CREATE_SESSIONDESCRIPTION_FAILED,
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MSG_GETSTATS,
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MSG_FREE_DATACHANNELS,
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};
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struct SetSessionDescriptionMsg : public rtc::MessageData {
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explicit SetSessionDescriptionMsg(
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webrtc::SetSessionDescriptionObserver* observer)
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: observer(observer) {
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}
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rtc::scoped_refptr<webrtc::SetSessionDescriptionObserver> observer;
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std::string error;
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};
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struct CreateSessionDescriptionMsg : public rtc::MessageData {
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explicit CreateSessionDescriptionMsg(
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webrtc::CreateSessionDescriptionObserver* observer)
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: observer(observer) {}
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rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserver> observer;
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std::string error;
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};
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struct GetStatsMsg : public rtc::MessageData {
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GetStatsMsg(webrtc::StatsObserver* observer,
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webrtc::MediaStreamTrackInterface* track)
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: observer(observer), track(track) {
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}
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rtc::scoped_refptr<webrtc::StatsObserver> observer;
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rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track;
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};
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// Check if we can send |new_stream| on a PeerConnection.
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bool CanAddLocalMediaStream(webrtc::StreamCollectionInterface* current_streams,
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webrtc::MediaStreamInterface* new_stream) {
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if (!new_stream || !current_streams) {
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return false;
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}
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if (current_streams->find(new_stream->label()) != nullptr) {
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RTC_LOG(LS_ERROR) << "MediaStream with label " << new_stream->label()
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<< " is already added.";
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return false;
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}
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return true;
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}
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// If the direction is "recvonly" or "inactive", treat the description
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// as containing no streams.
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// See: https://code.google.com/p/webrtc/issues/detail?id=5054
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std::vector<cricket::StreamParams> GetActiveStreams(
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const cricket::MediaContentDescription* desc) {
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return RtpTransceiverDirectionHasSend(desc->direction())
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? desc->streams()
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: std::vector<cricket::StreamParams>();
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}
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bool IsValidOfferToReceiveMedia(int value) {
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typedef PeerConnectionInterface::RTCOfferAnswerOptions Options;
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return (value >= Options::kUndefined) &&
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(value <= Options::kMaxOfferToReceiveMedia);
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}
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// Add options to |[audio/video]_media_description_options| from |senders|.
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void AddRtpSenderOptions(
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const std::vector<rtc::scoped_refptr<
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RtpSenderProxyWithInternal<RtpSenderInternal>>>& senders,
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cricket::MediaDescriptionOptions* audio_media_description_options,
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cricket::MediaDescriptionOptions* video_media_description_options) {
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for (const auto& sender : senders) {
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if (sender->media_type() == cricket::MEDIA_TYPE_AUDIO) {
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if (audio_media_description_options) {
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audio_media_description_options->AddAudioSender(
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sender->id(), sender->internal()->stream_ids());
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}
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} else {
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RTC_DCHECK(sender->media_type() == cricket::MEDIA_TYPE_VIDEO);
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if (video_media_description_options) {
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video_media_description_options->AddVideoSender(
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sender->id(), sender->internal()->stream_ids(), 1);
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}
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}
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}
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}
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// Add options to |session_options| from |rtp_data_channels|.
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void AddRtpDataChannelOptions(
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const std::map<std::string, rtc::scoped_refptr<DataChannel>>&
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rtp_data_channels,
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cricket::MediaDescriptionOptions* data_media_description_options) {
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if (!data_media_description_options) {
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return;
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}
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// Check for data channels.
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for (const auto& kv : rtp_data_channels) {
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const DataChannel* channel = kv.second;
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if (channel->state() == DataChannel::kConnecting ||
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channel->state() == DataChannel::kOpen) {
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// Legacy RTP data channels are signaled with the track/stream ID set to
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// the data channel's label.
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data_media_description_options->AddRtpDataChannel(channel->label(),
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channel->label());
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}
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}
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}
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uint32_t ConvertIceTransportTypeToCandidateFilter(
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PeerConnectionInterface::IceTransportsType type) {
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switch (type) {
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case PeerConnectionInterface::kNone:
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return cricket::CF_NONE;
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case PeerConnectionInterface::kRelay:
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return cricket::CF_RELAY;
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case PeerConnectionInterface::kNoHost:
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return (cricket::CF_ALL & ~cricket::CF_HOST);
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case PeerConnectionInterface::kAll:
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return cricket::CF_ALL;
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default:
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RTC_NOTREACHED();
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}
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return cricket::CF_NONE;
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}
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// Helper to set an error and return from a method.
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bool SafeSetError(webrtc::RTCErrorType type, webrtc::RTCError* error) {
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if (error) {
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error->set_type(type);
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}
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return type == webrtc::RTCErrorType::NONE;
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}
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bool SafeSetError(webrtc::RTCError error, webrtc::RTCError* error_out) {
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if (error_out) {
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*error_out = std::move(error);
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}
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return error.ok();
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}
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std::string GetSignalingStateString(
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PeerConnectionInterface::SignalingState state) {
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switch (state) {
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case PeerConnectionInterface::kStable:
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return "kStable";
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case PeerConnectionInterface::kHaveLocalOffer:
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return "kHaveLocalOffer";
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case PeerConnectionInterface::kHaveLocalPrAnswer:
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return "kHavePrAnswer";
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case PeerConnectionInterface::kHaveRemoteOffer:
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return "kHaveRemoteOffer";
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case PeerConnectionInterface::kHaveRemotePrAnswer:
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return "kHaveRemotePrAnswer";
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case PeerConnectionInterface::kClosed:
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return "kClosed";
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}
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RTC_NOTREACHED();
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return "";
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}
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IceCandidatePairType GetIceCandidatePairCounter(
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const cricket::Candidate& local,
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const cricket::Candidate& remote) {
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const auto& l = local.type();
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const auto& r = remote.type();
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const auto& host = LOCAL_PORT_TYPE;
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const auto& srflx = STUN_PORT_TYPE;
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const auto& relay = RELAY_PORT_TYPE;
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const auto& prflx = PRFLX_PORT_TYPE;
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if (l == host && r == host) {
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bool local_private = IPIsPrivate(local.address().ipaddr());
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bool remote_private = IPIsPrivate(remote.address().ipaddr());
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if (local_private) {
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if (remote_private) {
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return kIceCandidatePairHostPrivateHostPrivate;
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} else {
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return kIceCandidatePairHostPrivateHostPublic;
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}
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} else {
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if (remote_private) {
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return kIceCandidatePairHostPublicHostPrivate;
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} else {
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return kIceCandidatePairHostPublicHostPublic;
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}
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}
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}
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if (l == host && r == srflx)
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return kIceCandidatePairHostSrflx;
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if (l == host && r == relay)
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return kIceCandidatePairHostRelay;
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if (l == host && r == prflx)
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return kIceCandidatePairHostPrflx;
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if (l == srflx && r == host)
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return kIceCandidatePairSrflxHost;
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if (l == srflx && r == srflx)
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return kIceCandidatePairSrflxSrflx;
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if (l == srflx && r == relay)
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return kIceCandidatePairSrflxRelay;
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if (l == srflx && r == prflx)
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return kIceCandidatePairSrflxPrflx;
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if (l == relay && r == host)
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return kIceCandidatePairRelayHost;
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if (l == relay && r == srflx)
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return kIceCandidatePairRelaySrflx;
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if (l == relay && r == relay)
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return kIceCandidatePairRelayRelay;
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if (l == relay && r == prflx)
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return kIceCandidatePairRelayPrflx;
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if (l == prflx && r == host)
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return kIceCandidatePairPrflxHost;
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if (l == prflx && r == srflx)
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return kIceCandidatePairPrflxSrflx;
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if (l == prflx && r == relay)
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return kIceCandidatePairPrflxRelay;
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return kIceCandidatePairMax;
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}
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// Verify that the order of media sections in |new_desc| matches
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// |existing_desc|. The number of m= sections in |new_desc| should be no less
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// than |existing_desc|.
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bool MediaSectionsInSameOrder(const SessionDescription* existing_desc,
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const SessionDescription* new_desc) {
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if (!existing_desc || !new_desc) {
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return false;
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}
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if (existing_desc->contents().size() > new_desc->contents().size()) {
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return false;
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}
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for (size_t i = 0; i < existing_desc->contents().size(); ++i) {
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if (new_desc->contents()[i].name != existing_desc->contents()[i].name) {
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return false;
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}
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const MediaContentDescription* new_desc_mdesc =
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new_desc->contents()[i].media_description();
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const MediaContentDescription* existing_desc_mdesc =
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existing_desc->contents()[i].media_description();
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if (new_desc_mdesc->type() != existing_desc_mdesc->type()) {
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return false;
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}
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}
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return true;
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}
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bool MediaSectionsHaveSameCount(const SessionDescription* desc1,
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const SessionDescription* desc2) {
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if (!desc1 || !desc2) {
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return false;
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}
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return desc1->contents().size() == desc2->contents().size();
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}
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// Checks that each non-rejected content has SDES crypto keys or a DTLS
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// fingerprint, unless it's in a BUNDLE group, in which case only the
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// BUNDLE-tag section (first media section/description in the BUNDLE group)
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// needs a ufrag and pwd. Mismatches, such as replying with a DTLS fingerprint
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// to SDES keys, will be caught in JsepTransport negotiation, and backstopped
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// by Channel's |srtp_required| check.
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RTCError VerifyCrypto(const SessionDescription* desc, bool dtls_enabled) {
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const cricket::ContentGroup* bundle =
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desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
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for (const cricket::ContentInfo& content_info : desc->contents()) {
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if (content_info.rejected) {
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continue;
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}
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const std::string& mid = content_info.name;
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if (bundle && bundle->HasContentName(mid) &&
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mid != *(bundle->FirstContentName())) {
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// This isn't the first media section in the BUNDLE group, so it's not
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// required to have crypto attributes, since only the crypto attributes
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// from the first section actually get used.
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continue;
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}
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// If the content isn't rejected or bundled into another m= section, crypto
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// must be present.
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const MediaContentDescription* media = content_info.media_description();
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const TransportInfo* tinfo = desc->GetTransportInfoByName(mid);
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if (!media || !tinfo) {
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// Something is not right.
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LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, kInvalidSdp);
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}
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if (dtls_enabled) {
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if (!tinfo->description.identity_fingerprint) {
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RTC_LOG(LS_WARNING)
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<< "Session description must have DTLS fingerprint if "
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"DTLS enabled.";
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return RTCError(RTCErrorType::INVALID_PARAMETER,
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kSdpWithoutDtlsFingerprint);
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}
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} else {
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if (media->cryptos().empty()) {
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RTC_LOG(LS_WARNING)
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<< "Session description must have SDES when DTLS disabled.";
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return RTCError(RTCErrorType::INVALID_PARAMETER, kSdpWithoutSdesCrypto);
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}
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}
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}
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return RTCError::OK();
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}
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// Checks that each non-rejected content has ice-ufrag and ice-pwd set, unless
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// it's in a BUNDLE group, in which case only the BUNDLE-tag section (first
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// media section/description in the BUNDLE group) needs a ufrag and pwd.
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bool VerifyIceUfragPwdPresent(const SessionDescription* desc) {
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const cricket::ContentGroup* bundle =
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desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
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for (const cricket::ContentInfo& content_info : desc->contents()) {
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if (content_info.rejected) {
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continue;
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}
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const std::string& mid = content_info.name;
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if (bundle && bundle->HasContentName(mid) &&
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mid != *(bundle->FirstContentName())) {
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// This isn't the first media section in the BUNDLE group, so it's not
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// required to have ufrag/password, since only the ufrag/password from
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// the first section actually get used.
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continue;
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}
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// If the content isn't rejected or bundled into another m= section,
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// ice-ufrag and ice-pwd must be present.
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const TransportInfo* tinfo = desc->GetTransportInfoByName(mid);
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if (!tinfo) {
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// Something is not right.
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RTC_LOG(LS_ERROR) << kInvalidSdp;
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return false;
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}
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if (tinfo->description.ice_ufrag.empty() ||
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tinfo->description.ice_pwd.empty()) {
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RTC_LOG(LS_ERROR) << "Session description must have ice ufrag and pwd.";
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return false;
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}
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}
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return true;
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}
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bool GetTrackIdBySsrc(const SessionDescription* session_description,
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uint32_t ssrc,
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std::string* track_id) {
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RTC_DCHECK(track_id != NULL);
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const cricket::AudioContentDescription* audio_desc =
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cricket::GetFirstAudioContentDescription(session_description);
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if (audio_desc) {
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const auto* found = cricket::GetStreamBySsrc(audio_desc->streams(), ssrc);
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if (found) {
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*track_id = found->id;
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return true;
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}
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}
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const cricket::VideoContentDescription* video_desc =
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cricket::GetFirstVideoContentDescription(session_description);
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if (video_desc) {
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const auto* found = cricket::GetStreamBySsrc(video_desc->streams(), ssrc);
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if (found) {
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*track_id = found->id;
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return true;
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}
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}
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return false;
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}
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|
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// Get the SCTP port out of a SessionDescription.
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// Return -1 if not found.
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int GetSctpPort(const SessionDescription* session_description) {
|
|
const cricket::DataContentDescription* data_desc =
|
|
GetFirstDataContentDescription(session_description);
|
|
RTC_DCHECK(data_desc);
|
|
if (!data_desc) {
|
|
return -1;
|
|
}
|
|
std::string value;
|
|
cricket::DataCodec match_pattern(cricket::kGoogleSctpDataCodecPlType,
|
|
cricket::kGoogleSctpDataCodecName);
|
|
for (const cricket::DataCodec& codec : data_desc->codecs()) {
|
|
if (!codec.Matches(match_pattern)) {
|
|
continue;
|
|
}
|
|
if (codec.GetParam(cricket::kCodecParamPort, &value)) {
|
|
return rtc::FromString<int>(value);
|
|
}
|
|
}
|
|
return -1;
|
|
}
|
|
|
|
// Returns true if |new_desc| requests an ICE restart (i.e., new ufrag/pwd).
|
|
bool CheckForRemoteIceRestart(const SessionDescriptionInterface* old_desc,
|
|
const SessionDescriptionInterface* new_desc,
|
|
const std::string& content_name) {
|
|
if (!old_desc) {
|
|
return false;
|
|
}
|
|
const SessionDescription* new_sd = new_desc->description();
|
|
const SessionDescription* old_sd = old_desc->description();
|
|
const ContentInfo* cinfo = new_sd->GetContentByName(content_name);
|
|
if (!cinfo || cinfo->rejected) {
|
|
return false;
|
|
}
|
|
// If the content isn't rejected, check if ufrag and password has changed.
|
|
const cricket::TransportDescription* new_transport_desc =
|
|
new_sd->GetTransportDescriptionByName(content_name);
|
|
const cricket::TransportDescription* old_transport_desc =
|
|
old_sd->GetTransportDescriptionByName(content_name);
|
|
if (!new_transport_desc || !old_transport_desc) {
|
|
// No transport description exists. This is not an ICE restart.
|
|
return false;
|
|
}
|
|
if (cricket::IceCredentialsChanged(
|
|
old_transport_desc->ice_ufrag, old_transport_desc->ice_pwd,
|
|
new_transport_desc->ice_ufrag, new_transport_desc->ice_pwd)) {
|
|
RTC_LOG(LS_INFO) << "Remote peer requests ICE restart for " << content_name
|
|
<< ".";
|
|
return true;
|
|
}
|
|
return false;
|
|
}
|
|
|
|
} // namespace
|
|
|
|
// Upon completion, posts a task to execute the callback of the
|
|
// SetSessionDescriptionObserver asynchronously on the same thread. At this
|
|
// point, the state of the peer connection might no longer reflect the effects
|
|
// of the SetRemoteDescription operation, as the peer connection could have been
|
|
// modified during the post.
|
|
// TODO(hbos): Remove this class once we remove the version of
|
|
// PeerConnectionInterface::SetRemoteDescription() that takes a
|
|
// SetSessionDescriptionObserver as an argument.
|
|
class PeerConnection::SetRemoteDescriptionObserverAdapter
|
|
: public rtc::RefCountedObject<SetRemoteDescriptionObserverInterface> {
|
|
public:
|
|
SetRemoteDescriptionObserverAdapter(
|
|
rtc::scoped_refptr<PeerConnection> pc,
|
|
rtc::scoped_refptr<SetSessionDescriptionObserver> wrapper)
|
|
: pc_(std::move(pc)), wrapper_(std::move(wrapper)) {}
|
|
|
|
// SetRemoteDescriptionObserverInterface implementation.
|
|
void OnSetRemoteDescriptionComplete(RTCError error) override {
|
|
if (error.ok())
|
|
pc_->PostSetSessionDescriptionSuccess(wrapper_);
|
|
else
|
|
pc_->PostSetSessionDescriptionFailure(wrapper_, error.message());
|
|
}
|
|
|
|
private:
|
|
rtc::scoped_refptr<PeerConnection> pc_;
|
|
rtc::scoped_refptr<SetSessionDescriptionObserver> wrapper_;
|
|
};
|
|
|
|
bool PeerConnectionInterface::RTCConfiguration::operator==(
|
|
const PeerConnectionInterface::RTCConfiguration& o) const {
|
|
// This static_assert prevents us from accidentally breaking operator==.
|
|
// Note: Order matters! Fields must be ordered the same as RTCConfiguration.
|
|
struct stuff_being_tested_for_equality {
|
|
IceServers servers;
|
|
IceTransportsType type;
|
|
BundlePolicy bundle_policy;
|
|
RtcpMuxPolicy rtcp_mux_policy;
|
|
std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
|
|
int ice_candidate_pool_size;
|
|
bool disable_ipv6;
|
|
bool disable_ipv6_on_wifi;
|
|
int max_ipv6_networks;
|
|
bool enable_rtp_data_channel;
|
|
rtc::Optional<int> screencast_min_bitrate;
|
|
rtc::Optional<bool> combined_audio_video_bwe;
|
|
rtc::Optional<bool> enable_dtls_srtp;
|
|
TcpCandidatePolicy tcp_candidate_policy;
|
|
CandidateNetworkPolicy candidate_network_policy;
|
|
int audio_jitter_buffer_max_packets;
|
|
bool audio_jitter_buffer_fast_accelerate;
|
|
int ice_connection_receiving_timeout;
|
|
int ice_backup_candidate_pair_ping_interval;
|
|
ContinualGatheringPolicy continual_gathering_policy;
|
|
bool prioritize_most_likely_ice_candidate_pairs;
|
|
struct cricket::MediaConfig media_config;
|
|
bool prune_turn_ports;
|
|
bool presume_writable_when_fully_relayed;
|
|
bool enable_ice_renomination;
|
|
bool redetermine_role_on_ice_restart;
|
|
rtc::Optional<int> ice_check_min_interval;
|
|
rtc::Optional<rtc::IntervalRange> ice_regather_interval_range;
|
|
webrtc::TurnCustomizer* turn_customizer;
|
|
SdpSemantics sdp_semantics;
|
|
};
|
|
static_assert(sizeof(stuff_being_tested_for_equality) == sizeof(*this),
|
|
"Did you add something to RTCConfiguration and forget to "
|
|
"update operator==?");
|
|
return type == o.type && servers == o.servers &&
|
|
bundle_policy == o.bundle_policy &&
|
|
rtcp_mux_policy == o.rtcp_mux_policy &&
|
|
tcp_candidate_policy == o.tcp_candidate_policy &&
|
|
candidate_network_policy == o.candidate_network_policy &&
|
|
audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets &&
|
|
audio_jitter_buffer_fast_accelerate ==
|
|
o.audio_jitter_buffer_fast_accelerate &&
|
|
ice_connection_receiving_timeout ==
|
|
o.ice_connection_receiving_timeout &&
|
|
ice_backup_candidate_pair_ping_interval ==
|
|
o.ice_backup_candidate_pair_ping_interval &&
|
|
continual_gathering_policy == o.continual_gathering_policy &&
|
|
certificates == o.certificates &&
|
|
prioritize_most_likely_ice_candidate_pairs ==
|
|
o.prioritize_most_likely_ice_candidate_pairs &&
|
|
media_config == o.media_config && disable_ipv6 == o.disable_ipv6 &&
|
|
disable_ipv6_on_wifi == o.disable_ipv6_on_wifi &&
|
|
max_ipv6_networks == o.max_ipv6_networks &&
|
|
enable_rtp_data_channel == o.enable_rtp_data_channel &&
|
|
screencast_min_bitrate == o.screencast_min_bitrate &&
|
|
combined_audio_video_bwe == o.combined_audio_video_bwe &&
|
|
enable_dtls_srtp == o.enable_dtls_srtp &&
|
|
ice_candidate_pool_size == o.ice_candidate_pool_size &&
|
|
prune_turn_ports == o.prune_turn_ports &&
|
|
presume_writable_when_fully_relayed ==
|
|
o.presume_writable_when_fully_relayed &&
|
|
enable_ice_renomination == o.enable_ice_renomination &&
|
|
redetermine_role_on_ice_restart == o.redetermine_role_on_ice_restart &&
|
|
ice_check_min_interval == o.ice_check_min_interval &&
|
|
ice_regather_interval_range == o.ice_regather_interval_range &&
|
|
turn_customizer == o.turn_customizer &&
|
|
sdp_semantics == o.sdp_semantics;
|
|
}
|
|
|
|
bool PeerConnectionInterface::RTCConfiguration::operator!=(
|
|
const PeerConnectionInterface::RTCConfiguration& o) const {
|
|
return !(*this == o);
|
|
}
|
|
|
|
// Generate a RTCP CNAME when a PeerConnection is created.
|
|
std::string GenerateRtcpCname() {
|
|
std::string cname;
|
|
if (!rtc::CreateRandomString(kRtcpCnameLength, &cname)) {
|
|
RTC_LOG(LS_ERROR) << "Failed to generate CNAME.";
|
|
RTC_NOTREACHED();
|
|
}
|
|
return cname;
|
|
}
|
|
|
|
bool ValidateOfferAnswerOptions(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options) {
|
|
return IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_audio) &&
|
|
IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_video);
|
|
}
|
|
|
|
// From |rtc_options|, fill parts of |session_options| shared by all generated
|
|
// m= sections (in other words, nothing that involves a map/array).
|
|
void ExtractSharedMediaSessionOptions(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
session_options->vad_enabled = rtc_options.voice_activity_detection;
|
|
session_options->bundle_enabled = rtc_options.use_rtp_mux;
|
|
}
|
|
|
|
bool ConvertConstraintsToOfferAnswerOptions(
|
|
const MediaConstraintsInterface* constraints,
|
|
PeerConnectionInterface::RTCOfferAnswerOptions* offer_answer_options) {
|
|
if (!constraints) {
|
|
return true;
|
|
}
|
|
|
|
bool value = false;
|
|
size_t mandatory_constraints_satisfied = 0;
|
|
|
|
if (FindConstraint(constraints,
|
|
MediaConstraintsInterface::kOfferToReceiveAudio, &value,
|
|
&mandatory_constraints_satisfied)) {
|
|
offer_answer_options->offer_to_receive_audio =
|
|
value ? PeerConnectionInterface::RTCOfferAnswerOptions::
|
|
kOfferToReceiveMediaTrue
|
|
: 0;
|
|
}
|
|
|
|
if (FindConstraint(constraints,
|
|
MediaConstraintsInterface::kOfferToReceiveVideo, &value,
|
|
&mandatory_constraints_satisfied)) {
|
|
offer_answer_options->offer_to_receive_video =
|
|
value ? PeerConnectionInterface::RTCOfferAnswerOptions::
|
|
kOfferToReceiveMediaTrue
|
|
: 0;
|
|
}
|
|
if (FindConstraint(constraints,
|
|
MediaConstraintsInterface::kVoiceActivityDetection, &value,
|
|
&mandatory_constraints_satisfied)) {
|
|
offer_answer_options->voice_activity_detection = value;
|
|
}
|
|
if (FindConstraint(constraints, MediaConstraintsInterface::kUseRtpMux, &value,
|
|
&mandatory_constraints_satisfied)) {
|
|
offer_answer_options->use_rtp_mux = value;
|
|
}
|
|
if (FindConstraint(constraints, MediaConstraintsInterface::kIceRestart,
|
|
&value, &mandatory_constraints_satisfied)) {
|
|
offer_answer_options->ice_restart = value;
|
|
}
|
|
|
|
return mandatory_constraints_satisfied == constraints->GetMandatory().size();
|
|
}
|
|
|
|
PeerConnection::PeerConnection(PeerConnectionFactory* factory,
|
|
std::unique_ptr<RtcEventLog> event_log,
|
|
std::unique_ptr<Call> call)
|
|
: factory_(factory),
|
|
event_log_(std::move(event_log)),
|
|
rtcp_cname_(GenerateRtcpCname()),
|
|
local_streams_(StreamCollection::Create()),
|
|
remote_streams_(StreamCollection::Create()),
|
|
call_(std::move(call)) {}
|
|
|
|
PeerConnection::~PeerConnection() {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection");
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
|
|
// Need to stop transceivers before destroying the stats collector because
|
|
// AudioRtpSender has a reference to the StatsCollector it will update when
|
|
// stopping.
|
|
for (auto transceiver : transceivers_) {
|
|
transceiver->Stop();
|
|
}
|
|
|
|
stats_.reset(nullptr);
|
|
if (stats_collector_) {
|
|
stats_collector_->WaitForPendingRequest();
|
|
stats_collector_ = nullptr;
|
|
}
|
|
|
|
// Don't destroy BaseChannels until after stats has been cleaned up so that
|
|
// the last stats request can still read from the channels.
|
|
DestroyAllChannels();
|
|
|
|
RTC_LOG(LS_INFO) << "Session: " << session_id() << " is destroyed.";
|
|
|
|
webrtc_session_desc_factory_.reset();
|
|
sctp_invoker_.reset();
|
|
sctp_factory_.reset();
|
|
transport_controller_.reset();
|
|
|
|
// port_allocator_ lives on the network thread and should be destroyed there.
|
|
network_thread()->Invoke<void>(RTC_FROM_HERE,
|
|
[this] { port_allocator_.reset(); });
|
|
// call_ and event_log_ must be destroyed on the worker thread.
|
|
worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
|
|
call_.reset();
|
|
event_log_.reset();
|
|
});
|
|
}
|
|
|
|
void PeerConnection::DestroyAllChannels() {
|
|
// Destroy video channels first since they may have a pointer to a voice
|
|
// channel.
|
|
for (auto transceiver : transceivers_) {
|
|
if (transceiver->internal()->media_type() == cricket::MEDIA_TYPE_VIDEO) {
|
|
DestroyTransceiverChannel(transceiver);
|
|
}
|
|
}
|
|
for (auto transceiver : transceivers_) {
|
|
if (transceiver->internal()->media_type() == cricket::MEDIA_TYPE_AUDIO) {
|
|
DestroyTransceiverChannel(transceiver);
|
|
}
|
|
}
|
|
DestroyDataChannel();
|
|
}
|
|
|
|
bool PeerConnection::Initialize(
|
|
const PeerConnectionInterface::RTCConfiguration& configuration,
|
|
std::unique_ptr<cricket::PortAllocator> allocator,
|
|
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
|
|
PeerConnectionObserver* observer) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::Initialize");
|
|
|
|
RTCError config_error = ValidateConfiguration(configuration);
|
|
if (!config_error.ok()) {
|
|
RTC_LOG(LS_ERROR) << "Invalid configuration: " << config_error.message();
|
|
return false;
|
|
}
|
|
|
|
if (!allocator) {
|
|
RTC_LOG(LS_ERROR)
|
|
<< "PeerConnection initialized without a PortAllocator? "
|
|
<< "This shouldn't happen if using PeerConnectionFactory.";
|
|
return false;
|
|
}
|
|
|
|
if (!observer) {
|
|
// TODO(deadbeef): Why do we do this?
|
|
RTC_LOG(LS_ERROR) << "PeerConnection initialized without a "
|
|
<< "PeerConnectionObserver";
|
|
return false;
|
|
}
|
|
observer_ = observer;
|
|
port_allocator_ = std::move(allocator);
|
|
|
|
// The port allocator lives on the network thread and should be initialized
|
|
// there.
|
|
if (!network_thread()->Invoke<bool>(
|
|
RTC_FROM_HERE, rtc::Bind(&PeerConnection::InitializePortAllocator_n,
|
|
this, configuration))) {
|
|
return false;
|
|
}
|
|
|
|
// RFC 3264: The numeric value of the session id and version in the
|
|
// o line MUST be representable with a "64 bit signed integer".
|
|
// Due to this constraint session id |session_id_| is max limited to
|
|
// LLONG_MAX.
|
|
session_id_ = rtc::ToString(rtc::CreateRandomId64() & LLONG_MAX);
|
|
transport_controller_.reset(factory_->CreateTransportController(
|
|
port_allocator_.get(), configuration.redetermine_role_on_ice_restart));
|
|
transport_controller_->SetIceRole(cricket::ICEROLE_CONTROLLED);
|
|
transport_controller_->SignalConnectionState.connect(
|
|
this, &PeerConnection::OnTransportControllerConnectionState);
|
|
transport_controller_->SignalGatheringState.connect(
|
|
this, &PeerConnection::OnTransportControllerGatheringState);
|
|
transport_controller_->SignalCandidatesGathered.connect(
|
|
this, &PeerConnection::OnTransportControllerCandidatesGathered);
|
|
transport_controller_->SignalCandidatesRemoved.connect(
|
|
this, &PeerConnection::OnTransportControllerCandidatesRemoved);
|
|
transport_controller_->SignalDtlsHandshakeError.connect(
|
|
this, &PeerConnection::OnTransportControllerDtlsHandshakeError);
|
|
|
|
sctp_factory_ = factory_->CreateSctpTransportInternalFactory();
|
|
|
|
stats_.reset(new StatsCollector(this));
|
|
stats_collector_ = RTCStatsCollector::Create(this);
|
|
|
|
configuration_ = configuration;
|
|
|
|
const PeerConnectionFactoryInterface::Options& options = factory_->options();
|
|
|
|
transport_controller_->SetSslMaxProtocolVersion(options.ssl_max_version);
|
|
|
|
// Obtain a certificate from RTCConfiguration if any were provided (optional).
|
|
rtc::scoped_refptr<rtc::RTCCertificate> certificate;
|
|
if (!configuration.certificates.empty()) {
|
|
// TODO(hbos,torbjorng): Decide on certificate-selection strategy instead of
|
|
// just picking the first one. The decision should be made based on the DTLS
|
|
// handshake. The DTLS negotiations need to know about all certificates.
|
|
certificate = configuration.certificates[0];
|
|
}
|
|
|
|
transport_controller_->SetIceConfig(ParseIceConfig(configuration));
|
|
|
|
if (options.disable_encryption) {
|
|
dtls_enabled_ = false;
|
|
} else {
|
|
// Enable DTLS by default if we have an identity store or a certificate.
|
|
dtls_enabled_ = (cert_generator || certificate);
|
|
// |configuration| can override the default |dtls_enabled_| value.
|
|
if (configuration.enable_dtls_srtp) {
|
|
dtls_enabled_ = *(configuration.enable_dtls_srtp);
|
|
}
|
|
}
|
|
|
|
// Enable creation of RTP data channels if the kEnableRtpDataChannels is set.
|
|
// It takes precendence over the disable_sctp_data_channels
|
|
// PeerConnectionFactoryInterface::Options.
|
|
if (configuration.enable_rtp_data_channel) {
|
|
data_channel_type_ = cricket::DCT_RTP;
|
|
} else {
|
|
// DTLS has to be enabled to use SCTP.
|
|
if (!options.disable_sctp_data_channels && dtls_enabled_) {
|
|
data_channel_type_ = cricket::DCT_SCTP;
|
|
}
|
|
}
|
|
|
|
video_options_.screencast_min_bitrate_kbps =
|
|
configuration.screencast_min_bitrate;
|
|
audio_options_.combined_audio_video_bwe =
|
|
configuration.combined_audio_video_bwe;
|
|
|
|
audio_options_.audio_jitter_buffer_max_packets =
|
|
configuration.audio_jitter_buffer_max_packets;
|
|
|
|
audio_options_.audio_jitter_buffer_fast_accelerate =
|
|
configuration.audio_jitter_buffer_fast_accelerate;
|
|
|
|
// Whether the certificate generator/certificate is null or not determines
|
|
// what PeerConnectionDescriptionFactory will do, so make sure that we give it
|
|
// the right instructions by clearing the variables if needed.
|
|
if (!dtls_enabled_) {
|
|
cert_generator.reset();
|
|
certificate = nullptr;
|
|
} else if (certificate) {
|
|
// Favor generated certificate over the certificate generator.
|
|
cert_generator.reset();
|
|
}
|
|
|
|
webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
|
|
signaling_thread(), channel_manager(), this, session_id(),
|
|
std::move(cert_generator), certificate));
|
|
webrtc_session_desc_factory_->SignalCertificateReady.connect(
|
|
this, &PeerConnection::OnCertificateReady);
|
|
|
|
if (options.disable_encryption) {
|
|
webrtc_session_desc_factory_->SetSdesPolicy(cricket::SEC_DISABLED);
|
|
}
|
|
|
|
webrtc_session_desc_factory_->set_enable_encrypted_rtp_header_extensions(
|
|
options.crypto_options.enable_encrypted_rtp_header_extensions);
|
|
|
|
// Add default audio/video transceivers for Plan B SDP.
|
|
if (!IsUnifiedPlan()) {
|
|
transceivers_.push_back(
|
|
RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
|
|
signaling_thread(), new RtpTransceiver(cricket::MEDIA_TYPE_AUDIO)));
|
|
transceivers_.push_back(
|
|
RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
|
|
signaling_thread(), new RtpTransceiver(cricket::MEDIA_TYPE_VIDEO)));
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
RTCError PeerConnection::ValidateConfiguration(
|
|
const RTCConfiguration& config) const {
|
|
if (config.ice_regather_interval_range &&
|
|
config.continual_gathering_policy == GATHER_ONCE) {
|
|
return RTCError(RTCErrorType::INVALID_PARAMETER,
|
|
"ice_regather_interval_range specified but continual "
|
|
"gathering policy is GATHER_ONCE");
|
|
}
|
|
return RTCError::OK();
|
|
}
|
|
|
|
rtc::scoped_refptr<StreamCollectionInterface>
|
|
PeerConnection::local_streams() {
|
|
return local_streams_;
|
|
}
|
|
|
|
rtc::scoped_refptr<StreamCollectionInterface>
|
|
PeerConnection::remote_streams() {
|
|
return remote_streams_;
|
|
}
|
|
|
|
bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::AddStream");
|
|
if (IsClosed()) {
|
|
return false;
|
|
}
|
|
if (!CanAddLocalMediaStream(local_streams_, local_stream)) {
|
|
return false;
|
|
}
|
|
|
|
local_streams_->AddStream(local_stream);
|
|
MediaStreamObserver* observer = new MediaStreamObserver(local_stream);
|
|
observer->SignalAudioTrackAdded.connect(this,
|
|
&PeerConnection::OnAudioTrackAdded);
|
|
observer->SignalAudioTrackRemoved.connect(
|
|
this, &PeerConnection::OnAudioTrackRemoved);
|
|
observer->SignalVideoTrackAdded.connect(this,
|
|
&PeerConnection::OnVideoTrackAdded);
|
|
observer->SignalVideoTrackRemoved.connect(
|
|
this, &PeerConnection::OnVideoTrackRemoved);
|
|
stream_observers_.push_back(std::unique_ptr<MediaStreamObserver>(observer));
|
|
|
|
for (const auto& track : local_stream->GetAudioTracks()) {
|
|
AddAudioTrack(track.get(), local_stream);
|
|
}
|
|
for (const auto& track : local_stream->GetVideoTracks()) {
|
|
AddVideoTrack(track.get(), local_stream);
|
|
}
|
|
|
|
stats_->AddStream(local_stream);
|
|
observer_->OnRenegotiationNeeded();
|
|
return true;
|
|
}
|
|
|
|
void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::RemoveStream");
|
|
if (!IsClosed()) {
|
|
for (const auto& track : local_stream->GetAudioTracks()) {
|
|
RemoveAudioTrack(track.get(), local_stream);
|
|
}
|
|
for (const auto& track : local_stream->GetVideoTracks()) {
|
|
RemoveVideoTrack(track.get(), local_stream);
|
|
}
|
|
}
|
|
local_streams_->RemoveStream(local_stream);
|
|
stream_observers_.erase(
|
|
std::remove_if(
|
|
stream_observers_.begin(), stream_observers_.end(),
|
|
[local_stream](const std::unique_ptr<MediaStreamObserver>& observer) {
|
|
return observer->stream()->label().compare(local_stream->label()) ==
|
|
0;
|
|
}),
|
|
stream_observers_.end());
|
|
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
observer_->OnRenegotiationNeeded();
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpSenderInterface> PeerConnection::AddTrack(
|
|
MediaStreamTrackInterface* track,
|
|
std::vector<MediaStreamInterface*> streams) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::AddTrack");
|
|
std::vector<std::string> stream_labels;
|
|
for (auto* stream : streams) {
|
|
if (!stream) {
|
|
RTC_LOG(LS_ERROR) << "Stream list has null element.";
|
|
return nullptr;
|
|
}
|
|
stream_labels.push_back(stream->label());
|
|
}
|
|
auto sender_or_error = AddTrackWithStreamLabels(track, stream_labels);
|
|
if (!sender_or_error.ok()) {
|
|
return nullptr;
|
|
}
|
|
return sender_or_error.MoveValue();
|
|
}
|
|
|
|
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>
|
|
PeerConnection::AddTrackWithStreamLabels(
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
|
const std::vector<std::string>& stream_labels) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::AddTrackWithStreamLabels");
|
|
if (!track) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "Track is null.");
|
|
}
|
|
if (!(track->kind() == MediaStreamTrackInterface::kAudioKind ||
|
|
track->kind() == MediaStreamTrackInterface::kVideoKind)) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"Track has invalid kind: " + track->kind());
|
|
}
|
|
// TODO(bugs.webrtc.org/7932): Support adding a track to multiple streams.
|
|
if (stream_labels.size() > 1u) {
|
|
LOG_AND_RETURN_ERROR(
|
|
RTCErrorType::UNSUPPORTED_OPERATION,
|
|
"AddTrack with more than one stream is not currently supported.");
|
|
}
|
|
if (IsClosed()) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
|
|
"PeerConnection is closed.");
|
|
}
|
|
if (FindSenderForTrack(track)) {
|
|
LOG_AND_RETURN_ERROR(
|
|
RTCErrorType::INVALID_PARAMETER,
|
|
"Sender already exists for track " + track->id() + ".");
|
|
}
|
|
// TODO(bugs.webrtc.org/7933): MediaSession expects the sender to have exactly
|
|
// one stream. AddTrackInternal will return an error if there is more than one
|
|
// stream, but if the caller specifies none then we need to generate a random
|
|
// stream label.
|
|
std::vector<std::string> adjusted_stream_labels = stream_labels;
|
|
if (stream_labels.empty()) {
|
|
adjusted_stream_labels.push_back(rtc::CreateRandomUuid());
|
|
}
|
|
RTC_DCHECK_EQ(1, adjusted_stream_labels.size());
|
|
auto sender_or_error =
|
|
(IsUnifiedPlan() ? AddTrackUnifiedPlan(track, adjusted_stream_labels)
|
|
: AddTrackPlanB(track, adjusted_stream_labels));
|
|
if (sender_or_error.ok()) {
|
|
observer_->OnRenegotiationNeeded();
|
|
}
|
|
return sender_or_error;
|
|
}
|
|
|
|
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>
|
|
PeerConnection::AddTrackPlanB(
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
|
const std::vector<std::string>& stream_labels) {
|
|
if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
|
|
auto new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
|
signaling_thread(),
|
|
new AudioRtpSender(static_cast<AudioTrackInterface*>(track.get()),
|
|
voice_channel(), stats_.get()));
|
|
GetAudioTransceiver()->internal()->AddSender(new_sender);
|
|
new_sender->internal()->set_stream_ids(stream_labels);
|
|
const RtpSenderInfo* sender_info =
|
|
FindSenderInfo(local_audio_sender_infos_,
|
|
new_sender->internal()->stream_id(), track->id());
|
|
if (sender_info) {
|
|
new_sender->internal()->SetSsrc(sender_info->first_ssrc);
|
|
}
|
|
return rtc::scoped_refptr<RtpSenderInterface>(new_sender);
|
|
} else {
|
|
RTC_DCHECK_EQ(MediaStreamTrackInterface::kVideoKind, track->kind());
|
|
auto new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
|
signaling_thread(),
|
|
new VideoRtpSender(static_cast<VideoTrackInterface*>(track.get()),
|
|
video_channel()));
|
|
GetVideoTransceiver()->internal()->AddSender(new_sender);
|
|
new_sender->internal()->set_stream_ids(stream_labels);
|
|
const RtpSenderInfo* sender_info =
|
|
FindSenderInfo(local_video_sender_infos_,
|
|
new_sender->internal()->stream_id(), track->id());
|
|
if (sender_info) {
|
|
new_sender->internal()->SetSsrc(sender_info->first_ssrc);
|
|
}
|
|
return rtc::scoped_refptr<RtpSenderInterface>(new_sender);
|
|
}
|
|
}
|
|
|
|
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>
|
|
PeerConnection::AddTrackUnifiedPlan(
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
|
const std::vector<std::string>& stream_labels) {
|
|
auto transceiver = FindFirstTransceiverForAddedTrack(track);
|
|
if (transceiver) {
|
|
if (transceiver->direction() == RtpTransceiverDirection::kRecvOnly) {
|
|
transceiver->SetDirection(RtpTransceiverDirection::kSendRecv);
|
|
} else if (transceiver->direction() == RtpTransceiverDirection::kInactive) {
|
|
transceiver->SetDirection(RtpTransceiverDirection::kSendOnly);
|
|
}
|
|
} else {
|
|
cricket::MediaType media_type =
|
|
(track->kind() == MediaStreamTrackInterface::kAudioKind
|
|
? cricket::MEDIA_TYPE_AUDIO
|
|
: cricket::MEDIA_TYPE_VIDEO);
|
|
transceiver = CreateTransceiver(media_type);
|
|
transceiver->internal()->set_created_by_addtrack(true);
|
|
transceiver->SetDirection(RtpTransceiverDirection::kSendRecv);
|
|
}
|
|
transceiver->sender()->SetTrack(track);
|
|
transceiver->internal()->sender_internal()->set_stream_ids(stream_labels);
|
|
return transceiver->sender();
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
PeerConnection::FindFirstTransceiverForAddedTrack(
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track) {
|
|
RTC_DCHECK(track);
|
|
for (auto transceiver : transceivers_) {
|
|
if (!transceiver->sender()->track() &&
|
|
cricket::MediaTypeToString(transceiver->internal()->media_type()) ==
|
|
track->kind() &&
|
|
!transceiver->internal()->has_ever_been_used_to_send()) {
|
|
return transceiver;
|
|
}
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
bool PeerConnection::RemoveTrack(RtpSenderInterface* sender) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::RemoveTrack");
|
|
return RemoveTrackInternal(sender).ok();
|
|
}
|
|
|
|
RTCError PeerConnection::RemoveTrackInternal(
|
|
rtc::scoped_refptr<RtpSenderInterface> sender) {
|
|
if (!sender) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "Sender is null.");
|
|
}
|
|
if (IsClosed()) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
|
|
"PeerConnection is closed.");
|
|
}
|
|
if (IsUnifiedPlan()) {
|
|
auto transceiver = FindTransceiverBySender(sender);
|
|
if (!transceiver || !sender->track()) {
|
|
return RTCError::OK();
|
|
}
|
|
sender->SetTrack(nullptr);
|
|
if (transceiver->direction() == RtpTransceiverDirection::kSendRecv) {
|
|
transceiver->internal()->SetDirection(RtpTransceiverDirection::kRecvOnly);
|
|
} else if (transceiver->direction() == RtpTransceiverDirection::kSendOnly) {
|
|
transceiver->internal()->SetDirection(RtpTransceiverDirection::kInactive);
|
|
}
|
|
} else {
|
|
bool removed;
|
|
if (sender->media_type() == cricket::MEDIA_TYPE_AUDIO) {
|
|
removed = GetAudioTransceiver()->internal()->RemoveSender(sender);
|
|
} else {
|
|
RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, sender->media_type());
|
|
removed = GetVideoTransceiver()->internal()->RemoveSender(sender);
|
|
}
|
|
if (!removed) {
|
|
LOG_AND_RETURN_ERROR(
|
|
RTCErrorType::INVALID_PARAMETER,
|
|
"Couldn't find sender " + sender->id() + " to remove.");
|
|
}
|
|
}
|
|
observer_->OnRenegotiationNeeded();
|
|
return RTCError::OK();
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
PeerConnection::FindTransceiverBySender(
|
|
rtc::scoped_refptr<RtpSenderInterface> sender) {
|
|
for (auto transceiver : transceivers_) {
|
|
if (transceiver->sender() == sender) {
|
|
return transceiver;
|
|
}
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
|
|
PeerConnection::AddTransceiver(
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track) {
|
|
return AddTransceiver(track, RtpTransceiverInit());
|
|
}
|
|
|
|
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
|
|
PeerConnection::AddTransceiver(
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
|
const RtpTransceiverInit& init) {
|
|
if (!IsUnifiedPlan()) {
|
|
LOG_AND_RETURN_ERROR(
|
|
RTCErrorType::INTERNAL_ERROR,
|
|
"AddTransceiver only supported when Unified Plan is enabled.");
|
|
}
|
|
if (!track) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "track is null");
|
|
}
|
|
cricket::MediaType media_type;
|
|
if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
|
|
media_type = cricket::MEDIA_TYPE_AUDIO;
|
|
} else if (track->kind() == MediaStreamTrackInterface::kVideoKind) {
|
|
media_type = cricket::MEDIA_TYPE_VIDEO;
|
|
} else {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"Track kind is not audio or video");
|
|
}
|
|
return AddTransceiver(media_type, track, init);
|
|
}
|
|
|
|
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
|
|
PeerConnection::AddTransceiver(cricket::MediaType media_type) {
|
|
return AddTransceiver(media_type, RtpTransceiverInit());
|
|
}
|
|
|
|
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
|
|
PeerConnection::AddTransceiver(cricket::MediaType media_type,
|
|
const RtpTransceiverInit& init) {
|
|
if (!IsUnifiedPlan()) {
|
|
LOG_AND_RETURN_ERROR(
|
|
RTCErrorType::INTERNAL_ERROR,
|
|
"AddTransceiver only supported when Unified Plan is enabled.");
|
|
}
|
|
if (!(media_type == cricket::MEDIA_TYPE_AUDIO ||
|
|
media_type == cricket::MEDIA_TYPE_VIDEO)) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"media type is not audio or video");
|
|
}
|
|
return AddTransceiver(media_type, nullptr, init);
|
|
}
|
|
|
|
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
|
|
PeerConnection::AddTransceiver(
|
|
cricket::MediaType media_type,
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
|
const RtpTransceiverInit& init) {
|
|
RTC_DCHECK((media_type == cricket::MEDIA_TYPE_AUDIO ||
|
|
media_type == cricket::MEDIA_TYPE_VIDEO));
|
|
if (track) {
|
|
RTC_DCHECK_EQ(media_type,
|
|
(track->kind() == MediaStreamTrackInterface::kAudioKind
|
|
? cricket::MEDIA_TYPE_AUDIO
|
|
: cricket::MEDIA_TYPE_VIDEO));
|
|
}
|
|
|
|
// TODO(bugs.webrtc.org/7600): Verify init.
|
|
|
|
auto transceiver = CreateTransceiver(media_type);
|
|
transceiver->SetDirection(init.direction);
|
|
if (track) {
|
|
transceiver->sender()->SetTrack(track);
|
|
}
|
|
|
|
observer_->OnRenegotiationNeeded();
|
|
|
|
return rtc::scoped_refptr<RtpTransceiverInterface>(transceiver);
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
PeerConnection::CreateTransceiver(cricket::MediaType media_type) {
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender;
|
|
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
|
|
receiver;
|
|
std::string receiver_id = rtc::CreateRandomUuid();
|
|
// TODO(bugs.webrtc.org/7600): Initializing the sender/receiver with a null
|
|
// channel prevents users from calling SetParameters on them, which is needed
|
|
// to be in compliance with the spec.
|
|
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
|
|
sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
|
signaling_thread(), new AudioRtpSender(nullptr, stats_.get()));
|
|
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
|
|
signaling_thread(), new AudioRtpReceiver(receiver_id, {}, 0, nullptr));
|
|
} else {
|
|
RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, media_type);
|
|
sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
|
signaling_thread(), new VideoRtpSender(nullptr));
|
|
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
|
|
signaling_thread(),
|
|
new VideoRtpReceiver(receiver_id, {}, worker_thread(), 0, nullptr));
|
|
}
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
transceiver = RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
|
|
signaling_thread(), new RtpTransceiver(sender, receiver));
|
|
transceivers_.push_back(transceiver);
|
|
return transceiver;
|
|
}
|
|
|
|
rtc::scoped_refptr<DtmfSenderInterface> PeerConnection::CreateDtmfSender(
|
|
AudioTrackInterface* track) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::CreateDtmfSender");
|
|
if (IsClosed()) {
|
|
return nullptr;
|
|
}
|
|
if (!track) {
|
|
RTC_LOG(LS_ERROR) << "CreateDtmfSender - track is NULL.";
|
|
return nullptr;
|
|
}
|
|
auto track_sender = FindSenderForTrack(track);
|
|
if (!track_sender) {
|
|
RTC_LOG(LS_ERROR) << "CreateDtmfSender called with a non-added track.";
|
|
return nullptr;
|
|
}
|
|
|
|
return track_sender->GetDtmfSender();
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
|
|
const std::string& kind,
|
|
const std::string& stream_id) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::CreateSender");
|
|
if (IsClosed()) {
|
|
return nullptr;
|
|
}
|
|
|
|
// TODO(steveanton): Move construction of the RtpSenders to RtpTransceiver.
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender;
|
|
if (kind == MediaStreamTrackInterface::kAudioKind) {
|
|
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
|
signaling_thread(), new AudioRtpSender(voice_channel(), stats_.get()));
|
|
GetAudioTransceiver()->internal()->AddSender(new_sender);
|
|
} else if (kind == MediaStreamTrackInterface::kVideoKind) {
|
|
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
|
signaling_thread(), new VideoRtpSender(video_channel()));
|
|
GetVideoTransceiver()->internal()->AddSender(new_sender);
|
|
} else {
|
|
RTC_LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind;
|
|
return nullptr;
|
|
}
|
|
|
|
if (!stream_id.empty()) {
|
|
new_sender->internal()->set_stream_id(stream_id);
|
|
}
|
|
|
|
return new_sender;
|
|
}
|
|
|
|
std::vector<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders()
|
|
const {
|
|
std::vector<rtc::scoped_refptr<RtpSenderInterface>> ret;
|
|
for (auto sender : GetSendersInternal()) {
|
|
ret.push_back(sender);
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
|
|
PeerConnection::GetSendersInternal() const {
|
|
std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
|
|
all_senders;
|
|
for (auto transceiver : transceivers_) {
|
|
auto senders = transceiver->internal()->senders();
|
|
all_senders.insert(all_senders.end(), senders.begin(), senders.end());
|
|
}
|
|
return all_senders;
|
|
}
|
|
|
|
std::vector<rtc::scoped_refptr<RtpReceiverInterface>>
|
|
PeerConnection::GetReceivers() const {
|
|
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> ret;
|
|
for (const auto& receiver : GetReceiversInternal()) {
|
|
ret.push_back(receiver);
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
std::vector<
|
|
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
|
|
PeerConnection::GetReceiversInternal() const {
|
|
std::vector<
|
|
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
|
|
all_receivers;
|
|
for (auto transceiver : transceivers_) {
|
|
auto receivers = transceiver->internal()->receivers();
|
|
all_receivers.insert(all_receivers.end(), receivers.begin(),
|
|
receivers.end());
|
|
}
|
|
return all_receivers;
|
|
}
|
|
|
|
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
|
|
PeerConnection::GetTransceivers() const {
|
|
RTC_DCHECK(IsUnifiedPlan());
|
|
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> all_transceivers;
|
|
for (auto transceiver : transceivers_) {
|
|
all_transceivers.push_back(transceiver);
|
|
}
|
|
return all_transceivers;
|
|
}
|
|
|
|
bool PeerConnection::GetStats(StatsObserver* observer,
|
|
MediaStreamTrackInterface* track,
|
|
StatsOutputLevel level) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
if (!observer) {
|
|
RTC_LOG(LS_ERROR) << "GetStats - observer is NULL.";
|
|
return false;
|
|
}
|
|
|
|
stats_->UpdateStats(level);
|
|
// The StatsCollector is used to tell if a track is valid because it may
|
|
// remember tracks that the PeerConnection previously removed.
|
|
if (track && !stats_->IsValidTrack(track->id())) {
|
|
RTC_LOG(LS_WARNING) << "GetStats is called with an invalid track: "
|
|
<< track->id();
|
|
return false;
|
|
}
|
|
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_GETSTATS,
|
|
new GetStatsMsg(observer, track));
|
|
return true;
|
|
}
|
|
|
|
void PeerConnection::GetStats(RTCStatsCollectorCallback* callback) {
|
|
RTC_DCHECK(stats_collector_);
|
|
stats_collector_->GetStatsReport(callback);
|
|
}
|
|
|
|
PeerConnectionInterface::SignalingState PeerConnection::signaling_state() {
|
|
return signaling_state_;
|
|
}
|
|
|
|
PeerConnectionInterface::IceConnectionState
|
|
PeerConnection::ice_connection_state() {
|
|
return ice_connection_state_;
|
|
}
|
|
|
|
PeerConnectionInterface::IceGatheringState
|
|
PeerConnection::ice_gathering_state() {
|
|
return ice_gathering_state_;
|
|
}
|
|
|
|
rtc::scoped_refptr<DataChannelInterface>
|
|
PeerConnection::CreateDataChannel(
|
|
const std::string& label,
|
|
const DataChannelInit* config) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::CreateDataChannel");
|
|
|
|
bool first_datachannel = !HasDataChannels();
|
|
|
|
std::unique_ptr<InternalDataChannelInit> internal_config;
|
|
if (config) {
|
|
internal_config.reset(new InternalDataChannelInit(*config));
|
|
}
|
|
rtc::scoped_refptr<DataChannelInterface> channel(
|
|
InternalCreateDataChannel(label, internal_config.get()));
|
|
if (!channel.get()) {
|
|
return nullptr;
|
|
}
|
|
|
|
// Trigger the onRenegotiationNeeded event for every new RTP DataChannel, or
|
|
// the first SCTP DataChannel.
|
|
if (data_channel_type() == cricket::DCT_RTP || first_datachannel) {
|
|
observer_->OnRenegotiationNeeded();
|
|
}
|
|
|
|
return DataChannelProxy::Create(signaling_thread(), channel.get());
|
|
}
|
|
|
|
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
|
|
const MediaConstraintsInterface* constraints) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::CreateOffer");
|
|
|
|
PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options;
|
|
// Always create an offer even if |ConvertConstraintsToOfferAnswerOptions|
|
|
// returns false for now. Because |ConvertConstraintsToOfferAnswerOptions|
|
|
// compares the mandatory fields parsed with the mandatory fields added in the
|
|
// |constraints| and some downstream applications might create offers with
|
|
// mandatory fields which would not be parsed in the helper method. For
|
|
// example, in Chromium/remoting, |kEnableDtlsSrtp| is added to the
|
|
// |constraints| as a mandatory field but it is not parsed.
|
|
ConvertConstraintsToOfferAnswerOptions(constraints, &offer_answer_options);
|
|
|
|
CreateOffer(observer, offer_answer_options);
|
|
}
|
|
|
|
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
|
|
const RTCOfferAnswerOptions& options) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::CreateOffer");
|
|
|
|
if (!observer) {
|
|
RTC_LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
|
|
return;
|
|
}
|
|
|
|
if (IsClosed()) {
|
|
std::string error = "CreateOffer called when PeerConnection is closed.";
|
|
RTC_LOG(LS_ERROR) << error;
|
|
PostCreateSessionDescriptionFailure(observer, error);
|
|
return;
|
|
}
|
|
|
|
if (!ValidateOfferAnswerOptions(options)) {
|
|
std::string error = "CreateOffer called with invalid options.";
|
|
RTC_LOG(LS_ERROR) << error;
|
|
PostCreateSessionDescriptionFailure(observer, error);
|
|
return;
|
|
}
|
|
|
|
cricket::MediaSessionOptions session_options;
|
|
GetOptionsForOffer(options, &session_options);
|
|
webrtc_session_desc_factory_->CreateOffer(observer, options, session_options);
|
|
}
|
|
|
|
void PeerConnection::CreateAnswer(
|
|
CreateSessionDescriptionObserver* observer,
|
|
const MediaConstraintsInterface* constraints) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::CreateAnswer");
|
|
|
|
if (!observer) {
|
|
RTC_LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
|
|
return;
|
|
}
|
|
|
|
PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options;
|
|
if (!ConvertConstraintsToOfferAnswerOptions(constraints,
|
|
&offer_answer_options)) {
|
|
std::string error = "CreateAnswer called with invalid constraints.";
|
|
RTC_LOG(LS_ERROR) << error;
|
|
PostCreateSessionDescriptionFailure(observer, error);
|
|
return;
|
|
}
|
|
|
|
CreateAnswer(observer, offer_answer_options);
|
|
}
|
|
|
|
void PeerConnection::CreateAnswer(CreateSessionDescriptionObserver* observer,
|
|
const RTCOfferAnswerOptions& options) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::CreateAnswer");
|
|
if (!observer) {
|
|
RTC_LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
|
|
return;
|
|
}
|
|
|
|
if (IsClosed()) {
|
|
std::string error = "CreateAnswer called when PeerConnection is closed.";
|
|
RTC_LOG(LS_ERROR) << error;
|
|
PostCreateSessionDescriptionFailure(observer, error);
|
|
return;
|
|
}
|
|
|
|
if (remote_description() &&
|
|
remote_description()->GetType() != SdpType::kOffer) {
|
|
std::string error = "CreateAnswer called without remote offer.";
|
|
RTC_LOG(LS_ERROR) << error;
|
|
PostCreateSessionDescriptionFailure(observer, error);
|
|
return;
|
|
}
|
|
|
|
cricket::MediaSessionOptions session_options;
|
|
GetOptionsForAnswer(options, &session_options);
|
|
|
|
webrtc_session_desc_factory_->CreateAnswer(observer, session_options);
|
|
}
|
|
|
|
void PeerConnection::SetLocalDescription(
|
|
SetSessionDescriptionObserver* observer,
|
|
SessionDescriptionInterface* desc) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::SetLocalDescription");
|
|
|
|
if (!observer) {
|
|
RTC_LOG(LS_ERROR) << "SetLocalDescription - observer is NULL.";
|
|
return;
|
|
}
|
|
|
|
if (!desc) {
|
|
PostSetSessionDescriptionFailure(observer, "SessionDescription is NULL.");
|
|
return;
|
|
}
|
|
|
|
SdpType type = desc->GetType();
|
|
|
|
RTCError error = ApplyLocalDescription(rtc::WrapUnique(desc));
|
|
// |desc| may be destroyed at this point.
|
|
|
|
if (!error.ok()) {
|
|
std::ostringstream oss;
|
|
oss << "Failed to set local " << SdpTypeToString(type)
|
|
<< " sdp: " << error.message();
|
|
std::string error_message = oss.str();
|
|
RTC_LOG(LS_ERROR) << error_message << " (" << error.type() << ")";
|
|
PostSetSessionDescriptionFailure(observer, std::move(error_message));
|
|
return;
|
|
}
|
|
RTC_DCHECK(local_description());
|
|
|
|
PostSetSessionDescriptionSuccess(observer);
|
|
|
|
// According to JSEP, after setLocalDescription, changing the candidate pool
|
|
// size is not allowed, and changing the set of ICE servers will not result
|
|
// in new candidates being gathered.
|
|
port_allocator_->FreezeCandidatePool();
|
|
|
|
// MaybeStartGathering needs to be called after posting
|
|
// MSG_SET_SESSIONDESCRIPTION_SUCCESS, so that we don't signal any candidates
|
|
// before signaling that SetLocalDescription completed.
|
|
transport_controller_->MaybeStartGathering();
|
|
|
|
if (local_description()->GetType() == SdpType::kAnswer) {
|
|
// TODO(deadbeef): We already had to hop to the network thread for
|
|
// MaybeStartGathering...
|
|
network_thread()->Invoke<void>(
|
|
RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
|
|
port_allocator_.get()));
|
|
}
|
|
}
|
|
|
|
RTCError PeerConnection::ApplyLocalDescription(
|
|
std::unique_ptr<SessionDescriptionInterface> desc) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(desc);
|
|
|
|
RTCError error = ValidateSessionDescription(desc.get(), cricket::CS_LOCAL);
|
|
if (!error.ok()) {
|
|
return error;
|
|
}
|
|
|
|
// Update stats here so that we have the most recent stats for tracks and
|
|
// streams that might be removed by updating the session description.
|
|
stats_->UpdateStats(kStatsOutputLevelStandard);
|
|
|
|
// Update the initial_offerer flag if this session is the initial_offerer.
|
|
SdpType type = desc->GetType();
|
|
if (!initial_offerer_.has_value()) {
|
|
initial_offerer_.emplace(type == SdpType::kOffer);
|
|
if (*initial_offerer_) {
|
|
transport_controller_->SetIceRole(cricket::ICEROLE_CONTROLLING);
|
|
} else {
|
|
transport_controller_->SetIceRole(cricket::ICEROLE_CONTROLLED);
|
|
}
|
|
}
|
|
|
|
if (type == SdpType::kAnswer) {
|
|
current_local_description_ = std::move(desc);
|
|
pending_local_description_ = nullptr;
|
|
current_remote_description_ = std::move(pending_remote_description_);
|
|
} else {
|
|
pending_local_description_ = std::move(desc);
|
|
}
|
|
// The session description to apply now must be accessed by
|
|
// |local_description()|.
|
|
RTC_DCHECK(local_description());
|
|
|
|
// Transport and Media channels will be created only when offer is set.
|
|
if (type == SdpType::kOffer) {
|
|
// TODO(mallinath) - Handle CreateChannel failure, as new local description
|
|
// is applied. Restore back to old description.
|
|
RTCError error = CreateChannels(local_description()->description());
|
|
if (!error.ok()) {
|
|
return error;
|
|
}
|
|
}
|
|
|
|
// Remove unused channels if MediaContentDescription is rejected.
|
|
RemoveUnusedChannels(local_description()->description());
|
|
|
|
error = UpdateSessionState(type, cricket::CS_LOCAL);
|
|
if (!error.ok()) {
|
|
return error;
|
|
}
|
|
if (remote_description()) {
|
|
// Now that we have a local description, we can push down remote candidates.
|
|
UseCandidatesInSessionDescription(remote_description());
|
|
}
|
|
|
|
pending_ice_restarts_.clear();
|
|
if (session_error() != SessionError::kNone) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR, GetSessionErrorMsg());
|
|
}
|
|
|
|
// If setting the description decided our SSL role, allocate any necessary
|
|
// SCTP sids.
|
|
rtc::SSLRole role;
|
|
if (data_channel_type() == cricket::DCT_SCTP && GetSctpSslRole(&role)) {
|
|
AllocateSctpSids(role);
|
|
}
|
|
|
|
// Update state and SSRC of local MediaStreams and DataChannels based on the
|
|
// local session description.
|
|
const cricket::ContentInfo* audio_content =
|
|
GetFirstAudioContent(local_description()->description());
|
|
if (audio_content) {
|
|
if (audio_content->rejected) {
|
|
RemoveSenders(cricket::MEDIA_TYPE_AUDIO);
|
|
} else {
|
|
const cricket::AudioContentDescription* audio_desc =
|
|
audio_content->media_description()->as_audio();
|
|
UpdateLocalSenders(audio_desc->streams(), audio_desc->type());
|
|
}
|
|
}
|
|
|
|
const cricket::ContentInfo* video_content =
|
|
GetFirstVideoContent(local_description()->description());
|
|
if (video_content) {
|
|
if (video_content->rejected) {
|
|
RemoveSenders(cricket::MEDIA_TYPE_VIDEO);
|
|
} else {
|
|
const cricket::VideoContentDescription* video_desc =
|
|
video_content->media_description()->as_video();
|
|
UpdateLocalSenders(video_desc->streams(), video_desc->type());
|
|
}
|
|
}
|
|
|
|
const cricket::ContentInfo* data_content =
|
|
GetFirstDataContent(local_description()->description());
|
|
if (data_content) {
|
|
const cricket::DataContentDescription* data_desc =
|
|
data_content->media_description()->as_data();
|
|
if (rtc::starts_with(data_desc->protocol().data(),
|
|
cricket::kMediaProtocolRtpPrefix)) {
|
|
UpdateLocalRtpDataChannels(data_desc->streams());
|
|
}
|
|
}
|
|
|
|
return RTCError::OK();
|
|
}
|
|
|
|
void PeerConnection::SetRemoteDescription(
|
|
SetSessionDescriptionObserver* observer,
|
|
SessionDescriptionInterface* desc) {
|
|
SetRemoteDescription(
|
|
std::unique_ptr<SessionDescriptionInterface>(desc),
|
|
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface>(
|
|
new SetRemoteDescriptionObserverAdapter(this, observer)));
|
|
}
|
|
|
|
void PeerConnection::SetRemoteDescription(
|
|
std::unique_ptr<SessionDescriptionInterface> desc,
|
|
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::SetRemoteDescription");
|
|
|
|
if (!observer) {
|
|
RTC_LOG(LS_ERROR) << "SetRemoteDescription - observer is NULL.";
|
|
return;
|
|
}
|
|
|
|
if (!desc) {
|
|
observer->OnSetRemoteDescriptionComplete(RTCError(
|
|
RTCErrorType::INVALID_PARAMETER, "SessionDescription is NULL."));
|
|
return;
|
|
}
|
|
|
|
const SdpType type = desc->GetType();
|
|
|
|
RTCError error = ApplyRemoteDescription(std::move(desc));
|
|
// |desc| may be destroyed at this point.
|
|
|
|
if (!error.ok()) {
|
|
std::ostringstream oss;
|
|
oss << "Failed to set remote " << SdpTypeToString(type)
|
|
<< " sdp: " << error.message();
|
|
std::string error_message = oss.str();
|
|
RTC_LOG(LS_ERROR) << error_message << " (" << error.type() << ")";
|
|
observer->OnSetRemoteDescriptionComplete(
|
|
RTCError(error.type(), std::move(error_message)));
|
|
return;
|
|
}
|
|
|
|
if (remote_description()->GetType() == SdpType::kAnswer) {
|
|
// TODO(deadbeef): We already had to hop to the network thread for
|
|
// MaybeStartGathering...
|
|
network_thread()->Invoke<void>(
|
|
RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
|
|
port_allocator_.get()));
|
|
}
|
|
|
|
observer->OnSetRemoteDescriptionComplete(RTCError::OK());
|
|
}
|
|
|
|
RTCError PeerConnection::ApplyRemoteDescription(
|
|
std::unique_ptr<SessionDescriptionInterface> desc) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(desc);
|
|
|
|
RTCError error = ValidateSessionDescription(desc.get(), cricket::CS_REMOTE);
|
|
if (!error.ok()) {
|
|
return error;
|
|
}
|
|
|
|
// Update stats here so that we have the most recent stats for tracks and
|
|
// streams that might be removed by updating the session description.
|
|
stats_->UpdateStats(kStatsOutputLevelStandard);
|
|
// Takes the ownership of |desc|. On success, remote_description() is updated
|
|
// to reflect the description that was passed in.
|
|
|
|
const SessionDescriptionInterface* old_remote_description =
|
|
remote_description();
|
|
// Grab ownership of the description being replaced for the remainder of this
|
|
// method, since it's used below as |old_remote_description|.
|
|
std::unique_ptr<SessionDescriptionInterface> replaced_remote_description;
|
|
SdpType type = desc->GetType();
|
|
if (type == SdpType::kAnswer) {
|
|
replaced_remote_description = pending_remote_description_
|
|
? std::move(pending_remote_description_)
|
|
: std::move(current_remote_description_);
|
|
current_remote_description_ = std::move(desc);
|
|
pending_remote_description_ = nullptr;
|
|
current_local_description_ = std::move(pending_local_description_);
|
|
} else {
|
|
replaced_remote_description = std::move(pending_remote_description_);
|
|
pending_remote_description_ = std::move(desc);
|
|
}
|
|
// The session description to apply now must be accessed by
|
|
// |remote_description()|.
|
|
RTC_DCHECK(remote_description());
|
|
|
|
// Transport and Media channels will be created only when offer is set.
|
|
if (type == SdpType::kOffer) {
|
|
// TODO(mallinath) - Handle CreateChannel failure, as new local description
|
|
// is applied. Restore back to old description.
|
|
RTCError error = CreateChannels(remote_description()->description());
|
|
if (!error.ok()) {
|
|
return error;
|
|
}
|
|
}
|
|
|
|
// Remove unused channels if MediaContentDescription is rejected.
|
|
RemoveUnusedChannels(remote_description()->description());
|
|
|
|
// NOTE: Candidates allocation will be initiated only when SetLocalDescription
|
|
// is called.
|
|
error = UpdateSessionState(type, cricket::CS_REMOTE);
|
|
if (!error.ok()) {
|
|
return error;
|
|
}
|
|
|
|
if (local_description() &&
|
|
!UseCandidatesInSessionDescription(remote_description())) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, kInvalidCandidates);
|
|
}
|
|
|
|
if (old_remote_description) {
|
|
for (const cricket::ContentInfo& content :
|
|
old_remote_description->description()->contents()) {
|
|
// Check if this new SessionDescription contains new ICE ufrag and
|
|
// password that indicates the remote peer requests an ICE restart.
|
|
// TODO(deadbeef): When we start storing both the current and pending
|
|
// remote description, this should reset pending_ice_restarts and compare
|
|
// against the current description.
|
|
if (CheckForRemoteIceRestart(old_remote_description, remote_description(),
|
|
content.name)) {
|
|
if (type == SdpType::kOffer) {
|
|
pending_ice_restarts_.insert(content.name);
|
|
}
|
|
} else {
|
|
// We retain all received candidates only if ICE is not restarted.
|
|
// When ICE is restarted, all previous candidates belong to an old
|
|
// generation and should not be kept.
|
|
// TODO(deadbeef): This goes against the W3C spec which says the remote
|
|
// description should only contain candidates from the last set remote
|
|
// description plus any candidates added since then. We should remove
|
|
// this once we're sure it won't break anything.
|
|
WebRtcSessionDescriptionFactory::CopyCandidatesFromSessionDescription(
|
|
old_remote_description, content.name, mutable_remote_description());
|
|
}
|
|
}
|
|
}
|
|
|
|
if (session_error() != SessionError::kNone) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR, GetSessionErrorMsg());
|
|
}
|
|
|
|
// Set the the ICE connection state to connecting since the connection may
|
|
// become writable with peer reflexive candidates before any remote candidate
|
|
// is signaled.
|
|
// TODO(pthatcher): This is a short-term solution for crbug/446908. A real fix
|
|
// is to have a new signal the indicates a change in checking state from the
|
|
// transport and expose a new checking() member from transport that can be
|
|
// read to determine the current checking state. The existing SignalConnecting
|
|
// actually means "gathering candidates", so cannot be be used here.
|
|
if (remote_description()->GetType() != SdpType::kOffer &&
|
|
ice_connection_state() == PeerConnectionInterface::kIceConnectionNew) {
|
|
SetIceConnectionState(PeerConnectionInterface::kIceConnectionChecking);
|
|
}
|
|
|
|
// If setting the description decided our SSL role, allocate any necessary
|
|
// SCTP sids.
|
|
rtc::SSLRole role;
|
|
if (data_channel_type() == cricket::DCT_SCTP && GetSctpSslRole(&role)) {
|
|
AllocateSctpSids(role);
|
|
}
|
|
|
|
const cricket::ContentInfo* audio_content =
|
|
GetFirstAudioContent(remote_description()->description());
|
|
const cricket::ContentInfo* video_content =
|
|
GetFirstVideoContent(remote_description()->description());
|
|
const cricket::AudioContentDescription* audio_desc =
|
|
GetFirstAudioContentDescription(remote_description()->description());
|
|
const cricket::VideoContentDescription* video_desc =
|
|
GetFirstVideoContentDescription(remote_description()->description());
|
|
const cricket::DataContentDescription* data_desc =
|
|
GetFirstDataContentDescription(remote_description()->description());
|
|
|
|
// Check if the descriptions include streams, just in case the peer supports
|
|
// MSID, but doesn't indicate so with "a=msid-semantic".
|
|
if (remote_description()->description()->msid_supported() ||
|
|
(audio_desc && !audio_desc->streams().empty()) ||
|
|
(video_desc && !video_desc->streams().empty())) {
|
|
remote_peer_supports_msid_ = true;
|
|
}
|
|
|
|
// We wait to signal new streams until we finish processing the description,
|
|
// since only at that point will new streams have all their tracks.
|
|
rtc::scoped_refptr<StreamCollection> new_streams(StreamCollection::Create());
|
|
|
|
// TODO(steveanton): When removing RTP senders/receivers in response to a
|
|
// rejected media section, there is some cleanup logic that expects the voice/
|
|
// video channel to still be set. But in this method the voice/video channel
|
|
// would have been destroyed by the SetRemoteDescription caller above so the
|
|
// cleanup that relies on them fails to run. The RemoveSenders calls should be
|
|
// moved to right before the DestroyChannel calls to fix this.
|
|
|
|
// Find all audio rtp streams and create corresponding remote AudioTracks
|
|
// and MediaStreams.
|
|
if (audio_content) {
|
|
if (audio_content->rejected) {
|
|
RemoveSenders(cricket::MEDIA_TYPE_AUDIO);
|
|
} else {
|
|
bool default_audio_track_needed =
|
|
!remote_peer_supports_msid_ &&
|
|
RtpTransceiverDirectionHasSend(audio_desc->direction());
|
|
UpdateRemoteSendersList(GetActiveStreams(audio_desc),
|
|
default_audio_track_needed, audio_desc->type(),
|
|
new_streams);
|
|
}
|
|
}
|
|
|
|
// Find all video rtp streams and create corresponding remote VideoTracks
|
|
// and MediaStreams.
|
|
if (video_content) {
|
|
if (video_content->rejected) {
|
|
RemoveSenders(cricket::MEDIA_TYPE_VIDEO);
|
|
} else {
|
|
bool default_video_track_needed =
|
|
!remote_peer_supports_msid_ &&
|
|
RtpTransceiverDirectionHasSend(video_desc->direction());
|
|
UpdateRemoteSendersList(GetActiveStreams(video_desc),
|
|
default_video_track_needed, video_desc->type(),
|
|
new_streams);
|
|
}
|
|
}
|
|
|
|
// Update the DataChannels with the information from the remote peer.
|
|
if (data_desc) {
|
|
if (rtc::starts_with(data_desc->protocol().data(),
|
|
cricket::kMediaProtocolRtpPrefix)) {
|
|
UpdateRemoteRtpDataChannels(GetActiveStreams(data_desc));
|
|
}
|
|
}
|
|
|
|
// Iterate new_streams and notify the observer about new MediaStreams.
|
|
for (size_t i = 0; i < new_streams->count(); ++i) {
|
|
MediaStreamInterface* new_stream = new_streams->at(i);
|
|
stats_->AddStream(new_stream);
|
|
observer_->OnAddStream(
|
|
rtc::scoped_refptr<MediaStreamInterface>(new_stream));
|
|
}
|
|
|
|
UpdateEndedRemoteMediaStreams();
|
|
|
|
return RTCError::OK();
|
|
}
|
|
|
|
const cricket::ContentInfo* PeerConnection::FindMediaSectionForTransceiver(
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
transceiver,
|
|
const SessionDescriptionInterface* sdesc) const {
|
|
RTC_DCHECK(transceiver);
|
|
RTC_DCHECK(sdesc);
|
|
if (IsUnifiedPlan()) {
|
|
if (!transceiver->internal()->mid()) {
|
|
// This transceiver is not associated with a media section yet.
|
|
return nullptr;
|
|
}
|
|
return sdesc->description()->GetContentByName(
|
|
*transceiver->internal()->mid());
|
|
} else {
|
|
// Plan B only allows at most one audio and one video section, so use the
|
|
// first media section of that type.
|
|
return cricket::GetFirstMediaContent(sdesc->description()->contents(),
|
|
transceiver->internal()->media_type());
|
|
}
|
|
}
|
|
|
|
PeerConnectionInterface::RTCConfiguration PeerConnection::GetConfiguration() {
|
|
return configuration_;
|
|
}
|
|
|
|
bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration,
|
|
RTCError* error) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
|
|
|
|
if (local_description() && configuration.ice_candidate_pool_size !=
|
|
configuration_.ice_candidate_pool_size) {
|
|
RTC_LOG(LS_ERROR) << "Can't change candidate pool size after calling "
|
|
"SetLocalDescription.";
|
|
return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
|
|
}
|
|
|
|
// The simplest (and most future-compatible) way to tell if the config was
|
|
// modified in an invalid way is to copy each property we do support
|
|
// modifying, then use operator==. There are far more properties we don't
|
|
// support modifying than those we do, and more could be added.
|
|
RTCConfiguration modified_config = configuration_;
|
|
modified_config.servers = configuration.servers;
|
|
modified_config.type = configuration.type;
|
|
modified_config.ice_candidate_pool_size =
|
|
configuration.ice_candidate_pool_size;
|
|
modified_config.prune_turn_ports = configuration.prune_turn_ports;
|
|
modified_config.ice_check_min_interval = configuration.ice_check_min_interval;
|
|
modified_config.turn_customizer = configuration.turn_customizer;
|
|
if (configuration != modified_config) {
|
|
RTC_LOG(LS_ERROR) << "Modifying the configuration in an unsupported way.";
|
|
return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
|
|
}
|
|
|
|
// Validate the modified configuration.
|
|
RTCError validate_error = ValidateConfiguration(modified_config);
|
|
if (!validate_error.ok()) {
|
|
return SafeSetError(std::move(validate_error), error);
|
|
}
|
|
|
|
// Note that this isn't possible through chromium, since it's an unsigned
|
|
// short in WebIDL.
|
|
if (configuration.ice_candidate_pool_size < 0 ||
|
|
configuration.ice_candidate_pool_size > UINT16_MAX) {
|
|
return SafeSetError(RTCErrorType::INVALID_RANGE, error);
|
|
}
|
|
|
|
// Parse ICE servers before hopping to network thread.
|
|
cricket::ServerAddresses stun_servers;
|
|
std::vector<cricket::RelayServerConfig> turn_servers;
|
|
RTCErrorType parse_error =
|
|
ParseIceServers(configuration.servers, &stun_servers, &turn_servers);
|
|
if (parse_error != RTCErrorType::NONE) {
|
|
return SafeSetError(parse_error, error);
|
|
}
|
|
|
|
// In theory this shouldn't fail.
|
|
if (!network_thread()->Invoke<bool>(
|
|
RTC_FROM_HERE,
|
|
rtc::Bind(&PeerConnection::ReconfigurePortAllocator_n, this,
|
|
stun_servers, turn_servers, modified_config.type,
|
|
modified_config.ice_candidate_pool_size,
|
|
modified_config.prune_turn_ports,
|
|
modified_config.turn_customizer))) {
|
|
RTC_LOG(LS_ERROR) << "Failed to apply configuration to PortAllocator.";
|
|
return SafeSetError(RTCErrorType::INTERNAL_ERROR, error);
|
|
}
|
|
|
|
// As described in JSEP, calling setConfiguration with new ICE servers or
|
|
// candidate policy must set a "needs-ice-restart" bit so that the next offer
|
|
// triggers an ICE restart which will pick up the changes.
|
|
if (modified_config.servers != configuration_.servers ||
|
|
modified_config.type != configuration_.type ||
|
|
modified_config.prune_turn_ports != configuration_.prune_turn_ports) {
|
|
transport_controller_->SetNeedsIceRestartFlag();
|
|
}
|
|
|
|
if (modified_config.ice_check_min_interval !=
|
|
configuration_.ice_check_min_interval) {
|
|
transport_controller_->SetIceConfig(ParseIceConfig(modified_config));
|
|
}
|
|
|
|
configuration_ = modified_config;
|
|
return SafeSetError(RTCErrorType::NONE, error);
|
|
}
|
|
|
|
bool PeerConnection::AddIceCandidate(
|
|
const IceCandidateInterface* ice_candidate) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::AddIceCandidate");
|
|
if (IsClosed()) {
|
|
return false;
|
|
}
|
|
|
|
if (!remote_description()) {
|
|
RTC_LOG(LS_ERROR) << "ProcessIceMessage: ICE candidates can't be added "
|
|
<< "without any remote session description.";
|
|
return false;
|
|
}
|
|
|
|
if (!ice_candidate) {
|
|
RTC_LOG(LS_ERROR) << "ProcessIceMessage: Candidate is NULL.";
|
|
return false;
|
|
}
|
|
|
|
bool valid = false;
|
|
bool ready = ReadyToUseRemoteCandidate(ice_candidate, nullptr, &valid);
|
|
if (!valid) {
|
|
return false;
|
|
}
|
|
|
|
// Add this candidate to the remote session description.
|
|
if (!mutable_remote_description()->AddCandidate(ice_candidate)) {
|
|
RTC_LOG(LS_ERROR) << "ProcessIceMessage: Candidate cannot be used.";
|
|
return false;
|
|
}
|
|
|
|
if (ready) {
|
|
return UseCandidate(ice_candidate);
|
|
} else {
|
|
RTC_LOG(LS_INFO) << "ProcessIceMessage: Not ready to use candidate.";
|
|
return true;
|
|
}
|
|
}
|
|
|
|
bool PeerConnection::RemoveIceCandidates(
|
|
const std::vector<cricket::Candidate>& candidates) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::RemoveIceCandidates");
|
|
if (!remote_description()) {
|
|
RTC_LOG(LS_ERROR) << "RemoveRemoteIceCandidates: ICE candidates can't be "
|
|
<< "removed without any remote session description.";
|
|
return false;
|
|
}
|
|
|
|
if (candidates.empty()) {
|
|
RTC_LOG(LS_ERROR) << "RemoveRemoteIceCandidates: candidates are empty.";
|
|
return false;
|
|
}
|
|
|
|
size_t number_removed =
|
|
mutable_remote_description()->RemoveCandidates(candidates);
|
|
if (number_removed != candidates.size()) {
|
|
RTC_LOG(LS_ERROR)
|
|
<< "RemoveRemoteIceCandidates: Failed to remove candidates. "
|
|
<< "Requested " << candidates.size() << " but only " << number_removed
|
|
<< " are removed.";
|
|
}
|
|
|
|
// Remove the candidates from the transport controller.
|
|
std::string error;
|
|
bool res = transport_controller_->RemoveRemoteCandidates(candidates, &error);
|
|
if (!res && !error.empty()) {
|
|
RTC_LOG(LS_ERROR) << "Error when removing remote candidates: " << error;
|
|
}
|
|
return true;
|
|
}
|
|
|
|
void PeerConnection::RegisterUMAObserver(UMAObserver* observer) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::RegisterUmaObserver");
|
|
uma_observer_ = observer;
|
|
|
|
if (transport_controller()) {
|
|
transport_controller()->SetMetricsObserver(uma_observer_);
|
|
}
|
|
|
|
// Send information about IPv4/IPv6 status.
|
|
if (uma_observer_) {
|
|
port_allocator_->SetMetricsObserver(uma_observer_);
|
|
if (port_allocator_->flags() & cricket::PORTALLOCATOR_ENABLE_IPV6) {
|
|
uma_observer_->IncrementEnumCounter(
|
|
kEnumCounterAddressFamily, kPeerConnection_IPv6,
|
|
kPeerConnectionAddressFamilyCounter_Max);
|
|
} else {
|
|
uma_observer_->IncrementEnumCounter(
|
|
kEnumCounterAddressFamily, kPeerConnection_IPv4,
|
|
kPeerConnectionAddressFamilyCounter_Max);
|
|
}
|
|
}
|
|
}
|
|
|
|
RTCError PeerConnection::SetBitrate(const BitrateParameters& bitrate) {
|
|
if (!worker_thread()->IsCurrent()) {
|
|
return worker_thread()->Invoke<RTCError>(
|
|
RTC_FROM_HERE, rtc::Bind(&PeerConnection::SetBitrate, this, bitrate));
|
|
}
|
|
|
|
const bool has_min = static_cast<bool>(bitrate.min_bitrate_bps);
|
|
const bool has_current = static_cast<bool>(bitrate.current_bitrate_bps);
|
|
const bool has_max = static_cast<bool>(bitrate.max_bitrate_bps);
|
|
if (has_min && *bitrate.min_bitrate_bps < 0) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"min_bitrate_bps <= 0");
|
|
}
|
|
if (has_current) {
|
|
if (has_min && *bitrate.current_bitrate_bps < *bitrate.min_bitrate_bps) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"current_bitrate_bps < min_bitrate_bps");
|
|
} else if (*bitrate.current_bitrate_bps < 0) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"curent_bitrate_bps < 0");
|
|
}
|
|
}
|
|
if (has_max) {
|
|
if (has_current &&
|
|
*bitrate.max_bitrate_bps < *bitrate.current_bitrate_bps) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"max_bitrate_bps < current_bitrate_bps");
|
|
} else if (has_min && *bitrate.max_bitrate_bps < *bitrate.min_bitrate_bps) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"max_bitrate_bps < min_bitrate_bps");
|
|
} else if (*bitrate.max_bitrate_bps < 0) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"max_bitrate_bps < 0");
|
|
}
|
|
}
|
|
|
|
Call::Config::BitrateConfigMask mask;
|
|
mask.min_bitrate_bps = bitrate.min_bitrate_bps;
|
|
mask.start_bitrate_bps = bitrate.current_bitrate_bps;
|
|
mask.max_bitrate_bps = bitrate.max_bitrate_bps;
|
|
|
|
RTC_DCHECK(call_.get());
|
|
call_->SetBitrateConfigMask(mask);
|
|
|
|
return RTCError::OK();
|
|
}
|
|
|
|
void PeerConnection::SetBitrateAllocationStrategy(
|
|
std::unique_ptr<rtc::BitrateAllocationStrategy>
|
|
bitrate_allocation_strategy) {
|
|
rtc::Thread* worker_thread = factory_->worker_thread();
|
|
if (!worker_thread->IsCurrent()) {
|
|
rtc::BitrateAllocationStrategy* strategy_raw =
|
|
bitrate_allocation_strategy.release();
|
|
auto functor = [this, strategy_raw]() {
|
|
call_->SetBitrateAllocationStrategy(
|
|
rtc::WrapUnique<rtc::BitrateAllocationStrategy>(strategy_raw));
|
|
};
|
|
worker_thread->Invoke<void>(RTC_FROM_HERE, functor);
|
|
return;
|
|
}
|
|
RTC_DCHECK(call_.get());
|
|
call_->SetBitrateAllocationStrategy(std::move(bitrate_allocation_strategy));
|
|
}
|
|
|
|
void PeerConnection::SetAudioPlayout(bool playout) {
|
|
if (!worker_thread()->IsCurrent()) {
|
|
worker_thread()->Invoke<void>(
|
|
RTC_FROM_HERE,
|
|
rtc::Bind(&PeerConnection::SetAudioPlayout, this, playout));
|
|
return;
|
|
}
|
|
auto audio_state =
|
|
factory_->channel_manager()->media_engine()->GetAudioState();
|
|
audio_state->SetPlayout(playout);
|
|
}
|
|
|
|
void PeerConnection::SetAudioRecording(bool recording) {
|
|
if (!worker_thread()->IsCurrent()) {
|
|
worker_thread()->Invoke<void>(
|
|
RTC_FROM_HERE,
|
|
rtc::Bind(&PeerConnection::SetAudioRecording, this, recording));
|
|
return;
|
|
}
|
|
auto audio_state =
|
|
factory_->channel_manager()->media_engine()->GetAudioState();
|
|
audio_state->SetRecording(recording);
|
|
}
|
|
|
|
std::unique_ptr<rtc::SSLCertificate>
|
|
PeerConnection::GetRemoteAudioSSLCertificate() {
|
|
if (!voice_channel()) {
|
|
return nullptr;
|
|
}
|
|
return GetRemoteSSLCertificate(voice_channel()->transport_name());
|
|
}
|
|
|
|
bool PeerConnection::StartRtcEventLog(rtc::PlatformFile file,
|
|
int64_t max_size_bytes) {
|
|
// TODO(eladalon): It would be better to not allow negative values into PC.
|
|
const size_t max_size = (max_size_bytes < 0)
|
|
? RtcEventLog::kUnlimitedOutput
|
|
: rtc::saturated_cast<size_t>(max_size_bytes);
|
|
return StartRtcEventLog(
|
|
rtc::MakeUnique<RtcEventLogOutputFile>(file, max_size),
|
|
webrtc::RtcEventLog::kImmediateOutput);
|
|
}
|
|
|
|
bool PeerConnection::StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
|
|
int64_t output_period_ms) {
|
|
// TODO(eladalon): In C++14, this can be done with a lambda.
|
|
struct Functor {
|
|
bool operator()() {
|
|
return pc->StartRtcEventLog_w(std::move(output), output_period_ms);
|
|
}
|
|
PeerConnection* const pc;
|
|
std::unique_ptr<RtcEventLogOutput> output;
|
|
const int64_t output_period_ms;
|
|
};
|
|
return worker_thread()->Invoke<bool>(
|
|
RTC_FROM_HERE, Functor{this, std::move(output), output_period_ms});
|
|
}
|
|
|
|
void PeerConnection::StopRtcEventLog() {
|
|
worker_thread()->Invoke<void>(
|
|
RTC_FROM_HERE, rtc::Bind(&PeerConnection::StopRtcEventLog_w, this));
|
|
}
|
|
|
|
const SessionDescriptionInterface* PeerConnection::local_description() const {
|
|
return pending_local_description_ ? pending_local_description_.get()
|
|
: current_local_description_.get();
|
|
}
|
|
|
|
const SessionDescriptionInterface* PeerConnection::remote_description() const {
|
|
return pending_remote_description_ ? pending_remote_description_.get()
|
|
: current_remote_description_.get();
|
|
}
|
|
|
|
const SessionDescriptionInterface* PeerConnection::current_local_description()
|
|
const {
|
|
return current_local_description_.get();
|
|
}
|
|
|
|
const SessionDescriptionInterface* PeerConnection::current_remote_description()
|
|
const {
|
|
return current_remote_description_.get();
|
|
}
|
|
|
|
const SessionDescriptionInterface* PeerConnection::pending_local_description()
|
|
const {
|
|
return pending_local_description_.get();
|
|
}
|
|
|
|
const SessionDescriptionInterface* PeerConnection::pending_remote_description()
|
|
const {
|
|
return pending_remote_description_.get();
|
|
}
|
|
|
|
void PeerConnection::Close() {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::Close");
|
|
// Update stats here so that we have the most recent stats for tracks and
|
|
// streams before the channels are closed.
|
|
stats_->UpdateStats(kStatsOutputLevelStandard);
|
|
|
|
ChangeSignalingState(PeerConnectionInterface::kClosed);
|
|
|
|
for (auto transceiver : transceivers_) {
|
|
transceiver->Stop();
|
|
}
|
|
DestroyAllChannels();
|
|
|
|
network_thread()->Invoke<void>(
|
|
RTC_FROM_HERE,
|
|
rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
|
|
port_allocator_.get()));
|
|
|
|
worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
|
|
call_.reset();
|
|
// The event log must outlive call (and any other object that uses it).
|
|
event_log_.reset();
|
|
});
|
|
}
|
|
|
|
void PeerConnection::OnMessage(rtc::Message* msg) {
|
|
switch (msg->message_id) {
|
|
case MSG_SET_SESSIONDESCRIPTION_SUCCESS: {
|
|
SetSessionDescriptionMsg* param =
|
|
static_cast<SetSessionDescriptionMsg*>(msg->pdata);
|
|
param->observer->OnSuccess();
|
|
delete param;
|
|
break;
|
|
}
|
|
case MSG_SET_SESSIONDESCRIPTION_FAILED: {
|
|
SetSessionDescriptionMsg* param =
|
|
static_cast<SetSessionDescriptionMsg*>(msg->pdata);
|
|
param->observer->OnFailure(param->error);
|
|
delete param;
|
|
break;
|
|
}
|
|
case MSG_CREATE_SESSIONDESCRIPTION_FAILED: {
|
|
CreateSessionDescriptionMsg* param =
|
|
static_cast<CreateSessionDescriptionMsg*>(msg->pdata);
|
|
param->observer->OnFailure(param->error);
|
|
delete param;
|
|
break;
|
|
}
|
|
case MSG_GETSTATS: {
|
|
GetStatsMsg* param = static_cast<GetStatsMsg*>(msg->pdata);
|
|
StatsReports reports;
|
|
stats_->GetStats(param->track, &reports);
|
|
param->observer->OnComplete(reports);
|
|
delete param;
|
|
break;
|
|
}
|
|
case MSG_FREE_DATACHANNELS: {
|
|
sctp_data_channels_to_free_.clear();
|
|
break;
|
|
}
|
|
default:
|
|
RTC_NOTREACHED() << "Not implemented";
|
|
break;
|
|
}
|
|
}
|
|
|
|
void PeerConnection::CreateAudioReceiver(
|
|
MediaStreamInterface* stream,
|
|
const RtpSenderInfo& remote_sender_info) {
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams;
|
|
streams.push_back(rtc::scoped_refptr<MediaStreamInterface>(stream));
|
|
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
|
|
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
|
|
signaling_thread(),
|
|
new AudioRtpReceiver(remote_sender_info.sender_id, streams,
|
|
remote_sender_info.first_ssrc, voice_channel()));
|
|
stream->AddTrack(
|
|
static_cast<AudioTrackInterface*>(receiver->internal()->track().get()));
|
|
GetAudioTransceiver()->internal()->AddReceiver(receiver);
|
|
observer_->OnAddTrack(receiver, std::move(streams));
|
|
}
|
|
|
|
void PeerConnection::CreateVideoReceiver(
|
|
MediaStreamInterface* stream,
|
|
const RtpSenderInfo& remote_sender_info) {
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams;
|
|
streams.push_back(rtc::scoped_refptr<MediaStreamInterface>(stream));
|
|
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
|
|
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
|
|
signaling_thread(),
|
|
new VideoRtpReceiver(remote_sender_info.sender_id, streams,
|
|
worker_thread(), remote_sender_info.first_ssrc,
|
|
video_channel()));
|
|
stream->AddTrack(
|
|
static_cast<VideoTrackInterface*>(receiver->internal()->track().get()));
|
|
GetVideoTransceiver()->internal()->AddReceiver(receiver);
|
|
observer_->OnAddTrack(receiver, std::move(streams));
|
|
}
|
|
|
|
// TODO(deadbeef): Keep RtpReceivers around even if track goes away in remote
|
|
// description.
|
|
rtc::scoped_refptr<RtpReceiverInterface> PeerConnection::RemoveAndStopReceiver(
|
|
const RtpSenderInfo& remote_sender_info) {
|
|
auto receiver = FindReceiverById(remote_sender_info.sender_id);
|
|
if (!receiver) {
|
|
RTC_LOG(LS_WARNING) << "RtpReceiver for track with id "
|
|
<< remote_sender_info.sender_id << " doesn't exist.";
|
|
return nullptr;
|
|
}
|
|
if (receiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
|
|
GetAudioTransceiver()->internal()->RemoveReceiver(receiver);
|
|
} else {
|
|
GetVideoTransceiver()->internal()->RemoveReceiver(receiver);
|
|
}
|
|
return receiver;
|
|
}
|
|
|
|
void PeerConnection::AddAudioTrack(AudioTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
RTC_DCHECK(!IsClosed());
|
|
auto sender = FindSenderForTrack(track);
|
|
if (sender) {
|
|
// We already have a sender for this track, so just change the stream_id
|
|
// so that it's correct in the next call to CreateOffer.
|
|
sender->internal()->set_stream_id(stream->label());
|
|
return;
|
|
}
|
|
|
|
// Normal case; we've never seen this track before.
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender =
|
|
RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
|
signaling_thread(),
|
|
new AudioRtpSender(track, {stream->label()}, voice_channel(),
|
|
stats_.get()));
|
|
GetAudioTransceiver()->internal()->AddSender(new_sender);
|
|
// If the sender has already been configured in SDP, we call SetSsrc,
|
|
// which will connect the sender to the underlying transport. This can
|
|
// occur if a local session description that contains the ID of the sender
|
|
// is set before AddStream is called. It can also occur if the local
|
|
// session description is not changed and RemoveStream is called, and
|
|
// later AddStream is called again with the same stream.
|
|
const RtpSenderInfo* sender_info =
|
|
FindSenderInfo(local_audio_sender_infos_, stream->label(), track->id());
|
|
if (sender_info) {
|
|
new_sender->internal()->SetSsrc(sender_info->first_ssrc);
|
|
}
|
|
}
|
|
|
|
// TODO(deadbeef): Don't destroy RtpSenders here; they should be kept around
|
|
// indefinitely, when we have unified plan SDP.
|
|
void PeerConnection::RemoveAudioTrack(AudioTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
RTC_DCHECK(!IsClosed());
|
|
auto sender = FindSenderForTrack(track);
|
|
if (!sender) {
|
|
RTC_LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
|
|
<< " doesn't exist.";
|
|
return;
|
|
}
|
|
GetAudioTransceiver()->internal()->RemoveSender(sender);
|
|
}
|
|
|
|
void PeerConnection::AddVideoTrack(VideoTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
RTC_DCHECK(!IsClosed());
|
|
auto sender = FindSenderForTrack(track);
|
|
if (sender) {
|
|
// We already have a sender for this track, so just change the stream_id
|
|
// so that it's correct in the next call to CreateOffer.
|
|
sender->internal()->set_stream_id(stream->label());
|
|
return;
|
|
}
|
|
|
|
// Normal case; we've never seen this track before.
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender =
|
|
RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
|
signaling_thread(),
|
|
new VideoRtpSender(track, {stream->label()}, video_channel()));
|
|
GetVideoTransceiver()->internal()->AddSender(new_sender);
|
|
const RtpSenderInfo* sender_info =
|
|
FindSenderInfo(local_video_sender_infos_, stream->label(), track->id());
|
|
if (sender_info) {
|
|
new_sender->internal()->SetSsrc(sender_info->first_ssrc);
|
|
}
|
|
}
|
|
|
|
void PeerConnection::RemoveVideoTrack(VideoTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
RTC_DCHECK(!IsClosed());
|
|
auto sender = FindSenderForTrack(track);
|
|
if (!sender) {
|
|
RTC_LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
|
|
<< " doesn't exist.";
|
|
return;
|
|
}
|
|
GetVideoTransceiver()->internal()->RemoveSender(sender);
|
|
}
|
|
|
|
void PeerConnection::SetIceConnectionState(IceConnectionState new_state) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
if (ice_connection_state_ == new_state) {
|
|
return;
|
|
}
|
|
|
|
// After transitioning to "closed", ignore any additional states from
|
|
// TransportController (such as "disconnected").
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
|
|
RTC_LOG(LS_INFO) << "Changing IceConnectionState " << ice_connection_state_
|
|
<< " => " << new_state;
|
|
RTC_DCHECK(ice_connection_state_ !=
|
|
PeerConnectionInterface::kIceConnectionClosed);
|
|
|
|
ice_connection_state_ = new_state;
|
|
observer_->OnIceConnectionChange(ice_connection_state_);
|
|
}
|
|
|
|
void PeerConnection::OnIceGatheringChange(
|
|
PeerConnectionInterface::IceGatheringState new_state) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
ice_gathering_state_ = new_state;
|
|
observer_->OnIceGatheringChange(ice_gathering_state_);
|
|
}
|
|
|
|
void PeerConnection::OnIceCandidate(
|
|
std::unique_ptr<IceCandidateInterface> candidate) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
observer_->OnIceCandidate(candidate.get());
|
|
}
|
|
|
|
void PeerConnection::OnIceCandidatesRemoved(
|
|
const std::vector<cricket::Candidate>& candidates) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
observer_->OnIceCandidatesRemoved(candidates);
|
|
}
|
|
|
|
void PeerConnection::ChangeSignalingState(
|
|
PeerConnectionInterface::SignalingState signaling_state) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
if (signaling_state_ == signaling_state) {
|
|
return;
|
|
}
|
|
RTC_LOG(LS_INFO) << "Session: " << session_id() << " Old state: "
|
|
<< GetSignalingStateString(signaling_state_)
|
|
<< " New state: "
|
|
<< GetSignalingStateString(signaling_state);
|
|
signaling_state_ = signaling_state;
|
|
if (signaling_state == kClosed) {
|
|
ice_connection_state_ = kIceConnectionClosed;
|
|
observer_->OnIceConnectionChange(ice_connection_state_);
|
|
if (ice_gathering_state_ != kIceGatheringComplete) {
|
|
ice_gathering_state_ = kIceGatheringComplete;
|
|
observer_->OnIceGatheringChange(ice_gathering_state_);
|
|
}
|
|
}
|
|
observer_->OnSignalingChange(signaling_state_);
|
|
}
|
|
|
|
void PeerConnection::OnAudioTrackAdded(AudioTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
AddAudioTrack(track, stream);
|
|
observer_->OnRenegotiationNeeded();
|
|
}
|
|
|
|
void PeerConnection::OnAudioTrackRemoved(AudioTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
RemoveAudioTrack(track, stream);
|
|
observer_->OnRenegotiationNeeded();
|
|
}
|
|
|
|
void PeerConnection::OnVideoTrackAdded(VideoTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
AddVideoTrack(track, stream);
|
|
observer_->OnRenegotiationNeeded();
|
|
}
|
|
|
|
void PeerConnection::OnVideoTrackRemoved(VideoTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
RemoveVideoTrack(track, stream);
|
|
observer_->OnRenegotiationNeeded();
|
|
}
|
|
|
|
void PeerConnection::PostSetSessionDescriptionSuccess(
|
|
SetSessionDescriptionObserver* observer) {
|
|
SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
|
|
signaling_thread()->Post(RTC_FROM_HERE, this,
|
|
MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
|
|
}
|
|
|
|
void PeerConnection::PostSetSessionDescriptionFailure(
|
|
SetSessionDescriptionObserver* observer,
|
|
const std::string& error) {
|
|
SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
|
|
msg->error = error;
|
|
signaling_thread()->Post(RTC_FROM_HERE, this,
|
|
MSG_SET_SESSIONDESCRIPTION_FAILED, msg);
|
|
}
|
|
|
|
void PeerConnection::PostCreateSessionDescriptionFailure(
|
|
CreateSessionDescriptionObserver* observer,
|
|
const std::string& error) {
|
|
CreateSessionDescriptionMsg* msg = new CreateSessionDescriptionMsg(observer);
|
|
msg->error = error;
|
|
signaling_thread()->Post(RTC_FROM_HERE, this,
|
|
MSG_CREATE_SESSIONDESCRIPTION_FAILED, msg);
|
|
}
|
|
|
|
void PeerConnection::GetOptionsForOffer(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
ExtractSharedMediaSessionOptions(rtc_options, session_options);
|
|
|
|
// Figure out transceiver directional preferences.
|
|
bool send_audio = HasRtpSender(cricket::MEDIA_TYPE_AUDIO);
|
|
bool send_video = HasRtpSender(cricket::MEDIA_TYPE_VIDEO);
|
|
|
|
// By default, generate sendrecv/recvonly m= sections.
|
|
bool recv_audio = true;
|
|
bool recv_video = true;
|
|
|
|
// By default, only offer a new m= section if we have media to send with it.
|
|
bool offer_new_audio_description = send_audio;
|
|
bool offer_new_video_description = send_video;
|
|
bool offer_new_data_description = HasDataChannels();
|
|
|
|
// The "offer_to_receive_X" options allow those defaults to be overridden.
|
|
if (rtc_options.offer_to_receive_audio != RTCOfferAnswerOptions::kUndefined) {
|
|
recv_audio = (rtc_options.offer_to_receive_audio > 0);
|
|
offer_new_audio_description =
|
|
offer_new_audio_description || (rtc_options.offer_to_receive_audio > 0);
|
|
}
|
|
if (rtc_options.offer_to_receive_video != RTCOfferAnswerOptions::kUndefined) {
|
|
recv_video = (rtc_options.offer_to_receive_video > 0);
|
|
offer_new_video_description =
|
|
offer_new_video_description || (rtc_options.offer_to_receive_video > 0);
|
|
}
|
|
|
|
rtc::Optional<size_t> audio_index;
|
|
rtc::Optional<size_t> video_index;
|
|
rtc::Optional<size_t> data_index;
|
|
// If a current description exists, generate m= sections in the same order,
|
|
// using the first audio/video/data section that appears and rejecting
|
|
// extraneous ones.
|
|
if (local_description()) {
|
|
GenerateMediaDescriptionOptions(
|
|
local_description(),
|
|
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio),
|
|
RtpTransceiverDirectionFromSendRecv(send_video, recv_video),
|
|
&audio_index, &video_index, &data_index, session_options);
|
|
}
|
|
|
|
// Add audio/video/data m= sections to the end if needed.
|
|
if (!audio_index && offer_new_audio_description) {
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(
|
|
cricket::MEDIA_TYPE_AUDIO, cricket::CN_AUDIO,
|
|
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio),
|
|
false));
|
|
audio_index = session_options->media_description_options.size() - 1;
|
|
}
|
|
if (!video_index && offer_new_video_description) {
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(
|
|
cricket::MEDIA_TYPE_VIDEO, cricket::CN_VIDEO,
|
|
RtpTransceiverDirectionFromSendRecv(send_video, recv_video),
|
|
false));
|
|
video_index = session_options->media_description_options.size() - 1;
|
|
}
|
|
if (!data_index && offer_new_data_description) {
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(
|
|
cricket::MEDIA_TYPE_DATA, cricket::CN_DATA,
|
|
RtpTransceiverDirection::kSendRecv, false));
|
|
data_index = session_options->media_description_options.size() - 1;
|
|
}
|
|
|
|
cricket::MediaDescriptionOptions* audio_media_description_options =
|
|
!audio_index ? nullptr
|
|
: &session_options->media_description_options[*audio_index];
|
|
cricket::MediaDescriptionOptions* video_media_description_options =
|
|
!video_index ? nullptr
|
|
: &session_options->media_description_options[*video_index];
|
|
cricket::MediaDescriptionOptions* data_media_description_options =
|
|
!data_index ? nullptr
|
|
: &session_options->media_description_options[*data_index];
|
|
|
|
// Apply ICE restart flag and renomination flag.
|
|
for (auto& options : session_options->media_description_options) {
|
|
options.transport_options.ice_restart = rtc_options.ice_restart;
|
|
options.transport_options.enable_ice_renomination =
|
|
configuration_.enable_ice_renomination;
|
|
}
|
|
|
|
AddRtpSenderOptions(GetSendersInternal(), audio_media_description_options,
|
|
video_media_description_options);
|
|
AddRtpDataChannelOptions(rtp_data_channels_, data_media_description_options);
|
|
|
|
// Intentionally unset the data channel type for RTP data channel with the
|
|
// second condition. Otherwise the RTP data channels would be successfully
|
|
// negotiated by default and the unit tests in WebRtcDataBrowserTest will fail
|
|
// when building with chromium. We want to leave RTP data channels broken, so
|
|
// people won't try to use them.
|
|
if (!rtp_data_channels_.empty() || data_channel_type() != cricket::DCT_RTP) {
|
|
session_options->data_channel_type = data_channel_type();
|
|
}
|
|
|
|
session_options->rtcp_cname = rtcp_cname_;
|
|
session_options->crypto_options = factory_->options().crypto_options;
|
|
}
|
|
|
|
void PeerConnection::GetOptionsForAnswer(
|
|
const RTCOfferAnswerOptions& rtc_options,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
ExtractSharedMediaSessionOptions(rtc_options, session_options);
|
|
|
|
// Figure out transceiver directional preferences.
|
|
bool send_audio = HasRtpSender(cricket::MEDIA_TYPE_AUDIO);
|
|
bool send_video = HasRtpSender(cricket::MEDIA_TYPE_VIDEO);
|
|
|
|
// By default, generate sendrecv/recvonly m= sections. The direction is also
|
|
// restricted by the direction in the offer.
|
|
bool recv_audio = true;
|
|
bool recv_video = true;
|
|
|
|
// The "offer_to_receive_X" options allow those defaults to be overridden.
|
|
if (rtc_options.offer_to_receive_audio != RTCOfferAnswerOptions::kUndefined) {
|
|
recv_audio = (rtc_options.offer_to_receive_audio > 0);
|
|
}
|
|
if (rtc_options.offer_to_receive_video != RTCOfferAnswerOptions::kUndefined) {
|
|
recv_video = (rtc_options.offer_to_receive_video > 0);
|
|
}
|
|
|
|
rtc::Optional<size_t> audio_index;
|
|
rtc::Optional<size_t> video_index;
|
|
rtc::Optional<size_t> data_index;
|
|
if (remote_description()) {
|
|
// The pending remote description should be an offer.
|
|
RTC_DCHECK(remote_description()->GetType() == SdpType::kOffer);
|
|
// Generate m= sections that match those in the offer.
|
|
// Note that mediasession.cc will handle intersection our preferred
|
|
// direction with the offered direction.
|
|
GenerateMediaDescriptionOptions(
|
|
remote_description(),
|
|
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio),
|
|
RtpTransceiverDirectionFromSendRecv(send_video, recv_video),
|
|
&audio_index, &video_index, &data_index, session_options);
|
|
}
|
|
|
|
cricket::MediaDescriptionOptions* audio_media_description_options =
|
|
!audio_index ? nullptr
|
|
: &session_options->media_description_options[*audio_index];
|
|
cricket::MediaDescriptionOptions* video_media_description_options =
|
|
!video_index ? nullptr
|
|
: &session_options->media_description_options[*video_index];
|
|
cricket::MediaDescriptionOptions* data_media_description_options =
|
|
!data_index ? nullptr
|
|
: &session_options->media_description_options[*data_index];
|
|
|
|
// Apply ICE renomination flag.
|
|
for (auto& options : session_options->media_description_options) {
|
|
options.transport_options.enable_ice_renomination =
|
|
configuration_.enable_ice_renomination;
|
|
}
|
|
|
|
AddRtpSenderOptions(GetSendersInternal(), audio_media_description_options,
|
|
video_media_description_options);
|
|
AddRtpDataChannelOptions(rtp_data_channels_, data_media_description_options);
|
|
|
|
// Intentionally unset the data channel type for RTP data channel. Otherwise
|
|
// the RTP data channels would be successfully negotiated by default and the
|
|
// unit tests in WebRtcDataBrowserTest will fail when building with chromium.
|
|
// We want to leave RTP data channels broken, so people won't try to use them.
|
|
if (!rtp_data_channels_.empty() || data_channel_type() != cricket::DCT_RTP) {
|
|
session_options->data_channel_type = data_channel_type();
|
|
}
|
|
|
|
session_options->rtcp_cname = rtcp_cname_;
|
|
session_options->crypto_options = factory_->options().crypto_options;
|
|
}
|
|
|
|
void PeerConnection::GenerateMediaDescriptionOptions(
|
|
const SessionDescriptionInterface* session_desc,
|
|
RtpTransceiverDirection audio_direction,
|
|
RtpTransceiverDirection video_direction,
|
|
rtc::Optional<size_t>* audio_index,
|
|
rtc::Optional<size_t>* video_index,
|
|
rtc::Optional<size_t>* data_index,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
for (const cricket::ContentInfo& content :
|
|
session_desc->description()->contents()) {
|
|
if (IsAudioContent(&content)) {
|
|
// If we already have an audio m= section, reject this extra one.
|
|
if (*audio_index) {
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(
|
|
cricket::MEDIA_TYPE_AUDIO, content.name,
|
|
RtpTransceiverDirection::kInactive, true));
|
|
} else {
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(
|
|
cricket::MEDIA_TYPE_AUDIO, content.name, audio_direction,
|
|
audio_direction == RtpTransceiverDirection::kInactive));
|
|
*audio_index = session_options->media_description_options.size() - 1;
|
|
}
|
|
} else if (IsVideoContent(&content)) {
|
|
// If we already have an video m= section, reject this extra one.
|
|
if (*video_index) {
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(
|
|
cricket::MEDIA_TYPE_VIDEO, content.name,
|
|
RtpTransceiverDirection::kInactive, true));
|
|
} else {
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(
|
|
cricket::MEDIA_TYPE_VIDEO, content.name, video_direction,
|
|
video_direction == RtpTransceiverDirection::kInactive));
|
|
*video_index = session_options->media_description_options.size() - 1;
|
|
}
|
|
} else {
|
|
RTC_DCHECK(IsDataContent(&content));
|
|
// If we already have an data m= section, reject this extra one.
|
|
if (*data_index) {
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(
|
|
cricket::MEDIA_TYPE_DATA, content.name,
|
|
RtpTransceiverDirection::kInactive, true));
|
|
} else {
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(
|
|
cricket::MEDIA_TYPE_DATA, content.name,
|
|
// Direction for data sections is meaningless, but legacy
|
|
// endpoints might expect sendrecv.
|
|
RtpTransceiverDirection::kSendRecv, false));
|
|
*data_index = session_options->media_description_options.size() - 1;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::RemoveSenders(cricket::MediaType media_type) {
|
|
UpdateLocalSenders(std::vector<cricket::StreamParams>(), media_type);
|
|
UpdateRemoteSendersList(std::vector<cricket::StreamParams>(), false,
|
|
media_type, nullptr);
|
|
}
|
|
|
|
void PeerConnection::UpdateRemoteSendersList(
|
|
const cricket::StreamParamsVec& streams,
|
|
bool default_sender_needed,
|
|
cricket::MediaType media_type,
|
|
StreamCollection* new_streams) {
|
|
std::vector<RtpSenderInfo>* current_senders =
|
|
GetRemoteSenderInfos(media_type);
|
|
|
|
// Find removed senders. I.e., senders where the sender id or ssrc don't match
|
|
// the new StreamParam.
|
|
for (auto sender_it = current_senders->begin();
|
|
sender_it != current_senders->end();
|
|
/* incremented manually */) {
|
|
const RtpSenderInfo& info = *sender_it;
|
|
const cricket::StreamParams* params =
|
|
cricket::GetStreamBySsrc(streams, info.first_ssrc);
|
|
bool sender_exists = params && params->id == info.sender_id;
|
|
// If this is a default track, and we still need it, don't remove it.
|
|
if ((info.stream_label == kDefaultStreamLabel && default_sender_needed) ||
|
|
sender_exists) {
|
|
++sender_it;
|
|
} else {
|
|
OnRemoteSenderRemoved(info, media_type);
|
|
sender_it = current_senders->erase(sender_it);
|
|
}
|
|
}
|
|
|
|
// Find new and active senders.
|
|
for (const cricket::StreamParams& params : streams) {
|
|
// The sync_label is the MediaStream label and the |stream.id| is the
|
|
// sender id.
|
|
const std::string& stream_label = params.sync_label;
|
|
const std::string& sender_id = params.id;
|
|
uint32_t ssrc = params.first_ssrc();
|
|
|
|
rtc::scoped_refptr<MediaStreamInterface> stream =
|
|
remote_streams_->find(stream_label);
|
|
if (!stream) {
|
|
// This is a new MediaStream. Create a new remote MediaStream.
|
|
stream = MediaStreamProxy::Create(rtc::Thread::Current(),
|
|
MediaStream::Create(stream_label));
|
|
remote_streams_->AddStream(stream);
|
|
new_streams->AddStream(stream);
|
|
}
|
|
|
|
const RtpSenderInfo* sender_info =
|
|
FindSenderInfo(*current_senders, stream_label, sender_id);
|
|
if (!sender_info) {
|
|
current_senders->push_back(RtpSenderInfo(stream_label, sender_id, ssrc));
|
|
OnRemoteSenderAdded(current_senders->back(), media_type);
|
|
}
|
|
}
|
|
|
|
// Add default sender if necessary.
|
|
if (default_sender_needed) {
|
|
rtc::scoped_refptr<MediaStreamInterface> default_stream =
|
|
remote_streams_->find(kDefaultStreamLabel);
|
|
if (!default_stream) {
|
|
// Create the new default MediaStream.
|
|
default_stream = MediaStreamProxy::Create(
|
|
rtc::Thread::Current(), MediaStream::Create(kDefaultStreamLabel));
|
|
remote_streams_->AddStream(default_stream);
|
|
new_streams->AddStream(default_stream);
|
|
}
|
|
std::string default_sender_id = (media_type == cricket::MEDIA_TYPE_AUDIO)
|
|
? kDefaultAudioSenderId
|
|
: kDefaultVideoSenderId;
|
|
const RtpSenderInfo* default_sender_info = FindSenderInfo(
|
|
*current_senders, kDefaultStreamLabel, default_sender_id);
|
|
if (!default_sender_info) {
|
|
current_senders->push_back(
|
|
RtpSenderInfo(kDefaultStreamLabel, default_sender_id, 0));
|
|
OnRemoteSenderAdded(current_senders->back(), media_type);
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnRemoteSenderAdded(const RtpSenderInfo& sender_info,
|
|
cricket::MediaType media_type) {
|
|
MediaStreamInterface* stream =
|
|
remote_streams_->find(sender_info.stream_label);
|
|
|
|
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
|
|
CreateAudioReceiver(stream, sender_info);
|
|
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
|
|
CreateVideoReceiver(stream, sender_info);
|
|
} else {
|
|
RTC_NOTREACHED() << "Invalid media type";
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnRemoteSenderRemoved(const RtpSenderInfo& sender_info,
|
|
cricket::MediaType media_type) {
|
|
MediaStreamInterface* stream =
|
|
remote_streams_->find(sender_info.stream_label);
|
|
|
|
rtc::scoped_refptr<RtpReceiverInterface> receiver;
|
|
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
|
|
// When the MediaEngine audio channel is destroyed, the RemoteAudioSource
|
|
// will be notified which will end the AudioRtpReceiver::track().
|
|
receiver = RemoveAndStopReceiver(sender_info);
|
|
rtc::scoped_refptr<AudioTrackInterface> audio_track =
|
|
stream->FindAudioTrack(sender_info.sender_id);
|
|
if (audio_track) {
|
|
stream->RemoveTrack(audio_track);
|
|
}
|
|
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
|
|
// Stopping or destroying a VideoRtpReceiver will end the
|
|
// VideoRtpReceiver::track().
|
|
receiver = RemoveAndStopReceiver(sender_info);
|
|
rtc::scoped_refptr<VideoTrackInterface> video_track =
|
|
stream->FindVideoTrack(sender_info.sender_id);
|
|
if (video_track) {
|
|
// There's no guarantee the track is still available, e.g. the track may
|
|
// have been removed from the stream by an application.
|
|
stream->RemoveTrack(video_track);
|
|
}
|
|
} else {
|
|
RTC_NOTREACHED() << "Invalid media type";
|
|
}
|
|
if (receiver) {
|
|
observer_->OnRemoveTrack(receiver);
|
|
}
|
|
}
|
|
|
|
void PeerConnection::UpdateEndedRemoteMediaStreams() {
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_to_remove;
|
|
for (size_t i = 0; i < remote_streams_->count(); ++i) {
|
|
MediaStreamInterface* stream = remote_streams_->at(i);
|
|
if (stream->GetAudioTracks().empty() && stream->GetVideoTracks().empty()) {
|
|
streams_to_remove.push_back(stream);
|
|
}
|
|
}
|
|
|
|
for (auto& stream : streams_to_remove) {
|
|
remote_streams_->RemoveStream(stream);
|
|
observer_->OnRemoveStream(std::move(stream));
|
|
}
|
|
}
|
|
|
|
void PeerConnection::UpdateLocalSenders(
|
|
const std::vector<cricket::StreamParams>& streams,
|
|
cricket::MediaType media_type) {
|
|
std::vector<RtpSenderInfo>* current_senders = GetLocalSenderInfos(media_type);
|
|
|
|
// Find removed tracks. I.e., tracks where the track id, stream label or ssrc
|
|
// don't match the new StreamParam.
|
|
for (auto sender_it = current_senders->begin();
|
|
sender_it != current_senders->end();
|
|
/* incremented manually */) {
|
|
const RtpSenderInfo& info = *sender_it;
|
|
const cricket::StreamParams* params =
|
|
cricket::GetStreamBySsrc(streams, info.first_ssrc);
|
|
if (!params || params->id != info.sender_id ||
|
|
params->sync_label != info.stream_label) {
|
|
OnLocalSenderRemoved(info, media_type);
|
|
sender_it = current_senders->erase(sender_it);
|
|
} else {
|
|
++sender_it;
|
|
}
|
|
}
|
|
|
|
// Find new and active senders.
|
|
for (const cricket::StreamParams& params : streams) {
|
|
// The sync_label is the MediaStream label and the |stream.id| is the
|
|
// sender id.
|
|
const std::string& stream_label = params.sync_label;
|
|
const std::string& sender_id = params.id;
|
|
uint32_t ssrc = params.first_ssrc();
|
|
const RtpSenderInfo* sender_info =
|
|
FindSenderInfo(*current_senders, stream_label, sender_id);
|
|
if (!sender_info) {
|
|
current_senders->push_back(RtpSenderInfo(stream_label, sender_id, ssrc));
|
|
OnLocalSenderAdded(current_senders->back(), media_type);
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnLocalSenderAdded(const RtpSenderInfo& sender_info,
|
|
cricket::MediaType media_type) {
|
|
auto sender = FindSenderById(sender_info.sender_id);
|
|
if (!sender) {
|
|
RTC_LOG(LS_WARNING) << "An unknown RtpSender with id "
|
|
<< sender_info.sender_id
|
|
<< " has been configured in the local description.";
|
|
return;
|
|
}
|
|
|
|
if (sender->media_type() != media_type) {
|
|
RTC_LOG(LS_WARNING) << "An RtpSender has been configured in the local"
|
|
<< " description with an unexpected media type.";
|
|
return;
|
|
}
|
|
|
|
sender->internal()->set_stream_id(sender_info.stream_label);
|
|
sender->internal()->SetSsrc(sender_info.first_ssrc);
|
|
}
|
|
|
|
void PeerConnection::OnLocalSenderRemoved(const RtpSenderInfo& sender_info,
|
|
cricket::MediaType media_type) {
|
|
auto sender = FindSenderById(sender_info.sender_id);
|
|
if (!sender) {
|
|
// This is the normal case. I.e., RemoveStream has been called and the
|
|
// SessionDescriptions has been renegotiated.
|
|
return;
|
|
}
|
|
|
|
// A sender has been removed from the SessionDescription but it's still
|
|
// associated with the PeerConnection. This only occurs if the SDP doesn't
|
|
// match with the calls to CreateSender, AddStream and RemoveStream.
|
|
if (sender->media_type() != media_type) {
|
|
RTC_LOG(LS_WARNING) << "An RtpSender has been configured in the local"
|
|
<< " description with an unexpected media type.";
|
|
return;
|
|
}
|
|
|
|
sender->internal()->SetSsrc(0);
|
|
}
|
|
|
|
void PeerConnection::UpdateLocalRtpDataChannels(
|
|
const cricket::StreamParamsVec& streams) {
|
|
std::vector<std::string> existing_channels;
|
|
|
|
// Find new and active data channels.
|
|
for (const cricket::StreamParams& params : streams) {
|
|
// |it->sync_label| is actually the data channel label. The reason is that
|
|
// we use the same naming of data channels as we do for
|
|
// MediaStreams and Tracks.
|
|
// For MediaStreams, the sync_label is the MediaStream label and the
|
|
// track label is the same as |streamid|.
|
|
const std::string& channel_label = params.sync_label;
|
|
auto data_channel_it = rtp_data_channels_.find(channel_label);
|
|
if (data_channel_it == rtp_data_channels_.end()) {
|
|
RTC_LOG(LS_ERROR) << "channel label not found";
|
|
continue;
|
|
}
|
|
// Set the SSRC the data channel should use for sending.
|
|
data_channel_it->second->SetSendSsrc(params.first_ssrc());
|
|
existing_channels.push_back(data_channel_it->first);
|
|
}
|
|
|
|
UpdateClosingRtpDataChannels(existing_channels, true);
|
|
}
|
|
|
|
void PeerConnection::UpdateRemoteRtpDataChannels(
|
|
const cricket::StreamParamsVec& streams) {
|
|
std::vector<std::string> existing_channels;
|
|
|
|
// Find new and active data channels.
|
|
for (const cricket::StreamParams& params : streams) {
|
|
// The data channel label is either the mslabel or the SSRC if the mslabel
|
|
// does not exist. Ex a=ssrc:444330170 mslabel:test1.
|
|
std::string label = params.sync_label.empty()
|
|
? rtc::ToString(params.first_ssrc())
|
|
: params.sync_label;
|
|
auto data_channel_it = rtp_data_channels_.find(label);
|
|
if (data_channel_it == rtp_data_channels_.end()) {
|
|
// This is a new data channel.
|
|
CreateRemoteRtpDataChannel(label, params.first_ssrc());
|
|
} else {
|
|
data_channel_it->second->SetReceiveSsrc(params.first_ssrc());
|
|
}
|
|
existing_channels.push_back(label);
|
|
}
|
|
|
|
UpdateClosingRtpDataChannels(existing_channels, false);
|
|
}
|
|
|
|
void PeerConnection::UpdateClosingRtpDataChannels(
|
|
const std::vector<std::string>& active_channels,
|
|
bool is_local_update) {
|
|
auto it = rtp_data_channels_.begin();
|
|
while (it != rtp_data_channels_.end()) {
|
|
DataChannel* data_channel = it->second;
|
|
if (std::find(active_channels.begin(), active_channels.end(),
|
|
data_channel->label()) != active_channels.end()) {
|
|
++it;
|
|
continue;
|
|
}
|
|
|
|
if (is_local_update) {
|
|
data_channel->SetSendSsrc(0);
|
|
} else {
|
|
data_channel->RemotePeerRequestClose();
|
|
}
|
|
|
|
if (data_channel->state() == DataChannel::kClosed) {
|
|
rtp_data_channels_.erase(it);
|
|
it = rtp_data_channels_.begin();
|
|
} else {
|
|
++it;
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::CreateRemoteRtpDataChannel(const std::string& label,
|
|
uint32_t remote_ssrc) {
|
|
rtc::scoped_refptr<DataChannel> channel(
|
|
InternalCreateDataChannel(label, nullptr));
|
|
if (!channel.get()) {
|
|
RTC_LOG(LS_WARNING) << "Remote peer requested a DataChannel but"
|
|
<< "CreateDataChannel failed.";
|
|
return;
|
|
}
|
|
channel->SetReceiveSsrc(remote_ssrc);
|
|
rtc::scoped_refptr<DataChannelInterface> proxy_channel =
|
|
DataChannelProxy::Create(signaling_thread(), channel);
|
|
observer_->OnDataChannel(std::move(proxy_channel));
|
|
}
|
|
|
|
rtc::scoped_refptr<DataChannel> PeerConnection::InternalCreateDataChannel(
|
|
const std::string& label,
|
|
const InternalDataChannelInit* config) {
|
|
if (IsClosed()) {
|
|
return nullptr;
|
|
}
|
|
if (data_channel_type() == cricket::DCT_NONE) {
|
|
RTC_LOG(LS_ERROR)
|
|
<< "InternalCreateDataChannel: Data is not supported in this call.";
|
|
return nullptr;
|
|
}
|
|
InternalDataChannelInit new_config =
|
|
config ? (*config) : InternalDataChannelInit();
|
|
if (data_channel_type() == cricket::DCT_SCTP) {
|
|
if (new_config.id < 0) {
|
|
rtc::SSLRole role;
|
|
if ((GetSctpSslRole(&role)) &&
|
|
!sid_allocator_.AllocateSid(role, &new_config.id)) {
|
|
RTC_LOG(LS_ERROR)
|
|
<< "No id can be allocated for the SCTP data channel.";
|
|
return nullptr;
|
|
}
|
|
} else if (!sid_allocator_.ReserveSid(new_config.id)) {
|
|
RTC_LOG(LS_ERROR) << "Failed to create a SCTP data channel "
|
|
<< "because the id is already in use or out of range.";
|
|
return nullptr;
|
|
}
|
|
}
|
|
|
|
rtc::scoped_refptr<DataChannel> channel(
|
|
DataChannel::Create(this, data_channel_type(), label, new_config));
|
|
if (!channel) {
|
|
sid_allocator_.ReleaseSid(new_config.id);
|
|
return nullptr;
|
|
}
|
|
|
|
if (channel->data_channel_type() == cricket::DCT_RTP) {
|
|
if (rtp_data_channels_.find(channel->label()) != rtp_data_channels_.end()) {
|
|
RTC_LOG(LS_ERROR) << "DataChannel with label " << channel->label()
|
|
<< " already exists.";
|
|
return nullptr;
|
|
}
|
|
rtp_data_channels_[channel->label()] = channel;
|
|
} else {
|
|
RTC_DCHECK(channel->data_channel_type() == cricket::DCT_SCTP);
|
|
sctp_data_channels_.push_back(channel);
|
|
channel->SignalClosed.connect(this,
|
|
&PeerConnection::OnSctpDataChannelClosed);
|
|
}
|
|
|
|
SignalDataChannelCreated(channel.get());
|
|
return channel;
|
|
}
|
|
|
|
bool PeerConnection::HasDataChannels() const {
|
|
return !rtp_data_channels_.empty() || !sctp_data_channels_.empty();
|
|
}
|
|
|
|
void PeerConnection::AllocateSctpSids(rtc::SSLRole role) {
|
|
for (const auto& channel : sctp_data_channels_) {
|
|
if (channel->id() < 0) {
|
|
int sid;
|
|
if (!sid_allocator_.AllocateSid(role, &sid)) {
|
|
RTC_LOG(LS_ERROR) << "Failed to allocate SCTP sid.";
|
|
continue;
|
|
}
|
|
channel->SetSctpSid(sid);
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnSctpDataChannelClosed(DataChannel* channel) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
for (auto it = sctp_data_channels_.begin(); it != sctp_data_channels_.end();
|
|
++it) {
|
|
if (it->get() == channel) {
|
|
if (channel->id() >= 0) {
|
|
sid_allocator_.ReleaseSid(channel->id());
|
|
}
|
|
// Since this method is triggered by a signal from the DataChannel,
|
|
// we can't free it directly here; we need to free it asynchronously.
|
|
sctp_data_channels_to_free_.push_back(*it);
|
|
sctp_data_channels_.erase(it);
|
|
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_FREE_DATACHANNELS,
|
|
nullptr);
|
|
return;
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnDataChannelDestroyed() {
|
|
// Use a temporary copy of the RTP/SCTP DataChannel list because the
|
|
// DataChannel may callback to us and try to modify the list.
|
|
std::map<std::string, rtc::scoped_refptr<DataChannel>> temp_rtp_dcs;
|
|
temp_rtp_dcs.swap(rtp_data_channels_);
|
|
for (const auto& kv : temp_rtp_dcs) {
|
|
kv.second->OnTransportChannelDestroyed();
|
|
}
|
|
|
|
std::vector<rtc::scoped_refptr<DataChannel>> temp_sctp_dcs;
|
|
temp_sctp_dcs.swap(sctp_data_channels_);
|
|
for (const auto& channel : temp_sctp_dcs) {
|
|
channel->OnTransportChannelDestroyed();
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnDataChannelOpenMessage(
|
|
const std::string& label,
|
|
const InternalDataChannelInit& config) {
|
|
rtc::scoped_refptr<DataChannel> channel(
|
|
InternalCreateDataChannel(label, &config));
|
|
if (!channel.get()) {
|
|
RTC_LOG(LS_ERROR) << "Failed to create DataChannel from the OPEN message.";
|
|
return;
|
|
}
|
|
|
|
rtc::scoped_refptr<DataChannelInterface> proxy_channel =
|
|
DataChannelProxy::Create(signaling_thread(), channel);
|
|
observer_->OnDataChannel(std::move(proxy_channel));
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
PeerConnection::GetAudioTransceiver() const {
|
|
// This method only works with Plan B SDP, where there is a single
|
|
// audio/video transceiver.
|
|
RTC_DCHECK(!IsUnifiedPlan());
|
|
for (auto transceiver : transceivers_) {
|
|
if (transceiver->internal()->media_type() == cricket::MEDIA_TYPE_AUDIO) {
|
|
return transceiver;
|
|
}
|
|
}
|
|
RTC_NOTREACHED();
|
|
return nullptr;
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
PeerConnection::GetVideoTransceiver() const {
|
|
// This method only works with Plan B SDP, where there is a single
|
|
// audio/video transceiver.
|
|
RTC_DCHECK(!IsUnifiedPlan());
|
|
for (auto transceiver : transceivers_) {
|
|
if (transceiver->internal()->media_type() == cricket::MEDIA_TYPE_VIDEO) {
|
|
return transceiver;
|
|
}
|
|
}
|
|
RTC_NOTREACHED();
|
|
return nullptr;
|
|
}
|
|
|
|
// TODO(bugs.webrtc.org/7600): Remove this when multiple transceivers with
|
|
// individual transceiver directions are supported.
|
|
bool PeerConnection::HasRtpSender(cricket::MediaType type) const {
|
|
switch (type) {
|
|
case cricket::MEDIA_TYPE_AUDIO:
|
|
return !GetAudioTransceiver()->internal()->senders().empty();
|
|
case cricket::MEDIA_TYPE_VIDEO:
|
|
return !GetVideoTransceiver()->internal()->senders().empty();
|
|
case cricket::MEDIA_TYPE_DATA:
|
|
return false;
|
|
}
|
|
RTC_NOTREACHED();
|
|
return false;
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
|
|
PeerConnection::FindSenderForTrack(MediaStreamTrackInterface* track) const {
|
|
for (auto transceiver : transceivers_) {
|
|
for (auto sender : transceiver->internal()->senders()) {
|
|
if (sender->track() == track) {
|
|
return sender;
|
|
}
|
|
}
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
|
|
PeerConnection::FindSenderById(const std::string& sender_id) const {
|
|
for (auto transceiver : transceivers_) {
|
|
for (auto sender : transceiver->internal()->senders()) {
|
|
if (sender->id() == sender_id) {
|
|
return sender;
|
|
}
|
|
}
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
|
|
PeerConnection::FindReceiverById(const std::string& receiver_id) const {
|
|
for (auto transceiver : transceivers_) {
|
|
for (auto receiver : transceiver->internal()->receivers()) {
|
|
if (receiver->id() == receiver_id) {
|
|
return receiver;
|
|
}
|
|
}
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
std::vector<PeerConnection::RtpSenderInfo>*
|
|
PeerConnection::GetRemoteSenderInfos(cricket::MediaType media_type) {
|
|
RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
|
|
media_type == cricket::MEDIA_TYPE_VIDEO);
|
|
return (media_type == cricket::MEDIA_TYPE_AUDIO)
|
|
? &remote_audio_sender_infos_
|
|
: &remote_video_sender_infos_;
|
|
}
|
|
|
|
std::vector<PeerConnection::RtpSenderInfo>* PeerConnection::GetLocalSenderInfos(
|
|
cricket::MediaType media_type) {
|
|
RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
|
|
media_type == cricket::MEDIA_TYPE_VIDEO);
|
|
return (media_type == cricket::MEDIA_TYPE_AUDIO) ? &local_audio_sender_infos_
|
|
: &local_video_sender_infos_;
|
|
}
|
|
|
|
const PeerConnection::RtpSenderInfo* PeerConnection::FindSenderInfo(
|
|
const std::vector<PeerConnection::RtpSenderInfo>& infos,
|
|
const std::string& stream_label,
|
|
const std::string sender_id) const {
|
|
for (const RtpSenderInfo& sender_info : infos) {
|
|
if (sender_info.stream_label == stream_label &&
|
|
sender_info.sender_id == sender_id) {
|
|
return &sender_info;
|
|
}
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
DataChannel* PeerConnection::FindDataChannelBySid(int sid) const {
|
|
for (const auto& channel : sctp_data_channels_) {
|
|
if (channel->id() == sid) {
|
|
return channel;
|
|
}
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
bool PeerConnection::InitializePortAllocator_n(
|
|
const RTCConfiguration& configuration) {
|
|
cricket::ServerAddresses stun_servers;
|
|
std::vector<cricket::RelayServerConfig> turn_servers;
|
|
if (ParseIceServers(configuration.servers, &stun_servers, &turn_servers) !=
|
|
RTCErrorType::NONE) {
|
|
return false;
|
|
}
|
|
|
|
port_allocator_->Initialize();
|
|
|
|
// To handle both internal and externally created port allocator, we will
|
|
// enable BUNDLE here.
|
|
int portallocator_flags = port_allocator_->flags();
|
|
portallocator_flags |= cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
|
|
cricket::PORTALLOCATOR_ENABLE_IPV6 |
|
|
cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI;
|
|
// If the disable-IPv6 flag was specified, we'll not override it
|
|
// by experiment.
|
|
if (configuration.disable_ipv6) {
|
|
portallocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
|
|
} else if (webrtc::field_trial::FindFullName("WebRTC-IPv6Default")
|
|
.find("Disabled") == 0) {
|
|
portallocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
|
|
}
|
|
|
|
if (configuration.disable_ipv6_on_wifi) {
|
|
portallocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI);
|
|
RTC_LOG(LS_INFO) << "IPv6 candidates on Wi-Fi are disabled.";
|
|
}
|
|
|
|
if (configuration.tcp_candidate_policy == kTcpCandidatePolicyDisabled) {
|
|
portallocator_flags |= cricket::PORTALLOCATOR_DISABLE_TCP;
|
|
RTC_LOG(LS_INFO) << "TCP candidates are disabled.";
|
|
}
|
|
|
|
if (configuration.candidate_network_policy ==
|
|
kCandidateNetworkPolicyLowCost) {
|
|
portallocator_flags |= cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS;
|
|
RTC_LOG(LS_INFO) << "Do not gather candidates on high-cost networks";
|
|
}
|
|
|
|
port_allocator_->set_flags(portallocator_flags);
|
|
// No step delay is used while allocating ports.
|
|
port_allocator_->set_step_delay(cricket::kMinimumStepDelay);
|
|
port_allocator_->set_candidate_filter(
|
|
ConvertIceTransportTypeToCandidateFilter(configuration.type));
|
|
port_allocator_->set_max_ipv6_networks(configuration.max_ipv6_networks);
|
|
|
|
// Call this last since it may create pooled allocator sessions using the
|
|
// properties set above.
|
|
port_allocator_->SetConfiguration(stun_servers, turn_servers,
|
|
configuration.ice_candidate_pool_size,
|
|
configuration.prune_turn_ports,
|
|
configuration.turn_customizer);
|
|
return true;
|
|
}
|
|
|
|
bool PeerConnection::ReconfigurePortAllocator_n(
|
|
const cricket::ServerAddresses& stun_servers,
|
|
const std::vector<cricket::RelayServerConfig>& turn_servers,
|
|
IceTransportsType type,
|
|
int candidate_pool_size,
|
|
bool prune_turn_ports,
|
|
webrtc::TurnCustomizer* turn_customizer) {
|
|
port_allocator_->set_candidate_filter(
|
|
ConvertIceTransportTypeToCandidateFilter(type));
|
|
// Call this last since it may create pooled allocator sessions using the
|
|
// candidate filter set above.
|
|
return port_allocator_->SetConfiguration(
|
|
stun_servers, turn_servers, candidate_pool_size, prune_turn_ports,
|
|
turn_customizer);
|
|
}
|
|
|
|
cricket::ChannelManager* PeerConnection::channel_manager() const {
|
|
return factory_->channel_manager();
|
|
}
|
|
|
|
MetricsObserverInterface* PeerConnection::metrics_observer() const {
|
|
return uma_observer_;
|
|
}
|
|
|
|
bool PeerConnection::StartRtcEventLog_w(
|
|
std::unique_ptr<RtcEventLogOutput> output,
|
|
int64_t output_period_ms) {
|
|
if (!event_log_) {
|
|
return false;
|
|
}
|
|
return event_log_->StartLogging(std::move(output), output_period_ms);
|
|
}
|
|
|
|
void PeerConnection::StopRtcEventLog_w() {
|
|
if (event_log_) {
|
|
event_log_->StopLogging();
|
|
}
|
|
}
|
|
|
|
cricket::BaseChannel* PeerConnection::GetChannel(
|
|
const std::string& content_name) {
|
|
if (voice_channel() && voice_channel()->content_name() == content_name) {
|
|
return voice_channel();
|
|
}
|
|
if (video_channel() && video_channel()->content_name() == content_name) {
|
|
return video_channel();
|
|
}
|
|
if (rtp_data_channel() &&
|
|
rtp_data_channel()->content_name() == content_name) {
|
|
return rtp_data_channel();
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
bool PeerConnection::GetSctpSslRole(rtc::SSLRole* role) {
|
|
if (!local_description() || !remote_description()) {
|
|
RTC_LOG(LS_INFO)
|
|
<< "Local and Remote descriptions must be applied to get the "
|
|
<< "SSL Role of the SCTP transport.";
|
|
return false;
|
|
}
|
|
if (!sctp_transport_) {
|
|
RTC_LOG(LS_INFO) << "Non-rejected SCTP m= section is needed to get the "
|
|
<< "SSL Role of the SCTP transport.";
|
|
return false;
|
|
}
|
|
|
|
return transport_controller_->GetSslRole(*sctp_transport_name_, role);
|
|
}
|
|
|
|
bool PeerConnection::GetSslRole(const std::string& content_name,
|
|
rtc::SSLRole* role) {
|
|
if (!local_description() || !remote_description()) {
|
|
RTC_LOG(LS_INFO)
|
|
<< "Local and Remote descriptions must be applied to get the "
|
|
<< "SSL Role of the session.";
|
|
return false;
|
|
}
|
|
|
|
return transport_controller_->GetSslRole(GetTransportName(content_name),
|
|
role);
|
|
}
|
|
|
|
// TODO(steveanton): Eventually it'd be nice to store the channels as a single
|
|
// vector of BaseChannel pointers instead of separate voice and video channel
|
|
// vectors. At that point, this will become a simple getter.
|
|
std::vector<cricket::BaseChannel*> PeerConnection::Channels() const {
|
|
std::vector<cricket::BaseChannel*> channels;
|
|
if (voice_channel()) {
|
|
channels.push_back(voice_channel());
|
|
}
|
|
if (video_channel()) {
|
|
channels.push_back(video_channel());
|
|
}
|
|
if (rtp_data_channel_) {
|
|
channels.push_back(rtp_data_channel_);
|
|
}
|
|
return channels;
|
|
}
|
|
|
|
void PeerConnection::SetSessionError(SessionError error,
|
|
const std::string& error_desc) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (error != session_error_) {
|
|
session_error_ = error;
|
|
session_error_desc_ = error_desc;
|
|
}
|
|
}
|
|
|
|
RTCError PeerConnection::UpdateSessionState(SdpType type,
|
|
cricket::ContentSource source) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
|
|
// If there's already a pending error then no state transition should happen.
|
|
// But all call-sites should be verifying this before calling us!
|
|
RTC_DCHECK(session_error() == SessionError::kNone);
|
|
|
|
// If this is an answer then we know whether to BUNDLE or not. If both the
|
|
// local and remote side have agreed to BUNDLE, go ahead and enable it.
|
|
if (type == SdpType::kAnswer) {
|
|
const cricket::ContentGroup* local_bundle =
|
|
local_description()->description()->GetGroupByName(
|
|
cricket::GROUP_TYPE_BUNDLE);
|
|
const cricket::ContentGroup* remote_bundle =
|
|
remote_description()->description()->GetGroupByName(
|
|
cricket::GROUP_TYPE_BUNDLE);
|
|
if (local_bundle && remote_bundle) {
|
|
// The answerer decides the transport to bundle on.
|
|
const cricket::ContentGroup* answer_bundle =
|
|
(source == cricket::CS_LOCAL ? local_bundle : remote_bundle);
|
|
if (!EnableBundle(*answer_bundle)) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
kEnableBundleFailed);
|
|
}
|
|
}
|
|
}
|
|
|
|
// Only push down the transport description after potentially enabling BUNDLE;
|
|
// we don't want to push down a description on a transport about to be
|
|
// destroyed.
|
|
RTCError error = PushdownTransportDescription(source, type);
|
|
if (!error.ok()) {
|
|
return error;
|
|
}
|
|
|
|
// If this is answer-ish we're ready to let media flow.
|
|
if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
|
|
EnableSending();
|
|
}
|
|
|
|
// Update the signaling state according to the specified state machine (see
|
|
// https://w3c.github.io/webrtc-pc/#rtcsignalingstate-enum).
|
|
if (type == SdpType::kOffer) {
|
|
ChangeSignalingState(source == cricket::CS_LOCAL
|
|
? PeerConnectionInterface::kHaveLocalOffer
|
|
: PeerConnectionInterface::kHaveRemoteOffer);
|
|
} else if (type == SdpType::kPrAnswer) {
|
|
ChangeSignalingState(source == cricket::CS_LOCAL
|
|
? PeerConnectionInterface::kHaveLocalPrAnswer
|
|
: PeerConnectionInterface::kHaveRemotePrAnswer);
|
|
} else {
|
|
RTC_DCHECK(type == SdpType::kAnswer);
|
|
ChangeSignalingState(PeerConnectionInterface::kStable);
|
|
}
|
|
|
|
// Update internal objects according to the session description's media
|
|
// descriptions.
|
|
error = PushdownMediaDescription(type, source);
|
|
if (!error.ok()) {
|
|
SetSessionError(SessionError::kContent, error.message());
|
|
}
|
|
if (session_error() != SessionError::kNone) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR, GetSessionErrorMsg());
|
|
}
|
|
|
|
return RTCError::OK();
|
|
}
|
|
|
|
RTCError PeerConnection::PushdownMediaDescription(
|
|
SdpType type,
|
|
cricket::ContentSource source) {
|
|
const SessionDescriptionInterface* sdesc =
|
|
(source == cricket::CS_LOCAL ? local_description()
|
|
: remote_description());
|
|
RTC_DCHECK(sdesc);
|
|
|
|
// Push down the new SDP media section for each audio/video transceiver.
|
|
for (auto transceiver : transceivers_) {
|
|
const ContentInfo* content_info =
|
|
FindMediaSectionForTransceiver(transceiver, sdesc);
|
|
cricket::BaseChannel* channel = transceiver->internal()->channel();
|
|
if (!channel || !content_info || content_info->rejected) {
|
|
continue;
|
|
}
|
|
const MediaContentDescription* content_desc =
|
|
content_info->media_description();
|
|
if (!content_desc) {
|
|
continue;
|
|
}
|
|
std::string error;
|
|
bool success =
|
|
(source == cricket::CS_LOCAL)
|
|
? channel->SetLocalContent(content_desc, type, &error)
|
|
: channel->SetRemoteContent(content_desc, type, &error);
|
|
if (!success) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, std::move(error));
|
|
}
|
|
}
|
|
|
|
// If using the RtpDataChannel, push down the new SDP section for it too.
|
|
if (rtp_data_channel_) {
|
|
const ContentInfo* data_content =
|
|
cricket::GetFirstDataContent(sdesc->description());
|
|
if (data_content && !data_content->rejected) {
|
|
const MediaContentDescription* data_desc =
|
|
data_content->media_description();
|
|
if (data_desc) {
|
|
std::string error;
|
|
bool success =
|
|
(source == cricket::CS_LOCAL)
|
|
? rtp_data_channel_->SetLocalContent(data_desc, type, &error)
|
|
: rtp_data_channel_->SetRemoteContent(data_desc, type,
|
|
&error);
|
|
if (!success) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
std::move(error));
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
// Need complete offer/answer with an SCTP m= section before starting SCTP,
|
|
// according to https://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-19
|
|
if (sctp_transport_ && local_description() && remote_description() &&
|
|
cricket::GetFirstDataContent(local_description()->description()) &&
|
|
cricket::GetFirstDataContent(remote_description()->description())) {
|
|
bool success = network_thread()->Invoke<bool>(
|
|
RTC_FROM_HERE,
|
|
rtc::Bind(&PeerConnection::PushdownSctpParameters_n, this, source));
|
|
if (!success) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
|
|
"Failed to push down SCTP parameters.");
|
|
}
|
|
}
|
|
|
|
return RTCError::OK();
|
|
}
|
|
|
|
bool PeerConnection::PushdownSctpParameters_n(cricket::ContentSource source) {
|
|
RTC_DCHECK(network_thread()->IsCurrent());
|
|
RTC_DCHECK(local_description());
|
|
RTC_DCHECK(remote_description());
|
|
// Apply the SCTP port (which is hidden inside a DataCodec structure...)
|
|
// When we support "max-message-size", that would also be pushed down here.
|
|
return sctp_transport_->Start(
|
|
GetSctpPort(local_description()->description()),
|
|
GetSctpPort(remote_description()->description()));
|
|
}
|
|
|
|
RTCError PeerConnection::PushdownTransportDescription(
|
|
cricket::ContentSource source,
|
|
SdpType type) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
|
|
const SessionDescriptionInterface* sdesc =
|
|
(source == cricket::CS_LOCAL ? local_description()
|
|
: remote_description());
|
|
RTC_DCHECK(sdesc);
|
|
for (const cricket::TransportInfo& tinfo :
|
|
sdesc->description()->transport_infos()) {
|
|
std::string error;
|
|
bool success;
|
|
if (source == cricket::CS_LOCAL) {
|
|
success = transport_controller_->SetLocalTransportDescription(
|
|
tinfo.content_name, tinfo.description, type, &error);
|
|
} else {
|
|
success = transport_controller_->SetRemoteTransportDescription(
|
|
tinfo.content_name, tinfo.description, type, &error);
|
|
}
|
|
if (!success) {
|
|
LOG_AND_RETURN_ERROR(
|
|
RTCErrorType::INVALID_PARAMETER,
|
|
"Failed to push down transport description: " + error);
|
|
}
|
|
}
|
|
|
|
return RTCError::OK();
|
|
}
|
|
|
|
bool PeerConnection::GetTransportDescription(
|
|
const SessionDescription* description,
|
|
const std::string& content_name,
|
|
cricket::TransportDescription* tdesc) {
|
|
if (!description || !tdesc) {
|
|
return false;
|
|
}
|
|
const TransportInfo* transport_info =
|
|
description->GetTransportInfoByName(content_name);
|
|
if (!transport_info) {
|
|
return false;
|
|
}
|
|
*tdesc = transport_info->description;
|
|
return true;
|
|
}
|
|
|
|
bool PeerConnection::EnableBundle(const cricket::ContentGroup& bundle) {
|
|
const std::string* first_content_name = bundle.FirstContentName();
|
|
if (!first_content_name) {
|
|
RTC_LOG(LS_WARNING) << "Tried to BUNDLE with no contents.";
|
|
return false;
|
|
}
|
|
const std::string& transport_name = *first_content_name;
|
|
|
|
auto maybe_set_transport = [this, bundle,
|
|
transport_name](cricket::BaseChannel* ch) {
|
|
if (!ch || !bundle.HasContentName(ch->content_name())) {
|
|
return;
|
|
}
|
|
|
|
std::string old_transport_name = ch->transport_name();
|
|
if (old_transport_name == transport_name) {
|
|
RTC_LOG(LS_INFO) << "BUNDLE already enabled for " << ch->content_name()
|
|
<< " on " << transport_name << ".";
|
|
return;
|
|
}
|
|
|
|
cricket::DtlsTransportInternal* rtp_dtls_transport =
|
|
transport_controller_->CreateDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
bool need_rtcp = (ch->rtcp_dtls_transport() != nullptr);
|
|
cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;
|
|
if (need_rtcp) {
|
|
rtcp_dtls_transport = transport_controller_->CreateDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
|
|
}
|
|
|
|
ch->SetTransports(rtp_dtls_transport, rtcp_dtls_transport);
|
|
RTC_LOG(LS_INFO) << "Enabled BUNDLE for " << ch->content_name() << " on "
|
|
<< transport_name << ".";
|
|
transport_controller_->DestroyDtlsTransport(
|
|
old_transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
// If the channel needs rtcp, it means that the channel used to have a
|
|
// rtcp transport which needs to be deleted now.
|
|
if (need_rtcp) {
|
|
transport_controller_->DestroyDtlsTransport(
|
|
old_transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
|
|
}
|
|
};
|
|
|
|
for (auto transceiver : transceivers_) {
|
|
maybe_set_transport(transceiver->internal()->channel());
|
|
}
|
|
maybe_set_transport(rtp_data_channel_);
|
|
|
|
// For SCTP, transport creation/deletion happens here instead of in the
|
|
// object itself.
|
|
if (sctp_transport_) {
|
|
RTC_DCHECK(sctp_transport_name_);
|
|
RTC_DCHECK(sctp_content_name_);
|
|
if (transport_name != *sctp_transport_name_ &&
|
|
bundle.HasContentName(*sctp_content_name_)) {
|
|
network_thread()->Invoke<void>(
|
|
RTC_FROM_HERE, rtc::Bind(&PeerConnection::ChangeSctpTransport_n, this,
|
|
transport_name));
|
|
}
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
cricket::IceConfig PeerConnection::ParseIceConfig(
|
|
const PeerConnectionInterface::RTCConfiguration& config) const {
|
|
cricket::ContinualGatheringPolicy gathering_policy;
|
|
// TODO(honghaiz): Add the third continual gathering policy in
|
|
// PeerConnectionInterface and map it to GATHER_CONTINUALLY_AND_RECOVER.
|
|
switch (config.continual_gathering_policy) {
|
|
case PeerConnectionInterface::GATHER_ONCE:
|
|
gathering_policy = cricket::GATHER_ONCE;
|
|
break;
|
|
case PeerConnectionInterface::GATHER_CONTINUALLY:
|
|
gathering_policy = cricket::GATHER_CONTINUALLY;
|
|
break;
|
|
default:
|
|
RTC_NOTREACHED();
|
|
gathering_policy = cricket::GATHER_ONCE;
|
|
}
|
|
cricket::IceConfig ice_config;
|
|
ice_config.receiving_timeout = config.ice_connection_receiving_timeout;
|
|
ice_config.prioritize_most_likely_candidate_pairs =
|
|
config.prioritize_most_likely_ice_candidate_pairs;
|
|
ice_config.backup_connection_ping_interval =
|
|
config.ice_backup_candidate_pair_ping_interval;
|
|
ice_config.continual_gathering_policy = gathering_policy;
|
|
ice_config.presume_writable_when_fully_relayed =
|
|
config.presume_writable_when_fully_relayed;
|
|
ice_config.ice_check_min_interval = config.ice_check_min_interval;
|
|
ice_config.regather_all_networks_interval_range =
|
|
config.ice_regather_interval_range;
|
|
return ice_config;
|
|
}
|
|
|
|
bool PeerConnection::GetLocalTrackIdBySsrc(uint32_t ssrc,
|
|
std::string* track_id) {
|
|
if (!local_description()) {
|
|
return false;
|
|
}
|
|
return webrtc::GetTrackIdBySsrc(local_description()->description(), ssrc,
|
|
track_id);
|
|
}
|
|
|
|
bool PeerConnection::GetRemoteTrackIdBySsrc(uint32_t ssrc,
|
|
std::string* track_id) {
|
|
if (!remote_description()) {
|
|
return false;
|
|
}
|
|
return webrtc::GetTrackIdBySsrc(remote_description()->description(), ssrc,
|
|
track_id);
|
|
}
|
|
|
|
bool PeerConnection::SendData(const cricket::SendDataParams& params,
|
|
const rtc::CopyOnWriteBuffer& payload,
|
|
cricket::SendDataResult* result) {
|
|
if (!rtp_data_channel_ && !sctp_transport_) {
|
|
RTC_LOG(LS_ERROR) << "SendData called when rtp_data_channel_ "
|
|
<< "and sctp_transport_ are NULL.";
|
|
return false;
|
|
}
|
|
return rtp_data_channel_
|
|
? rtp_data_channel_->SendData(params, payload, result)
|
|
: network_thread()->Invoke<bool>(
|
|
RTC_FROM_HERE,
|
|
Bind(&cricket::SctpTransportInternal::SendData,
|
|
sctp_transport_.get(), params, payload, result));
|
|
}
|
|
|
|
bool PeerConnection::ConnectDataChannel(DataChannel* webrtc_data_channel) {
|
|
if (!rtp_data_channel_ && !sctp_transport_) {
|
|
// Don't log an error here, because DataChannels are expected to call
|
|
// ConnectDataChannel in this state. It's the only way to initially tell
|
|
// whether or not the underlying transport is ready.
|
|
return false;
|
|
}
|
|
if (rtp_data_channel_) {
|
|
rtp_data_channel_->SignalReadyToSendData.connect(
|
|
webrtc_data_channel, &DataChannel::OnChannelReady);
|
|
rtp_data_channel_->SignalDataReceived.connect(webrtc_data_channel,
|
|
&DataChannel::OnDataReceived);
|
|
} else {
|
|
SignalSctpReadyToSendData.connect(webrtc_data_channel,
|
|
&DataChannel::OnChannelReady);
|
|
SignalSctpDataReceived.connect(webrtc_data_channel,
|
|
&DataChannel::OnDataReceived);
|
|
SignalSctpStreamClosedRemotely.connect(
|
|
webrtc_data_channel, &DataChannel::OnStreamClosedRemotely);
|
|
}
|
|
return true;
|
|
}
|
|
|
|
void PeerConnection::DisconnectDataChannel(DataChannel* webrtc_data_channel) {
|
|
if (!rtp_data_channel_ && !sctp_transport_) {
|
|
RTC_LOG(LS_ERROR)
|
|
<< "DisconnectDataChannel called when rtp_data_channel_ and "
|
|
"sctp_transport_ are NULL.";
|
|
return;
|
|
}
|
|
if (rtp_data_channel_) {
|
|
rtp_data_channel_->SignalReadyToSendData.disconnect(webrtc_data_channel);
|
|
rtp_data_channel_->SignalDataReceived.disconnect(webrtc_data_channel);
|
|
} else {
|
|
SignalSctpReadyToSendData.disconnect(webrtc_data_channel);
|
|
SignalSctpDataReceived.disconnect(webrtc_data_channel);
|
|
SignalSctpStreamClosedRemotely.disconnect(webrtc_data_channel);
|
|
}
|
|
}
|
|
|
|
void PeerConnection::AddSctpDataStream(int sid) {
|
|
if (!sctp_transport_) {
|
|
RTC_LOG(LS_ERROR)
|
|
<< "AddSctpDataStream called when sctp_transport_ is NULL.";
|
|
return;
|
|
}
|
|
network_thread()->Invoke<void>(
|
|
RTC_FROM_HERE, rtc::Bind(&cricket::SctpTransportInternal::OpenStream,
|
|
sctp_transport_.get(), sid));
|
|
}
|
|
|
|
void PeerConnection::RemoveSctpDataStream(int sid) {
|
|
if (!sctp_transport_) {
|
|
RTC_LOG(LS_ERROR) << "RemoveSctpDataStream called when sctp_transport_ is "
|
|
<< "NULL.";
|
|
return;
|
|
}
|
|
network_thread()->Invoke<void>(
|
|
RTC_FROM_HERE, rtc::Bind(&cricket::SctpTransportInternal::ResetStream,
|
|
sctp_transport_.get(), sid));
|
|
}
|
|
|
|
bool PeerConnection::ReadyToSendData() const {
|
|
return (rtp_data_channel_ && rtp_data_channel_->ready_to_send_data()) ||
|
|
sctp_ready_to_send_data_;
|
|
}
|
|
|
|
std::unique_ptr<SessionStats> PeerConnection::GetSessionStats_s() {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
ChannelNamePairs channel_name_pairs;
|
|
if (voice_channel()) {
|
|
channel_name_pairs.voice = ChannelNamePair(
|
|
voice_channel()->content_name(), voice_channel()->transport_name());
|
|
}
|
|
if (video_channel()) {
|
|
channel_name_pairs.video = ChannelNamePair(
|
|
video_channel()->content_name(), video_channel()->transport_name());
|
|
}
|
|
if (rtp_data_channel()) {
|
|
channel_name_pairs.data =
|
|
ChannelNamePair(rtp_data_channel()->content_name(),
|
|
rtp_data_channel()->transport_name());
|
|
}
|
|
if (sctp_transport_) {
|
|
RTC_DCHECK(sctp_content_name_);
|
|
RTC_DCHECK(sctp_transport_name_);
|
|
channel_name_pairs.data =
|
|
ChannelNamePair(*sctp_content_name_, *sctp_transport_name_);
|
|
}
|
|
return GetSessionStats(channel_name_pairs);
|
|
}
|
|
|
|
std::unique_ptr<SessionStats> PeerConnection::GetSessionStats(
|
|
const ChannelNamePairs& channel_name_pairs) {
|
|
if (network_thread()->IsCurrent()) {
|
|
return GetSessionStats_n(channel_name_pairs);
|
|
}
|
|
return network_thread()->Invoke<std::unique_ptr<SessionStats>>(
|
|
RTC_FROM_HERE,
|
|
rtc::Bind(&PeerConnection::GetSessionStats_n, this, channel_name_pairs));
|
|
}
|
|
|
|
bool PeerConnection::GetLocalCertificate(
|
|
const std::string& transport_name,
|
|
rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
|
|
return transport_controller_->GetLocalCertificate(transport_name,
|
|
certificate);
|
|
}
|
|
|
|
std::unique_ptr<rtc::SSLCertificate> PeerConnection::GetRemoteSSLCertificate(
|
|
const std::string& transport_name) {
|
|
return transport_controller_->GetRemoteSSLCertificate(transport_name);
|
|
}
|
|
|
|
cricket::DataChannelType PeerConnection::data_channel_type() const {
|
|
return data_channel_type_;
|
|
}
|
|
|
|
bool PeerConnection::IceRestartPending(const std::string& content_name) const {
|
|
return pending_ice_restarts_.find(content_name) !=
|
|
pending_ice_restarts_.end();
|
|
}
|
|
|
|
bool PeerConnection::NeedsIceRestart(const std::string& content_name) const {
|
|
return transport_controller_->NeedsIceRestart(content_name);
|
|
}
|
|
|
|
void PeerConnection::OnCertificateReady(
|
|
const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
|
|
transport_controller_->SetLocalCertificate(certificate);
|
|
}
|
|
|
|
void PeerConnection::OnDtlsSrtpSetupFailure(cricket::BaseChannel*, bool rtcp) {
|
|
SetSessionError(SessionError::kTransport,
|
|
rtcp ? kDtlsSrtpSetupFailureRtcp : kDtlsSrtpSetupFailureRtp);
|
|
}
|
|
|
|
void PeerConnection::OnTransportControllerConnectionState(
|
|
cricket::IceConnectionState state) {
|
|
switch (state) {
|
|
case cricket::kIceConnectionConnecting:
|
|
// If the current state is Connected or Completed, then there were
|
|
// writable channels but now there are not, so the next state must
|
|
// be Disconnected.
|
|
// kIceConnectionConnecting is currently used as the default,
|
|
// un-connected state by the TransportController, so its only use is
|
|
// detecting disconnections.
|
|
if (ice_connection_state_ ==
|
|
PeerConnectionInterface::kIceConnectionConnected ||
|
|
ice_connection_state_ ==
|
|
PeerConnectionInterface::kIceConnectionCompleted) {
|
|
SetIceConnectionState(
|
|
PeerConnectionInterface::kIceConnectionDisconnected);
|
|
}
|
|
break;
|
|
case cricket::kIceConnectionFailed:
|
|
SetIceConnectionState(PeerConnectionInterface::kIceConnectionFailed);
|
|
break;
|
|
case cricket::kIceConnectionConnected:
|
|
RTC_LOG(LS_INFO) << "Changing to ICE connected state because "
|
|
<< "all transports are writable.";
|
|
SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
|
|
break;
|
|
case cricket::kIceConnectionCompleted:
|
|
RTC_LOG(LS_INFO) << "Changing to ICE completed state because "
|
|
<< "all transports are complete.";
|
|
if (ice_connection_state_ !=
|
|
PeerConnectionInterface::kIceConnectionConnected) {
|
|
// If jumping directly from "checking" to "connected",
|
|
// signal "connected" first.
|
|
SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
|
|
}
|
|
SetIceConnectionState(PeerConnectionInterface::kIceConnectionCompleted);
|
|
if (metrics_observer()) {
|
|
ReportTransportStats();
|
|
}
|
|
break;
|
|
default:
|
|
RTC_NOTREACHED();
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnTransportControllerCandidatesGathered(
|
|
const std::string& transport_name,
|
|
const cricket::Candidates& candidates) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
int sdp_mline_index;
|
|
if (!GetLocalCandidateMediaIndex(transport_name, &sdp_mline_index)) {
|
|
RTC_LOG(LS_ERROR)
|
|
<< "OnTransportControllerCandidatesGathered: content name "
|
|
<< transport_name << " not found";
|
|
return;
|
|
}
|
|
|
|
for (cricket::Candidates::const_iterator citer = candidates.begin();
|
|
citer != candidates.end(); ++citer) {
|
|
// Use transport_name as the candidate media id.
|
|
std::unique_ptr<JsepIceCandidate> candidate(
|
|
new JsepIceCandidate(transport_name, sdp_mline_index, *citer));
|
|
if (local_description()) {
|
|
mutable_local_description()->AddCandidate(candidate.get());
|
|
}
|
|
OnIceCandidate(std::move(candidate));
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnTransportControllerCandidatesRemoved(
|
|
const std::vector<cricket::Candidate>& candidates) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
// Sanity check.
|
|
for (const cricket::Candidate& candidate : candidates) {
|
|
if (candidate.transport_name().empty()) {
|
|
RTC_LOG(LS_ERROR) << "OnTransportControllerCandidatesRemoved: "
|
|
<< "empty content name in candidate "
|
|
<< candidate.ToString();
|
|
return;
|
|
}
|
|
}
|
|
|
|
if (local_description()) {
|
|
mutable_local_description()->RemoveCandidates(candidates);
|
|
}
|
|
OnIceCandidatesRemoved(candidates);
|
|
}
|
|
|
|
void PeerConnection::OnTransportControllerDtlsHandshakeError(
|
|
rtc::SSLHandshakeError error) {
|
|
if (metrics_observer()) {
|
|
metrics_observer()->IncrementEnumCounter(
|
|
webrtc::kEnumCounterDtlsHandshakeError, static_cast<int>(error),
|
|
static_cast<int>(rtc::SSLHandshakeError::MAX_VALUE));
|
|
}
|
|
}
|
|
|
|
void PeerConnection::EnableSending() {
|
|
for (auto transceiver : transceivers_) {
|
|
cricket::BaseChannel* channel = transceiver->internal()->channel();
|
|
if (channel && !channel->enabled()) {
|
|
channel->Enable(true);
|
|
}
|
|
}
|
|
|
|
if (rtp_data_channel_ && !rtp_data_channel_->enabled()) {
|
|
rtp_data_channel_->Enable(true);
|
|
}
|
|
}
|
|
|
|
// Returns the media index for a local ice candidate given the content name.
|
|
bool PeerConnection::GetLocalCandidateMediaIndex(
|
|
const std::string& content_name,
|
|
int* sdp_mline_index) {
|
|
if (!local_description() || !sdp_mline_index) {
|
|
return false;
|
|
}
|
|
|
|
bool content_found = false;
|
|
const ContentInfos& contents = local_description()->description()->contents();
|
|
for (size_t index = 0; index < contents.size(); ++index) {
|
|
if (contents[index].name == content_name) {
|
|
*sdp_mline_index = static_cast<int>(index);
|
|
content_found = true;
|
|
break;
|
|
}
|
|
}
|
|
return content_found;
|
|
}
|
|
|
|
bool PeerConnection::UseCandidatesInSessionDescription(
|
|
const SessionDescriptionInterface* remote_desc) {
|
|
if (!remote_desc) {
|
|
return true;
|
|
}
|
|
bool ret = true;
|
|
|
|
for (size_t m = 0; m < remote_desc->number_of_mediasections(); ++m) {
|
|
const IceCandidateCollection* candidates = remote_desc->candidates(m);
|
|
for (size_t n = 0; n < candidates->count(); ++n) {
|
|
const IceCandidateInterface* candidate = candidates->at(n);
|
|
bool valid = false;
|
|
if (!ReadyToUseRemoteCandidate(candidate, remote_desc, &valid)) {
|
|
if (valid) {
|
|
RTC_LOG(LS_INFO)
|
|
<< "UseCandidatesInSessionDescription: Not ready to use "
|
|
<< "candidate.";
|
|
}
|
|
continue;
|
|
}
|
|
ret = UseCandidate(candidate);
|
|
if (!ret) {
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
bool PeerConnection::UseCandidate(const IceCandidateInterface* candidate) {
|
|
size_t mediacontent_index = static_cast<size_t>(candidate->sdp_mline_index());
|
|
size_t remote_content_size =
|
|
remote_description()->description()->contents().size();
|
|
if (mediacontent_index >= remote_content_size) {
|
|
RTC_LOG(LS_ERROR) << "UseCandidate: Invalid candidate media index.";
|
|
return false;
|
|
}
|
|
|
|
cricket::ContentInfo content =
|
|
remote_description()->description()->contents()[mediacontent_index];
|
|
std::vector<cricket::Candidate> candidates;
|
|
candidates.push_back(candidate->candidate());
|
|
// Invoking BaseSession method to handle remote candidates.
|
|
std::string error;
|
|
if (transport_controller_->AddRemoteCandidates(content.name, candidates,
|
|
&error)) {
|
|
// Candidates successfully submitted for checking.
|
|
if (ice_connection_state_ == PeerConnectionInterface::kIceConnectionNew ||
|
|
ice_connection_state_ ==
|
|
PeerConnectionInterface::kIceConnectionDisconnected) {
|
|
// If state is New, then the session has just gotten its first remote ICE
|
|
// candidates, so go to Checking.
|
|
// If state is Disconnected, the session is re-using old candidates or
|
|
// receiving additional ones, so go to Checking.
|
|
// If state is Connected, stay Connected.
|
|
// TODO(bemasc): If state is Connected, and the new candidates are for a
|
|
// newly added transport, then the state actually _should_ move to
|
|
// checking. Add a way to distinguish that case.
|
|
SetIceConnectionState(PeerConnectionInterface::kIceConnectionChecking);
|
|
}
|
|
// TODO(bemasc): If state is Completed, go back to Connected.
|
|
} else {
|
|
if (!error.empty()) {
|
|
RTC_LOG(LS_WARNING) << error;
|
|
}
|
|
}
|
|
return true;
|
|
}
|
|
|
|
void PeerConnection::RemoveUnusedChannels(const SessionDescription* desc) {
|
|
// Destroy video channel first since it may have a pointer to the
|
|
// voice channel.
|
|
const cricket::ContentInfo* video_info = cricket::GetFirstVideoContent(desc);
|
|
if (!video_info || video_info->rejected) {
|
|
DestroyTransceiverChannel(GetVideoTransceiver());
|
|
}
|
|
|
|
const cricket::ContentInfo* audio_info = cricket::GetFirstAudioContent(desc);
|
|
if (!audio_info || audio_info->rejected) {
|
|
DestroyTransceiverChannel(GetAudioTransceiver());
|
|
}
|
|
|
|
const cricket::ContentInfo* data_info = cricket::GetFirstDataContent(desc);
|
|
if (!data_info || data_info->rejected) {
|
|
DestroyDataChannel();
|
|
}
|
|
}
|
|
|
|
std::string PeerConnection::GetTransportNameForMediaSection(
|
|
const std::string& mid,
|
|
const cricket::ContentGroup* bundle_group) const {
|
|
if (!bundle_group) {
|
|
return mid;
|
|
}
|
|
const std::string* first_content_name = bundle_group->FirstContentName();
|
|
if (!first_content_name) {
|
|
RTC_LOG(LS_WARNING) << "Tried to BUNDLE with no contents.";
|
|
return mid;
|
|
}
|
|
if (!bundle_group->HasContentName(mid)) {
|
|
RTC_LOG(LS_WARNING) << mid << " is not part of any bundle group";
|
|
return mid;
|
|
}
|
|
RTC_LOG(LS_INFO) << "Bundling " << mid << " on " << *first_content_name;
|
|
return *first_content_name;
|
|
}
|
|
|
|
RTCError PeerConnection::CreateChannels(const SessionDescription* desc) {
|
|
RTC_DCHECK(desc);
|
|
|
|
const cricket::ContentGroup* bundle_group = nullptr;
|
|
if (configuration_.bundle_policy ==
|
|
PeerConnectionInterface::kBundlePolicyMaxBundle) {
|
|
bundle_group = desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
|
|
if (!bundle_group) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"max-bundle configured but session description "
|
|
"has no BUNDLE group");
|
|
}
|
|
}
|
|
|
|
// Creating the media channels and transport proxies.
|
|
const cricket::ContentInfo* voice = cricket::GetFirstAudioContent(desc);
|
|
if (voice && !voice->rejected &&
|
|
!GetAudioTransceiver()->internal()->channel()) {
|
|
cricket::VoiceChannel* voice_channel = CreateVoiceChannel(
|
|
voice->name,
|
|
GetTransportNameForMediaSection(voice->name, bundle_group));
|
|
if (!voice_channel) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
|
|
"Failed to create voice channel.");
|
|
}
|
|
GetAudioTransceiver()->internal()->SetChannel(voice_channel);
|
|
}
|
|
|
|
const cricket::ContentInfo* video = cricket::GetFirstVideoContent(desc);
|
|
if (video && !video->rejected &&
|
|
!GetVideoTransceiver()->internal()->channel()) {
|
|
cricket::VideoChannel* video_channel = CreateVideoChannel(
|
|
video->name,
|
|
GetTransportNameForMediaSection(video->name, bundle_group));
|
|
if (!video_channel) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
|
|
"Failed to create video channel.");
|
|
}
|
|
GetVideoTransceiver()->internal()->SetChannel(video_channel);
|
|
}
|
|
|
|
const cricket::ContentInfo* data = cricket::GetFirstDataContent(desc);
|
|
if (data_channel_type_ != cricket::DCT_NONE && data && !data->rejected &&
|
|
!rtp_data_channel_ && !sctp_transport_) {
|
|
if (!CreateDataChannel(data->name, GetTransportNameForMediaSection(
|
|
data->name, bundle_group))) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
|
|
"Failed to create data channel.");
|
|
}
|
|
}
|
|
|
|
return RTCError::OK();
|
|
}
|
|
|
|
// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
|
|
cricket::VoiceChannel* PeerConnection::CreateVoiceChannel(
|
|
const std::string& mid,
|
|
const std::string& transport_name) {
|
|
cricket::DtlsTransportInternal* rtp_dtls_transport =
|
|
transport_controller_->CreateDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;
|
|
if (configuration_.rtcp_mux_policy !=
|
|
PeerConnectionInterface::kRtcpMuxPolicyRequire) {
|
|
rtcp_dtls_transport = transport_controller_->CreateDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
|
|
}
|
|
|
|
cricket::VoiceChannel* voice_channel = channel_manager()->CreateVoiceChannel(
|
|
call_.get(), configuration_.media_config, rtp_dtls_transport,
|
|
rtcp_dtls_transport, signaling_thread(), mid, SrtpRequired(),
|
|
audio_options_);
|
|
if (!voice_channel) {
|
|
transport_controller_->DestroyDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
if (rtcp_dtls_transport) {
|
|
transport_controller_->DestroyDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
|
|
}
|
|
return nullptr;
|
|
}
|
|
voice_channel->SignalRtcpMuxFullyActive.connect(
|
|
this, &PeerConnection::DestroyRtcpTransport_n);
|
|
voice_channel->SignalDtlsSrtpSetupFailure.connect(
|
|
this, &PeerConnection::OnDtlsSrtpSetupFailure);
|
|
voice_channel->SignalSentPacket.connect(this,
|
|
&PeerConnection::OnSentPacket_w);
|
|
|
|
return voice_channel;
|
|
}
|
|
|
|
// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
|
|
cricket::VideoChannel* PeerConnection::CreateVideoChannel(
|
|
const std::string& mid,
|
|
const std::string& transport_name) {
|
|
cricket::DtlsTransportInternal* rtp_dtls_transport =
|
|
transport_controller_->CreateDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;
|
|
if (configuration_.rtcp_mux_policy !=
|
|
PeerConnectionInterface::kRtcpMuxPolicyRequire) {
|
|
rtcp_dtls_transport = transport_controller_->CreateDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
|
|
}
|
|
|
|
cricket::VideoChannel* video_channel = channel_manager()->CreateVideoChannel(
|
|
call_.get(), configuration_.media_config, rtp_dtls_transport,
|
|
rtcp_dtls_transport, signaling_thread(), mid, SrtpRequired(),
|
|
video_options_);
|
|
|
|
if (!video_channel) {
|
|
transport_controller_->DestroyDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
if (rtcp_dtls_transport) {
|
|
transport_controller_->DestroyDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
|
|
}
|
|
return nullptr;
|
|
}
|
|
video_channel->SignalRtcpMuxFullyActive.connect(
|
|
this, &PeerConnection::DestroyRtcpTransport_n);
|
|
video_channel->SignalDtlsSrtpSetupFailure.connect(
|
|
this, &PeerConnection::OnDtlsSrtpSetupFailure);
|
|
video_channel->SignalSentPacket.connect(this,
|
|
&PeerConnection::OnSentPacket_w);
|
|
|
|
return video_channel;
|
|
}
|
|
|
|
bool PeerConnection::CreateDataChannel(const std::string& mid,
|
|
const std::string& transport_name) {
|
|
bool sctp = (data_channel_type_ == cricket::DCT_SCTP);
|
|
if (sctp) {
|
|
if (!sctp_factory_) {
|
|
RTC_LOG(LS_ERROR)
|
|
<< "Trying to create SCTP transport, but didn't compile with "
|
|
"SCTP support (HAVE_SCTP)";
|
|
return false;
|
|
}
|
|
if (!network_thread()->Invoke<bool>(
|
|
RTC_FROM_HERE, rtc::Bind(&PeerConnection::CreateSctpTransport_n,
|
|
this, mid, transport_name))) {
|
|
return false;
|
|
}
|
|
for (const auto& channel : sctp_data_channels_) {
|
|
channel->OnTransportChannelCreated();
|
|
}
|
|
} else {
|
|
cricket::DtlsTransportInternal* rtp_dtls_transport =
|
|
transport_controller_->CreateDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;
|
|
if (configuration_.rtcp_mux_policy !=
|
|
PeerConnectionInterface::kRtcpMuxPolicyRequire) {
|
|
rtcp_dtls_transport = transport_controller_->CreateDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
|
|
}
|
|
|
|
rtp_data_channel_ = channel_manager()->CreateRtpDataChannel(
|
|
configuration_.media_config, rtp_dtls_transport, rtcp_dtls_transport,
|
|
signaling_thread(), mid, SrtpRequired());
|
|
|
|
if (!rtp_data_channel_) {
|
|
transport_controller_->DestroyDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
if (rtcp_dtls_transport) {
|
|
transport_controller_->DestroyDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
|
|
}
|
|
return false;
|
|
}
|
|
|
|
rtp_data_channel_->SignalRtcpMuxFullyActive.connect(
|
|
this, &PeerConnection::DestroyRtcpTransport_n);
|
|
rtp_data_channel_->SignalDtlsSrtpSetupFailure.connect(
|
|
this, &PeerConnection::OnDtlsSrtpSetupFailure);
|
|
rtp_data_channel_->SignalSentPacket.connect(
|
|
this, &PeerConnection::OnSentPacket_w);
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
Call::Stats PeerConnection::GetCallStats() {
|
|
if (!worker_thread()->IsCurrent()) {
|
|
return worker_thread()->Invoke<Call::Stats>(
|
|
RTC_FROM_HERE, rtc::Bind(&PeerConnection::GetCallStats, this));
|
|
}
|
|
if (call_) {
|
|
return call_->GetStats();
|
|
} else {
|
|
return Call::Stats();
|
|
}
|
|
}
|
|
|
|
std::unique_ptr<SessionStats> PeerConnection::GetSessionStats_n(
|
|
const ChannelNamePairs& channel_name_pairs) {
|
|
RTC_DCHECK(network_thread()->IsCurrent());
|
|
std::unique_ptr<SessionStats> session_stats(new SessionStats());
|
|
for (const auto channel_name_pair :
|
|
{&channel_name_pairs.voice, &channel_name_pairs.video,
|
|
&channel_name_pairs.data}) {
|
|
if (*channel_name_pair) {
|
|
cricket::TransportStats transport_stats;
|
|
if (!transport_controller_->GetStats((*channel_name_pair)->transport_name,
|
|
&transport_stats)) {
|
|
return nullptr;
|
|
}
|
|
session_stats->transport_stats[(*channel_name_pair)->transport_name] =
|
|
std::move(transport_stats);
|
|
}
|
|
}
|
|
return session_stats;
|
|
}
|
|
|
|
bool PeerConnection::CreateSctpTransport_n(const std::string& content_name,
|
|
const std::string& transport_name) {
|
|
RTC_DCHECK(network_thread()->IsCurrent());
|
|
RTC_DCHECK(sctp_factory_);
|
|
cricket::DtlsTransportInternal* tc =
|
|
transport_controller_->CreateDtlsTransport_n(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
sctp_transport_ = sctp_factory_->CreateSctpTransport(tc);
|
|
RTC_DCHECK(sctp_transport_);
|
|
sctp_invoker_.reset(new rtc::AsyncInvoker());
|
|
sctp_transport_->SignalReadyToSendData.connect(
|
|
this, &PeerConnection::OnSctpTransportReadyToSendData_n);
|
|
sctp_transport_->SignalDataReceived.connect(
|
|
this, &PeerConnection::OnSctpTransportDataReceived_n);
|
|
sctp_transport_->SignalStreamClosedRemotely.connect(
|
|
this, &PeerConnection::OnSctpStreamClosedRemotely_n);
|
|
sctp_transport_name_ = transport_name;
|
|
sctp_content_name_ = content_name;
|
|
return true;
|
|
}
|
|
|
|
void PeerConnection::ChangeSctpTransport_n(const std::string& transport_name) {
|
|
RTC_DCHECK(network_thread()->IsCurrent());
|
|
RTC_DCHECK(sctp_transport_);
|
|
RTC_DCHECK(sctp_transport_name_);
|
|
std::string old_sctp_transport_name = *sctp_transport_name_;
|
|
sctp_transport_name_ = transport_name;
|
|
cricket::DtlsTransportInternal* tc =
|
|
transport_controller_->CreateDtlsTransport_n(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
sctp_transport_->SetTransportChannel(tc);
|
|
transport_controller_->DestroyDtlsTransport_n(
|
|
old_sctp_transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
}
|
|
|
|
void PeerConnection::DestroySctpTransport_n() {
|
|
RTC_DCHECK(network_thread()->IsCurrent());
|
|
sctp_transport_.reset(nullptr);
|
|
sctp_content_name_.reset();
|
|
sctp_transport_name_.reset();
|
|
sctp_invoker_.reset(nullptr);
|
|
sctp_ready_to_send_data_ = false;
|
|
}
|
|
|
|
void PeerConnection::OnSctpTransportReadyToSendData_n() {
|
|
RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
|
|
RTC_DCHECK(network_thread()->IsCurrent());
|
|
// Note: Cannot use rtc::Bind here because it will grab a reference to
|
|
// PeerConnection and potentially cause PeerConnection to live longer than
|
|
// expected. It is safe not to grab a reference since the sctp_invoker_ will
|
|
// be destroyed before PeerConnection is destroyed, and at that point all
|
|
// pending tasks will be cleared.
|
|
sctp_invoker_->AsyncInvoke<void>(RTC_FROM_HERE, signaling_thread(), [this] {
|
|
OnSctpTransportReadyToSendData_s(true);
|
|
});
|
|
}
|
|
|
|
void PeerConnection::OnSctpTransportReadyToSendData_s(bool ready) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
sctp_ready_to_send_data_ = ready;
|
|
SignalSctpReadyToSendData(ready);
|
|
}
|
|
|
|
void PeerConnection::OnSctpTransportDataReceived_n(
|
|
const cricket::ReceiveDataParams& params,
|
|
const rtc::CopyOnWriteBuffer& payload) {
|
|
RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
|
|
RTC_DCHECK(network_thread()->IsCurrent());
|
|
// Note: Cannot use rtc::Bind here because it will grab a reference to
|
|
// PeerConnection and potentially cause PeerConnection to live longer than
|
|
// expected. It is safe not to grab a reference since the sctp_invoker_ will
|
|
// be destroyed before PeerConnection is destroyed, and at that point all
|
|
// pending tasks will be cleared.
|
|
sctp_invoker_->AsyncInvoke<void>(
|
|
RTC_FROM_HERE, signaling_thread(), [this, params, payload] {
|
|
OnSctpTransportDataReceived_s(params, payload);
|
|
});
|
|
}
|
|
|
|
void PeerConnection::OnSctpTransportDataReceived_s(
|
|
const cricket::ReceiveDataParams& params,
|
|
const rtc::CopyOnWriteBuffer& payload) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
if (params.type == cricket::DMT_CONTROL && IsOpenMessage(payload)) {
|
|
// Received OPEN message; parse and signal that a new data channel should
|
|
// be created.
|
|
std::string label;
|
|
InternalDataChannelInit config;
|
|
config.id = params.ssrc;
|
|
if (!ParseDataChannelOpenMessage(payload, &label, &config)) {
|
|
RTC_LOG(LS_WARNING) << "Failed to parse the OPEN message for sid "
|
|
<< params.ssrc;
|
|
return;
|
|
}
|
|
config.open_handshake_role = InternalDataChannelInit::kAcker;
|
|
OnDataChannelOpenMessage(label, config);
|
|
} else {
|
|
// Otherwise just forward the signal.
|
|
SignalSctpDataReceived(params, payload);
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnSctpStreamClosedRemotely_n(int sid) {
|
|
RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
|
|
RTC_DCHECK(network_thread()->IsCurrent());
|
|
sctp_invoker_->AsyncInvoke<void>(
|
|
RTC_FROM_HERE, signaling_thread(),
|
|
rtc::Bind(&sigslot::signal1<int>::operator(),
|
|
&SignalSctpStreamClosedRemotely, sid));
|
|
}
|
|
|
|
// Returns false if bundle is enabled and rtcp_mux is disabled.
|
|
bool PeerConnection::ValidateBundleSettings(const SessionDescription* desc) {
|
|
bool bundle_enabled = desc->HasGroup(cricket::GROUP_TYPE_BUNDLE);
|
|
if (!bundle_enabled)
|
|
return true;
|
|
|
|
const cricket::ContentGroup* bundle_group =
|
|
desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
|
|
RTC_DCHECK(bundle_group != NULL);
|
|
|
|
const cricket::ContentInfos& contents = desc->contents();
|
|
for (cricket::ContentInfos::const_iterator citer = contents.begin();
|
|
citer != contents.end(); ++citer) {
|
|
const cricket::ContentInfo* content = (&*citer);
|
|
RTC_DCHECK(content != NULL);
|
|
if (bundle_group->HasContentName(content->name) && !content->rejected &&
|
|
content->type == MediaProtocolType::kRtp) {
|
|
if (!HasRtcpMuxEnabled(content))
|
|
return false;
|
|
}
|
|
}
|
|
// RTCP-MUX is enabled in all the contents.
|
|
return true;
|
|
}
|
|
|
|
bool PeerConnection::HasRtcpMuxEnabled(const cricket::ContentInfo* content) {
|
|
return content->media_description()->rtcp_mux();
|
|
}
|
|
|
|
RTCError PeerConnection::ValidateSessionDescription(
|
|
const SessionDescriptionInterface* sdesc,
|
|
cricket::ContentSource source) {
|
|
if (session_error() != SessionError::kNone) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR, GetSessionErrorMsg());
|
|
}
|
|
|
|
if (!sdesc || !sdesc->description()) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, kInvalidSdp);
|
|
}
|
|
|
|
SdpType type = sdesc->GetType();
|
|
if ((source == cricket::CS_LOCAL && !ExpectSetLocalDescription(type)) ||
|
|
(source == cricket::CS_REMOTE && !ExpectSetRemoteDescription(type))) {
|
|
LOG_AND_RETURN_ERROR(
|
|
RTCErrorType::INVALID_PARAMETER,
|
|
"Called in wrong state: " + GetSignalingStateString(signaling_state()));
|
|
}
|
|
|
|
// Verify crypto settings.
|
|
std::string crypto_error;
|
|
if (webrtc_session_desc_factory_->SdesPolicy() == cricket::SEC_REQUIRED ||
|
|
dtls_enabled_) {
|
|
RTCError crypto_error = VerifyCrypto(sdesc->description(), dtls_enabled_);
|
|
if (!crypto_error.ok()) {
|
|
return crypto_error;
|
|
}
|
|
}
|
|
|
|
// Verify ice-ufrag and ice-pwd.
|
|
if (!VerifyIceUfragPwdPresent(sdesc->description())) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
kSdpWithoutIceUfragPwd);
|
|
}
|
|
|
|
if (!ValidateBundleSettings(sdesc->description())) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
kBundleWithoutRtcpMux);
|
|
}
|
|
|
|
// TODO(skvlad): When the local rtcp-mux policy is Require, reject any
|
|
// m-lines that do not rtcp-mux enabled.
|
|
|
|
// Verify m-lines in Answer when compared against Offer.
|
|
if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
|
|
const cricket::SessionDescription* offer_desc =
|
|
(source == cricket::CS_LOCAL) ? remote_description()->description()
|
|
: local_description()->description();
|
|
if (!MediaSectionsHaveSameCount(offer_desc, sdesc->description()) ||
|
|
!MediaSectionsInSameOrder(offer_desc, sdesc->description())) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
kMlineMismatchInAnswer);
|
|
}
|
|
} else {
|
|
const cricket::SessionDescription* current_desc = nullptr;
|
|
if (source == cricket::CS_LOCAL && local_description()) {
|
|
current_desc = local_description()->description();
|
|
} else if (source == cricket::CS_REMOTE && remote_description()) {
|
|
current_desc = remote_description()->description();
|
|
}
|
|
// The re-offers should respect the order of m= sections in current
|
|
// description. See RFC3264 Section 8 paragraph 4 for more details.
|
|
if (current_desc &&
|
|
!MediaSectionsInSameOrder(current_desc, sdesc->description())) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
kMlineMismatchInSubsequentOffer);
|
|
}
|
|
}
|
|
|
|
return RTCError::OK();
|
|
}
|
|
|
|
bool PeerConnection::ExpectSetLocalDescription(SdpType type) {
|
|
PeerConnectionInterface::SignalingState state = signaling_state();
|
|
if (type == SdpType::kOffer) {
|
|
return (state == PeerConnectionInterface::kStable) ||
|
|
(state == PeerConnectionInterface::kHaveLocalOffer);
|
|
} else {
|
|
RTC_DCHECK(type == SdpType::kPrAnswer || type == SdpType::kAnswer);
|
|
return (state == PeerConnectionInterface::kHaveRemoteOffer) ||
|
|
(state == PeerConnectionInterface::kHaveLocalPrAnswer);
|
|
}
|
|
}
|
|
|
|
bool PeerConnection::ExpectSetRemoteDescription(SdpType type) {
|
|
PeerConnectionInterface::SignalingState state = signaling_state();
|
|
if (type == SdpType::kOffer) {
|
|
return (state == PeerConnectionInterface::kStable) ||
|
|
(state == PeerConnectionInterface::kHaveRemoteOffer);
|
|
} else {
|
|
RTC_DCHECK(type == SdpType::kPrAnswer || type == SdpType::kAnswer);
|
|
return (state == PeerConnectionInterface::kHaveLocalOffer) ||
|
|
(state == PeerConnectionInterface::kHaveRemotePrAnswer);
|
|
}
|
|
}
|
|
|
|
const char* PeerConnection::SessionErrorToString(SessionError error) const {
|
|
switch (error) {
|
|
case SessionError::kNone:
|
|
return "ERROR_NONE";
|
|
case SessionError::kContent:
|
|
return "ERROR_CONTENT";
|
|
case SessionError::kTransport:
|
|
return "ERROR_TRANSPORT";
|
|
}
|
|
RTC_NOTREACHED();
|
|
return "";
|
|
}
|
|
|
|
std::string PeerConnection::GetSessionErrorMsg() {
|
|
std::ostringstream desc;
|
|
desc << kSessionError << SessionErrorToString(session_error()) << ". ";
|
|
desc << kSessionErrorDesc << session_error_desc() << ".";
|
|
return desc.str();
|
|
}
|
|
|
|
// We need to check the local/remote description for the Transport instead of
|
|
// the session, because a new Transport added during renegotiation may have
|
|
// them unset while the session has them set from the previous negotiation.
|
|
// Not doing so may trigger the auto generation of transport description and
|
|
// mess up DTLS identity information, ICE credential, etc.
|
|
bool PeerConnection::ReadyToUseRemoteCandidate(
|
|
const IceCandidateInterface* candidate,
|
|
const SessionDescriptionInterface* remote_desc,
|
|
bool* valid) {
|
|
*valid = true;
|
|
|
|
const SessionDescriptionInterface* current_remote_desc =
|
|
remote_desc ? remote_desc : remote_description();
|
|
|
|
if (!current_remote_desc) {
|
|
return false;
|
|
}
|
|
|
|
size_t mediacontent_index = static_cast<size_t>(candidate->sdp_mline_index());
|
|
size_t remote_content_size =
|
|
current_remote_desc->description()->contents().size();
|
|
if (mediacontent_index >= remote_content_size) {
|
|
RTC_LOG(LS_ERROR)
|
|
<< "ReadyToUseRemoteCandidate: Invalid candidate media index "
|
|
<< mediacontent_index;
|
|
|
|
*valid = false;
|
|
return false;
|
|
}
|
|
|
|
cricket::ContentInfo content =
|
|
current_remote_desc->description()->contents()[mediacontent_index];
|
|
|
|
const std::string transport_name = GetTransportName(content.name);
|
|
if (transport_name.empty()) {
|
|
return false;
|
|
}
|
|
return transport_controller_->ReadyForRemoteCandidates(transport_name);
|
|
}
|
|
|
|
bool PeerConnection::SrtpRequired() const {
|
|
return dtls_enabled_ ||
|
|
webrtc_session_desc_factory_->SdesPolicy() == cricket::SEC_REQUIRED;
|
|
}
|
|
|
|
void PeerConnection::OnTransportControllerGatheringState(
|
|
cricket::IceGatheringState state) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
if (state == cricket::kIceGatheringGathering) {
|
|
OnIceGatheringChange(PeerConnectionInterface::kIceGatheringGathering);
|
|
} else if (state == cricket::kIceGatheringComplete) {
|
|
OnIceGatheringChange(PeerConnectionInterface::kIceGatheringComplete);
|
|
}
|
|
}
|
|
|
|
void PeerConnection::ReportTransportStats() {
|
|
// Use a set so we don't report the same stats twice if two channels share
|
|
// a transport.
|
|
std::set<std::string> transport_names;
|
|
if (voice_channel()) {
|
|
transport_names.insert(voice_channel()->transport_name());
|
|
}
|
|
if (video_channel()) {
|
|
transport_names.insert(video_channel()->transport_name());
|
|
}
|
|
if (rtp_data_channel()) {
|
|
transport_names.insert(rtp_data_channel()->transport_name());
|
|
}
|
|
if (sctp_transport_name_) {
|
|
transport_names.insert(*sctp_transport_name_);
|
|
}
|
|
for (const auto& name : transport_names) {
|
|
cricket::TransportStats stats;
|
|
if (transport_controller_->GetStats(name, &stats)) {
|
|
ReportBestConnectionState(stats);
|
|
ReportNegotiatedCiphers(stats);
|
|
}
|
|
}
|
|
}
|
|
// Walk through the ConnectionInfos to gather best connection usage
|
|
// for IPv4 and IPv6.
|
|
void PeerConnection::ReportBestConnectionState(
|
|
const cricket::TransportStats& stats) {
|
|
RTC_DCHECK(metrics_observer());
|
|
for (cricket::TransportChannelStatsList::const_iterator it =
|
|
stats.channel_stats.begin();
|
|
it != stats.channel_stats.end(); ++it) {
|
|
for (cricket::ConnectionInfos::const_iterator it_info =
|
|
it->connection_infos.begin();
|
|
it_info != it->connection_infos.end(); ++it_info) {
|
|
if (!it_info->best_connection) {
|
|
continue;
|
|
}
|
|
|
|
PeerConnectionEnumCounterType type = kPeerConnectionEnumCounterMax;
|
|
const cricket::Candidate& local = it_info->local_candidate;
|
|
const cricket::Candidate& remote = it_info->remote_candidate;
|
|
|
|
// Increment the counter for IceCandidatePairType.
|
|
if (local.protocol() == cricket::TCP_PROTOCOL_NAME ||
|
|
(local.type() == RELAY_PORT_TYPE &&
|
|
local.relay_protocol() == cricket::TCP_PROTOCOL_NAME)) {
|
|
type = kEnumCounterIceCandidatePairTypeTcp;
|
|
} else if (local.protocol() == cricket::UDP_PROTOCOL_NAME) {
|
|
type = kEnumCounterIceCandidatePairTypeUdp;
|
|
} else {
|
|
RTC_CHECK(0);
|
|
}
|
|
metrics_observer()->IncrementEnumCounter(
|
|
type, GetIceCandidatePairCounter(local, remote),
|
|
kIceCandidatePairMax);
|
|
|
|
// Increment the counter for IP type.
|
|
if (local.address().family() == AF_INET) {
|
|
metrics_observer()->IncrementEnumCounter(
|
|
kEnumCounterAddressFamily, kBestConnections_IPv4,
|
|
kPeerConnectionAddressFamilyCounter_Max);
|
|
|
|
} else if (local.address().family() == AF_INET6) {
|
|
metrics_observer()->IncrementEnumCounter(
|
|
kEnumCounterAddressFamily, kBestConnections_IPv6,
|
|
kPeerConnectionAddressFamilyCounter_Max);
|
|
} else {
|
|
RTC_CHECK(0);
|
|
}
|
|
|
|
return;
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::ReportNegotiatedCiphers(
|
|
const cricket::TransportStats& stats) {
|
|
RTC_DCHECK(metrics_observer());
|
|
if (!dtls_enabled_ || stats.channel_stats.empty()) {
|
|
return;
|
|
}
|
|
|
|
int srtp_crypto_suite = stats.channel_stats[0].srtp_crypto_suite;
|
|
int ssl_cipher_suite = stats.channel_stats[0].ssl_cipher_suite;
|
|
if (srtp_crypto_suite == rtc::SRTP_INVALID_CRYPTO_SUITE &&
|
|
ssl_cipher_suite == rtc::TLS_NULL_WITH_NULL_NULL) {
|
|
return;
|
|
}
|
|
|
|
PeerConnectionEnumCounterType srtp_counter_type;
|
|
PeerConnectionEnumCounterType ssl_counter_type;
|
|
if (stats.transport_name == cricket::CN_AUDIO) {
|
|
srtp_counter_type = kEnumCounterAudioSrtpCipher;
|
|
ssl_counter_type = kEnumCounterAudioSslCipher;
|
|
} else if (stats.transport_name == cricket::CN_VIDEO) {
|
|
srtp_counter_type = kEnumCounterVideoSrtpCipher;
|
|
ssl_counter_type = kEnumCounterVideoSslCipher;
|
|
} else if (stats.transport_name == cricket::CN_DATA) {
|
|
srtp_counter_type = kEnumCounterDataSrtpCipher;
|
|
ssl_counter_type = kEnumCounterDataSslCipher;
|
|
} else {
|
|
RTC_NOTREACHED();
|
|
return;
|
|
}
|
|
|
|
if (srtp_crypto_suite != rtc::SRTP_INVALID_CRYPTO_SUITE) {
|
|
metrics_observer()->IncrementSparseEnumCounter(srtp_counter_type,
|
|
srtp_crypto_suite);
|
|
}
|
|
if (ssl_cipher_suite != rtc::TLS_NULL_WITH_NULL_NULL) {
|
|
metrics_observer()->IncrementSparseEnumCounter(ssl_counter_type,
|
|
ssl_cipher_suite);
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnSentPacket_w(const rtc::SentPacket& sent_packet) {
|
|
RTC_DCHECK(worker_thread()->IsCurrent());
|
|
RTC_DCHECK(call_);
|
|
call_->OnSentPacket(sent_packet);
|
|
}
|
|
|
|
const std::string PeerConnection::GetTransportName(
|
|
const std::string& content_name) {
|
|
cricket::BaseChannel* channel = GetChannel(content_name);
|
|
if (channel) {
|
|
return channel->transport_name();
|
|
}
|
|
if (sctp_transport_) {
|
|
RTC_DCHECK(sctp_content_name_);
|
|
RTC_DCHECK(sctp_transport_name_);
|
|
if (content_name == *sctp_content_name_) {
|
|
return *sctp_transport_name_;
|
|
}
|
|
}
|
|
// Return an empty string if failed to retrieve the transport name.
|
|
return "";
|
|
}
|
|
|
|
void PeerConnection::DestroyRtcpTransport_n(const std::string& transport_name) {
|
|
RTC_DCHECK(network_thread()->IsCurrent());
|
|
transport_controller_->DestroyDtlsTransport_n(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
|
|
}
|
|
|
|
void PeerConnection::DestroyTransceiverChannel(
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
transceiver) {
|
|
RTC_DCHECK(transceiver);
|
|
|
|
cricket::BaseChannel* channel = transceiver->internal()->channel();
|
|
if (channel) {
|
|
transceiver->internal()->SetChannel(nullptr);
|
|
DestroyBaseChannel(channel);
|
|
}
|
|
}
|
|
|
|
void PeerConnection::DestroyDataChannel() {
|
|
if (rtp_data_channel_) {
|
|
OnDataChannelDestroyed();
|
|
DestroyBaseChannel(rtp_data_channel_);
|
|
rtp_data_channel_ = nullptr;
|
|
}
|
|
|
|
// Note: Cannot use rtc::Bind to create a functor to invoke because it will
|
|
// grab a reference to this PeerConnection. If this is called from the
|
|
// PeerConnection destructor, the RefCountedObject vtable will have already
|
|
// been destroyed (since it is a subclass of PeerConnection) and using
|
|
// rtc::Bind will cause "Pure virtual function called" error to appear.
|
|
|
|
if (sctp_transport_) {
|
|
OnDataChannelDestroyed();
|
|
network_thread()->Invoke<void>(RTC_FROM_HERE,
|
|
[this] { DestroySctpTransport_n(); });
|
|
}
|
|
}
|
|
|
|
void PeerConnection::DestroyBaseChannel(cricket::BaseChannel* channel) {
|
|
RTC_DCHECK(channel);
|
|
RTC_DCHECK(channel->rtp_dtls_transport());
|
|
|
|
// Need to cache these before destroying the base channel so that we do not
|
|
// access uninitialized memory.
|
|
const std::string transport_name =
|
|
channel->rtp_dtls_transport()->transport_name();
|
|
const bool need_to_delete_rtcp = (channel->rtcp_dtls_transport() != nullptr);
|
|
|
|
switch (channel->media_type()) {
|
|
case cricket::MEDIA_TYPE_AUDIO:
|
|
channel_manager()->DestroyVoiceChannel(
|
|
static_cast<cricket::VoiceChannel*>(channel));
|
|
break;
|
|
case cricket::MEDIA_TYPE_VIDEO:
|
|
channel_manager()->DestroyVideoChannel(
|
|
static_cast<cricket::VideoChannel*>(channel));
|
|
break;
|
|
case cricket::MEDIA_TYPE_DATA:
|
|
channel_manager()->DestroyRtpDataChannel(
|
|
static_cast<cricket::RtpDataChannel*>(channel));
|
|
break;
|
|
default:
|
|
RTC_NOTREACHED() << "Unknown media type: " << channel->media_type();
|
|
break;
|
|
}
|
|
|
|
// |channel| can no longer be used.
|
|
|
|
transport_controller_->DestroyDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
|
|
if (need_to_delete_rtcp) {
|
|
transport_controller_->DestroyDtlsTransport(
|
|
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
|
|
}
|
|
}
|
|
|
|
} // namespace webrtc
|